From cehoyos at ag.or.at  Thu Aug  1 00:12:19 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 31 Jul 2013 22:12:19 +0000 (UTC)
Subject: [FFmpeg-user] Offlist,
	Is mixing of audio channels happening by default in ffmpeg?
References: <51E7E8A6.8070906@ringsidecreative.com>
	<loom.20130718T151116-7@post.gmane.org>
	<51E7ED02.5080707@ringsidecreative.com>
	<loom.20130718T153016-321@post.gmane.org>
	<51E7EF4E.8020100@ringsidecreative.com>
	<loom.20130718T154053-475@post.gmane.org>
	<0EF41E1BE75A4DBB80A1E74873E4DC91@HPKANTOOR>
	<loom.20130719T101328-635@post.gmane.org>
	<EA3514503FCB4A868E0BB71A67D7A8A7@HPKANTOOR>
	<loom.20130719T120527-55@post.gmane.org>
	<CACegdbJ3EsCjRZZGnPa0VXFdwjagM2wX7fAAtPLzNVCfa6RFyg@mail.gmail.com>
Message-ID: <loom.20130801T001127-852@post.gmane.org>

Nathan Lewis <nlewis <at> crawford.com> writes:

> I'd be glad to help test this but I'm not clear on 
> what needs to change.
> Does avio_wb24(pb, track->secondary ? 0x2 : 0xf); 
> need to become 
> avio_wb24(pb, track->secondary ? 0x2 : 0xF);  ?

No, please test:
avio_wb24(pb, 0xF);

Thank you, Carl Eugen


From jiachielee at live.com  Thu Aug  1 04:13:36 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 31 Jul 2013 19:13:36 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51F6A399.4020400@gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
Message-ID: <1375323216242-4660379.post@n4.nabble.com>

Andy Furniss-2 wrote
> thljcl wrote:
> 
>> The short answer: YUV444 can fully represent all color information in
>> RGB.
> 
> Did you even read the pdf I linked to?
> 
> Maybe you know better than him - a fellow of smpte according to
> 
> http://en.wikipedia.org/wiki/Charles_Poynton
> 
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Now that I think about it. Many of us are familiar with lossy compression
with both audio and video but not lossless compression. Clearly the best of
the codec implementation is only up to how accurate and fast the decoding
and encoding process goes. In the early days when H.264 codec is
implemented, even though the lossless encoding is available in High 4:4:4
Predictive Profile (Hi444PP) according to the H.264 specification; the
mathematical intensity and demanding computing resources requirement
prevents a lot of people to actually test it or use it. It?s not unusual to
see many streaming videos are encoded in Baseline Profile or other simpler
codecs.
There is no denying that badly designed implementation of particular
encoders would result in the substantial loss of image quality; which is
particularly true when it comes to re-encoding the color information in
different representation, such as in YCbCr. The rounding error, noise, and
other complexities would cause the information to be lost during the
encoding process. A simple way to illustrate this is that 1/3 ? 0.33. Also,
there would be issues related to the correct decoding of signals from
encoded information. As we said before, YCbCr itself is not an absolute
color space. Without a well-defined standard for a given codec
implementation, the incorrect decoding process would result in loss of
information.
If a lossy encoding scheme is used, the use of YCbCr 4:4:4 would not change
the fact that some information is lost during the encoding process. I?m not
absolutely certain that the implementation of lossless encoding by x264 is
flawless. But the lossless encoding of H.264 is done with the use of YCbCr
4:4:4; H.264 does not directly support sRGB. Even though in Windows, one can
argue that the color information would be decoded to and encoded from sRGB
anyway; it?s the absolute color space, after all.
The initial version of H.264 version, published on May 30, 2003, contains
only Baseline, Main, and Extended profiles; none of them supports lossless
encoding or chroma format of 4:4:4. H.264 2.0 Specification (version three),
published on March 3, 2005, adds new four profiles, which are High, High 10,
High 4:2:2, and High 4:4:4. 
In particular, I quotes
?With the exception of the transform bypass mode of operation for lossless
coding in the High 4:4:4 profile and the I_PCM mode of operation in all
profiles, the algorithm is typically not lossless.?
H.264 High 4:4:4 profile explicitly supports lossless coding in High 4:4:4
with the chroma format of 4:4:4. H.264 High 4:4:4 profile is later removed
in H.264 2.2 Specification, published on June, 2006. Since H.264 2.3
Specification, published on April, 2007, it adds High 4:4:4 Predictive
Profile. At H.264 8.0 Specification, published on April 2013, I quotes
?With the exception of the transform bypass mode of operation for lossless
coding in the High 4:4:4 Intra, CAVLC 4:4:4 Intra, and High 4:4:4 Predictive
profiles, and the I_PCM mode of operation in all profiles, the algorithm is
typically not lossless, as the exact source sample values are typically not
preserved through the encoding and decoding processes?
In the current specification (Ed 8.0), there are three profiles of H.264
that support lossless coding. Among those three, one of them, which is High
4:4:4 Predictive, is currently implemented by x264, which is used by ffmpeg
as H.264 encoder. 
Poynton?s article you mentioned at
http://www.poynton.com/papers/Discreet_Logic/index.html was published on
July 28, 1997; it was a time not even the first edition of H.264
Specification was published.
It?s fair to say that while some statements stated by him at the time remain
valid today, considering the pace of change in technology, we cannot take
everything he stated at the time for granted today.
Whether or not -pix_fmt ?yuv444p? is required to be specified explicitly
when it comes to lossless coding depends on the sources, if sources are PNG
images, ?yuv444p? would be chosen automatically. If not, you may need to
specify ?yuv444p? for lossless coding.
As usual, you do not have to take my words for granted. Go to
http://www.itu.int/ITU-T/recommendations/rec.aspx?rec=6312 read the H.264
specifications yourself. The current edition as well as the past editions
are all available online. You can verify yourself whether I said is true or
not.
Of course, you can claim that the lossless coding in H.264 is not ?truly
lossless?. If you can submit proofs to prove your case, many other people
and I will be glad to accept the truth. Until then, let us the accept the
currently known theories.




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From adf.lists at gmail.com  Thu Aug  1 12:12:10 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Thu, 01 Aug 2013 11:12:10 +0100
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375323216242-4660379.post@n4.nabble.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
Message-ID: <51FA347A.6080103@gmail.com>

thljcl wrote:

> In the current specification (Ed 8.0), there are three profiles of H.264
> that support lossless coding. Among those three, one of them, which is High
> 4:4:4 Predictive, is currently implemented by x264, which is used by ffmpeg
> as H.264 encoder.
> Poynton?s article you mentioned at
> http://www.poynton.com/papers/Discreet_Logic/index.html was published on
> July 28, 1997; it was a time not even the first edition of H.264
> Specification was published.
> It?s fair to say that while some statements stated by him at the time remain
> valid today, considering the pace of change in technology, we cannot take
> everything he stated at the time for granted today.
> Whether or not -pix_fmt ?yuv444p? is required to be specified explicitly
> when it comes to lossless coding depends on the sources, if sources are PNG
> images, ?yuv444p? would be chosen automatically. If not, you may need to
> specify ?yuv444p? for lossless coding.
> As usual, you do not have to take my words for granted. Go to
> http://www.itu.int/ITU-T/recommendations/rec.aspx?rec=6312 read the H.264
> specifications yourself. The current edition as well as the past editions
> are all available online. You can verify yourself whether I said is true or
> not.
> Of course, you can claim that the lossless coding in H.264 is not ?truly
> lossless?. If you can submit proofs to prove your case, many other people
> and I will be glad to accept the truth. Until then, let us the accept the
> currently known theories.

Well you maths is better than mine, and the effort you put into 
presenting your case is hugely more than me, but -

You don't seem to have listened to what Nicolas said and now above you 
seem to to be comparing something totally un-related to what was stated 
in the Poynton paper.

He is talking about full range RGB to 601 studio levels Y'CbCr, but the 
stuff about lossless H264 above has  absolutely nothing to do with that.

H264 is talking about lossless where input is and output are the same 
444 - Y'CbCr - there is no conversion by H264 from RGB as your PNG 
example would need - that would be done prior to H264 by ffmpeg.






From jiachielee at live.com  Thu Aug  1 13:00:27 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 04:00:27 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51FA347A.6080103@gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
Message-ID: <1375354827157-4660382.post@n4.nabble.com>

At first, reading at the PDF file written by Poynton seems to lead us to
conclude that color space conversion alone, before chroma subsampling is
done, would reduce the number of possible colors, thus reducing color
information. If that is true, that?s what we called ?lossy encoding?, i.e.
the encoding process where the original information is irrecoverable. 
Hm? Seems like I?ve given out a lot of information without arranging them
systematically.
Let me try to make it simpler.
I first tried to define what YCbCr really means and tell what ?absolute
color space? and ?relative color space? really means.
As I said before, YCbCr alone cannot define colors, that is without a
well-defined mapping to an absolute color space.
Then the question naturally leads to how does such mapping work? Does it
one-to-one or many to one?
That?s why I specifically choose a conversion matrix and try to derive its
inverse to show that its relationship is truly one-to-one, at least
mathematically.
However, mathematically one-to-one function does not guarantee the actual
encoding process is lossless.
As I said before, it?s very common for studios or producers to use lossy
encoding scheme; then the point of use YCbCr 4:4:4 would not have any effect
on preserving the color information. To put it simply, the bits for color
information have been reduced, of course there would be less color
information.
For computer or software to read YCbCr signal correctly, it has to be
decoded to an absolute color space, where sRGB is the most common one,
co-developed by Microsoft and HP.
Due to the widespread popularity of sRGB, the color profiles or the color
space assumed to be used by lossless image files is always sRGB; if but
software is able to do conversion between absolute color space. But YCbCr is
not absolute color space, it?s just the question of rearranging the
information and doing the right mapping.
In practice, the encoding process is very complex, often with noise,
rounding errors, and other issues. That?s why codec design is important.
Speaking of lossless compression of video creation using H.264 High 4:4:4
Predictive Profile, how do you think a video is being created?
It?s created from individual (presumably lossless) images encoded in an
absolute color space (presumably sRGB). In fact, you can do that with ffmpeg
alone.
The simple command is something like this:
ffmpeg -r "24" -i "frames\f_%%1d.png" -vcodec "libx264" -crf "0"
"output.mkv"
Here the lossless h.264 video is created from individual frames, which are
PNG files with the filenames of ?f_1.png, f_2.png??
Of course, for this operation to work, the image files have to share the
same resolution and bit depth.
If color space conversion would somehow cause the loss of color information,
such an encoding process should not be called ?lossless?.
Besides, mathematically, the affine map which I used to derive the inverse
affine map does not show any loss of information. Mathematically, when a
rounding occurs; or the function itself is not one-to-one; it?s generally
not possible find the inverse function; that?s what we call irreversible
conversion. That?s why trigonometric function or exponential function,
strictly speaking, does not have inverse functions. When they are defined in
terms of complex numbers, there are simply no ways to know what the original
input number is. One way for such ?inverse? to work is to simply ?choose?
the one of the many possibilities; again that option does not necessarily
equal what input is; that?s why we specify a range of possible values for
such ?inverse? function, instead of the whole field of complex numbers.
Based on well-defined mapping, the color information in your videos, despite
being encoded in YCbCr, is defined by sRGB color space in your Windows PC.
Keep that in mind. Although re-encoding to YCbCr from sRGB is commonly
called ?color space conversion? or ?pixel format conversion?, it?s entirely
different from ?absolute color space conversion?; where the colors
themselves are being defined differently. For example, in Adobe RGB, the set
of possible colors is different from what is being defined by sRGB; if a
?color space conversion? occurs between them, of course there would be
noticeably difference of what we would see in terms of ?perceptual colors?
after the encoding is done. But YCbCr does not define ?colors?. It?s up to
the codec implementation to see if their design and mapping works
flawlessly. It?s also very common to call sRGB ?4:4:4?; that is for a
reason. If the mathematical operation being used is buggy and the codec
design is not good, there is no denying that re-encoding the information
itself would lead to information loss.




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From lulebo at gmail.com  Thu Aug  1 14:06:56 2013
From: lulebo at gmail.com (Carl Lindqvist)
Date: Thu, 1 Aug 2013 14:06:56 +0200
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375354827157-4660382.post@n4.nabble.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375354827157-4660382.post@n4.nabble.com>
Message-ID: <CAJ6A8t+_0TFmwV+vkzcAB6CU22-fYTQ5N8+6+bZH=yLDi296ew@mail.gmail.com>

It's quite easy to see that no YCbCr 444 can't be one to one with RGB,
even when the full values of Y (0-255 instead of 16-235) are used.

Think of a Y value of 0. Cb and Cr can have 256x256=65536 different
combination of values, but they all represent the same black color in
RGB: 0,0,0

So there MUST be some loss of color information when converting between the two.

From jiachielee at live.com  Thu Aug  1 14:19:50 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 05:19:50 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <CAJ6A8t+_0TFmwV+vkzcAB6CU22-fYTQ5N8+6+bZH=yLDi296ew@mail.gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375354827157-4660382.post@n4.nabble.com>
	<CAJ6A8t+_0TFmwV+vkzcAB6CU22-fYTQ5N8+6+bZH=yLDi296ew@mail.gmail.com>
Message-ID: <1375359590166-4660385.post@n4.nabble.com>

Carl Lindqvist wrote
> It's quite easy to see that no YCbCr 444 can't be one to one with RGB,
> even when the full values of Y (0-255 instead of 16-235) are used.
> 
> Think of a Y value of 0. Cb and Cr can have 256x256=65536 different
> combination of values, but they all represent the same black color in
> RGB: 0,0,0
> 
> So there MUST be some loss of color information when converting between
> the two.
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

What you said sounds reasonable, except that you forgot about one thing: you
cannot simply assign a real number and tried to use the inverse
transformation matrix on YCbCr. The operation is only well-defined for a set
of 16,777,216 input values. As I said before, the YCbCr itself is defined as
an affine map from an absolute color space, not necessarily sRGB, but I
always cite sRGB as an example.
If you want to which numbers are valid for such operation, you have to
transform from sRGB ? YCbCr first, after getting the output value; you can
then use the output value as input value and apply the inverse
transformation on it, of course what you got is the original input value
from the first operation, assuming that your calculation is accurate enough
to avoid rounding-off error problem.
The set of valid input values is not define as equal to the set of positive
integers, for your information. I?m using 24-bit color space as an example
in this case. But you can easily see that the argument is still valid for
other bit depths, such as 30 bits. Only if the bit depth is larger, the
number of valid input values also increases accordingly.



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From jiachielee at live.com  Thu Aug  1 14:36:12 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 05:36:12 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <CAJ6A8t+_0TFmwV+vkzcAB6CU22-fYTQ5N8+6+bZH=yLDi296ew@mail.gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375354827157-4660382.post@n4.nabble.com>
	<CAJ6A8t+_0TFmwV+vkzcAB6CU22-fYTQ5N8+6+bZH=yLDi296ew@mail.gmail.com>
Message-ID: <1375360572796-4660386.post@n4.nabble.com>

There is also another mistake you made. I said previously, you cannot know
colors from YCbCr alone without mapping from an absolute color space;
because YCbCr does not define colors. Say, if you have three (valid) values
from YCbCr representations, instead of trying to guess what color it might
be, why don?t you simply apply the inverse transformation matrix on it to
see what are their corresponding sRGB representation? As the colors are all
defined in absolute color space, which sRGB is one of the most popular.



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From lulebo at gmail.com  Thu Aug  1 15:03:39 2013
From: lulebo at gmail.com (Carl Lindqvist)
Date: Thu, 1 Aug 2013 15:03:39 +0200
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375359590166-4660385.post@n4.nabble.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375354827157-4660382.post@n4.nabble.com>
	<CAJ6A8t+_0TFmwV+vkzcAB6CU22-fYTQ5N8+6+bZH=yLDi296ew@mail.gmail.com>
	<1375359590166-4660385.post@n4.nabble.com>
Message-ID: <CAJ6A8tLPwspE9xghUQZVRyh0vTheJ1XefgHu+NfwXkTHdRNCyw@mail.gmail.com>

For the purpose of this thread my argument stands. It shows that there
can not be a 1:1 mapping between 24 bit RGB and 8 bit (24 bit) YCbCr
color spaces. If you have an RGB image, convert it to YCbCr, you can
not represent all possible RGB values that were originally there. (For
example one huge 16,7 megapixel image that happened to include every
color in RGB)

If we all agree that 24 bit RGB has 16777216 possible colors, and that
8 bit YCbCr also has 16777216 possible colors, then what I said must
be true. Because I have just shown that at least 65536 of the possible
colors in YCbCr all represent the same color in RGB. That means that a
conversion must lose colors.

From jiachielee at live.com  Thu Aug  1 14:09:26 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 05:09:26 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51FA347A.6080103@gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
Message-ID: <1375358966168-4660384.post@n4.nabble.com>

Andy Furniss-2 wrote
> H264 is talking about lossless where input is and output are the same 
> 444 - Y'CbCr - there is no conversion by H264 from RGB as your PNG 
> example would need - that would be done prior to H264 by ffmpeg.
> 
> 
> 
> 
> 
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


Speaking of ?YCbCr ?YCbCr? for H.264 lossless encoding, you obviously did
not read the H.264 

specifications Ed 8.0, did you?
Let me quote directly from H.264 Specifications,
?The source and decoded pictures (frames or fields) are each comprised of
one or more sample arrays:
? Luma (Y) only (monochrome), with or without an auxiliary array.
? Luma and two Chroma (YCbCr or YCgCo), with or without an auxiliary array.
? Green, Blue and Red (GBR, also known as RGB), with or without an auxiliary
array.
? Arrays representing other unspecified monochrome or tri-stimulus colour
samplings (for example, YZX,
also known as XYZ), with or without an auxiliary array.?
That is at Page 42.
On Page 25, I quote
?With the exception of the transform bypass mode of operation for lossless
coding in the High 4:4:4 

Intra, CAVLC 4:4:4 Intra, and High 4:4:4 Predictive profiles, and the I_PCM
mode of operation in all profiles, the algorithm is typically not lossless,
as the exact source sample values are typically not preserved through the
encoding and decoding processes.?

Also look at Page 408 at Table E-3 - Color Primaries 

?The same 444?? Yeah, considering that sRGB is also regarded as 444
correctly.

H.264 specification explicitly says that the source is image encoded in
sRGB. Anything to you want say to response?



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From nisar.med at gmail.com  Thu Aug  1 14:58:58 2013
From: nisar.med at gmail.com (nisar)
Date: Thu, 1 Aug 2013 05:58:58 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec adts/aac decoder
Message-ID: <1375361938390-4660387.post@n4.nabble.com>

Hello,

I am trying to decode adts/aac packets using libavcodec. the input format is 
sample rate: 48000
2 channels
256kb/sec

Each packet I send for decode is around 684 bytes long and the first packet
is usually 20 bytes long.

The decoded packet is around 8192 bytes and sample output format is
AV_SAMPLE_FMT_FLTP. When I send this data for playout using SDL I can hear
noise in the output

SDL_AudioSpec desiredSpec;

    desiredSpec.freq = 48000;
    desiredSpec.format = AUDIO_F32SYS;
    desiredSpec.channels = 2;
    desiredSpec.samples = 2048;
    desiredSpec.callback = audio_callback;
    desiredSpec.userdata = this;

I need to know why am I getting noise in the sound output and how to fix it.

Thanks
Nisar



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From jiachielee at live.com  Thu Aug  1 15:20:43 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 06:20:43 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <CAJ6A8tLPwspE9xghUQZVRyh0vTheJ1XefgHu+NfwXkTHdRNCyw@mail.gmail.com>
References: <1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375354827157-4660382.post@n4.nabble.com>
	<CAJ6A8t+_0TFmwV+vkzcAB6CU22-fYTQ5N8+6+bZH=yLDi296ew@mail.gmail.com>
	<1375359590166-4660385.post@n4.nabble.com>
	<CAJ6A8tLPwspE9xghUQZVRyh0vTheJ1XefgHu+NfwXkTHdRNCyw@mail.gmail.com>
Message-ID: <1375363243963-4660389.post@n4.nabble.com>

No, you haven?t. 
?Think of a Y value of 0. Cb and Cr can have 256x256=65536 different
combination of values, but they all represent the same black color in RGB:
0,0,0?
This argument is wrong. Plain and simple.
First of all, I do not know what ?Y=0? would represent. It is not
necessarily black. Each color in YCbCr is represented by three numbers. I
think you got this idea because of you think of ?Y? as luma component, if
there is ?no? luma information, it must be black, right? That is the
misunderstanding. As a matter of fact, I explicitly told you that you only
know which three numbers from YCbCr represent from a well-defined mapping
from an absolute color space.
By the way, ?Y? is called luma component not for the reason you think it is.
That is, for grayscale image, both Cr and Cr would represented by two unique
but maybe different numbers, only ?Y? component is allowed to have different
numbers, which is 256 in total. That would be 256?1?1=256.
No, I do not claim that we can know which color YCbCr represents without
mapping. It?s just that YCbCr affine mapping (of different conversion
standards) is defined specifically to behave in such manner.
To answer your question directly, if Cb and Cr changes but Y remains
unchanged, there would be different colors, not ?black? as you claimed; and
I don?t even suggest ?Y=0? leads to black.
In short, yes, in 24-bit YCbCr, there would be exactly 16,777,216 number of
colors




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From nlewis at crawford.com  Thu Aug  1 15:53:02 2013
From: nlewis at crawford.com (Nathan Lewis)
Date: Thu, 1 Aug 2013 09:53:02 -0400
Subject: [FFmpeg-user] Offlist,
 Is mixing of audio channels happening by default in ffmpeg?
In-Reply-To: <loom.20130801T001127-852@post.gmane.org>
References: <51E7E8A6.8070906@ringsidecreative.com>
	<loom.20130718T151116-7@post.gmane.org>
	<51E7ED02.5080707@ringsidecreative.com>
	<loom.20130718T153016-321@post.gmane.org>
	<51E7EF4E.8020100@ringsidecreative.com>
	<loom.20130718T154053-475@post.gmane.org>
	<0EF41E1BE75A4DBB80A1E74873E4DC91@HPKANTOOR>
	<loom.20130719T101328-635@post.gmane.org>
	<EA3514503FCB4A868E0BB71A67D7A8A7@HPKANTOOR>
	<loom.20130719T120527-55@post.gmane.org>
	<CACegdbJ3EsCjRZZGnPa0VXFdwjagM2wX7fAAtPLzNVCfa6RFyg@mail.gmail.com>
	<loom.20130801T001127-852@post.gmane.org>
Message-ID: <CACegdbLGON3dQ-Ce4wkZtb0grf6Vrbv96D7_HOsMbFr3ZX02mQ@mail.gmail.com>

On Wed, Jul 31, 2013 at 6:12 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> No, please test:
> avio_wb24(pb, 0xF);
>
> Thank you, Carl Eugen


That does not  have any effect on the output file.  In QT Player, Sound
Track 1 is the only one enabled.   I used this command:

 ./ffmpeg -i A.mov -c:v copy -c:a copy -map 0:1 -map 0:2 -map 0:3 -map 0:4
-map 0:5 -ss 120 -t 60 -r 30000/1001 B.mov


ffmpeg version N-55126-g69fe25c Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug  1 2013 09:35:41 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
  configuration: --enable-libopenjpeg --enable-nonfree --enable-gpl
--enable-libfaac --enable-libmp3lame --enable-libvpx --enable-pthreads
--enable-libx264 --enable-version3
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.2 : mono
Guessed Channel Layout for  Input Stream #0.3 : mono
Guessed Channel Layout for  Input Stream #0.4 : mono
Guessed Channel Layout for  Input Stream #0.5 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'A.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537134592
    compatible_brands: qt
    creation_time   : 2012-10-18 09:59:25
  Duration: 00:25:20.72, start: 0.000000, bitrate: 45253 kb/s
    Stream #0:0(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:1(eng): Video: prores (apcn / 0x6E637061), yuv422p10le,
720x486, 42159 kb/s, SAR 8:9 DAR 320:243, 29.97 fps, 29.97 tbr, 30k tbn,
30k tbc
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:2(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:3(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:4(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:5(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:6(eng): Subtitle: eia_608 (c608 / 0x38303663)
File 'B.mov' already exists. Overwrite ? [y/N] y
Output #0, mov, to 'B.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537134592
    compatible_brands: qt
    encoder         : Lavf55.12.102
    Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le,
720x486 [SAR 8:9 DAR 320:243], q=2-31, 42159 kb/s, 29.97 fps, 30k tbn,
29.97 tbc
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:1(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:2(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:3(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:4(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
Stream mapping:
  Stream #0:1 -> #0:0 (copy)
  Stream #0:2 -> #0:1 (copy)
  Stream #0:3 -> #0:2 (copy)
  Stream #0:4 -> #0:3 (copy)
  Stream #0:5 -> #0:4 (copy)
Press [q] to stop, [?] for help
    Last message repeated 23 times     0kB time=00:00:00.00 bitrate=N/A
frame= 1798 fps= 44 q=-1.0 Lsize=  334449kB time=00:01:00.01
bitrate=45653.5kbits/s
video:311882kB audio:22498kB subtitle:0 global headers:0kB muxing overhead
0.020677%

From cehoyos at ag.or.at  Thu Aug  1 16:04:52 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 1 Aug 2013 14:04:52 +0000 (UTC)
Subject: [FFmpeg-user] libavcodec adts/aac decoder
References: <1375361938390-4660387.post@n4.nabble.com>
Message-ID: <loom.20130801T160344-712@post.gmane.org>

nisar <nisar.med <at> gmail.com> writes:

> sample output format is AV_SAMPLE_FMT_FLTP

>     desiredSpec.format = AUDIO_F32SYS;

Did you run data through libswresample or the 
aresample filter in-between to convert from 
FLTP to FLT?

Your question may be better suited for the 
libav-user mailing list, Carl Eugen


From cehoyos at ag.or.at  Thu Aug  1 16:06:36 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 1 Aug 2013 14:06:36 +0000 (UTC)
Subject: [FFmpeg-user] Offlist,
	Is mixing of audio channels happening by default in ffmpeg?
References: <51E7E8A6.8070906@ringsidecreative.com>
	<loom.20130718T151116-7@post.gmane.org>
	<51E7ED02.5080707@ringsidecreative.com>
	<loom.20130718T153016-321@post.gmane.org>
	<51E7EF4E.8020100@ringsidecreative.com>
	<loom.20130718T154053-475@post.gmane.org>
	<0EF41E1BE75A4DBB80A1E74873E4DC91@HPKANTOOR>
	<loom.20130719T101328-635@post.gmane.org>
	<EA3514503FCB4A868E0BB71A67D7A8A7@HPKANTOOR>
	<loom.20130719T120527-55@post.gmane.org>
	<CACegdbJ3EsCjRZZGnPa0VXFdwjagM2wX7fAAtPLzNVCfa6RFyg@mail.gmail.com>
	<loom.20130801T001127-852@post.gmane.org>
	<CACegdbLGON3dQ-Ce4wkZtb0grf6Vrbv96D7_HOsMbFr3ZX02mQ@mail.gmail.com>
Message-ID: <loom.20130801T160519-989@post.gmane.org>

Nathan Lewis <nlewis <at> crawford.com> writes:

> On Wed, Jul 31, 2013 at 6:12 PM, Carl Eugen Hoyos wrote:
> 
> > No, please test:
> > avio_wb24(pb, 0xF);
> >
> > Thank you, Carl Eugen
> 
> That does not have any effect on the output file.

Do you mean the output files are binary identical 
no matter if you use the original line or 
avio_wb24(pb, 0xF);
?

Alternatively, you could test (unchanged) 454c5d1

Carl Eugen


From adf.lists at gmail.com  Thu Aug  1 16:50:27 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Thu, 01 Aug 2013 15:50:27 +0100
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375358966168-4660384.post@n4.nabble.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
Message-ID: <51FA75B3.2020204@gmail.com>

thljcl wrote:
> Andy Furniss-2 wrote
>> H264 is talking about lossless where input is and output are the same
>> 444 - Y'CbCr - there is no conversion by H264 from RGB as your PNG
>> example would need - that would be done prior to H264 by ffmpeg.
>>
>>
>>
>>
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>
>> ffmpeg-user@
>
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>
> Speaking of ?YCbCr ?YCbCr? for H.264 lossless encoding, you obviously did
> not read the H.264
>
> specifications Ed 8.0, did you?
> Let me quote directly from H.264 Specifications,
> ?The source and decoded pictures (frames or fields) are each comprised of
> one or more sample arrays:
> ? Luma (Y) only (monochrome), with or without an auxiliary array.
> ? Luma and two Chroma (YCbCr or YCgCo), with or without an auxiliary array.
> ? Green, Blue and Red (GBR, also known as RGB), with or without an auxiliary
> array.
> ? Arrays representing other unspecified monochrome or tri-stimulus colour
> samplings (for example, YZX,
> also known as XYZ), with or without an auxiliary array.?
> That is at Page 42.
> On Page 25, I quote
> ?With the exception of the transform bypass mode of operation for lossless
> coding in the High 4:4:4
>
> Intra, CAVLC 4:4:4 Intra, and High 4:4:4 Predictive profiles, and the I_PCM
> mode of operation in all profiles, the algorithm is typically not lossless,
> as the exact source sample values are typically not preserved through the
> encoding and decoding processes.?
>
> Also look at Page 408 at Table E-3 - Color Primaries
>
> ?The same 444?? Yeah, considering that sRGB is also regarded as 444
> correctly.
>
> H.264 specification explicitly says that the source is image encoded in
> sRGB. Anything to you want say to response?

Yes, it does not say source is encoded sRGB it says it can be - it can 
be other things as well.

If the three planes represent RGB and are encoded losslessly then after 
decoding you will have your RGB back.

There is nothing in what you quote to say that CSC to Y'CbCr then back 
to RGB would be involved.

A quick search will show you that "transform bypass mode" is nothing to 
do with CSC.




From jiachielee at live.com  Thu Aug  1 17:01:34 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 08:01:34 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51FA75B3.2020204@gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
	<51FA75B3.2020204@gmail.com>
Message-ID: <1375369292433-4660394.post@n4.nabble.com>

Andy Furniss-2 wrote
> thljcl wrote:
>> Andy Furniss-2 wrote
>>> H264 is talking about lossless where input is and output are the same
>>> 444 - Y'CbCr - there is no conversion by H264 from RGB as your PNG
>>> example would need - that would be done prior to H264 by ffmpeg.
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> ffmpeg-user mailing list
>>
>>> ffmpeg-user@
>>
>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>>
>> Speaking of ?YCbCr ?YCbCr? for H.264 lossless encoding, you obviously did
>> not read the H.264
>>
>> specifications Ed 8.0, did you?
>> Let me quote directly from H.264 Specifications,
>> ?The source and decoded pictures (frames or fields) are each comprised of
>> one or more sample arrays:
>> ? Luma (Y) only (monochrome), with or without an auxiliary array.
>> ? Luma and two Chroma (YCbCr or YCgCo), with or without an auxiliary
>> array.
>> ? Green, Blue and Red (GBR, also known as RGB), with or without an
>> auxiliary
>> array.
>> ? Arrays representing other unspecified monochrome or tri-stimulus colour
>> samplings (for example, YZX,
>> also known as XYZ), with or without an auxiliary array.?
>> That is at Page 42.
>> On Page 25, I quote
>> ?With the exception of the transform bypass mode of operation for
>> lossless
>> coding in the High 4:4:4
>>
>> Intra, CAVLC 4:4:4 Intra, and High 4:4:4 Predictive profiles, and the
>> I_PCM
>> mode of operation in all profiles, the algorithm is typically not
>> lossless,
>> as the exact source sample values are typically not preserved through the
>> encoding and decoding processes.?
>>
>> Also look at Page 408 at Table E-3 - Color Primaries
>>
>> ?The same 444?? Yeah, considering that sRGB is also regarded as 444
>> correctly.
>>
>> H.264 specification explicitly says that the source is image encoded in
>> sRGB. Anything to you want say to response?
> 
> Yes, it does not say source is encoded sRGB it says it can be - it can 
> be other things as well.
> 
> If the three planes represent RGB and are encoded losslessly then after 
> decoding you will have your RGB back.
> 
> There is nothing in what you quote to say that CSC to Y'CbCr then back 
> to RGB would be involved.
> 
> A quick search will show you that "transform bypass mode" is nothing to 
> do with CSC.
> 
> 
> 
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

You are missing the point. The H.264 specifications do not attempt to
explain some of basic notions such as ?absolute color space? and ?relative
color space?. I?ve repeatedly told you about it. What H.264 specifications
explicitly said that one of the supported sources is the image encoded in
RGB and it does support lossless coding in three profiles, one of them is
called ?High 4:4:4 Predictive? which is used by x264.
The fact that encoded information is required to be decoded is based on the
how ?absolute color space? and ?relative color space? are related. But H.264
Specification does explicitly say that the video is to be decoded to RGB.
Perhaps you should read what is written at
http://en.wikipedia.org/wiki/Absolute_color_space
Or, if you can afford it, just buy the books which are cited as sources by
Wikipedia, which are Java 2D graphics, written by Jonathan Knudsen, and
Human Vision and Electronic Imaging XII. SPIE. ISBN 0-8194-6605-0.




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From nlewis at crawford.com  Thu Aug  1 17:39:22 2013
From: nlewis at crawford.com (Nathan Lewis)
Date: Thu, 1 Aug 2013 11:39:22 -0400
Subject: [FFmpeg-user] Offlist,
 Is mixing of audio channels happening by default in ffmpeg?
In-Reply-To: <loom.20130801T160519-989@post.gmane.org>
References: <51E7E8A6.8070906@ringsidecreative.com>
	<loom.20130718T151116-7@post.gmane.org>
	<51E7ED02.5080707@ringsidecreative.com>
	<loom.20130718T153016-321@post.gmane.org>
	<51E7EF4E.8020100@ringsidecreative.com>
	<loom.20130718T154053-475@post.gmane.org>
	<0EF41E1BE75A4DBB80A1E74873E4DC91@HPKANTOOR>
	<loom.20130719T101328-635@post.gmane.org>
	<EA3514503FCB4A868E0BB71A67D7A8A7@HPKANTOOR>
	<loom.20130719T120527-55@post.gmane.org>
	<CACegdbJ3EsCjRZZGnPa0VXFdwjagM2wX7fAAtPLzNVCfa6RFyg@mail.gmail.com>
	<loom.20130801T001127-852@post.gmane.org>
	<CACegdbLGON3dQ-Ce4wkZtb0grf6Vrbv96D7_HOsMbFr3ZX02mQ@mail.gmail.com>
	<loom.20130801T160519-989@post.gmane.org>
Message-ID: <CACegdbJT=hRS+4bPoZvgXm+iea5OtyjQSmJL6RiD2Und8VdpLQ@mail.gmail.com>

On Thu, Aug 1, 2013 at 10:06 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Do you mean the output files are binary identical
> no matter if you use the original line or
> avio_wb24(pb, 0xF);
> ?
>

Sorry, I didn't.  I meant that no matter what line I used the resultant
output files had the same audio configuration with only channel one
enabled.  Channels 2-4 were disabled.  As for the files being binary
identical md5 returns different checksums.


> Alternatively, you could test (unchanged) 454c5d1


 This too results in a file with only channel 1 (QT player "Sound Track 1")
enabled.  Channels 2-4 are disabled.

# ./ffmpeg -i A.mov -c:v copy -c:a copy -map 0:1 -map 0:2 -map 0:3 -map 0:4
-map 0:5 -ss 120 -t 60 -r 30000/1001 C_4545cd1.mov
ffmpeg version 1.1.git-454c5d1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  1 2013 11:30:46 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
  configuration: --enable-libopenjpeg --enable-nonfree --enable-gpl
--enable-libfaac --enable-libmp3lame --enable-libvpx --enable-pthreads
--enable-libx264 --enable-version3
  libavutil      52. 17.103 / 52. 17.103
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.102 / 54. 63.102
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 41.100 /  3. 41.100
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
Guessed Channel Layout for  Input Stream #0.2 : mono
Guessed Channel Layout for  Input Stream #0.3 : mono
Guessed Channel Layout for  Input Stream #0.4 : mono
Guessed Channel Layout for  Input Stream #0.5 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'A.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537134592
    compatible_brands: qt
    creation_time   : 2012-10-18 09:59:25
  Duration: 00:25:20.72, start: 0.000000, bitrate: 45253 kb/s
    Stream #0:0(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:1(eng): Video: prores (apcn / 0x6E637061), yuv422p10le,
720x486, 42159 kb/s, SAR 8:9 DAR 320:243, 29.97 fps, 29.97 tbr, 30k tbn,
30k tbc
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:2(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:3(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:4(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:5(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
s16, 768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:6(eng): Subtitle: eia_608 (c608 / 0x38303663)
Output #0, mov, to 'C_4545cd1.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537134592
    compatible_brands: qt
    encoder         : Lavf54.63.102
    Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le,
720x486 [SAR 8:9 DAR 320:243], q=2-31, 42159 kb/s, 29.97 fps, 30k tbn,
29.97 tbc
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:1(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:2(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:3(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
    Stream #0:4(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, mono,
768 kb/s
    Metadata:
      timecode        : 01:25:05;00
Stream mapping:
  Stream #0:1 -> #0:0 (copy)
  Stream #0:2 -> #0:1 (copy)
  Stream #0:3 -> #0:2 (copy)
  Stream #0:4 -> #0:3 (copy)
  Stream #0:5 -> #0:4 (copy)
Press [q] to stop, [?] for help
frame= 1798 fps= 50 q=-1.0 Lsize=  334449kB time=00:01:00.01
bitrate=45653.5kbits/s
video:311882kB audio:22498kB subtitle:0 global headers:0kB muxing overhead
0.020677%

From nisar.med at gmail.com  Thu Aug  1 18:07:35 2013
From: nisar.med at gmail.com (nisar)
Date: Thu, 1 Aug 2013 09:07:35 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec adts/aac decoder
In-Reply-To: <loom.20130801T160344-712@post.gmane.org>
References: <1375361938390-4660387.post@n4.nabble.com>
	<loom.20130801T160344-712@post.gmane.org>
Message-ID: <CAL1+ejgyTD7BSGdM3SHY0rqgccq1HxOS22HHHpJq+1iHH35mWQ@mail.gmail.com>

Thank you for you response.

The sound is better but the rate is slow and it seems that there is some of
the samples in the output are missing (silence) each frame. I think there
is some error in my calculations please help


On Thu, Aug 1, 2013 at 7:07 PM, Carl Eugen Hoyos [via FFmpeg-users] <
ml-node+s933282n4660391h40 at n4.nabble.com> wrote:

> nisar <nisar.med <at> gmail.com> writes:
>
> > sample output format is AV_SAMPLE_FMT_FLTP
>
> >     desiredSpec.format = AUDIO_F32SYS;
>
> Did you run data through libswresample or the
> aresample filter in-between to convert from
> FLTP to FLT?
>
> Your question may be better suited for the
> libav-user mailing list, Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=4660391&i=0>
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>
> ------------------------------
>  If you reply to this email, your message will be added to the discussion
> below:
>
> http://ffmpeg-users.933282.n4.nabble.com/libavcodec-adts-aac-decoder-tp4660387p4660391.html
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From nisar.med at gmail.com  Thu Aug  1 18:29:42 2013
From: nisar.med at gmail.com (Nisar Ahmed)
Date: Thu, 1 Aug 2013 21:29:42 +0500
Subject: [FFmpeg-user] libavcodec adts/aac decoder
In-Reply-To: <CAL1+ejgyTD7BSGdM3SHY0rqgccq1HxOS22HHHpJq+1iHH35mWQ@mail.gmail.com>
References: <1375361938390-4660387.post@n4.nabble.com>
	<loom.20130801T160344-712@post.gmane.org>
	<CAL1+ejgyTD7BSGdM3SHY0rqgccq1HxOS22HHHpJq+1iHH35mWQ@mail.gmail.com>
Message-ID: <CAL1+ejiOcBAZ5QK6CLPwYUerj1xAwWe1XDYaCoTUnh9cLt+9eg@mail.gmail.com>

Ok, I changed samples parameter in SDL_AudioSpec to 1024, and multiplied
length of buffer returned in the callback by 2. It took me a while to
understand SDL_AudioSpec properly

Thank you for you help
Nisar


On Thu, Aug 1, 2013 at 9:07 PM, nisar <nisar.med at gmail.com> wrote:

> Thank you for you response.
>
> The sound is better but the rate is slow and it seems that there is some of
> the samples in the output are missing (silence) each frame. I think there
> is some error in my calculations please help
>
>
> On Thu, Aug 1, 2013 at 7:07 PM, Carl Eugen Hoyos [via FFmpeg-users] <
> ml-node+s933282n4660391h40 at n4.nabble.com> wrote:
>
> > nisar <nisar.med <at> gmail.com> writes:
> >
> > > sample output format is AV_SAMPLE_FMT_FLTP
> >
> > >     desiredSpec.format = AUDIO_F32SYS;
> >
> > Did you run data through libswresample or the
> > aresample filter in-between to convert from
> > FLTP to FLT?
> >
> > Your question may be better suited for the
> > libav-user mailing list, Carl Eugen
> >
> > _______________________________________________
> > ffmpeg-user mailing list
> > [hidden email] <http://user/SendEmail.jtp?type=node&node=4660391&i=0>
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> >
> > ------------------------------
> >  If you reply to this email, your message will be added to the discussion
> > below:
> >
> >
> http://ffmpeg-users.933282.n4.nabble.com/libavcodec-adts-aac-decoder-tp4660387p4660391.html
> >  To unsubscribe from libavcodec adts/aac decoder, click here<
> http://ffmpeg-users.933282.n4.nabble.com/template/NamlServlet.jtp?macro=unsubscribe_by_code&node=4660387&code=bmlzYXIubWVkQGdtYWlsLmNvbXw0NjYwMzg3fDQxNzA0OTk1Ng==
> >
> > .
> > NAML<
> http://ffmpeg-users.933282.n4.nabble.com/template/NamlServlet.jtp?macro=macro_viewer&id=instant_html%21nabble%3Aemail.naml&base=nabble.naml.namespaces.BasicNamespace-nabble.view.web.template.NabbleNamespace-nabble.view.web.template.NodeNamespace&breadcrumbs=notify_subscribers%21nabble%3Aemail.naml-instant_emails%21nabble%3Aemail.naml-send_instant_email%21nabble%3Aemail.naml
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> >
>
>
>
>
> --
> View this message in context:
> http://ffmpeg-users.933282.n4.nabble.com/libavcodec-adts-aac-decoder-tp4660387p4660396.html
> Sent from the FFmpeg-users mailing list archive at Nabble.com.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From jiachielee at live.com  Thu Aug  1 18:49:41 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 09:49:41 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51FA75B3.2020204@gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
	<51FA75B3.2020204@gmail.com>
Message-ID: <1375375781673-4660398.post@n4.nabble.com>

Look at Page 42 of the H.264 Specification Ed 8.0

?For convenience of notation and terminology in this Specification, the
variables and terms associated with these arrays are referred to as luma (or
L or Y) and chroma, where the two chroma arrays are referred to as Cb and
Cr; regardless of the actual color representation method in use. The actual
color representation method in use can be indicated in syntax that is
specified in Annex E. The (monochrome) auxiliary arrays, which may or may
not be present as auxiliary pictures in a coded video sequence, are optional
for decoding and can be used for such purposes as alpha blending.?

It?s asking you to look at Annex E. So go to Annex E, Look at table E-3.

It specifically mentioned sRGB.

Of course, there are many RGB-based color spaces being mentioned, with sRGB
being one of them. Either way, it just shows that YCbCr is just a "relative
color space".



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Does-converting-to-yuv444p-by-default-make-sense-tp4660219p4660398.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From bhavesh.vthink at gmail.com  Thu Aug  1 15:23:50 2013
From: bhavesh.vthink at gmail.com (bhavesh bodar)
Date: Thu, 1 Aug 2013 18:53:50 +0530
Subject: [FFmpeg-user] ffmpeg help
Message-ID: <CAFXmZcYeWVXxHoBdQHx_N6V9Ox2UD+9z90K6SGTYUYQ6MFCzGQ@mail.gmail.com>

Dear sir,

               i want to merge tow images or video using php with ffmpeg. i
have successfully installed ffmpeg in my localhost wamp server also i can
convert video usigng ffmpeg in php
       but i can not merge two image or video so i want to complete php
code of merge two images or video.

Thanx

From kalpesh.g at quantumtechnolabs.com  Thu Aug  1 17:10:13 2013
From: kalpesh.g at quantumtechnolabs.com (Kalpesh Gohel)
Date: Thu, 1 Aug 2013 20:40:13 +0530
Subject: [FFmpeg-user] arm-linux-androideabi-gcc is unable to create an
	executable file.
Message-ID: <CAJ4Cjsk--4OJXLjY0OWv4LJVn3NQHyEKch2x5xfR4qgc+XC85Q@mail.gmail.com>

Helo,

os : *windows 8*
software : *cygwin *terminal

I am using *latest ffmpeg from git *and *android-ndk-r9*

run following file as describe below

kalpesh at qtm12 /cygdrive/d/ffmpeg-master
$ ./script.sh
Auto-config: --toolchain=arm-linux-androideabi-4.6
Copying prebuilt binaries...
Copying sysroot headers and libraries...
Copying libstdc++ headers and libraries...
Copying files to: /cygdrive/d/tmp/ffmpeg
Cleaning up...
Done.
*arm-linux-androideabi-gcc is unable to create an executable file.*
*C compiler test failed.*

If you think configure made a mistake, make sure you are using the latest
version from Git.  If the latest version fails, report the problem to the
ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net.
Include the log file "config.log" produced by configure as this will help
solving the problem.

--------------------
PLS PLS help us


-- 



Thanks & regards

kalpesh gohel
-------------- next part --------------
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From bobm-ffmpeg at burner.com  Thu Aug  1 20:24:05 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Thu, 01 Aug 2013 12:24:05 -0600
Subject: [FFmpeg-user] uyvy422 psuedo 10-bit?
In-Reply-To: <loom.20130727T121406-38@post.gmane.org>
References: <51F38EF2.9090207@burner.com>
	<loom.20130727T121406-38@post.gmane.org>
Message-ID: <51FAA7C5.1000606@burner.com>

> The additional information may be there for "Avid Packed" 
> (AVup) but a lossless reference (v210?) will be needed to 
> be able to interpret it.

Nevermind.. over the last week I've been learning a lot about Avid's
busted codecs.  The Avid Packed codec apparently is somewhat
"special-use" from Media Composer, my test cases were invalid.

Bob

From cehoyos at ag.or.at  Thu Aug  1 20:56:46 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 1 Aug 2013 18:56:46 +0000 (UTC)
Subject: [FFmpeg-user] arm-linux-androideabi-gcc is unable to create an
	executable file.
References: <CAJ4Cjsk--4OJXLjY0OWv4LJVn3NQHyEKch2x5xfR4qgc+XC85Q@mail.gmail.com>
Message-ID: <loom.20130801T205507-144@post.gmane.org>

Kalpesh Gohel <kalpesh.g <at> quantumtechnolabs.com> writes:

> kalpesh <at> qtm12 /cygdrive/d/ffmpeg-master
> $ ./script.sh

Only running FFmpeg's configure script is 
supported on this mailing list, external 
scripts are not recommended (and not 
supported).

Carl Eugen


From cehoyos at ag.or.at  Thu Aug  1 20:58:13 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 1 Aug 2013 18:58:13 +0000 (UTC)
Subject: [FFmpeg-user] uyvy422 psuedo 10-bit?
References: <51F38EF2.9090207@burner.com>
	<loom.20130727T121406-38@post.gmane.org>
	<51FAA7C5.1000606@burner.com>
Message-ID: <loom.20130801T205712-16@post.gmane.org>

Bob Maple <bobm-ffmpeg <at> burner.com> writes:

> > The additional information may be there for "Avid Packed" 
> > (AVup) but a lossless reference (v210?) will be needed to 
> > be able to interpret it.
> 
> Nevermind.. over the last week I've been learning a lot 
> about Avid's busted codecs.

Could you elaborate?
No matter how busted they are, it would be interesting to 
completely support them...

Thank you, Carl Eugen


From barsnick at gmx.net  Thu Aug  1 21:01:14 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Thu, 1 Aug 2013 21:01:14 +0200
Subject: [FFmpeg-user] arm-linux-androideabi-gcc is unable to create an
 executable file.
In-Reply-To: <CAJ4Cjsk--4OJXLjY0OWv4LJVn3NQHyEKch2x5xfR4qgc+XC85Q@mail.gmail.com>
References: <CAJ4Cjsk--4OJXLjY0OWv4LJVn3NQHyEKch2x5xfR4qgc+XC85Q@mail.gmail.com>
Message-ID: <20130801190114.GA27696@sunshine.barsnick.net>

Kalpesh,
did you read this properly?:

On Thu, Aug 01, 2013 at 20:40:13 +0530, Kalpesh Gohel wrote:
> Include the log file "config.log" produced by configure as this will help
> solving the problem.

You didn't attach that file.

Moritz

From bobm-ffmpeg at burner.com  Thu Aug  1 22:20:08 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Thu, 01 Aug 2013 14:20:08 -0600
Subject: [FFmpeg-user] uyvy422 psuedo 10-bit?
In-Reply-To: <loom.20130801T205712-16@post.gmane.org>
References: <51F38EF2.9090207@burner.com>
	<loom.20130727T121406-38@post.gmane.org>
	<51FAA7C5.1000606@burner.com>
	<loom.20130801T205712-16@post.gmane.org>
Message-ID: <51FAC2F8.2040900@burner.com>

On 8/1/2013 12:58 PM, Carl Eugen Hoyos wrote:

> Could you elaborate?
> No matter how busted they are, it would be interesting to 
> completely support them...

Well, Media Composer has an export function called 'Same as Source'
where they jam whatever they want into a Quicktime wrapper.  The 'Avid
Packed' Quicktime codecs for Windows/MacOS are really only meant for
decoding these files, not encoding them, and all the tests I was making
were starting through Quicktime and weren't actually 10-bit data
(although the file sizes looked like they were 10-bit if you did the
math. Nice and confusing.)

So, I need to do some more investigation first to get some actual 10-bit
images in there, and then see if ffmpeg still reads it as 8bit or not.
I'll report back when I have a moment to do some more testing.

Thanks!

Bob

From adf.lists at gmail.com  Thu Aug  1 22:22:53 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Thu, 01 Aug 2013 21:22:53 +0100
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375369292433-4660394.post@n4.nabble.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
	<51FA75B3.2020204@gmail.com>
	<1375369292433-4660394.post@n4.nabble.com>
Message-ID: <51FAC39D.7040404@gmail.com>

thljcl wrote:

> You are missing the point..

Well I may be missing your points, but you have really gone well away 
from the point that Poynton makes and I linked to, which was 
specifically about rgb24 to 8bit Y'CbCr.

It doesn't matter what could be, it matters what is and you are not 
going to be changing the way every decoder/display converts yuv to rgb.

You said initially -

"?yuv444p? is a way of re-encoding the entire 24-bit RGB information"

and went on to say

"To me personally, I kind of like the way ffmpeg handle color space
conversion. Ffmpeg does have issues within its algorithm. YUV444p is not 
one of them"

So do you think that if you made a 4096x4096 rgb24 image using C with 
all the possible variations, that you could, using ffmpeg, convert it to 
yuv444p and back to rgb24 without loss?

Try it - it's trivial in C to generate and also trivial to write 
something that counts the unique pixels.

I've just done it but my C is so bad I may make the programmers on here 
feel ill if I posted it - and it's probably still buggy :-)


From jiachielee at live.com  Thu Aug  1 22:53:16 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 13:53:16 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51FAC39D.7040404@gmail.com>
References: <1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
	<51FA75B3.2020204@gmail.com>
	<1375369292433-4660394.post@n4.nabble.com>
	<51FAC39D.7040404@gmail.com>
Message-ID: <1375390396474-4660407.post@n4.nabble.com>

Andy Furniss-2 wrote
> thljcl wrote:
> 
>> You are missing the point..
> 
> Well I may be missing your points, but you have really gone well away 
> from the point that Poynton makes and I linked to, which was 
> specifically about rgb24 to 8bit Y'CbCr.
> 
> It doesn't matter what could be, it matters what is and you are not 
> going to be changing the way every decoder/display converts yuv to rgb.
> 
> You said initially -
> 
> "?yuv444p? is a way of re-encoding the entire 24-bit RGB information"
> 
> and went on to say
> 
> "To me personally, I kind of like the way ffmpeg handle color space
> conversion. Ffmpeg does have issues within its algorithm. YUV444p is not 
> one of them"
> 
> So do you think that if you made a 4096x4096 rgb24 image using C with 
> all the possible variations, that you could, using ffmpeg, convert it to 
> yuv444p and back to rgb24 without loss?
> 
> Try it - it's trivial in C to generate and also trivial to write 
> something that counts the unique pixels.
> 
> I've just done it but my C is so bad I may make the programmers on here 
> feel ill if I posted it - and it's probably still buggy :-)
> 
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Of course, what I've been saying is the feature of H.264 and what YCbCr
4:4:4 means. Whether or not x264 can actually meet H.264 specification;
whether or not H.264 design is with bugs; that is entirely different issue.
I never once said that I assumed that the encoder itself is without bugs.
What I really said was what it was supposed to be. It?s a question of
theoretical capability and actual implementation.
By the way, did you not read my previous post? YCbCr, as stated by H.264
specification Ed 8.0, is not the actual color representation. YCbCr 4:4:4 is
just another way to say full color information is being encoded. (x264 calls
it yuv444p; anyway the name is not the point here) You can say that there is
?color space conversion? if re-encoding the information with affine map is
what you mean.
This may seem surprising to you. The actual absolute color space currently
supported by H.264 is
1.	Rec. ITU-R BT.709-5
2.	Rec. ITU-R BT.1361
3.	IEC 61966-2-1 (sRGB or sYCC)
4.	IEC 61966-2-4
5.	Society of Motion Picture and Television Engineers RP 177 (1993) Annex B
6.	Rec. ITU-R BT.470-6 System M
7.	United States National Television System Committee 1953 Recommendation
for transmission standards for color television
8.	United States Federal Communications Commission Title 47 Code of Federal
Regulations
9.	Rec. ITU-R BT.470-6 System B, G
10.	Rec. ITU-R BT.601-6 625
11.	Rec. ITU-R BT.1358 625
12.	Rec. ITU-R BT.1700 625 PAL and 625 SECAM
13.	Rec. ITU-R BT.601-6 525
14.	Rec. ITU-R BT.1358 525
15.	Rec. ITU-R BT.1700 NTSC
16.	Society of Motion Picture and Television Engineers 170M (2004)
(functionally the same as the value 7)
17.	Society of Motion Picture and Television Engineers 240M (1999)
(functionally the same as the value 6)
18.	Generic film (color filters using Illuminant C)
19.	Rec. ITU-R BT.2020
This is true as of Ed. 8.0 I?m not sure how much specs of H.264 being
implemented by x264, it?s not full; that?s for sure. It?s not practical to
implement every spec anyway. Let me quote once again from page 42

?For convenience of notation and terminology in this Specification, the
variables and terms associated with these arrays are referred to as luma (or
L or Y) and chroma, where the two chroma arrays are referred to as Cb and
Cr; regardless of the actual color representation method in use. The actual
color representation method in use can be indicated in syntax that is
specified in Annex E. The (monochrome) auxiliary arrays, which may or may
not be present as auxiliary pictures in a coded video sequence, are optional
for decoding and can be used for such purposes as alpha blending.?
In other words, the reported YUV444p is not the actual absolute color space
being used. How to know which one is being used? That?s a good question. The
short answer is I don?t know.
According to H.264 specification, the ?colour_primaries? syntax element is
being used to specify which absolute color space is being used. Regardless
of which absolute color space is being encoded in, the decoder still need to
decode it to sRGB in Windows anyway.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Does-converting-to-yuv444p-by-default-make-sense-tp4660219p4660407.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From cehoyos at ag.or.at  Thu Aug  1 23:15:59 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 1 Aug 2013 21:15:59 +0000 (UTC)
Subject: [FFmpeg-user] uyvy422 psuedo 10-bit?
References: <51F38EF2.9090207@burner.com>
	<loom.20130727T121406-38@post.gmane.org>
	<51FAA7C5.1000606@burner.com>
	<loom.20130801T205712-16@post.gmane.org>
	<51FAC2F8.2040900@burner.com>
Message-ID: <loom.20130801T231530-31@post.gmane.org>

Bob Maple <bobm-ffmpeg <at> burner.com> writes:

> So, I need to do some more investigation first 
> to get some actual 10-bit images in there

Perhaps you can use a v210 sample as input.

Carl Eugen


From onemda at gmail.com  Thu Aug  1 23:51:11 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 1 Aug 2013 21:51:11 +0000
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375390396474-4660407.post@n4.nabble.com>
References: <1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
	<51FA75B3.2020204@gmail.com>
	<1375369292433-4660394.post@n4.nabble.com>
	<51FAC39D.7040404@gmail.com>
	<1375390396474-4660407.post@n4.nabble.com>
Message-ID: <CAPYw7P7ENkEVGUbhgQiqOQQh5+Dy7i4h6TvUZn1Rt+mq4H9-Mg@mail.gmail.com>

On 8/1/13, thljcl <jiachielee at live.com> wrote:
> Andy Furniss-2 wrote
>> thljcl wrote:
>>
>>> You are missing the point..
>>
>> Well I may be missing your points, but you have really gone well away
>> from the point that Poynton makes and I linked to, which was
>> specifically about rgb24 to 8bit Y'CbCr.
>>
>> It doesn't matter what could be, it matters what is and you are not
>> going to be changing the way every decoder/display converts yuv to rgb.
>>
>> You said initially -
>>
>> ""yuv444p" is a way of re-encoding the entire 24-bit RGB information"
>>
>> and went on to say
>>
>> "To me personally, I kind of like the way ffmpeg handle color space
>> conversion. Ffmpeg does have issues within its algorithm. YUV444p is not
>> one of them"
>>
>> So do you think that if you made a 4096x4096 rgb24 image using C with
>> all the possible variations, that you could, using ffmpeg, convert it to
>> yuv444p and back to rgb24 without loss?
>>
>> Try it - it's trivial in C to generate and also trivial to write
>> something that counts the unique pixels.
>>
>> I've just done it but my C is so bad I may make the programmers on here
>> feel ill if I posted it - and it's probably still buggy :-)
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>
>> ffmpeg-user@
>
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> Of course, what I've been saying is the feature of H.264 and what YCbCr
> 4:4:4 means. Whether or not x264 can actually meet H.264 specification;
> whether or not H.264 design is with bugs; that is entirely different issue.
> I never once said that I assumed that the encoder itself is without bugs.
> What I really said was what it was supposed to be. It's a question of
> theoretical capability and actual implementation.
> By the way, did you not read my previous post? YCbCr, as stated by H.264
> specification Ed 8.0, is not the actual color representation. YCbCr 4:4:4 is
> just another way to say full color information is being encoded. (x264 calls
> it yuv444p; anyway the name is not the point here) You can say that there is
> "color space conversion" if re-encoding the information with affine map is
> what you mean.
> This may seem surprising to you. The actual absolute color space currently
> supported by H.264 is
> 1.	Rec. ITU-R BT.709-5
> 2.	Rec. ITU-R BT.1361
> 3.	IEC 61966-2-1 (sRGB or sYCC)
> 4.	IEC 61966-2-4
> 5.	Society of Motion Picture and Television Engineers RP 177 (1993) Annex B
> 6.	Rec. ITU-R BT.470-6 System M
> 7.	United States National Television System Committee 1953 Recommendation
> for transmission standards for color television
> 8.	United States Federal Communications Commission Title 47 Code of Federal
> Regulations
> 9.	Rec. ITU-R BT.470-6 System B, G
> 10.	Rec. ITU-R BT.601-6 625
> 11.	Rec. ITU-R BT.1358 625
> 12.	Rec. ITU-R BT.1700 625 PAL and 625 SECAM
> 13.	Rec. ITU-R BT.601-6 525
> 14.	Rec. ITU-R BT.1358 525
> 15.	Rec. ITU-R BT.1700 NTSC
> 16.	Society of Motion Picture and Television Engineers 170M (2004)
> (functionally the same as the value 7)
> 17.	Society of Motion Picture and Television Engineers 240M (1999)
> (functionally the same as the value 6)
> 18.	Generic film (color filters using Illuminant C)
> 19.	Rec. ITU-R BT.2020
> This is true as of Ed. 8.0 I'm not sure how much specs of H.264 being
> implemented by x264, it's not full; that's for sure. It's not practical to
> implement every spec anyway. Let me quote once again from page 42
>
> "For convenience of notation and terminology in this Specification, the
> variables and terms associated with these arrays are referred to as luma (or
> L or Y) and chroma, where the two chroma arrays are referred to as Cb and
> Cr; regardless of the actual color representation method in use. The actual
> color representation method in use can be indicated in syntax that is
> specified in Annex E. The (monochrome) auxiliary arrays, which may or may
> not be present as auxiliary pictures in a coded video sequence, are optional
> for decoding and can be used for such purposes as alpha blending."
> In other words, the reported YUV444p is not the actual absolute color space
> being used. How to know which one is being used? That's a good question. The
> short answer is I don't know.
> According to H.264 specification, the "colour_primaries" syntax element is
> being used to specify which absolute color space is being used. Regardless
> of which absolute color space is being encoded in, the decoder still need to
> decode it to sRGB in Windows anyway.

OK, so what was original thread about?

From shadowing71 at gmail.com  Fri Aug  2 00:23:31 2013
From: shadowing71 at gmail.com (Young Kim)
Date: Thu, 1 Aug 2013 15:23:31 -0700
Subject: [FFmpeg-user] De-interlacing and Inverse Telecining Questions
Message-ID: <91574A7267144E66BC9820C75A530CB7@gmail.com>

Hello, 

I've been reading through the ffmpeg documentation about deinterlacing and inverse telecining, and it's a bit confusing. In particular, there seems to be a myriad ways of achieving this (i.e. yadif, fieldmatch, and kerndeint). Does anyone happen to know what the difference is among these, and if there is a way to automatically deinterlace / inverse telecine an input accordingly?

Thanks,
Young Kim


From cehoyos at ag.or.at  Fri Aug  2 00:34:29 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 1 Aug 2013 22:34:29 +0000 (UTC)
Subject: [FFmpeg-user] De-interlacing and Inverse Telecining Questions
References: <91574A7267144E66BC9820C75A530CB7@gmail.com>
Message-ID: <loom.20130802T002600-665@post.gmane.org>

Young Kim <shadowing71 <at> gmail.com> writes:

> I've been reading through the ffmpeg documentation 
> about deinterlacing and inverse telecining, and it's
> a bit confusing. In particular, there seems to be a 
> myriad ways of achieving this (i.e. yadif, fieldmatch,
> and kerndeint). Does anyone happen to know what the 
> difference is among these

You first have to understand that while interlaced and 
telecined video look similar they are fundamentally 
different: Deinterlacing means inventing 50% of the 
image where no image was, inverse telecine brings the 
original frames back that were used as input for the 
telecine process.
(Or in other words: The telecine process only duplicates 
some information while interlacing means throwing 
away 50% of the video information - this may even 
happen within camera equipment.)

You should not deinterlace telecined material and you 
cannot inverse telecine interlaced videos.

kerndeint is an old deinterlacer that you should only 
use if you have a specific reason. (performance on arm?)

yadif is a good deinterlacer and has simd optimization 
on x86.

fieldmatch is a modern inverse telecine filter that is 
said to beat MPlayer's pullup filter for every kind of 
source file (but needs more resources).

> and if there is a way to automatically deinterlace / 
> inverse telecine an input accordingly?

Not really because while it is technically possible to 
detect if your video is interlaced or telecined (see 
the idet filter), it is significantly easier and less 
error-prone if you do it (visually) before transcoding.

Carl Eugen


From adf.lists at gmail.com  Fri Aug  2 01:38:29 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Fri, 02 Aug 2013 00:38:29 +0100
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375390396474-4660407.post@n4.nabble.com>
References: <1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
	<1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
	<51FA75B3.2020204@gmail.com>
	<1375369292433-4660394.post@n4.nabble.com>
	<51FAC39D.7040404@gmail.com>
	<1375390396474-4660407.post@n4.nabble.com>
Message-ID: <51FAF175.30803@gmail.com>

thljcl wrote:
> Andy Furniss-2 wrote
>> thljcl wrote:
>>
>>> You are missing the point..
>>
>> Well I may be missing your points, but you have really gone well away
>> from the point that Poynton makes and I linked to, which was
>> specifically about rgb24 to 8bit Y'CbCr.
>>
>> It doesn't matter what could be, it matters what is and you are not
>> going to be changing the way every decoder/display converts yuv to rgb.
>>
>> You said initially -
>>
>> "?yuv444p? is a way of re-encoding the entire 24-bit RGB information"
>>
>> and went on to say
>>
>> "To me personally, I kind of like the way ffmpeg handle color space
>> conversion. Ffmpeg does have issues within its algorithm. YUV444p is not
>> one of them"
>>
>> So do you think that if you made a 4096x4096 rgb24 image using C with
>> all the possible variations, that you could, using ffmpeg, convert it to
>> yuv444p and back to rgb24 without loss?
>>
>> Try it - it's trivial in C to generate and also trivial to write
>> something that counts the unique pixels.
>>
>> I've just done it but my C is so bad I may make the programmers on here
>> feel ill if I posted it - and it's probably still buggy :-)
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>
>> ffmpeg-user@
>
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> Of course, what I've been saying is the feature of H.264 and what YCbCr
> 4:4:4 means. Whether or not x264 can actually meet H.264 specification;
> whether or not H.264 design is with bugs; that is entirely different issue.
> I never once said that I assumed that the encoder itself is without bugs.
> What I really said was what it was supposed to be. It?s a question of
> theoretical capability and actual implementation.

This is the ffmpeg mailing list not x264 and when you used the term 
YUV444p that means a very specific thing which I pointed out is not 
lossless when used to convert from rgb24, and it is ffmpeg that does the 
conversion.

> By the way, did you not read my previous post? YCbCr, as stated by H.264
> specification Ed 8.0, is not the actual color representation. YCbCr 4:4:4 is
> just another way to say full color information is being encoded. (x264 calls
> it yuv444p; anyway the name is not the point here) You can say that there is
> ?color space conversion? if re-encoding the information with affine map is
> what you mean.

No, you can't say there is "color space conversion" or re-encoding 
because x264 doesn't do that - it takes what you give it and for the 
purpose of reading the spec what you give it is referred to as YCbCr.


> This may seem surprising to you. The actual absolute color space currently
> supported by H.264 is

Supported meaning that you feed it something, tell it what it is, so 
that what it is can be conveyed to the decoder.


> This is true as of Ed. 8.0 I?m not sure how much specs of H.264 being
> implemented by x264, it?s not full; that?s for sure. It?s not practical to
> implement every spec anyway. Let me quote once again from page 42
>
> ?For convenience of notation and terminology in this Specification, the
> variables and terms associated with these arrays are referred to as luma (or
> L or Y) and chroma, where the two chroma arrays are referred to as Cb and
> Cr; regardless of the actual color representation method in use. The actual
> color representation method in use can be indicated in syntax that is
> specified in Annex E. The (monochrome) auxiliary arrays, which may or may
> not be present as auxiliary pictures in a coded video sequence, are optional
> for decoding and can be used for such purposes as alpha blending.?
> In other words, the reported YUV444p is not the actual absolute color space
> being used.

As above - this is ffmpeg list and when you use the term yuv444p that 
means something specific and totally unrelated to how you should read 
the h264 spec.


  How to know which one is being used? That?s a good question. The
> short answer is I don?t know.

The one that is being used is the one you (ffmpeg) gave to it and 
hopefully told it so it could pass on the info to the decoder.

> According to H.264 specification, the ?colour_primaries? syntax element is
> being used to specify which absolute color space is being used. Regardless
> of which absolute color space is being encoded in, the decoder still need to
> decode it to sRGB in Windows anyway.

In practice on a computer assuming yuv is the input then the decoder 
will send yuv to the graphics card/chip for csc and scaling.


From jiachielee at live.com  Fri Aug  2 01:46:21 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 1 Aug 2013 16:46:21 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51FAF175.30803@gmail.com>
References: <1375096444082-4660295.post@n4.nabble.com>
	<51F6A399.4020400@gmail.com>
	<1375323216242-4660379.post@n4.nabble.com>
	<51FA347A.6080103@gmail.com>
	<1375358966168-4660384.post@n4.nabble.com>
	<51FA75B3.2020204@gmail.com>
	<1375369292433-4660394.post@n4.nabble.com>
	<51FAC39D.7040404@gmail.com>
	<1375390396474-4660407.post@n4.nabble.com>
	<51FAF175.30803@gmail.com>
Message-ID: <1375400781203-4660413.post@n4.nabble.com>

"yuv" itself is a misnomer. It's commonly understood to actually mean "YCbCr"
for the encoder. By the way, ffmpeg does not do the encoding on its own; it
uses x264. x264 calls it YCbCr and follows ITU-T's recommendation (H.264
Specs). See http://mewiki.project357.com/wiki/X264_Settings#colormatrix here
for yourself. By the way, bt709 is different from sRGB, but they do share
the same color primaries.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Does-converting-to-yuv444p-by-default-make-sense-tp4660219p4660413.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From brad.clemetson at codeprogrammers.net  Fri Aug  2 04:12:28 2013
From: brad.clemetson at codeprogrammers.net (Bradley Clemetson)
Date: Thu, 1 Aug 2013 19:12:28 -0700
Subject: [FFmpeg-user] Decoding from a file in memory (std::vector<uint8_t>)
Message-ID: <379C7B63-CB26-4ABF-B77A-A2BB9C3D8B85@codeprogrammers.net>

Hello,
I am new to FFMPEG is general and am going through the decoding_encoding.c example from the sources. I was able to successfully encode the sample code in my own implementation; but am struggling with decoding files. For my usage of FFMPEG, the audio file is already loaded into memory via std::vector<uint8_t> I am working to adapt the static void audio_decode_example(const char *outfilename, const char *filename) to fit this need. Unfortunately it seems the data I am writing out is "loosing" packets in a consistent pattern. I'm not sure if it is the way I am passing the incorrect data to avcodec_decode_audio4 or incorrectly writing the result.

I have put my current progress on paste bin @ http://pastebin.com/M7K1287V

Thank You :)
Bradley Clemetson
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From tstrike34 at gmail.com  Fri Aug  2 05:40:23 2013
From: tstrike34 at gmail.com (JT Edwards)
Date: Thu, 1 Aug 2013 22:40:23 -0500
Subject: [FFmpeg-user] FFMPEG decoding Dailymotion Videos
Message-ID: <CAHr_b4G0iuM7Vxv0a42mWpAVGF8OtV2EmvBfwzbCdCkWWoPgYg@mail.gmail.com>

Some videos I can, and some videos I cant... Weird aint it?

Here is some output:

ffmpeg -i
http://www.dailymotion.com/cdn/H264-512x384/video/x3n3tn.mp4?auth=1375587534-9675cf2684c10d505ac89357ab3060a1
ffmpeg version N-54946-gaf2a196-syslint Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 26 2013 08:08:22 with gcc 4.4.7 (GCC) 20120313 (Red Hat
4.4.7-3)
  configuration: --prefix=/usr/local/cpffmpeg --enable-shared
--enable-nonfree --enable-gpl --enable-pthreads --enable-libopencore-amrnb
--enable-decoder=liba52 --enable-libopencore-amrwb --enable-libfaac
--enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264
--enable-libxvid --extra-cflags=-I/usr/local/cpffmpeg/include/
--extra-ldflags=-L/usr/local/cpffmpeg/lib --enable-version3
--extra-version=syslint
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 81.103 /  3. 81.103
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
[http @ 0x235d600] HTTP error 404 Not Found
http://www.dailymotion.com/cdn/H264-512x384/video/x3n3tn.mp4?auth=1375587534-9675cf2684c10d505ac89357ab3060a1:
Input/output error

--

From cehoyos at ag.or.at  Fri Aug  2 09:39:53 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 2 Aug 2013 07:39:53 +0000 (UTC)
Subject: [FFmpeg-user] FFMPEG decoding Dailymotion Videos
References: <CAHr_b4G0iuM7Vxv0a42mWpAVGF8OtV2EmvBfwzbCdCkWWoPgYg@mail.gmail.com>
Message-ID: <loom.20130802T093922-430@post.gmane.org>

JT Edwards <tstrike34 <at> gmail.com> writes:

> Some videos I can, and some videos I cant... Weird aint it?

Does it work with wget?
Did you try -user-agent ?

Carl Eugen


From hendry at iki.fi  Fri Aug  2 09:51:50 2013
From: hendry at iki.fi (Kai Hendry)
Date: Fri, 2 Aug 2013 15:51:50 +0800
Subject: [FFmpeg-user] Efficient Audio encoding from a PI
Message-ID: <CAF8XF0d1bVBtMRS=9VkQbpgahimmgVXPyDGRwNPHCYyVvamVYQ@mail.gmail.com>

Hi there,

I'm trying to stream audio from a Raspberry PI with a USB attached
GOMIC. The problem is when I try HTTP Live Streaming [1] with h/w
encoded video & aac audio I get ALSA xruns:
http://ix.io/70K

Not quite sure what a "xrun" actually is. I'm going to guess an under
run, since the PI 's CPU seems maxed out.


If I just run the audio process, without video:
ffmpeg -f alsa -ac 1 -i hw:1,0 -acodec aac -strict experimental -y
RecordingMYvoice.aac
http://s.natalian.org/2013-08-02/1375429223_1366x768.png

I can see it's taking 70% of my PI's CPU! Is there some good
strategies to make the voice perhaps lower quality and hence less
likely max CPU and pause / stutter on playback?
I don't know what settings to play with.

I believe I have to use the "aac" codec, since we want it to work from IOS.

Many thanks!

[1] https://github.com/AndyA/psips

From cehoyos at ag.or.at  Fri Aug  2 10:03:37 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 2 Aug 2013 08:03:37 +0000 (UTC)
Subject: [FFmpeg-user] Efficient Audio encoding from a PI
References: <CAF8XF0d1bVBtMRS=9VkQbpgahimmgVXPyDGRwNPHCYyVvamVYQ@mail.gmail.com>
Message-ID: <loom.20130802T100103-232@post.gmane.org>

Kai Hendry <hendry <at> iki.fi> writes:

> ffmpeg -f alsa -ac 1 -i hw:1,0 -acodec aac 
> -strict experimental -y RecordingMYvoice.aac

How does the cpu usage look if you encode to wav?

Try setting another bitrate and the -aac_coder option.

The alternative is to implement SIMD functions for 
the encoder but this is not only non-trivial but may 
be bad for an encoder that is not finished yet.

Carl Eugen


From cehoyos at ag.or.at  Fri Aug  2 10:06:28 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 2 Aug 2013 08:06:28 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?Decoding_from_a_file_in_memory_=28std=3A?=
	=?utf-8?b?OnZlY3Rvcjx1aW50OF90Pik=?=
References: <379C7B63-CB26-4ABF-B77A-A2BB9C3D8B85@codeprogrammers.net>
Message-ID: <loom.20130802T100440-876@post.gmane.org>

Bradley Clemetson <brad.clemetson <at> codeprogrammers.net> writes:

> For my usage of FFMPEG, the audio file is already loaded 
> into memory via std::vector<uint8_t>

I fear you will first have to implement the "mem" protocol 
within FFmpeg, this may not be so difficult, an ancient 
patch exists:
http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/49711

> I have put my current progress on paste bin 
> http://pastebin.com/M7K1287V

Does not compile here.

libav-user may be the more appropriate mailing list.

Carl Eugen


From highgod0401 at gmail.com  Fri Aug  2 10:17:45 2013
From: highgod0401 at gmail.com (Wei Gao)
Date: Fri, 2 Aug 2013 16:17:45 +0800
Subject: [FFmpeg-user] question: Hi,
 I want to ask a question about capture audio from audio card
In-Reply-To: <CAL1QdWfXseON2nCSpXhWNgB3h8N94APV5AABcx8C9DGUGKJgyA@mail.gmail.com>
References: <CAAGF25gj93ogvjUh5dh3+LSu-j4m_HJZrGHJCb6d=A6GHX09+Q@mail.gmail.com>
	<CAL1QdWdSZJJdthc8pA0OyCWyNCx634LJDEDqd_EdcPbDTeEHPA@mail.gmail.com>
	<CAAGF25jZKXiYiBufCG1r5dZN=Y7J-pGoeeetf6-h4HtjmdLpsA@mail.gmail.com>
	<CAL1QdWfXseON2nCSpXhWNgB3h8N94APV5AABcx8C9DGUGKJgyA@mail.gmail.com>
Message-ID: <CAAGF25gkomsr=Pw5Wg6Xbt6Wc=yosXwYmXT2dkFtMDK96kFR8g@mail.gmail.com>

2013/7/8 Roger Pack <rogerdpack2 at gmail.com>

> you may need to intsall the visual studio redistributable runtime 64 bit
> first?
>
> On 7/8/13, Wei Gao <highgod0401 at gmail.com> wrote:
> > 2013/7/3 Roger Pack <rogerdpack2 at gmail.com>
> >
> >> perhaps try googling "ffmpeg dshow" ?
> >>
> >>
> > Hi, I use ffmpeg 32bit and  virtial-audio-capture, the program run
> correct,
> > but when I swtich to 64bit and use regsvr32 to
> > register audio_sniffer-x64.dll, error occured says that the dll is not
> > suiltable for my computer.
> >
> >
> > On 4/28/13, Wei Gao <highgod0401 at gmail.com> wrote:
> >> > Hi,
> >> >
> >> > I want to capture audio from audio card, not from mic, the data is the
> >> > output audio from audio card
> >> >
> >> > Funman has told me to use asla to capture on linux in mirc, how can I
> >> > capture on windows?
> >> >
> >> > Thanks
> >> > Best regards.
>
Hi, I about the audio capture, I tested on win7 both 32 and 64 is working
perfect,  using  virtual-audio-capturer. But when I switch it to win8, If I
play the audio, the program run correct, if I stop the audio or pause the
audio, the program will block at function "av_read_frame", my question is
can it run on win8?

Thanks
Best regards

> >> > _______________________________________________
> >> > ffmpeg-devel mailing list
> >> > ffmpeg-devel at ffmpeg.org
> >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> >> >
> >>
> >
>

From 99191957 at qq.com  Fri Aug  2 10:26:01 2013
From: 99191957 at qq.com (99191957 at qq.com)
Date: Fri, 2 Aug 2013 16:26:01 +0800
Subject: [FFmpeg-user] question: Hi,
	I want to ask a question about capture audio from audio card
References: <CAAGF25gj93ogvjUh5dh3+LSu-j4m_HJZrGHJCb6d=A6GHX09+Q@mail.gmail.com>
	<CAL1QdWdSZJJdthc8pA0OyCWyNCx634LJDEDqd_EdcPbDTeEHPA@mail.gmail.com>
	<CAAGF25jZKXiYiBufCG1r5dZN=Y7J-pGoeeetf6-h4HtjmdLpsA@mail.gmail.com>
	<CAL1QdWfXseON2nCSpXhWNgB3h8N94APV5AABcx8C9DGUGKJgyA@mail.gmail.com>,
	<CAAGF25gkomsr=Pw5Wg6Xbt6Wc=yosXwYmXT2dkFtMDK96kFR8g@mail.gmail.com>
Message-ID: <201308021626008341243@qq.com>

you can just ignore the return value of AV_read_frame.
For example: no not use " While(Av_read_frame(...)>=0)",
                      then, if you get a wrong packet ,it won't stop reading next one.




99191957 at qq.com

From: Wei Gao
Date: 2013-08-02 16:17
To: FFmpeg user questions and RTFMs; FFmpeg development discussions and patches
Subject: Re: [FFmpeg-user] question: Hi, I want to ask a question about capture audio from audio card
2013/7/8 Roger Pack <rogerdpack2 at gmail.com>

> you may need to intsall the visual studio redistributable runtime 64 bit
> first?
>
> On 7/8/13, Wei Gao <highgod0401 at gmail.com> wrote:
> > 2013/7/3 Roger Pack <rogerdpack2 at gmail.com>
> >
> >> perhaps try googling "ffmpeg dshow" ?
> >>
> >>
> > Hi, I use ffmpeg 32bit and  virtial-audio-capture, the program run
> correct,
> > but when I swtich to 64bit and use regsvr32 to
> > register audio_sniffer-x64.dll, error occured says that the dll is not
> > suiltable for my computer.
> >
> >
> > On 4/28/13, Wei Gao <highgod0401 at gmail.com> wrote:
> >> > Hi,
> >> >
> >> > I want to capture audio from audio card, not from mic, the data is the
> >> > output audio from audio card
> >> >
> >> > Funman has told me to use asla to capture on linux in mirc, how can I
> >> > capture on windows?
> >> >
> >> > Thanks
> >> > Best regards.
>
Hi, I about the audio capture, I tested on win7 both 32 and 64 is working
perfect,  using  virtual-audio-capturer. But when I switch it to win8, If I
play the audio, the program run correct, if I stop the audio or pause the
audio, the program will block at function "av_read_frame", my question is
can it run on win8?

Thanks
Best regards

> >> > _______________________________________________
> >> > ffmpeg-devel mailing list
> >> > ffmpeg-devel at ffmpeg.org
> >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> >> >
> >>
> >
>
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
.

From hendry at iki.fi  Fri Aug  2 10:54:53 2013
From: hendry at iki.fi (Kai Hendry)
Date: Fri, 2 Aug 2013 16:54:53 +0800
Subject: [FFmpeg-user] Efficient Audio encoding from a PI
In-Reply-To: <loom.20130802T100103-232@post.gmane.org>
References: <CAF8XF0d1bVBtMRS=9VkQbpgahimmgVXPyDGRwNPHCYyVvamVYQ@mail.gmail.com>
	<loom.20130802T100103-232@post.gmane.org>
Message-ID: <CAF8XF0f0A=agoBTDnLDpuw0YYNh+TjDZjXjPbiREeGkVWKBASQ@mail.gmail.com>

On 2 August 2013 16:03, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> How does the cpu usage look if you encode to wav?

Recording just to WAV on Raspberry PI is about ~50% of CPU:
http://s.natalian.org/2013-08-02/1375433595_1366x768.png

What's a good bitrate candidate? Sorry I don't know.

From cehoyos at ag.or.at  Fri Aug  2 11:02:18 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 2 Aug 2013 09:02:18 +0000 (UTC)
Subject: [FFmpeg-user] Efficient Audio encoding from a PI
References: <CAF8XF0d1bVBtMRS=9VkQbpgahimmgVXPyDGRwNPHCYyVvamVYQ@mail.gmail.com>
	<loom.20130802T100103-232@post.gmane.org>
	<CAF8XF0f0A=agoBTDnLDpuw0YYNh+TjDZjXjPbiREeGkVWKBASQ@mail.gmail.com>
Message-ID: <loom.20130802T110159-762@post.gmane.org>

Kai Hendry <hendry <at> iki.fi> writes:

> What's a good bitrate candidate? Sorry I don't know.

Me neither, I just wanted to suggest things you can try.

Carl Eugen


From highgod0401 at gmail.com  Fri Aug  2 11:24:42 2013
From: highgod0401 at gmail.com (Wei Gao)
Date: Fri, 2 Aug 2013 17:24:42 +0800
Subject: [FFmpeg-user] question: Hi,
 I want to ask a question about capture audio from audio card
In-Reply-To: <201308021626008341243@qq.com>
References: <CAAGF25gj93ogvjUh5dh3+LSu-j4m_HJZrGHJCb6d=A6GHX09+Q@mail.gmail.com>
	<CAL1QdWdSZJJdthc8pA0OyCWyNCx634LJDEDqd_EdcPbDTeEHPA@mail.gmail.com>
	<CAAGF25jZKXiYiBufCG1r5dZN=Y7J-pGoeeetf6-h4HtjmdLpsA@mail.gmail.com>
	<CAL1QdWfXseON2nCSpXhWNgB3h8N94APV5AABcx8C9DGUGKJgyA@mail.gmail.com>
	<CAAGF25gkomsr=Pw5Wg6Xbt6Wc=yosXwYmXT2dkFtMDK96kFR8g@mail.gmail.com>
	<201308021626008341243@qq.com>
Message-ID: <CAAGF25gykkmmtEOt-aDZM-rxPObCC-MBDsu6qS+NCu4cQ-sf_g@mail.gmail.com>

2013/8/2 99191957 at qq.com <99191957 at qq.com>

> you can just ignore the return value of AV_read_frame.
> For example: no not use " While(Av_read_frame(...)>=0)",
>                       then, if you get a wrong packet ,it won't stop
> reading next one.
>
> Hi, thanks

yes, I ignore it, but our problem is that the av_read_frame can not return
on win8. it dead and without any error report, the program block in the
function.

Thanks for your reply

Best regards

>
>
>
> 99191957 at qq.com
>
> From: Wei Gao
> Date: 2013-08-02 16:17
> To: FFmpeg user questions and RTFMs; FFmpeg development discussions and
> patches
> Subject: Re: [FFmpeg-user] question: Hi, I want to ask a question about
> capture audio from audio card
> 2013/7/8 Roger Pack <rogerdpack2 at gmail.com>
>
> > you may need to intsall the visual studio redistributable runtime 64 bit
> > first?
> >
> > On 7/8/13, Wei Gao <highgod0401 at gmail.com> wrote:
> > > 2013/7/3 Roger Pack <rogerdpack2 at gmail.com>
> > >
> > >> perhaps try googling "ffmpeg dshow" ?
> > >>
> > >>
> > > Hi, I use ffmpeg 32bit and  virtial-audio-capture, the program run
> > correct,
> > > but when I swtich to 64bit and use regsvr32 to
> > > register audio_sniffer-x64.dll, error occured says that the dll is not
> > > suiltable for my computer.
> > >
> > >
> > > On 4/28/13, Wei Gao <highgod0401 at gmail.com> wrote:
> > >> > Hi,
> > >> >
> > >> > I want to capture audio from audio card, not from mic, the data is
> the
> > >> > output audio from audio card
> > >> >
> > >> > Funman has told me to use asla to capture on linux in mirc, how can
> I
> > >> > capture on windows?
> > >> >
> > >> > Thanks
> > >> > Best regards.
> >
> Hi, I about the audio capture, I tested on win7 both 32 and 64 is working
> perfect,  using  virtual-audio-capturer. But when I switch it to win8, If I
> play the audio, the program run correct, if I stop the audio or pause the
> audio, the program will block at function "av_read_frame", my question is
> can it run on win8?
>
> Thanks
> Best regards
>
> > >> > _______________________________________________
> > >> > ffmpeg-devel mailing list
> > >> > ffmpeg-devel at ffmpeg.org
> > >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> > >> >
> > >>
> > >
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> .
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From soho123.2012 at gmail.com  Fri Aug  2 11:38:16 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Fri, 2 Aug 2013 17:38:16 +0800
Subject: [FFmpeg-user] ffmpeg can not post new video data to ffserver
Message-ID: <CAFd2cZWMxuu+fMAFnh=A9VJ6n6qTE9BwxWX9yo5qjnX2oqMtEw@mail.gmail.com>

Hi All,

I got the problem about ffmpeg post fail to ffserver.
I use the solution ffmpeg+ ffserver for a stream server.
the stream can work for a few minutes, then ffmpeg can not post video data
via http://localhost:8090/feed1.ffm
the command line I use for ffmpeg:
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format h264
-i /dev/video1 -vcodec
copy http://localhost:8090/feed1.ffm


ffserver can not receive new video data from ffmpeg if the stream is
running for about 30~60 minutes.
I check the errno from ffmpeg av_interleaved_write_frame()
it says "No such file or directory"
errno is 2

Does anyone have ideas?

From cehoyos at ag.or.at  Fri Aug  2 12:13:12 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 2 Aug 2013 10:13:12 +0000 (UTC)
Subject: [FFmpeg-user] question: Hi,
	I want to ask a question about capture audio from audio card
References: <CAAGF25gj93ogvjUh5dh3+LSu-j4m_HJZrGHJCb6d=A6GHX09+Q@mail.gmail.com>
	<CAL1QdWdSZJJdthc8pA0OyCWyNCx634LJDEDqd_EdcPbDTeEHPA@mail.gmail.com>
	<CAAGF25jZKXiYiBufCG1r5dZN=Y7J-pGoeeetf6-h4HtjmdLpsA@mail.gmail.com>
	<CAL1QdWfXseON2nCSpXhWNgB3h8N94APV5AABcx8C9DGUGKJgyA@mail.gmail.com>
	<CAAGF25gkomsr=Pw5Wg6Xbt6Wc=yosXwYmXT2dkFtMDK96kFR8g@mail.gmail.com>
	<201308021626008341243@qq.com>
	<CAAGF25gykkmmtEOt-aDZM-rxPObCC-MBDsu6qS+NCu4cQ-sf_g@mail.gmail.com>
Message-ID: <loom.20130802T121250-833@post.gmane.org>

Wei Gao <highgod0401 <at> gmail.com> writes:

> our problem is that the av_read_frame can not return
> on win8. it dead and without any error report, the 
> program block in the function.

What does gdb report?

Carl Eugen


From andrey.aleksandrovich at googlemail.com  Fri Aug  2 13:51:38 2013
From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich)
Date: Fri, 2 Aug 2013 14:51:38 +0300
Subject: [FFmpeg-user] De-interlacing and Inverse Telecining Questions
In-Reply-To: <loom.20130802T002600-665@post.gmane.org>
References: <91574A7267144E66BC9820C75A530CB7@gmail.com>
	<loom.20130802T002600-665@post.gmane.org>
Message-ID: <CAPGkfYGx-buBO64FLCf_HyMN=vb-LUpvF+9zesWcNKJ8nnVj7g@mail.gmail.com>

> fieldmatch is a modern inverse telecine filter that is
> said to beat MPlayer's pullup filter for every kind of
> source file (but needs more resources).

FFmpeg 1.0.7 - I don't have that filter (fieldmatch) available.
Is that filter new feature of latest versions?

On 8/2/13, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Young Kim <shadowing71 <at> gmail.com> writes:
>
>> I've been reading through the ffmpeg documentation
>> about deinterlacing and inverse telecining, and it's
>> a bit confusing. In particular, there seems to be a
>> myriad ways of achieving this (i.e. yadif, fieldmatch,
>> and kerndeint). Does anyone happen to know what the
>> difference is among these
>
> You first have to understand that while interlaced and
> telecined video look similar they are fundamentally
> different: Deinterlacing means inventing 50% of the
> image where no image was, inverse telecine brings the
> original frames back that were used as input for the
> telecine process.
> (Or in other words: The telecine process only duplicates
> some information while interlacing means throwing
> away 50% of the video information - this may even
> happen within camera equipment.)
>
> You should not deinterlace telecined material and you
> cannot inverse telecine interlaced videos.
>
> kerndeint is an old deinterlacer that you should only
> use if you have a specific reason. (performance on arm?)
>
> yadif is a good deinterlacer and has simd optimization
> on x86.
>
> fieldmatch is a modern inverse telecine filter that is
> said to beat MPlayer's pullup filter for every kind of
> source file (but needs more resources).
>
>> and if there is a way to automatically deinterlace /
>> inverse telecine an input accordingly?
>
> Not really because while it is technically possible to
> detect if your video is interlaced or telecined (see
> the idet filter), it is significantly easier and less
> error-prone if you do it (visually) before transcoding.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Fri Aug  2 14:07:10 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 2 Aug 2013 12:07:10 +0000 (UTC)
Subject: [FFmpeg-user] De-interlacing and Inverse Telecining Questions
References: <91574A7267144E66BC9820C75A530CB7@gmail.com>
	<loom.20130802T002600-665@post.gmane.org>
	<CAPGkfYGx-buBO64FLCf_HyMN=vb-LUpvF+9zesWcNKJ8nnVj7g@mail.gmail.com>
Message-ID: <loom.20130802T140611-292@post.gmane.org>

Andrey Aleksandrovich <andrey.aleksandrovich <at> googlemail.com> writes:

> FFmpeg 1.0.7 - I don't have that filter (fieldmatch) available.
> Is that filter new feature of latest versions?

It is actually more than seven months old.

Unrelated: If you are not a distributor yourself, you are 
strongly encouraged not to use a release.

Carl Eugen


From brad.clemetson at codeprogrammers.net  Fri Aug  2 15:36:11 2013
From: brad.clemetson at codeprogrammers.net (Bradley Clemetson)
Date: Fri, 2 Aug 2013 06:36:11 -0700
Subject: [FFmpeg-user] Decoding from a file in memory
	(std::vector<uint8_t>)
In-Reply-To: <loom.20130802T100440-876@post.gmane.org>
References: <379C7B63-CB26-4ABF-B77A-A2BB9C3D8B85@codeprogrammers.net>
	<loom.20130802T100440-876@post.gmane.org>
Message-ID: <E5BA7581-0307-474E-A4C3-8288E39BBD8F@codeprogrammers.net>

My mistake, I forgot to include the headers in the paste, this will compile.

http://pastebin.com/CDydt5vB

On Aug 2, 2013, at 1:06 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Bradley Clemetson <brad.clemetson <at> codeprogrammers.net> writes:
> 
>> For my usage of FFMPEG, the audio file is already loaded 
>> into memory via std::vector<uint8_t>
> 
> I fear you will first have to implement the "mem" protocol 
> within FFmpeg, this may not be so difficult, an ancient 
> patch exists:
> http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/49711
> 
>> I have put my current progress on paste bin 
>> http://pastebin.com/M7K1287V
> 
> Does not compile here.
> 
> libav-user may be the more appropriate mailing list.
> 
> Carl Eugen
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

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From francois.visagie at gmail.com  Fri Aug  2 16:09:19 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Fri, 2 Aug 2013 16:09:19 +0200
Subject: [FFmpeg-user] Preserving preserved loudness when downmixing audio
	from 5.1 AC3 to stereo AAC
Message-ID: <000601ce8f89$e60c9980$b225cc80$@gmail.com>

Hi All,

 

What is the simplest way of preserving perceived loudness when rematrixing
audio from 5.1 AC3 to stereo AAC? There seem to be many different options
available, not all are fully documented, and results vary:

.         output from '-ac 2' is noticeably softer than the original

.         '-filter:a aformat=channel_layouts=stereo' seems to have the same
volume as the original

.         '-filter:a_aresample=ocl=3' is noticeably softer than the original

 

I'd have expected the resampler's rematrixing to do the best job, but given
the above results, is it OK going with '-filter:a
aformat=channel_layouts=stereo'? Or is some other way recommended?

 

Console output:

 

C:\Users\fvisagie\Videos\Home Videos\Testing\x264\AC3>ffmpeg -i inac3.mpg
-ac 2 test-ac2.mp4

ffmpeg version N-54277-gff130d7 Copyright (c) 2000-2013 the FFmpeg
developers

  built on Jun 30 2013 01:20:12 with gcc 4.7.3 (GCC)

  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enab

le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --ena

ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp

eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --

enable-libxvid --enable-zlib

  libavutil      52. 37.101 / 52. 37.101

  libavcodec     55. 17.100 / 55. 17.100

  libavformat    55. 10.100 / 55. 10.100

  libavdevice    55.  2.100 / 55.  2.100

  libavfilter     3. 77.101 /  3. 77.101

  libswscale      2.  3.100 /  2.  3.100

  libswresample   0. 17.102 /  0. 17.102

  libpostproc    52.  3.100 / 52.  3.100

Input #0, mpeg, from 'inac3.mpg':

  Duration: 00:00:08.99, start: 0.184656, bitrate: 7204 kb/s

    Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR
64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc

    Stream #0:1[0x80]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s

[libx264 @ 027a79e0] using SAR=64/45

[libx264 @ 027a79e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX

[libx264 @ 027a79e0] profile High, level 3.0

[libx264 @ 027a79e0] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1
ref=3 deb

lock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11
fast_pskip=1 chro

ma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0
decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3
b_pyramid=2 b_adapt=1

b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25
scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0
qcomp=0.60 qpmin=

0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00

Output #0, mp4, to 'test-ac2.mp4':

  Metadata:

    encoder         : Lavf55.10.100

    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
720x576 [SAR 64:45 DAR 16:9], q=-1--1, 12800 tbn, 25 tbc

    Stream #0:1: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 48000
Hz, stereo, s16, 128 kb/s

Stream mapping:

  Stream #0:0 -> #0:0 (mpeg2video -> libx264)

  Stream #0:1 -> #0:1 (ac3 -> libvo_aacenc)

Press [q] to stop, [?] for help

frame=  222 fps= 42 q=-1.0 Lsize=    2370kB time=00:00:09.03
bitrate=2149.2kbits/s

video:2220kB audio:142kB subtitle:0 global headers:0kB muxing overhead
0.327047%

[libx264 @ 027a79e0] frame I:3     Avg QP:19.91  size: 26627

[libx264 @ 027a79e0] frame P:136   Avg QP:24.64  size: 14045

[libx264 @ 027a79e0] frame B:83    Avg QP:29.17  size:  3406

[libx264 @ 027a79e0] consecutive B-frames: 26.6% 70.3%  1.4%  1.8%

[libx264 @ 027a79e0] mb I  I16..4: 10.9% 84.9%  4.2%

[libx264 @ 027a79e0] mb P  I16..4:  4.7%  3.9%  0.6%  P16..4: 37.8% 26.6%
15.8%  0.0%  0.0%    skip:10.5%

[libx264 @ 027a79e0] mb B  I16..4:  0.0%  0.1%  0.0%  B16..8: 43.4%  7.8%
1.8%  direct: 2.4%  skip:44.4%  L0:26.8% L1:47.8% BI:25.4%

[libx264 @ 027a79e0] 8x8 transform intra:50.8% inter:69.2%

[libx264 @ 027a79e0] coded y,uvDC,uvAC intra: 38.9% 51.9% 29.5% inter: 31.3%
38.6% 9.0%

[libx264 @ 027a79e0] i16 v,h,dc,p: 80% 11%  5%  4%

[libx264 @ 027a79e0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 10% 25%  4%  7%  9%
5%  8%  6%

[libx264 @ 027a79e0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 13% 16%  6% 11% 13%
7%  9%  4%

[libx264 @ 027a79e0] i8c dc,h,v,p: 67% 12% 16%  5%

[libx264 @ 027a79e0] Weighted P-Frames: Y:5.9% UV:2.9%

[libx264 @ 027a79e0] ref P L0: 72.1% 24.1%  3.0%  0.8%  0.0%

[libx264 @ 027a79e0] ref B L0: 98.5%  1.5%

[libx264 @ 027a79e0] ref B L1: 100.0%  0.0%

[libx264 @ 027a79e0] kb/s:2047.52

 

C:\Users\fvisagie\Videos\Home Videos\Testing\x264\AC3>ffmpeg -i inac3.mpg
-filter:a aformat=channel_layouts=stereo
test-filter.a_aformat=channel_layouts=stereo.

mp4

ffmpeg version N-54277-gff130d7 Copyright (c) 2000-2013 the FFmpeg
developers

  built on Jun 30 2013 01:20:12 with gcc 4.7.3 (GCC)

  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enab

le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --ena

ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp

eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --

enable-libxvid --enable-zlib

  libavutil      52. 37.101 / 52. 37.101

  libavcodec     55. 17.100 / 55. 17.100

  libavformat    55. 10.100 / 55. 10.100

  libavdevice    55.  2.100 / 55.  2.100

  libavfilter     3. 77.101 /  3. 77.101

  libswscale      2.  3.100 /  2.  3.100

  libswresample   0. 17.102 /  0. 17.102

  libpostproc    52.  3.100 / 52.  3.100

Input #0, mpeg, from 'inac3.mpg':

  Duration: 00:00:08.99, start: 0.184656, bitrate: 7204 kb/s

    Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR
64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc

    Stream #0:1[0x80]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s

[libx264 @ 035be3a0] using SAR=64/45

[libx264 @ 035be3a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX

[libx264 @ 035be3a0] profile High, level 3.0

[libx264 @ 035be3a0] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1
ref=3 deb

lock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11
fast_pskip=1 chro

ma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0
decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3
b_pyramid=2 b_adapt=1

b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25
scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0
qcomp=0.60 qpmin=

0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00

Output #0, mp4, to 'test-filter.a_aformat=channel_layouts=stereo.mp4':

  Metadata:

    encoder         : Lavf55.10.100

    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
720x576 [SAR 64:45 DAR 16:9], q=-1--1, 12800 tbn, 25 tbc

    Stream #0:1: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 48000
Hz, stereo, s16, 128 kb/s

Stream mapping:

  Stream #0:0 -> #0:0 (mpeg2video -> libx264)

  Stream #0:1 -> #0:1 (ac3 -> libvo_aacenc)

Press [q] to stop, [?] for help

frame=  222 fps= 42 q=-1.0 Lsize=    2370kB time=00:00:09.03
bitrate=2149.2kbits/s

video:2220kB audio:142kB subtitle:0 global headers:0kB muxing overhead
0.327056%

[libx264 @ 035be3a0] frame I:3     Avg QP:19.91  size: 26627

[libx264 @ 035be3a0] frame P:136   Avg QP:24.64  size: 14045

[libx264 @ 035be3a0] frame B:83    Avg QP:29.17  size:  3406

[libx264 @ 035be3a0] consecutive B-frames: 26.6% 70.3%  1.4%  1.8%

[libx264 @ 035be3a0] mb I  I16..4: 10.9% 84.9%  4.2%

[libx264 @ 035be3a0] mb P  I16..4:  4.7%  3.9%  0.6%  P16..4: 37.8% 26.6%
15.8%  0.0%  0.0%    skip:10.5%

[libx264 @ 035be3a0] mb B  I16..4:  0.0%  0.1%  0.0%  B16..8: 43.4%  7.8%
1.8%  direct: 2.4%  skip:44.4%  L0:26.8% L1:47.8% BI:25.4%

[libx264 @ 035be3a0] 8x8 transform intra:50.8% inter:69.2%

[libx264 @ 035be3a0] coded y,uvDC,uvAC intra: 38.9% 51.9% 29.5% inter: 31.3%
38.6% 9.0%

[libx264 @ 035be3a0] i16 v,h,dc,p: 80% 11%  5%  4%

[libx264 @ 035be3a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 10% 25%  4%  7%  9%
5%  8%  6%

[libx264 @ 035be3a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 13% 16%  6% 11% 13%
7%  9%  4%

[libx264 @ 035be3a0] i8c dc,h,v,p: 67% 12% 16%  5%

[libx264 @ 035be3a0] Weighted P-Frames: Y:5.9% UV:2.9%

[libx264 @ 035be3a0] ref P L0: 72.1% 24.1%  3.0%  0.8%  0.0%

[libx264 @ 035be3a0] ref B L0: 98.5%  1.5%

[libx264 @ 035be3a0] ref B L1: 100.0%  0.0%

[libx264 @ 035be3a0] kb/s:2047.52

 

C:\Users\fvisagie\Videos\Home Videos\Testing\x264\AC3>ffmpeg -i inac3.mpg
-filter:a aresample=ocl=3 test-filter.a_aresample=ocl=3.mp4

ffmpeg version N-54277-gff130d7 Copyright (c) 2000-2013 the FFmpeg
developers

  built on Jun 30 2013 01:20:12 with gcc 4.7.3 (GCC)

  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enab

le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --ena

ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp

eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --

enable-libxvid --enable-zlib

  libavutil      52. 37.101 / 52. 37.101

  libavcodec     55. 17.100 / 55. 17.100

  libavformat    55. 10.100 / 55. 10.100

  libavdevice    55.  2.100 / 55.  2.100

  libavfilter     3. 77.101 /  3. 77.101

  libswscale      2.  3.100 /  2.  3.100

  libswresample   0. 17.102 /  0. 17.102

  libpostproc    52.  3.100 / 52.  3.100

Input #0, mpeg, from 'inac3.mpg':

  Duration: 00:00:08.99, start: 0.184656, bitrate: 7204 kb/s

    Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR
64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc

    Stream #0:1[0x80]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s

[libx264 @ 0032b2c0] using SAR=64/45

[libx264 @ 0032b2c0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX

[libx264 @ 0032b2c0] profile High, level 3.0

[libx264 @ 0032b2c0] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1
ref=3 deb

lock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11
fast_pskip=1 chro

ma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0
decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3
b_pyramid=2 b_adapt=1

b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25
scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0
qcomp=0.60 qpmin=

0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00

Output #0, mp4, to 'test-filter.a_aresample=ocl=3.mp4':

  Metadata:

    encoder         : Lavf55.10.100

    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
720x576 [SAR 64:45 DAR 16:9], q=-1--1, 12800 tbn, 25 tbc

    Stream #0:1: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 48000
Hz, stereo, s16, 128 kb/s

Stream mapping:

  Stream #0:0 -> #0:0 (mpeg2video -> libx264)

  Stream #0:1 -> #0:1 (ac3 -> libvo_aacenc)

Press [q] to stop, [?] for help

frame=  222 fps= 42 q=-1.0 Lsize=    2370kB time=00:00:09.03
bitrate=2149.2kbits/s

video:2220kB audio:142kB subtitle:0 global headers:0kB muxing overhead
0.327047%

[libx264 @ 0032b2c0] frame I:3     Avg QP:19.91  size: 26627

[libx264 @ 0032b2c0] frame P:136   Avg QP:24.64  size: 14045

[libx264 @ 0032b2c0] frame B:83    Avg QP:29.17  size:  3406

[libx264 @ 0032b2c0] consecutive B-frames: 26.6% 70.3%  1.4%  1.8%

[libx264 @ 0032b2c0] mb I  I16..4: 10.9% 84.9%  4.2%

[libx264 @ 0032b2c0] mb P  I16..4:  4.7%  3.9%  0.6%  P16..4: 37.8% 26.6%
15.8%  0.0%  0.0%    skip:10.5%

[libx264 @ 0032b2c0] mb B  I16..4:  0.0%  0.1%  0.0%  B16..8: 43.4%  7.8%
1.8%  direct: 2.4%  skip:44.4%  L0:26.8% L1:47.8% BI:25.4%

[libx264 @ 0032b2c0] 8x8 transform intra:50.8% inter:69.2%

[libx264 @ 0032b2c0] coded y,uvDC,uvAC intra: 38.9% 51.9% 29.5% inter: 31.3%
38.6% 9.0%

[libx264 @ 0032b2c0] i16 v,h,dc,p: 80% 11%  5%  4%

[libx264 @ 0032b2c0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 10% 25%  4%  7%  9%
5%  8%  6%

[libx264 @ 0032b2c0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 13% 16%  6% 11% 13%
7%  9%  4%

[libx264 @ 0032b2c0] i8c dc,h,v,p: 67% 12% 16%  5%

[libx264 @ 0032b2c0] Weighted P-Frames: Y:5.9% UV:2.9%

[libx264 @ 0032b2c0] ref P L0: 72.1% 24.1%  3.0%  0.8%  0.0%

[libx264 @ 0032b2c0] ref B L0: 98.5%  1.5%

[libx264 @ 0032b2c0] ref B L1: 100.0%  0.0%

[libx264 @ 0032b2c0] kb/s:2047.52

 

Thanks,

Francois

 

Francois Visagie

 

 

T: +27 (83) 326-4358

F: +27 (86) 511-3490

E: francois.visagie at gmail.com

 

 


From jshupert at pps-inc.com  Fri Aug  2 18:46:59 2013
From: jshupert at pps-inc.com (Jim Shupert, Jr)
Date: Fri, 2 Aug 2013 12:46:59 -0400
Subject: [FFmpeg-user] 50 Mbps Ifram mpg2 with scaling and pad --sync problem
Message-ID: <99b707c624bfd985e2248ec04e42b10b.squirrel@webmail.theppsgroup.com>

Friends,
I wish to take 720x486 Unc 8 bit rawVideo with 2 chn  pcm audio
and make 50 mbps Iframe mpg2 422 1920x1080i with mpg2 audio

I realize that upScaling is not best - but I do need to in this situation.

I have something that I think is close but the audio goes out of sync.
I wonder why?

You will see I have my buffer set rather high - i did that to avoid
...
[mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
[mpeg @ 0x307afa0] packet too large, ignoring buffer limits to mux it
[mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
...

here is my command

jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
"scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
+ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
-async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg

jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
"scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
+ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
-async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg
ffmpeg version git-2012-09-24-fd63c2f Copyright (c) 2000-2012 the FFmpeg
developers
  built on Sep 24 2012 14:09:13 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
  configuration: --enable-gpl --enable-libfaac --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora
--enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
--enable-version3 --enable-x11grab
  libavutil      51. 73.101 / 51. 73.101
  libavcodec     54. 58.100 / 54. 58.100
  libavformat    54. 28.101 / 54. 28.101
  libavdevice    54.  2.101 / 54.  2.101
  libavfilter     3. 17.100 /  3. 17.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[avi @ 0x28b5240] non-interleaved AVI
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, avi, from '/media/data/z_parts/dog.avi':
  Duration: 00:00:05.03, start: 0.000000, bitrate: 170941 kb/s
    Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x486,
29.97 tbr, 29.97 tbn, 29.97 tbc
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo, s16, 1536 kb/s
File '/media/data/jtest/dog-3a_1080i.mpg' already exists. Overwrite ? [y/N] y
-async is forwarded to lavfi similarly to -af
aresample=min_comp=0.001:min_hard_comp=0.100000.
Output #0, mpeg, to '/media/data/jtest/dog-3a_1080i.mpg':
  Metadata:
    encoder         : Lavf54.28.101
    Stream #0:0: Video: mpeg2video, yuv422p, 1920x1080, q=1-3, 50000 kb/s,
90k tbn, 29.97 tbc
    Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 224 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> mpeg2video)
  Stream #0:1 -> #0:1 (pcm_s16le -> mp2)
Press [q] to stop, [?] for help
[mpeg2video @ 0x28b6800] rc buffer underflow
    Last message repeated 24 times
frame=   25 fps=0.0 q=24.8 size=    6190kB time=00:00:00.83
bitrate=60789.4kbitsrc buffer underflow
[mpeg2video @ 0x28b6800] rc buffer underflow
    Last message repeated 25 times
frame=   52 fps= 51 q=24.8 size=   13722kB time=00:00:01.73
bitrate=64787.5kbitsrc buffer underflow
[mpeg2video @ 0x28b6800] rc buffer underflow
    Last message repeated 25 times
frame=   79 fps= 52 q=24.8 size=   21908kB time=00:00:02.63
bitrate=68085.2kbitsrc buffer underflow
[mpeg2video @ 0x28b6800] rc buffer underflow
    Last message repeated 25 times
frame=  106 fps= 52 q=24.8 size=   29746kB time=00:00:03.53
bitrate=68896.9kbitsrc buffer underflow
[mpeg2video @ 0x28b6800] rc buffer underflow
    Last message repeated 24 times
frame=  132 fps= 52 q=24.8 size=   37322kB time=00:00:04.40
bitrate=69417.4kbitsrc buffer underflow
[mpeg2video @ 0x28b6800] rc buffer underflow
    Last message repeated 17 times
frame=  151 fps= 52 q=24.8 Lsize=   44048kB time=00:00:05.03
bitrate=71618.7kbits/s
video:43755kB audio:138kB subtitle:0 global headers:0kB muxing overhead
0.353431%


I notice the
-async is forwarded to lavfi similarly to -af
aresample=min_comp=0.001:min_hard_comp=0.100000.
I am uncertain as to the wisdom of that....

This cmd does make a file ; and I think it is ok except for the audio sync
problem
note: i added the -async 1 in attempt to correct the sync.

Thank you


From cehoyos at ag.or.at  Fri Aug  2 20:00:38 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 2 Aug 2013 18:00:38 +0000 (UTC)
Subject: [FFmpeg-user] Preserving preserved loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
References: <000601ce8f89$e60c9980$b225cc80$@gmail.com>
Message-ID: <loom.20130802T193936-710@post.gmane.org>

Francois Visagie <francois.visagie <at> gmail.com> writes:

> . output from '-ac 2' is noticeably softer than the original
>
> . '-filter:a aformat=channel_layouts=stereo' seems to 
> have the same volume as the original
> 
> . '-filter:a_aresample=ocl=3' is noticeably softer 
> than the original

I tested the following three command lines with random 
5.1 ac3 input:
$ ffmpeg -i input -ac 2 out1.wav
$ ffmpeg -i input -filter:a format=channel_layouts=stereo out2.wav
$ ffmpeg -i input -filter:a aresample=ocl=3 out3.wav

As expected, the three output files are bit-identical.
(Both format and ac 2 auto-insert the aresample filter 
as loglevel debug will tell you.)

The following makes a difference, it does only work 
for certain input codecs though (not aac):
$ ffmpeg -request_channels 2 -i input out4.wav

Carl Eugen


From phpdev at ehrhardt.nl  Fri Aug  2 20:21:57 2013
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Fri, 02 Aug 2013 20:21:57 +0200
Subject: [FFmpeg-user] pkg-config --static ?
References: <CAL1QdWcNmZPRWkjx1iq223Orriu9rAerunmu_+kaMHxO+qXvdg@mail.gmail.com>
	<loom.20120910T232409-529@post.gmane.org>
	<CAL1QdWcvnHg_hb_cSJ+x=9bcTvJnkpS+D=wP59rGXhU6t1_oQA@mail.gmail.com>
Message-ID: <bkrnv8lqhioq7uc2mb98qlek2lho6k2tsd@4ax.com>

Roger Pack in gmane.comp.video.ffmpeg.user (Tue, 18 Sep 2012 17:18:57
-0600):
>Apparently nobody (including myself) really understand pkg-config.

I do not either...

>Here is an answer from fontconfig people:
>
>> I guess my only question mark here is "since I built fontconfig with
>> --disable-shared --enable-static" would you say that fontconfig is
>> still correct here?
>
>>AFAICT It's pkg-config's design no matter what the library has been built with.
>
>I'm really not sure how FFmpeg is supposed to "know" it's linking
>against a static library, and thus know to pass the "--static" flag to
>pkg-config.

I was having troubles today compiling fontconfig for Windows (cross
compile on Ubuntu) and ran into the same errors as you. Finally Google
pointed me to your remarks on this subject.

I made a special pair of require_pkg_config() and check_pkg_config()
functions for --enable-fontconfig in FFMpeg's configure, which did (for
fontconfig only) nothing more than

-    pkg_libs=$($pkg_config --libs $pkg)
+    pkg_libs=$($pkg_config --static --libs $pkg)

No more errors and finally a ffmpeg.exe that embeds subtitles in the
video. Weird issue, but thanks for reporting this almost a year ago!

Jan


From mnileshgupta at gmail.com  Fri Aug  2 21:24:28 2013
From: mnileshgupta at gmail.com (Nilesh Gupta)
Date: Sat, 3 Aug 2013 00:54:28 +0530
Subject: [FFmpeg-user] ffmpeg help
In-Reply-To: <CAFXmZcYeWVXxHoBdQHx_N6V9Ox2UD+9z90K6SGTYUYQ6MFCzGQ@mail.gmail.com>
References: <CAFXmZcYeWVXxHoBdQHx_N6V9Ox2UD+9z90K6SGTYUYQ6MFCzGQ@mail.gmail.com>
Message-ID: <CAEc8cJC-rn1VoYRy99FjcmQzNfASX+hwqLDV21Z_1HS+3tFKPg@mail.gmail.com>

Hi,
First you need dedicated server for ffmpeg which should be installed ffmpeg
properly

then use this command:
ffmpeg -f image2 -r 1/5 -i img%03d.png -vcodec libx264 out.mp4

It will take files as example: img001.png, img002.png, img003.png

Thanks.




On Thu, Aug 1, 2013 at 6:53 PM, bhavesh bodar <bhavesh.vthink at gmail.com>wrote:

> Dear sir,
>
>                i want to merge tow images or video using php with ffmpeg. i
> have successfully installed ffmpeg in my localhost wamp server also i can
> convert video usigng ffmpeg in php
>        but i can not merge two image or video so i want to complete php
> code of merge two images or video.
>
> Thanx
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>



-- 
Best Regards,
*Nilesh Gupta * | Web Developer (PHP) *
*Handphone:- +91 8347872125*
*Home Town:- Jabalpur (M.P.) | Current Location:- Vadodara (G.J.)
* *

From thilo.borgmann at googlemail.com  Fri Aug  2 22:28:45 2013
From: thilo.borgmann at googlemail.com (Thilo Borgmann)
Date: Fri, 02 Aug 2013 22:28:45 +0200
Subject: [FFmpeg-user] FFMPEG decoding Dailymotion Videos
In-Reply-To: <CAHr_b4G0iuM7Vxv0a42mWpAVGF8OtV2EmvBfwzbCdCkWWoPgYg@mail.gmail.com>
References: <CAHr_b4G0iuM7Vxv0a42mWpAVGF8OtV2EmvBfwzbCdCkWWoPgYg@mail.gmail.com>
Message-ID: <51FC167D.6060407@googlemail.com>

Am 02.08.13 05:40, schrieb JT Edwards:
> Some videos I can, and some videos I cant... Weird aint it?

> ffmpeg -i
> http://www.dailymotion.com/cdn/H264-512x384/video/x3n3tn.mp4?auth=1375587534-9675cf2684c10d505ac89357ab3060a1
> [...]
> [http @ 0x235d600] HTTP error 404 Not Found
> http://www.dailymotion.com/cdn/H264-512x384/video/x3n3tn.mp4?auth=1375587534-9675cf2684c10d505ac89357ab3060a1:
> Input/output error

are there differences in the URL?

404 error comes from the server, for example if there are further arguments to
the URL (http://.../x3n3tn.mp4?<arg1>&<arg2>&<arg3>) etc. that are cut before
calling ffmpeg.

If so, correct quoting of the URL can help because shells tend to cut at
plain-text '&'.

-Thilo


From tstrike34 at gmail.com  Fri Aug  2 22:54:20 2013
From: tstrike34 at gmail.com (JT Edwards)
Date: Fri, 2 Aug 2013 15:54:20 -0500
Subject: [FFmpeg-user] FFMPEG decoding Dailymotion Videos
In-Reply-To: <51FC167D.6060407@googlemail.com>
References: <CAHr_b4G0iuM7Vxv0a42mWpAVGF8OtV2EmvBfwzbCdCkWWoPgYg@mail.gmail.com>
	<51FC167D.6060407@googlemail.com>
Message-ID: <CAHr_b4EB9epg_5VG8wOn=DVkD8Sj3HuDFJNQ_QvAWc7z0CDSzA@mail.gmail.com>

Ok I will get those and circle back to you guys.

Thanks for the help (very cool friendly list).


On Fri, Aug 2, 2013 at 3:28 PM, Thilo Borgmann <
thilo.borgmann at googlemail.com> wrote:

> Am 02.08.13 05:40, schrieb JT Edwards:
> > Some videos I can, and some videos I cant... Weird aint it?
>
> > ffmpeg -i
> >
> http://www.dailymotion.com/cdn/H264-512x384/video/x3n3tn.mp4?auth=1375587534-9675cf2684c10d505ac89357ab3060a1
> > [...]
> > [http @ 0x235d600] HTTP error 404 Not Found
> >
> http://www.dailymotion.com/cdn/H264-512x384/video/x3n3tn.mp4?auth=1375587534-9675cf2684c10d505ac89357ab3060a1
> :
> > Input/output error
>
> are there differences in the URL?
>
> 404 error comes from the server, for example if there are further
> arguments to
> the URL (http://.../x3n3tn.mp4?<arg1>&<arg2>&<arg3>) etc. that are cut
> before
> calling ffmpeg.
>
> If so, correct quoting of the URL can help because shells tend to cut at
> plain-text '&'.
>
> -Thilo
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cmharwoo at umich.edu  Sat Aug  3 01:59:13 2013
From: cmharwoo at umich.edu (cmharwood89)
Date: Fri, 2 Aug 2013 16:59:13 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
Message-ID: <1375487953644-4660437.post@n4.nabble.com>

Hi all,

I'm a fairly new ffmpeg user, but I've found it really helpful for
processing videos as part of my graduate research. I've got a mess of videos
(recordings of experiments from different angles) that I'm trying to
synchronize. I coded my data acquisition system such that an LED turns on
when I begin collecting data, giving me a visual indicator of where I can
sync the videos, which should be accurate to within ~1/2 frame.

So I've got a script to trim my videos to the frames at which I see the LED
turn on (in HH:MM:SS[.ms] format). I tried it with -vcodec copy, but
(predictably) I found my outputs began at the nearest I-frame, up to 1s away
from my desired trim point.

My understanding is that, if I re-encode, I should be able to trim at
exactly the desired time and place an I-frame at the beginning of the saved
output. An example command is using a gopro video shot at 59.94 fps, with
h.264 codec:

ffmpeg -t 30 -i GOPR0103.mp4 -vcodec libx264 ?threads 0 -preset ultrafast
-an -y -ss 00:03:09[.606] Run041_UW1.avi

Here's where I'm confused. If I execute the above command, I do get a
trimmed output, but the resulting video starts 40 frames away from the
desired frame. In contrast, if I use -vcodec copy on the same file, the
output is only off by ~26 frames.

What am I doing wrong here? Is there any way for me to get frame-accurate
splitting? I'm okay with re-encoding if that's what it takes. I have about
800 videos to chew through, but I can run my lab workstation for a week
straight to get it done.

Thanks very much!



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Inaccurate-trimming-with-re-encoding-tp4660437.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From mythtv at american-hero.com  Sat Aug  3 03:01:55 2013
From: mythtv at american-hero.com (Larry Roberts)
Date: Fri, 02 Aug 2013 21:01:55 -0400
Subject: [FFmpeg-user] getting latest version of ffmpeg to work with Roku
Message-ID: <51FC5683.9090405@american-hero.com>

Trying to use ffmpeg to convert videos to a roku compatible format. I 
have a script that has worked fine for the better part of 6 months, but 
I just upgraded from an older machine to a new virtual machine and along 
with this upgraded my ffmpeg version.
I went from what I believe is 0.4.8 (see below) to the latest distro 
version.

## Working Version ##
ffmpeg version N-48402-gaa86d2d
built on Jan  2 2013 18:15:58 with gcc 4.4.6 (GCC) 20120305 (Red Hat 
4.4.6-4)
configuration: --enable-gpl --enable-libmp3lame --enable-libx264 
--enable-libfaac --enable-nonfree
libavutil      52. 13.100 / 52. 13.100
libavcodec     54. 85.100 / 54. 85.100
libavformat    54. 59.100 / 54. 59.100
libavdevice    54.  3.102 / 54.  3.102
libavfilter     3. 30.102 /  3. 30.102
libswscale      2.  1.103 /  2.  1.103
libswresample   0. 17.102 /  0. 17.102
libpostproc    52.  2.100 / 52.  2.100

## Latest Distro version ##
ffmpeg version 1.2.1
built on May 10 2013 15:14:14 with gcc 4.4.6 (GCC) 20120305 (Red Hat 
4.4.6-4)
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 
--mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect 
--enable-gpl --enable-version3 --enable-postproc --enable-avfilter 
--enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth 
--enable-frei0r --enable-libopencv --enable-libdc1394 --enable-libgsm 
--enable-libmp3lame --enable-libnut --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp 
--enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx 
--enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 
-g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector 
--param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping
libavutil      52. 18.100 / 52. 18.100
libavcodec     54. 92.100 / 54. 92.100
libavformat    54. 63.104 / 54. 63.104
libavdevice    54.  3.103 / 54.  3.103
libavfilter     3. 42.103 /  3. 42.103
libswscale      2.  2.100 /  2.  2.100
libswresample   0. 17.102 /  0. 17.102
libpostproc    52.  2.100 / 52.  2.100


While no changes have been made to the script , a once working solution 
now produces videos that wont work on the Roku.  In googling I found 
some emails that seem to point to anything 5.10 or before as working fine.

What I would like to do is see if I can compile the latest version from 
source and get this to work, or perhaps go back and compile the 5.10 
version.  I did what I thought was a successful build of ffmpeg but most 
of the options that were working are now longer available (presets)

So I'm looking for a little guidance.  I'm trying to find a proper 
support forum for Roku as well since I think its a problem on their end 
most likely (the videos play fine on an iPAD or computer) but I thought 
I would ask here as well in case someone has some guidance to offer.

Thanks for any advice.

Larry





From h.reindl at thelounge.net  Sat Aug  3 10:37:04 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Sat, 03 Aug 2013 10:37:04 +0200
Subject: [FFmpeg-user] getting latest version of ffmpeg to work with Roku
In-Reply-To: <51FC5683.9090405@american-hero.com>
References: <51FC5683.9090405@american-hero.com>
Message-ID: <51FCC130.3050001@thelounge.net>



Am 03.08.2013 03:01, schrieb Larry Roberts:
> I went from what I believe is 0.4.8 (see below) to the latest distro version.
> 
> ## Working Version ##
> ffmpeg version N-48402-gaa86d2d.100
> 
> ## Latest Distro version ##
> ffmpeg version 1.2.1
>
> While no changes have been made to the script , a once working solution now produces videos that wont work on the
> Roku. In googling I found some emails that seem to point to anything 5.10 or before as working fine.
> 
> What I would like to do is see if I can compile the latest version from source and get this to work, or perhaps go
> back and compile the 5.10 version.  I did what I thought was a successful build of ffmpeg but most of the options
> that were working are now longer available (presets)
> 
> So I'm looking for a little guidance

you need to provide the full command line and uncutted output

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From francois.visagie at gmail.com  Sat Aug  3 11:04:11 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Sat, 3 Aug 2013 11:04:11 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
Message-ID: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos
> Sent: 02 August 2013 20:01
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] Preserving preserved loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Francois Visagie <francois.visagie <at> gmail.com> writes:
> 
> > . output from '-ac 2' is noticeably softer than the original
> >
> > . '-filter:a aformat=channel_layouts=stereo' seems to
> > have the same volume as the original
> >
> > . '-filter:a_aresample=ocl=3' is noticeably softer
> > than the original
> 
> I tested the following three command lines with random
> 5.1 ac3 input:
> $ ffmpeg -i input -ac 2 out1.wav
> $ ffmpeg -i input -filter:a format=channel_layouts=stereo out2.wav

You mean aformat?

> $ ffmpeg -i input -filter:a aresample=ocl=3 out3.wav
> 
> As expected, the three output files are bit-identical.

Thanks for doing this testing, which I repeated here.

With
http://ffmpeg.zeranoe.com/builds/win32/static/ffmpeg-20130630-git-ff130d7-wi
n32-static.7z which I was using up to now, out2.wav differs from the other
two. However, it's the only one equally loud as the input file.

With
http://ffmpeg.zeranoe.com/builds/win32/static/ffmpeg-20130801-git-f118b41-wi
n32-static.7z the files are bit-identical as you found. However, all three
are now significantly softer than the original.

I don't think this is the desired outcome, to achieve unified audio
processing but with significant reduction in perceived volume in the
process. Is there any further testing I can do to help investigate and
hopefully improve this situation?

> (Both format and ac 2 auto-insert the aresample filter
> as loglevel debug will tell you.)
> 
> The following makes a difference, it does only work
> for certain input codecs though (not aac):
> $ ffmpeg -request_channels 2 -i input out4.wav
> 
> Carl Eugen
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From jhbell50 at hotmail.com  Sat Aug  3 17:02:25 2013
From: jhbell50 at hotmail.com (john Bell)
Date: Sat, 3 Aug 2013 11:02:25 -0400
Subject: [FFmpeg-user] Submitting bugs
Message-ID: <BLU178-W17DB303926DB4D5698892BCF520@phx.gbl>

Can anyone help me with advice on how to get approval to submit bugs?
I have enrolled in Trac and have an id, but no approval email has comeback despite multiple requests to "resend the email".   As a result I cannot submit a bug until my email address has been confirmed.
 		 	   		  

From cehoyos at ag.or.at  Sat Aug  3 22:57:17 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 3 Aug 2013 20:57:17 +0000 (UTC)
Subject: [FFmpeg-user] Submitting bugs
References: <BLU178-W17DB303926DB4D5698892BCF520@phx.gbl>
Message-ID: <loom.20130803T225525-259@post.gmane.org>

john Bell <jhbell50 <at> hotmail.com> writes:

> Can anyone help me with advice on how to get 
> approval to submit bugs? I have enrolled in Trac 
> and have an id, but no approval email has 
> comeback despite multiple requests to "resend 
> the email".   As a result I cannot submit a bug 
> until my email address has been confirmed.

Please consider reporting the problem here 
on the mailing list.

Carl Eugen




From mythtv at american-hero.com  Sat Aug  3 23:34:43 2013
From: mythtv at american-hero.com (Larry Roberts)
Date: Sat, 03 Aug 2013 17:34:43 -0400
Subject: [FFmpeg-user] getting latest version of ffmpeg to work with Roku
In-Reply-To: <51FCC130.3050001@thelounge.net>
References: <51FC5683.9090405@american-hero.com>
	<51FCC130.3050001@thelounge.net>
Message-ID: <51FD7773.6040609@american-hero.com>

On 8/3/2013 4:37 AM, Reindl Harald wrote:
>
> Am 03.08.2013 03:01, schrieb Larry Roberts:
>> I went from what I believe is 0.4.8 (see below) to the latest distro version.
>>
>> ## Working Version ##
>> ffmpeg version N-48402-gaa86d2d.100
>>
>> ## Latest Distro version ##
>> ffmpeg version 1.2.1
>>
>> While no changes have been made to the script , a once working solution now produces videos that wont work on the
>> Roku. In googling I found some emails that seem to point to anything 5.10 or before as working fine.
>>
>> What I would like to do is see if I can compile the latest version from source and get this to work, or perhaps go
>> back and compile the 5.10 version.  I did what I thought was a successful build of ffmpeg but most of the options
>> that were working are now longer available (presets)
>>
>> So I'm looking for a little guidance
> you need to provide the full command line and uncutted output
>
>
So, in the process of crafting your requested info I discovered 
something that is pointing me to a new direction.

I went from encoding videos that were sourced from a PVR-350, to 
encoding video sourced from an HDHomeRun box.

I just fired up the old box and ran the old ffmpeg against a new 
recording and it DOESN'T work on the Roku.  I then ran the new ffmpeg 
against an old recording from the PVR-350 and it DOES work.

I hadn't bothered to try to convert any older stuff as it was already 
converted.   next step had me thinking about SD vs. HD. What I found is 
that I can convert new SD recordings and so far they work, while new HD 
recordings dont on the ROKU.
I should again point out that I can play either just fine on a pc/iPAD....

HD conversion output

[root at mythbe1 bin]# ffmpeg -i /mythtvstorage/2013_20130802170000.mpg   
-vcodec libx264  -s 640x480 -ss 00:00:00 -t 00:00:15 -preset ultrafast 
-crf 23 -acodec aac -strict -2 -ab 128k /roku/test6.m4v
ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
   built on May 10 2013 15:14:14 with gcc 4.4.6 (GCC) 20120305 (Red Hat 
4.4.6-4)
   configuration: --prefix=/usr --libdir=/usr/lib64 
--shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared 
--enable-runtime-cpudetect --enable-gpl --enable-version3 
--enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab 
--enable-vdpau --disable-avisynth --enable-frei0r --enable-libopencv 
--enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libnut 
--enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenjpeg --enable-librtmp --enable-libspeex 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 
--enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall 
-Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector 
--param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping
   libavutil      52. 18.100 / 52. 18.100
   libavcodec     54. 92.100 / 54. 92.100
   libavformat    54. 63.104 / 54. 63.104
   libavdevice    54.  3.103 / 54.  3.103
   libavfilter     3. 42.103 /  3. 42.103
   libswscale      2.  2.100 /  2.  2.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  2.100 / 52.  2.100
[mpegts @ 0x978a40] PES packet size mismatch
Input #0, mpegts, from '/mythtvstorage/2013_20130802170000.mpg':
   Duration: 00:59:59.47, start: 48658.910067, bitrate: 11381 kb/s
   Program 1
     Stream #0:0[0x7c0]: Video: mpeg2video (Main) ([2][0][0][0] / 
0x0002), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 
90k tbn, 59.94 tbc
     Stream #0:1[0x7c1](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 
5.1(side), fltp, 384 kb/s
     Stream #0:2[0x7c2](spa): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 
mono, fltp, 96 kb/s (visual impaired)
[libx264 @ 0x97f040] using SAR=4/3
[libx264 @ 0x97f040] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x97f040] profile Constrained Baseline, level 3.0
[libx264 @ 0x97f040] 264 - core 130 - H.264/MPEG-4 AVC codec - Copyleft 
2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 
deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 
mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 
deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=1 
lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 
bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 
keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 
qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
Output #0, ipod, to '/roku/test6.m4v':
   Metadata:
     encoder         : Lavf54.63.104
     Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p, 640x480 [SAR 
4:3 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc
     Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 
5.1(side), fltp, 128 kb/s
Stream mapping:
   Stream #0:0 -> #0:0 (mpeg2video -> libx264)
   Stream #0:1 -> #0:1 (ac3 -> aac)
Press [q] to stop, [?] for help
[mpeg2video @ 0x97ca40] warning: first frame is no keyframe
frame=  450 fps= 76 q=22.0 Lsize=    3544kB time=00:00:15.01 
bitrate=1933.1kbits/s dup=81 drop=0
video:3400kB audio:130kB subtitle:0 global headers:0kB muxing overhead 
0.398015%
[libx264 @ 0x97f040] frame I:2     Avg QP:20.00  size: 36056
[libx264 @ 0x97f040] frame P:448   Avg QP:22.27  size:  7608
[libx264 @ 0x97f040] mb I  I16..4: 100.0%  0.0%  0.0%
[libx264 @ 0x97f040] mb P  I16..4: 10.6%  0.0%  0.0%  P16..4: 48.2% 
0.0%  0.0%  0.0%  0.0%    skip:41.2%
[libx264 @ 0x97f040] coded y,uvDC,uvAC intra: 30.7% 20.0% 9.4% inter: 
27.2% 12.8% 2.2%
[libx264 @ 0x97f040] i16 v,h,dc,p: 54% 27% 10% 10%
[libx264 @ 0x97f040] i8c dc,h,v,p: 64% 14% 19%  2%
[libx264 @ 0x97f040] kb/s:1854.52

Working SD conversion

[root at mythbe1 bin]# ffmpeg -i /mythtvstorage/1029_20130802093000.mpg   
-vcodec libx264  -s 640x480 -ss 00:00:00 -t 00:00:30 -preset ultrafast 
-crf 23 -acodec aac -strict -2 -ab 128k /roku/test5.m4v
ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
   built on May 10 2013 15:14:14 with gcc 4.4.6 (GCC) 20120305 (Red Hat 
4.4.6-4)
   configuration: --prefix=/usr --libdir=/usr/lib64 
--shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared 
--enable-runtime-cpudetect --enable-gpl --enable-version3 
--enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab 
--enable-vdpau --disable-avisynth --enable-frei0r --enable-libopencv 
--enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libnut 
--enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenjpeg --enable-librtmp --enable-libspeex 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 
--enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall 
-Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector 
--param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping
   libavutil      52. 18.100 / 52. 18.100
   libavcodec     54. 92.100 / 54. 92.100
   libavformat    54. 63.104 / 54. 63.104
   libavdevice    54.  3.103 / 54.  3.103
   libavfilter     3. 42.103 /  3. 42.103
   libswscale      2.  2.100 /  2.  2.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  2.100 / 52.  2.100
[mpegts @ 0x1f9ca40] max_analyze_duration 5000000 reached at 5024000 
microseconds
[mpegts @ 0x1f9ca40] Could not find codec parameters for stream 3 
(Unknown: none ([134][0][0][0] / 0x0086)): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' 
options
[NULL @ 0x1fa2840] start time is not set in estimate_timings_from_pts
Input #0, mpegts, from '/mythtvstorage/1029_20130802093000.mpg':
   Duration: 00:29:59.10, start: 55495.128611, bitrate: 2589 kb/s
   Program 1
     Stream #0:0[0x8c0]: Video: mpeg2video (Main) ([2][0][0][0] / 
0x0002), yuv420p, 704x480 [SAR 10:11 DAR 4:3], 29.97 fps, 59.94 tbr, 90k 
tbn, 59.94 tbc
     Stream #0:1[0x8c1](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 
48000 Hz, stereo, fltp, 192 kb/s
     Stream #0:2[0x8c2](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 
48000 Hz, stereo, fltp, 192 kb/s
     Stream #0:3[0x8c3]: Unknown: none ([134][0][0][0] / 0x0086)
File '/roku/test5.m4v' already exists. Overwrite ? [y/N] y
using SAR=1/1
[libx264 @ 0x1fa4300] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x1fa4300] profile Constrained Baseline, level 3.0
[libx264 @ 0x1fa4300] 264 - core 130 - H.264/MPEG-4 AVC codec - Copyleft 
2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 
deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 
mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 
deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=1 
lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 
bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 
keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 
qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
Output #0, ipod, to '/roku/test5.m4v':
   Metadata:
     encoder         : Lavf54.63.104
     Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p, 640x480 [SAR 
1:1 DAR 4:3], q=-1--1, 30k tbn, 29.97 tbc
     Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 
fltp, 128 kb/s
Stream mapping:
   Stream #0:0 -> #0:0 (mpeg2video -> libx264)
   Stream #0:1 -> #0:1 (ac3 -> aac)
Press [q] to stop, [?] for help
frame=  900 fps=150 q=20.0 Lsize=    5903kB time=00:00:30.03 
bitrate=1610.2kbits/s dup=153 drop=0
video:5401kB audio:474kB subtitle:0 global headers:0kB muxing overhead 
0.469127%
[libx264 @ 0x1fa4300] frame I:4     Avg QP:15.50  size: 59440
[libx264 @ 0x1fa4300] frame P:896   Avg QP:19.68  size:  5906
[libx264 @ 0x1fa4300] mb I  I16..4: 100.0%  0.0%  0.0%
[libx264 @ 0x1fa4300] mb P  I16..4:  6.7%  0.0%  0.0%  P16..4: 31.2%  
0.0%  0.0%  0.0%  0.0%    skip:62.1%
[libx264 @ 0x1fa4300] coded y,uvDC,uvAC intra: 31.0% 63.9% 28.6% inter: 
15.1% 19.3% 5.4%
[libx264 @ 0x1fa4300] i16 v,h,dc,p: 33% 35% 16% 16%
[libx264 @ 0x1fa4300] i8c dc,h,v,p: 34% 33% 18% 15%
[libx264 @ 0x1fa4300] kb/s:1473.17

What do I need to tweak to get HD conversions working?  I have NO idea 
what I should even be testing at this point...



Thanks!

>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From mythtv at american-hero.com  Sun Aug  4 00:53:32 2013
From: mythtv at american-hero.com (Larry Roberts)
Date: Sat, 03 Aug 2013 18:53:32 -0400
Subject: [FFmpeg-user] getting latest version of ffmpeg to work with Roku
In-Reply-To: <51FD7773.6040609@american-hero.com>
References: <51FC5683.9090405@american-hero.com>
	<51FCC130.3050001@thelounge.net>
	<51FD7773.6040609@american-hero.com>
Message-ID: <51FD89EC.5000101@american-hero.com>

adding some additional info on files that work and dont work

File 1 that DOESNT work
[root at mythbe1 bin]# ffprobe /roku/test5.m4v
ffprobe version 1.2.1 Copyright (c) 2007-2013 the FFmpeg developers
   built on May 10 2013 15:14:14 with gcc 4.4.6 (GCC) 20120305 (Red Hat 
4.4.6-4)
   configuration: --prefix=/usr --libdir=/usr/lib64 
--shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared 
--enable-runtime-cpudetect --enable-gpl --enable-version3 
--enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab 
--enable-vdpau --disable-avisynth --enable-frei0r --enable-libopencv 
--enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libnut 
--enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenjpeg --enable-librtmp --enable-libspeex 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 
--enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall 
-Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector 
--param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping
   libavutil      52. 18.100 / 52. 18.100
   libavcodec     54. 92.100 / 54. 92.100
   libavformat    54. 63.104 / 54. 63.104
   libavdevice    54.  3.103 / 54.  3.103
   libavfilter     3. 42.103 /  3. 42.103
   libswscale      2.  2.100 /  2.  2.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  2.100 / 52.  2.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/roku/test5.m4v':
   Metadata:
     major_brand     : M4V
     minor_version   : 512
     compatible_brands: isomiso2avc1
     encoder         : Lavf54.63.104
   Duration: 00:00:15.04, start: 0.021333, bitrate: 8193 kb/s
     Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 
0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 8128 kb/s, 29.97 
fps, 29.97 tbr, 30k tbn, 59.94 tbc
     Metadata:
       handler_name    : VideoHandler
     Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, 
fltp, 70 kb/s
     Metadata:
       handler_name    : SoundHandler
file 2 that DOESNT work
[root at mythbe1 bin]# ffprobe /roku/test12.mp4
ffprobe version 1.2.1 Copyright (c) 2007-2013 the FFmpeg developers
   built on May 10 2013 15:14:14 with gcc 4.4.6 (GCC) 20120305 (Red Hat 
4.4.6-4)
   configuration: --prefix=/usr --libdir=/usr/lib64 
--shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared 
--enable-runtime-cpudetect --enable-gpl --enable-version3 
--enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab 
--enable-vdpau --disable-avisynth --enable-frei0r --enable-libopencv 
--enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libnut 
--enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenjpeg --enable-librtmp --enable-libspeex 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 
--enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall 
-Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector 
--param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping
   libavutil      52. 18.100 / 52. 18.100
   libavcodec     54. 92.100 / 54. 92.100
   libavformat    54. 63.104 / 54. 63.104
   libavdevice    54.  3.103 / 54.  3.103
   libavfilter     3. 42.103 /  3. 42.103
   libswscale      2.  2.100 /  2.  2.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  2.100 / 52.  2.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/roku/test12.mp4':
   Metadata:
     major_brand     : isom
     minor_version   : 512
     compatible_brands: isomiso2avc1mp41
     encoder         : Lavf54.63.104
   Duration: 00:00:15.04, start: 0.021333, bitrate: 669 kb/s
     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 
768x432 [SAR 1:1 DAR 16:9], 632 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 
59.94 tbc
     Metadata:
       handler_name    : VideoHandler
     Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, 
fltp, 28 kb/s
     Metadata:
       handler_name    : SoundHandler
[root at mythbe1 bin]


file that DOES work

[root at mythbe1 bin]# ffprobe /roku/Untitled_8-1\ -\ airvideo.m4v
ffprobe version 1.2.1 Copyright (c) 2007-2013 the FFmpeg developers
   built on May 10 2013 15:14:14 with gcc 4.4.6 (GCC) 20120305 (Red Hat 
4.4.6-4)
   configuration: --prefix=/usr --libdir=/usr/lib64 
--shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared 
--enable-runtime-cpudetect --enable-gpl --enable-version3 
--enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab 
--enable-vdpau --disable-avisynth --enable-frei0r --enable-libopencv 
--enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libnut 
--enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenjpeg --enable-librtmp --enable-libspeex 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 
--enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall 
-Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector 
--param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping
   libavutil      52. 18.100 / 52. 18.100
   libavcodec     54. 92.100 / 54. 92.100
   libavformat    54. 63.104 / 54. 63.104
   libavdevice    54.  3.103 / 54.  3.103
   libavfilter     3. 42.103 /  3. 42.103
   libswscale      2.  2.100 /  2.  2.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  2.100 / 52.  2.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/roku/Untitled_8-1 - airvideo.m4v':
   Metadata:
     major_brand     : isom
     minor_version   : 512
     compatible_brands: isomiso2avc1mp41
     encoder         : Lavf54.9.0
   Duration: 00:59:58.20, start: 0.000000, bitrate: 1767 kb/s
     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 
768x432 [SAR 1:1 DAR 16:9], 1594 kb/s, 29.97 fps, 29.97 tbr, 2997 tbn, 
59.94 tbc
     Metadata:
       handler_name    : VideoHandler
     Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 
fltp, 165 kb/s
     Metadata:
       handler_name    : SoundHandler
[root at mythbe1 bin]#


From blacktrash at gmx.net  Sun Aug  4 11:37:42 2013
From: blacktrash at gmx.net (Christian Ebert)
Date: Sun, 4 Aug 2013 10:37:42 +0100
Subject: [FFmpeg-user] De-interlacing and Inverse Telecining Questions
In-Reply-To: <loom.20130802T002600-665@post.gmane.org>
References: <91574A7267144E66BC9820C75A530CB7@gmail.com>
	<loom.20130802T002600-665@post.gmane.org>
Message-ID: <20130804093742.GD1508@krille.blacktrash.org>

* Carl Eugen Hoyos on Thursday, August 01, 2013 at 22:34:29 +0000
> Not really because while it is technically possible to 
> detect if your video is interlaced or telecined (see 
> the idet filter), it is significantly easier and less 
> error-prone if you do it (visually) before transcoding.

I know - not least thanks to your help - how to detect interlaced
content with idet, how would I go about detecting telecined
content?

thx

-- 
theatre - books - texts - movies
Black Trash Productions at home: http://www.blacktrash.org
Black Trash Productions on Facebook:
http://www.facebook.com/blacktrashproductions

From adf.lists at gmail.com  Sun Aug  4 22:11:29 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sun, 04 Aug 2013 21:11:29 +0100
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
Message-ID: <51FEB571.2080701@gmail.com>

Francois Visagie wrote:

> I don't think this is the desired outcome, to achieve unified audio
> processing but with significant reduction in perceived volume in the
> process. Is there any further testing I can do to help investigate and
> hopefully improve this situation?

Down mixing is not ideal. Lower volume is expected.

If you don't want the result to clip then you have to normalise enough 
to allow for all input channels being full level - even though that may 
be rare in practice.

As you are dealing with ac3 you should also be aware that a studio 
produced stream will usually carry mata data specifying dynamic range 
control and by default ffmpeg applies this fully so you get the 
restricted the range output. Just turning this off with

-drc_scale 0

will however produce something that IMHO has too much range - it's not 
actually on/off it's float so you can do in between 0 and 1.

ac3 also has meta data for downmix so using -request_channels 2 should 
in theory get a downmix as the studio intended - but of course it's 
still normalised to prevent clipping and it's still compromised by the 
fact it's a downmix.

I don't know if there's a way with ffmpeg to analyse and adjust the 
whole downmixed track to boost the levels to take advantage of any headroom.

You can certainly do it with sox (working on pcm)

Slight digression - last time I looked at dts/dca codecs default downmix 
it was a clipping stereo squashing mess - nice and loud though, but 
really. don't think it's OK or anything to aim for should you happen to 
ever test.

Not much use for your need, but as for the future - the nice thing about 
Dolby Truehd and DTS MA is that they mix up, so a stereo user like me 
should get an "artistic" studio mix rather than something kludged down.

AFAIK dts ma isn't supported yet truehd is (but no drc support) - and 
this is just decode.







From phpdev at ehrhardt.nl  Mon Aug  5 00:00:13 2013
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Mon, 05 Aug 2013 00:00:13 +0200
Subject: [FFmpeg-user] pkg-config --static ?
References: <CAL1QdWcNmZPRWkjx1iq223Orriu9rAerunmu_+kaMHxO+qXvdg@mail.gmail.com>
	<loom.20120910T232409-529@post.gmane.org>
	<CAL1QdWcvnHg_hb_cSJ+x=9bcTvJnkpS+D=wP59rGXhU6t1_oQA@mail.gmail.com>
Message-ID: <rbjtv810kd8l6gr83ocafa0q739us2f75h@4ax.com>

Roger Pack in gmane.comp.video.ffmpeg.user (Tue, 18 Sep 2012 17:18:57
-0600):
>Apparently nobody (including myself) really understand pkg-config.
>
>Here is an answer from fontconfig people:
>
>> I guess my only question mark here is "since I built fontconfig with
>> --disable-shared --enable-static" would you say that fontconfig is
>> still correct here?
>
>>AFAICT It's pkg-config's design no matter what the library has been built with.
>
>I'm really not sure how FFmpeg is supposed to "know" it's linking
>against a static library, and thus know to pass the "--static" flag to
>pkg-config.

To follow-up once again on this message: you can tell FFmpeg to use
--static for all libraries. Just put "pkg_config='pkg-config --static'"
at the beginning of your ./configure command-line.

See http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=5&t=4&start=10#p2173
and http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=5&t=4&start=20#p5338

Take a look at man pkg-config as well.

Jan


From jiachielee at live.com  Mon Aug  5 02:12:42 2013
From: jiachielee at live.com (thljcl)
Date: Sun, 4 Aug 2013 17:12:42 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375487953644-4660437.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
Message-ID: <1375661562225-4660448.post@n4.nabble.com>

Did you try to cut out a portion of video from a longer video?



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Inaccurate-trimming-with-re-encoding-tp4660437p4660448.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From feanor2112 at covad.net  Mon Aug  5 03:27:50 2013
From: feanor2112 at covad.net (Ned Haskin)
Date: Sun, 04 Aug 2013 21:27:50 -0400
Subject: [FFmpeg-user] ffmpeg Unknown encoder 'libx264' error
Message-ID: <1375666070.5067.41.camel@videoathlon>

Hi, I compiled x264-0.135, lame-3.99.5 & ffmpeg-2 on debian-7.1. No errors, lots of
warnings on ffmpeg's make, but no errors. I used --enable-shared on all 3. I used these pages as reference. 

https://trac.ffmpeg.org/wiki/How%20to%20quickly%20compile%20libx264 

https://trac.ffmpeg.org/wiki/How%20to%20quickly%20compile%20FFmpeg%20with%20mp3%2C%20aac%2B%20and%20x264 

See below for my ./configure for ffmpeg, I added a bunch of --enables- all were part of the deb compile. 
That I thought, I may need in the future. 

I get an error on trying to encode Unknown encoder 'libx264' 

ned337 at vid:~/video-final/bicycle$ ffmpeg -f concat -i bike02-2012.txt -acodec libfaac -ab 384k -ar 44100 -vcodec libx264 -s 720x480 -r 29.48 -aspect 4:3 -profile:v high444 -b:v 700 bike020212.mp4
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 29 2013 13:49:23 with gcc 4.7 (Debian 4.7.2-5)
  configuration: 
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 81.103 /  3. 81.103
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x2fad700] Stream #1: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x2fad700] Could not find codec parameters for stream 1 (Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 223 kb/s): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[concat @ 0x2fa4aa0] Stream #1: not enough frames to estimate rate; consider increasing probesize
[concat @ 0x2fa4aa0] Estimating duration from bitrate, this may be inaccurate
[concat @ 0x2fa4aa0] Could not find codec parameters for stream 1 (Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, 2 channels, s16p, 223 kb/s): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, concat, from 'bike02-2012.txt':
  Duration: N/A, start: 0.000000, bitrate: 244739 kb/s
    Stream #0:0: Video: rawvideo (raw  / 0x20776172), rgb24, 720x480, 244515 kb/s, 29.48 fps, 29.48 tbr, 29509 tbn, 29509 tbc
    Stream #0:1: Audio: mp3 (.mp3 / 0x33706D2E), 44100 Hz, stereo, s16p, 223 kb/s
Unknown encoder 'libx264'
ned337 at vid:~/video-final/bicycle$ 

So after confirming that make install actually put libx264 in /usr/local/lib/ 

ned337 at vid:~/x264-0135$ gksu make install
install -d /usr/local/bin
install x264 /usr/local/bin
install -d /usr/local/include
install -d /usr/local/lib
install -d /usr/local/lib/pkgconfig
install -m 644 ./x264.h /usr/local/include
install -m 644 x264_config.h /usr/local/include
install -m 644 x264.pc /usr/local/lib/pkgconfig
ln -f -s libx264.so.135 /usr/local/lib/libx264.so
install -m 755 libx264.so.135 /usr/local/lib
install -m 644 libx264.a /usr/local/lib
ranlib /usr/local/lib/libx264.a

I tried encoding to xvid that didn't work either. Same error just
libxvid unknown. Yes libxvid-dev is installed. Encoding to mjpeg works
with pcm audio. But, not with libmp3lame, libtwolame or libfaac. Yes
their -dev files are installed. 

This is my ffmpeg ./configure which ran without warnings or errors. 


ned337 at vid:~/ffmpeg$ make clean
ned337 at vid:~/ffmpeg$ ./configure --enable-libdc1394 --prefix=/usr
--extra-cflags='-Wall -g ' --enable-gpl --enable-shared --enable-nonfree
--enable-libx264  --enable-encoder=libx264 --enable-libmp3lame
--enable-libvorbis --enable-libtwolame --enable-libv4l2
--enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc
--enable-x11grab --enable-libtheora --disable-stripping
--enable-avfilter --enable-libfreetype --enable-opencl --enable-version3
--enable-libopenjpeg --disable-altivec --disable-armv5te --disable-armv6
--disable-vis 

# skipped some stuff

Enabled encoders:
a64multi		libopenjpeg		pgm
a64multi5		libtheora		pgmyuv
aac			libtwolame		png
ac3			libvorbis		ppm
ac3_fixed		libx264			prores
adpcm_adx		libx264rgb		prores_aw
adpcm_g722		libxvid			prores_ks
adpcm_g726		ljpeg			qtrle
adpcm_ima_qt		mjpeg			r10k
adpcm_ima_wav		movtext			r210

# skipped some stuff

License: nonfree and unredistributable
Creating config.mak, config.h, and doc/config.texi...
libavutil/avconfig.h is unchanged
ned337 at vid:~/ffmpeg$ 

No errors or warnings on ./configure 

ned337 at vid:~/ffmpeg$ gksu make
CC	libavdevice/alldevices.o
CC	libavdevice/alsa-audio-common.o
CC	libavdevice/alsa-audio-dec.o
CC	libavdevice/alsa-audio-enc.o
CC	libavdevice/avdevice.o
CC	libavdevice/dv1394.o
CC	libavdevice/fbdev.o
CC	libavdevice/lavfi.o
CC	libavdevice/libdc1394.o
CC	libavdevice/oss_audio.o
CC	libavdevice/sdl.o
CC	libavdevice/timefilter.o
CC	libavdevice/v4l2-common.o
CC	libavdevice/v4l2.o
libavdevice/v4l2.c: In function ?mmap_read_frame?:
libavdevice/v4l2.c:554:9: warning: ?destruct? is deprecated (declared
at ./libavcodec/avcodec.h:1076) [-Wdeprecated-declarations]
CC	libavdevice/v4l2enc.o
CC	libavdevice/x11grab.o
AR	libavdevice/libavdevice.a

# skipped lots of stuff 

LD	ffmpeg_g
CP	ffmpeg
STRIP	ffmpeg
skipping strip ffmpeg
CC	ffplay.o
LD	ffplay_g
CP	ffplay
STRIP	ffplay
skipping strip ffplay
CC	ffprobe.o
LD	ffprobe_g
CP	ffprobe
STRIP	ffprobe
skipping strip ffprobe
CC	ffserver.o
LD	ffserver_g
CP	ffserver
STRIP	ffserver
skipping strip ffserver
ned337 at vid:~/ffmpeg$


If you want all of the cl output, I have it.


ned337 at vid:~/ffmpeg$ gksu make install
INSTALL	libavdevice/libavdevice.a
INSTALL	libavdevice/libavdevice.so
STRIP	install-libavdevice-shared
skipping strip /usr/lib/libavdevice.so.55.3.100
INSTALL	libavfilter/libavfilter.a
INSTALL	libavfilter/libavfilter.so
STRIP	install-libavfilter-shared
skipping strip /usr/lib/libavfilter.so.3.81.103
INSTALL	libavformat/libavformat.a
INSTALL	libavformat/libavformat.so
STRIP	install-libavformat-shared
skipping strip /usr/lib/libavformat.so.55.12.102
INSTALL	libavcodec/libavcodec.a
INSTALL	libavcodec/libavcodec.so
STRIP	install-libavcodec-shared
skipping strip /usr/lib/libavcodec.so.55.19.100
INSTALL	libpostproc/libpostproc.a
INSTALL	libpostproc/libpostproc.so
STRIP	install-libpostproc-shared
skipping strip /usr/lib/libpostproc.so.52.3.100
INSTALL	libswresample/libswresample.a
INSTALL	libswresample/libswresample.so
STRIP	install-libswresample-shared
skipping strip /usr/lib/libswresample.so.0.17.103
INSTALL	libswscale/libswscale.a
INSTALL	libswscale/libswscale.so
STRIP	install-libswscale-shared
skipping strip /usr/lib/libswscale.so.2.4.100
INSTALL	libavutil/libavutil.a
INSTALL	libavutil/libavutil.so
STRIP	install-libavutil-shared
skipping strip /usr/lib/libavutil.so.52.40.100
INSTALL	doc/ffmpeg.1
INSTALL	doc/ffplay.1
INSTALL	doc/ffprobe.1
INSTALL	doc/ffserver.1
INSTALL	doc/ffmpeg-all.1
INSTALL	doc/ffplay-all.1
INSTALL	doc/ffprobe-all.1
INSTALL	doc/ffserver-all.1
INSTALL	doc/ffmpeg-utils.1
INSTALL	doc/ffmpeg-scaler.1
INSTALL	doc/ffmpeg-resampler.1
INSTALL	doc/ffmpeg-codecs.1
INSTALL	doc/ffmpeg-bitstream-filters.1
INSTALL	doc/ffmpeg-formats.1
INSTALL	doc/ffmpeg-protocols.1
INSTALL	doc/ffmpeg-devices.1
INSTALL	doc/ffmpeg-filters.1
INSTALL	doc/libavutil.3
INSTALL	doc/libswscale.3
INSTALL	doc/libswresample.3
INSTALL	doc/libavcodec.3
INSTALL	doc/libavformat.3
INSTALL	doc/libavdevice.3
INSTALL	doc/libavfilter.3
INSTALL	doc/ffmpeg.1
INSTALL	doc/ffplay.1
INSTALL	doc/ffprobe.1
INSTALL	doc/ffserver.1
INSTALL	doc/ffmpeg-all.1
INSTALL	doc/ffplay-all.1
INSTALL	doc/ffprobe-all.1
INSTALL	doc/ffserver-all.1
INSTALL	doc/ffmpeg-utils.1
INSTALL	doc/ffmpeg-scaler.1
INSTALL	doc/ffmpeg-resampler.1
INSTALL	doc/ffmpeg-codecs.1
INSTALL	doc/ffmpeg-bitstream-filters.1
INSTALL	doc/ffmpeg-formats.1
INSTALL	doc/ffmpeg-protocols.1
INSTALL	doc/ffmpeg-devices.1
INSTALL	doc/ffmpeg-filters.1
INSTALL	doc/libavutil.3
INSTALL	doc/libswscale.3
INSTALL	doc/libswresample.3
INSTALL	doc/libavcodec.3
INSTALL	doc/libavformat.3
INSTALL	doc/libavdevice.3
INSTALL	doc/libavfilter.3
INSTALL	install-progs-yes
INSTALL	ffmpeg
INSTALL	ffplay
INSTALL	ffprobe
INSTALL	ffserver
INSTALL	presets/libvpx-1080p50_60.ffpreset
INSTALL	presets/libvpx-1080p.ffpreset
INSTALL	presets/libvpx-360p.ffpreset
INSTALL	presets/libvpx-720p50_60.ffpreset
INSTALL	presets/libvpx-720p.ffpreset
INSTALL	presets/libx264-ipod320.ffpreset
INSTALL	presets/libx264-ipod640.ffpreset
INSTALL	doc/ffprobe.xsd
INSTALL	doc/examples/decoding_encoding.c
INSTALL	doc/examples/demuxing.c
INSTALL	doc/examples/filtering_audio.c
INSTALL	doc/examples/filtering_video.c
INSTALL	doc/examples/metadata.c
INSTALL	doc/examples/muxing.c
INSTALL	doc/examples/resampling_audio.c
INSTALL	doc/examples/scaling_video.c
INSTALL	doc/examples/Makefile
INSTALL	doc/examples/README
INSTALL	presets/libvpx-1080p50_60.ffpreset
INSTALL	presets/libvpx-1080p.ffpreset
INSTALL	presets/libvpx-360p.ffpreset
INSTALL	presets/libvpx-720p50_60.ffpreset
INSTALL	presets/libvpx-720p.ffpreset
INSTALL	presets/libx264-ipod320.ffpreset
INSTALL	presets/libx264-ipod640.ffpreset
INSTALL	doc/ffprobe.xsd
INSTALL	doc/examples/decoding_encoding.c
INSTALL	doc/examples/demuxing.c
INSTALL	doc/examples/filtering_audio.c
INSTALL	doc/examples/filtering_video.c
INSTALL	doc/examples/metadata.c
INSTALL	doc/examples/muxing.c
INSTALL	doc/examples/resampling_audio.c
INSTALL	doc/examples/scaling_video.c
INSTALL	doc/examples/Makefile
INSTALL	doc/examples/README
INSTALL	libavdevice/avdevice.h
INSTALL	libavdevice/version.h
INSTALL	libavdevice/libavdevice.pc
INSTALL	libavfilter/asrc_abuffer.h
INSTALL	libavfilter/avcodec.h
INSTALL	libavfilter/avfilter.h
INSTALL	libavfilter/avfiltergraph.h
INSTALL	libavfilter/buffersink.h
INSTALL	libavfilter/buffersrc.h
INSTALL	libavfilter/version.h
INSTALL	libavfilter/libavfilter.pc
INSTALL	libavformat/avformat.h
INSTALL	libavformat/avio.h
INSTALL	libavformat/version.h
INSTALL	libavformat/libavformat.pc
INSTALL	libavcodec/avcodec.h
INSTALL	libavcodec/avfft.h
INSTALL	libavcodec/dxva2.h
INSTALL	libavcodec/old_codec_ids.h
INSTALL	libavcodec/vaapi.h
INSTALL	libavcodec/vda.h
INSTALL	libavcodec/vdpau.h
INSTALL	libavcodec/version.h
INSTALL	libavcodec/xvmc.h
INSTALL	libavcodec/libavcodec.pc
INSTALL	libpostproc/postprocess.h
INSTALL	libpostproc/version.h
INSTALL	libpostproc/libpostproc.pc
INSTALL	libswresample/swresample.h
INSTALL	libswresample/version.h
INSTALL	libswresample/libswresample.pc
INSTALL	libswscale/swscale.h
INSTALL	libswscale/version.h
INSTALL	libswscale/libswscale.pc
INSTALL	libavutil/adler32.h
INSTALL	libavutil/aes.h
INSTALL	libavutil/attributes.h
INSTALL	libavutil/audio_fifo.h
INSTALL	libavutil/audioconvert.h
INSTALL	libavutil/avassert.h
INSTALL	libavutil/avstring.h
INSTALL	libavutil/avutil.h
INSTALL	libavutil/base64.h
INSTALL	libavutil/blowfish.h
INSTALL	libavutil/bprint.h
INSTALL	libavutil/bswap.h
INSTALL	libavutil/buffer.h
INSTALL	libavutil/channel_layout.h
INSTALL	libavutil/common.h
INSTALL	libavutil/cpu.h
INSTALL	libavutil/crc.h
INSTALL	libavutil/error.h
INSTALL	libavutil/eval.h
INSTALL	libavutil/fifo.h
INSTALL	libavutil/file.h
INSTALL	libavutil/frame.h
INSTALL	libavutil/hmac.h
INSTALL	libavutil/imgutils.h
INSTALL	libavutil/intfloat.h
INSTALL	libavutil/intfloat_readwrite.h
INSTALL	libavutil/intreadwrite.h
INSTALL	libavutil/lfg.h
INSTALL	libavutil/log.h
INSTALL	libavutil/mathematics.h
INSTALL	libavutil/md5.h
INSTALL	libavutil/mem.h
INSTALL	libavutil/murmur3.h
INSTALL	libavutil/dict.h
INSTALL	libavutil/old_pix_fmts.h
INSTALL	libavutil/opt.h
INSTALL	libavutil/parseutils.h
INSTALL	libavutil/pixdesc.h
INSTALL	libavutil/pixfmt.h
INSTALL	libavutil/random_seed.h
INSTALL	libavutil/rational.h
INSTALL	libavutil/ripemd.h
INSTALL	libavutil/samplefmt.h
INSTALL	libavutil/sha.h
INSTALL	libavutil/sha512.h
INSTALL	libavutil/time.h
INSTALL	libavutil/timecode.h
INSTALL	libavutil/timestamp.h
INSTALL	libavutil/version.h
INSTALL	libavutil/xtea.h
INSTALL	libavutil/lzo.h
INSTALL	libavutil/opencl.h
INSTALL	libavutil/avconfig.h
INSTALL	libavutil/libavutil.pc
ned337 at vid:~/ffmpeg$ 

Thanks in Advance, 
Ned 


From mark.himsley at gmail.com  Mon Aug  5 08:44:15 2013
From: mark.himsley at gmail.com (Mark Himsley)
Date: Mon, 5 Aug 2013 07:44:15 +0100
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375487953644-4660437.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
Message-ID: <CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>

The square brackets around the decimal parts of seconds is to indicate to
you that they are optional. Remove the square brackets from your command.

From ajsinclair at gmail.com  Mon Aug  5 08:54:35 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Mon, 5 Aug 2013 16:54:35 +1000
Subject: [FFmpeg-user] getting latest version of ffmpeg to work with Roku
In-Reply-To: <51FD89EC.5000101@american-hero.com>
References: <51FC5683.9090405@american-hero.com>
	<51FCC130.3050001@thelounge.net>
	<51FD7773.6040609@american-hero.com>
	<51FD89EC.5000101@american-hero.com>
Message-ID: <CAOeAR15Q8tKdo-y5JPPoUon47qNmO9+31uz-Ka4J3FH0XTpSog@mail.gmail.com>

Not definitive but constrained baseline is an unusual profile for a HD
encode and stands out in your "DOESN'T WORK" option, should try main or high

If you want HD you should also drop the -s 640x360 bit

>
>

From leo.currie at gmail.com  Mon Aug  5 12:30:52 2013
From: leo.currie at gmail.com (Leo Currie)
Date: Mon, 5 Aug 2013 11:30:52 +0100
Subject: [FFmpeg-user] audio-only flv over rtmp
Message-ID: <CANBFD3oOjoiHKWFWLyc0EKM9hzqZhqDFUYXF_q80h=-BpaXhEQ@mail.gmail.com>

Hi -

I'm trying to generate an RTMP live stream for Flash clients that contains
audio only.
Is there a special option to prevent the flv muxer generating a video
stream?

Here's an example of the kind of stream I'd like to generate:
rtmp://87.117.250.4/radiosix3/

I'm trying this:
ffmpeg -re -i (my input stream) -acodec aac -strict experimental -ab 96k
-ar 44100 -vn -f flv rtmp://127.0.0.1/live/test
..which connects to my local nginx instance running the rtmp module.

When I try to play the stream in VLC, it complains about an 'undf' video
format, but audio plays fine. Looking at the stream properties in VLC I can
see there is indeed an 'undf' video stream 0, with the audio being stream 1.

Is there an option for the flv muxer to prevent this video stream being
generated? I tried -vn but this doesn't do what I expect.

Or am I going about this the wrong way?

Thanks.

Leo

From blacktrash at gmx.net  Mon Aug  5 12:50:09 2013
From: blacktrash at gmx.net (Christian Ebert)
Date: Mon, 5 Aug 2013 11:50:09 +0100
Subject: [FFmpeg-user] audio-only flv over rtmp
In-Reply-To: <CANBFD3oOjoiHKWFWLyc0EKM9hzqZhqDFUYXF_q80h=-BpaXhEQ@mail.gmail.com>
References: <CANBFD3oOjoiHKWFWLyc0EKM9hzqZhqDFUYXF_q80h=-BpaXhEQ@mail.gmail.com>
Message-ID: <20130805105008.GF1508@krille.blacktrash.org>

* Leo Currie on Monday, August 05, 2013 at 11:30:52 +0100
> I'm trying to generate an RTMP live stream for Flash clients that contains
> audio only.

Why don't you stream just MP3?

> Is there a special option to prevent the flv muxer generating a video
> stream?
> 
> Here's an example of the kind of stream I'd like to generate:
> rtmp://87.117.250.4/radiosix3/
> 
> I'm trying this:
> ffmpeg -re -i (my input stream) -acodec aac -strict experimental -ab 96k
> -ar 44100 -vn -f flv rtmp://127.0.0.1/live/test
> ..which connects to my local nginx instance running the rtmp module.
> 
> When I try to play the stream in VLC, it complains about an 'undf' video
> format, but audio plays fine. Looking at the stream properties in VLC I can
> see there is indeed an 'undf' video stream 0, with the audio being stream 1.
> 
> Is there an option for the flv muxer to prevent this video stream being
> generated? I tried -vn but this doesn't do what I expect.
> 
> Or am I going about this the wrong way?

FLV is Flash Video, so a missing video track does not make sense
imho. Furthermore I don't think you need this contortion for RTMP
(see above). I packed audio into a FLV container for
*pseudo*streaming, but again then you need a _minimal_ video
track to be able to inject metadata.

-- 
  Was hei?t hier Dogma, ich bin Underdogma!
[ What the hell do you mean dogma, I am underdogma. ]
free movies       --->>> http://www.blacktrash.org/underdogma
http://itunes.apple.com/podcast/underdogma-movies/id363423596

From francois.visagie at gmail.com  Mon Aug  5 12:58:08 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Mon, 5 Aug 2013 12:58:08 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
In-Reply-To: <51FEB571.2080701@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
Message-ID: <006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>

> -----Original Message-----
> From: Andy Furniss [mailto:adf.lists at gmail.com]
> Sent: 04 August 2013 22:11
> To: FFmpeg user questions
> Cc: Francois Visagie
> Subject: Re: [FFmpeg-user] Preserving perceived loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Francois Visagie wrote:
> 
> > I don't think this is the desired outcome, to achieve unified audio
> > processing but with significant reduction in perceived volume in the
> > process. Is there any further testing I can do to help investigate and
> > hopefully improve this situation?
> 
> Down mixing is not ideal. Lower volume is expected.
> 
> If you don't want the result to clip then you have to normalise enough to
> allow for all input channels being full level - even though that may be
rare in
> practice.
> 
> As you are dealing with ac3

Therein lies part of the problem, not all input files are AC3. Up to at
least 30 June -filter:a aformat=channel_layouts=stereo could be used in a
standard command line to produce stereo from multi-channel inputs with input
and output volumes perceivably equal. Now each encode needs to be inspected
individually for input/output differences, and the remedy will in each case
also differ according to input type and/or volume differences. Really
sub-optimal in my view, one which I expect to be more widely shared once
these implications are more widely understood.

I sincerely appreciate the trouble you took with outlining various
principles involved, but, on a more practical level: rather than making
-filter:a aformat=channel_layouts=stereo now share the mechanism of -ac 2
and -filter:a aresample=ocl=3 (incorrectly so wrt. volume levels in my
view), what is the feasibility of making the other two behave like -filter:a
aformat=channel_layouts=stereo instead?

> you should also be aware that a studio produced
> stream will usually carry mata data specifying dynamic range control and
by
> default ffmpeg applies this fully so you get the restricted the range
output.
> Just turning this off with
> 
> -drc_scale 0
> 
> will however produce something that IMHO has too much range - it's not
> actually on/off it's float so you can do in between 0 and 1.
> 
> ac3 also has meta data for downmix so using -request_channels 2 should in
> theory get a downmix as the studio intended - but of course it's still
> normalised to prevent clipping and it's still compromised by the fact it's
a
> downmix.
> 
> I don't know if there's a way with ffmpeg to analyse and adjust the whole
> downmixed track to boost the levels to take advantage of any headroom.
> 
> You can certainly do it with sox (working on pcm)
> 
> Slight digression - last time I looked at dts/dca codecs default downmix
it was
> a clipping stereo squashing mess - nice and loud though, but really. don't
> think it's OK or anything to aim for should you happen to ever test.
> 
> Not much use for your need, but as for the future - the nice thing about
> Dolby Truehd and DTS MA is that they mix up, so a stereo user like me
should
> get an "artistic" studio mix rather than something kludged down.
> 
> AFAIK dts ma isn't supported yet truehd is (but no drc support) - and this
is
> just decode.
> 
> 
> 
> 



From cehoyos at ag.or.at  Mon Aug  5 14:31:01 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 5 Aug 2013 12:31:01 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg Unknown encoder 'libx264' error
References: <1375666070.5067.41.camel@videoathlon>
Message-ID: <loom.20130805T141926-122@post.gmane.org>

Ned Haskin <feanor2112 <at> covad.net> writes:

> ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers

If you are a user (and not a distributor) please use git head.

>   built on Jul 29 2013 13:49:23 with gcc 4.7 (Debian 4.7.2-5)
>   configuration: 

This means you compiled without x264 support ...

> Unknown encoder 'libx264'

... so it cannot be used.

Carl Eugen Hoyos



From jshupert at pps-inc.com  Mon Aug  5 15:10:37 2013
From: jshupert at pps-inc.com (Jim Shupert)
Date: Mon, 05 Aug 2013 09:10:37 -0400
Subject: [FFmpeg-user] 50 Mbps Ifram mpg2 with scaling and pad --sync
 problem
In-Reply-To: <99b707c624bfd985e2248ec04e42b10b.squirrel@webmail.theppsgroup.com>
References: <99b707c624bfd985e2248ec04e42b10b.squirrel@webmail.theppsgroup.com>
Message-ID: <51FFA44D.1010104@pps-inc.com>


On 8/2/2013 12:46 PM, Jim Shupert, Jr wrote:
> Friends,
> I wish to take 720x486 Unc 8 bit rawVideo with 2 chn  pcm audio
> and make 50 mbps Iframe mpg2 422 1920x1080i with mpg2 audio
>
> I realize that upScaling is not best - but I do need to in this situation.
>
> I have something that I think is close but the audio goes out of sync.
> I wonder why?
>
> You will see I have my buffer set rather high - i did that to avoid
> ...
> [mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
> [mpeg @ 0x307afa0] packet too large, ignoring buffer limits to mux it
> [mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
> ...
>
> here is my command
>
> jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
> "scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
> mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
> +ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
> 5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
> -async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg
>
> jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
> "scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
> mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
> +ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
> 5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
> -async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg
> ffmpeg version git-2012-09-24-fd63c2f Copyright (c) 2000-2012 the FFmpeg
> developers
>    built on Sep 24 2012 14:09:13 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
>    configuration: --enable-gpl --enable-libfaac --enable-libmp3lame
> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora
> --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
> --enable-version3 --enable-x11grab
>    libavutil      51. 73.101 / 51. 73.101
>    libavcodec     54. 58.100 / 54. 58.100
>    libavformat    54. 28.101 / 54. 28.101
>    libavdevice    54.  2.101 / 54.  2.101
>    libavfilter     3. 17.100 /  3. 17.100
>    libswscale      2.  1.101 /  2.  1.101
>    libswresample   0. 15.100 /  0. 15.100
>    libpostproc    52.  0.100 / 52.  0.100
> [avi @ 0x28b5240] non-interleaved AVI
> Guessed Channel Layout for  Input Stream #0.1 : stereo
> Input #0, avi, from '/media/data/z_parts/dog.avi':
>    Duration: 00:00:05.03, start: 0.000000, bitrate: 170941 kb/s
>      Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x486,
> 29.97 tbr, 29.97 tbn, 29.97 tbc
>      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
> stereo, s16, 1536 kb/s
> File '/media/data/jtest/dog-3a_1080i.mpg' already exists. Overwrite ? [y/N] y
> -async is forwarded to lavfi similarly to -af
> aresample=min_comp=0.001:min_hard_comp=0.100000.
> Output #0, mpeg, to '/media/data/jtest/dog-3a_1080i.mpg':
>    Metadata:
>      encoder         : Lavf54.28.101
>      Stream #0:0: Video: mpeg2video, yuv422p, 1920x1080, q=1-3, 50000 kb/s,
> 90k tbn, 29.97 tbc
>      Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 224 kb/s
> Stream mapping:
>    Stream #0:0 -> #0:0 (rawvideo -> mpeg2video)
>    Stream #0:1 -> #0:1 (pcm_s16le -> mp2)
> Press [q] to stop, [?] for help
> [mpeg2video @ 0x28b6800] rc buffer underflow
>      Last message repeated 24 times
> frame=   25 fps=0.0 q=24.8 size=    6190kB time=00:00:00.83
> bitrate=60789.4kbitsrc buffer underflow
> [mpeg2video @ 0x28b6800] rc buffer underflow
>
Maybe I could ask another way..
what might be the suggested means to scale raw video , uncompressed d1 
720x486 ntsc w pcm audio to 50 Mbps I fram mpg2.
I am scaling and padding . My audio comes out of sync.
Likely this is unrelated to the scaling and padding - but maybe not.
and
Does anyone have an opinion on buffer setting?

Thank you

best regards

From lists.davep at gmail.com  Mon Aug  5 15:58:08 2013
From: lists.davep at gmail.com (Dave P)
Date: Mon, 5 Aug 2013 09:58:08 -0400
Subject: [FFmpeg-user] 50 Mbps Ifram mpg2 with scaling and pad --sync
	problem
In-Reply-To: <51FFA44D.1010104@pps-inc.com>
References: <99b707c624bfd985e2248ec04e42b10b.squirrel@webmail.theppsgroup.com>
	<51FFA44D.1010104@pps-inc.com>
Message-ID: <CANPFk1fqgggYhiXkS-e1s51d9wnTGrVb1iPU6oPR2vcU1pPMfw@mail.gmail.com>

On Aug 5, 2013 9:11 AM, "Jim Shupert" <jshupert at pps-inc.com> wrote:
>
>
> On 8/2/2013 12:46 PM, Jim Shupert, Jr wrote:
>>
>> Friends,
>> I wish to take 720x486 Unc 8 bit rawVideo with 2 chn  pcm audio
>> and make 50 mbps Iframe mpg2 422 1920x1080i with mpg2 audio
>>
>> I realize that upScaling is not best - but I do need to in this
situation.
>>
>> I have something that I think is close but the audio goes out of sync.
>> I wonder why?
>>
>> You will see I have my buffer set rather high - i did that to avoid
>> ...
>> [mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
>> [mpeg @ 0x307afa0] packet too large, ignoring buffer limits to mux it
>> [mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
>> ...
>>
>> here is my command
>>
>> jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
>> "scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
>> mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
>> +ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
>> 5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
>> -async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg
>>
>> jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
>> "scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
>> mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
>> +ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
>> 5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
>> -async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg
>> ffmpeg version git-2012-09-24-fd63c2f Copyright (c) 2000-2012 the FFmpeg
>> developers
>>    built on Sep 24 2012 14:09:13 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
>>    configuration: --enable-gpl --enable-libfaac --enable-libmp3lame
>> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora
>> --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
>> --enable-version3 --enable-x11grab
>>    libavutil      51. 73.101 / 51. 73.101
>>    libavcodec     54. 58.100 / 54. 58.100
>>    libavformat    54. 28.101 / 54. 28.101
>>    libavdevice    54.  2.101 / 54.  2.101
>>    libavfilter     3. 17.100 /  3. 17.100
>>    libswscale      2.  1.101 /  2.  1.101
>>    libswresample   0. 15.100 /  0. 15.100
>>    libpostproc    52.  0.100 / 52.  0.100
>> [avi @ 0x28b5240] non-interleaved AVI
>> Guessed Channel Layout for  Input Stream #0.1 : stereo
>> Input #0, avi, from '/media/data/z_parts/dog.avi':
>>    Duration: 00:00:05.03, start: 0.000000, bitrate: 170941 kb/s
>>      Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x486,
>> 29.97 tbr, 29.97 tbn, 29.97 tbc
>>      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
>> stereo, s16, 1536 kb/s
>> File '/media/data/jtest/dog-3a_1080i.mpg' already exists. Overwrite ?
[y/N] y
>> -async is forwarded to lavfi similarly to -af
>> aresample=min_comp=0.001:min_hard_comp=0.100000.
>> Output #0, mpeg, to '/media/data/jtest/dog-3a_1080i.mpg':
>>    Metadata:
>>      encoder         : Lavf54.28.101
>>      Stream #0:0: Video: mpeg2video, yuv422p, 1920x1080, q=1-3, 50000
kb/s,
>> 90k tbn, 29.97 tbc
>>      Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 224 kb/s
>> Stream mapping:
>>    Stream #0:0 -> #0:0 (rawvideo -> mpeg2video)
>>    Stream #0:1 -> #0:1 (pcm_s16le -> mp2)
>> Press [q] to stop, [?] for help
>> [mpeg2video @ 0x28b6800] rc buffer underflow
>>      Last message repeated 24 times
>> frame=   25 fps=0.0 q=24.8 size=    6190kB time=00:00:00.83
>> bitrate=60789.4kbitsrc buffer underflow
>> [mpeg2video @ 0x28b6800] rc buffer underflow
>>
> Maybe I could ask another way..
> what might be the suggested means to scale raw video , uncompressed d1
720x486 ntsc w pcm audio to 50 Mbps I fram mpg2.
> I am scaling and padding . My audio comes out of sync.
> Likely this is unrelated to the scaling and padding - but maybe not.
> and
> Does anyone have an opinion on buffer setting?

Try .ts or .m2t instead of .mpg

From cmharwoo at umich.edu  Mon Aug  5 16:03:40 2013
From: cmharwoo at umich.edu (cmharwood89)
Date: Mon, 5 Aug 2013 07:03:40 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375710620712-4660457.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
Message-ID: <1375711420502-4660459.post@n4.nabble.com>

Or am I misunderstanding what you're saying? Do you mean to include the
milliseconds, but without the square brackets? I thought you were suggesting
to truncate the time down to HH:MM:SS format. 



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From soho123.2012 at gmail.com  Mon Aug  5 16:17:09 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 5 Aug 2013 22:17:09 +0800
Subject: [FFmpeg-user] VLC CANNOT play rtp stream that output by ffmpeg +
	ffserver
Message-ID: <CAFd2cZWTDqgW5amm681779CzSrWb+oXoVhFaYs8hbRBCaU6qiw@mail.gmail.com>

Hi All,


The problem is :
when  I use the command to capture video and audio:
ffmpeg -loglevel debug -debug_ts -f video4linux2 -r 30 -s 640x480
-input_format h264 -i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec
copy -acodec copy  http://localhost:8090/feed1.ffm

then ffserver can service the VLC client request for live stream.

video can be displayed,
but there is no audio output because VLC says "PTS out of range"

Does anyone have idea?

From cmharwoo at umich.edu  Mon Aug  5 15:50:20 2013
From: cmharwoo at umich.edu (cmharwood89)
Date: Mon, 5 Aug 2013 06:50:20 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
Message-ID: <1375710620712-4660457.post@n4.nabble.com>

@thljcl - Yes, I'm cutting a section from a longer video. I have an camera
running continuously during physics experiments and I need to segment the
master video into individual movies for each experimental condition.

@Mark Himsley - I know that the square brackets are optional, but I need the
millisecond resolution to precisely locate the frame I want. I need to cut
the video at exactly the frame where an LED turns on. From my reading, it
looks like there's not a way to specify a cutting point by frame number, so
the HH:MM:SS[.ms] format is the only way to get the precision I need.





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From soho123.2012 at gmail.com  Mon Aug  5 16:34:48 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 5 Aug 2013 22:34:48 +0800
Subject: [FFmpeg-user] [HELP]capture audio question about Non-monotonous
	DTS??????
Message-ID: <CAFd2cZWY65Fhaf7KCT5aVnFWpc1iK-3zLBdykiRwpL79UhaR5w@mail.gmail.com>

HI all,

when I try to capture audio and video from USB web cam and usb sound card.
the command :
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy
http://localhost:8090/feed1.ffm

I got the warning message from ffmpeg:

Non-monotonous DTS in output stream 0:0; previous: 254883622, current:
254883581; changing to 254883623. This may result in incorrect timestamps
in the output file


and
the VLC player can not play audio data  , but  video data can be displayed.

Does anyone have idea and the timestamp ?

From jiachielee at live.com  Mon Aug  5 16:36:51 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 5 Aug 2013 07:36:51 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375711420502-4660459.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
Message-ID: <1375713411619-4660462.post@n4.nabble.com>

You do know that how small the video can be cut out from the source depends
on your camera frame rate, do you not? I do not think that there is a
consumer-application camera that can take 1000 high-resolution pictures per
second. The maximum frame rate supported by H.264 is 172 FPS. For normal
playback though, it would be necessary to convert the frame rate to 24 FPS,
25 FPS, or 30 FPS for smooth playback. Storing video in 172 FPS is mainly
for editing purposes and being kept as master copy.
Yes, it?s possible to cut out the video from the source based on frame
numbers, assuming that your video source is of constant frame rate. Only
with constant frame rate, you can then tell the length of video by counting
the number of available frames. Is cutting out a portion of video based on
number of frames is what you are looking for? In fact, that?s the most
accurate we can get in terms of cutting out portions of video from a longer
video.
Tell me exactly if I get your question right. If you do want to know how to
cut the video based on the number of frames, I would tell you later on this
forum.




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From lists.davep at gmail.com  Mon Aug  5 17:03:30 2013
From: lists.davep at gmail.com (Dave P)
Date: Mon, 5 Aug 2013 11:03:30 -0400
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375710620712-4660457.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
Message-ID: <CANPFk1dXrFM3DtS2DtNu_D3C_p6u7e0E7F=_EvRMC7q60T2rKw@mail.gmail.com>

On Aug 5, 2013 10:26 AM, "cmharwood89" <cmharwoo at umich.edu> wrote:
>
> @thljcl - Yes, I'm cutting a section from a longer video. I have an camera
> running continuously during physics experiments and I need to segment the
> master video into individual movies for each experimental condition.
>
> @Mark Himsley - I know that the square brackets are optional, but I need
the
> millisecond resolution to precisely locate the frame I want. I need to cut
> the video at exactly the frame where an LED turns on. From my reading, it
> looks like there's not a way to specify a cutting point by frame number,
so
> the HH:MM:SS[.ms] format is the only way to get the precision I need.

The select filter mentioned in this recent thread might help:
lists.ffmpeg.org/pipermail/ffmpeg-user/2013-July/016291.html

From cmharwoo at umich.edu  Mon Aug  5 17:45:40 2013
From: cmharwoo at umich.edu (cmharwood89)
Date: Mon, 5 Aug 2013 08:45:40 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375717043247-4660465.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
	<1375717043247-4660465.post@n4.nabble.com>
Message-ID: <CAKh93wOxWcF5tYNzWFaKvCUUVsJ5oDRVhyu93X=1GPfcBcuZBA@mail.gmail.com>

I'm able to get it done using time markers, yes. However, if cutting from a
frame number is a more precise method, I'd like to do that instead.


On Mon, Aug 5, 2013 at 10:37 AM, thljcl [via FFmpeg-users] <
ml-node+s933282n4660465h39 at n4.nabble.com> wrote:

> You said that you were already able to get the job done but still want to
> know the way to cut a video from a specific frame number?
>
> ------------------------------
>  If you reply to this email, your message will be added to the discussion
> below:
>
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-- 
Casey M. Harwood
Ph.D. Candidate
NSF GRFP Fellow
Naval Architecture & Marine Engineering
University of Michigan
Ann Arbor, MI
cmharwoo at umich.edu
360-589-9626




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From jiachielee at live.com  Mon Aug  5 17:37:23 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 5 Aug 2013 08:37:23 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375716335502-4660464.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
Message-ID: <1375717043247-4660465.post@n4.nabble.com>

You said that you were already able to get the job done but still want to
know the way to cut a video from a specific frame number?



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From jpboard2 at yahoo.com  Mon Aug  5 18:07:55 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 5 Aug 2013 09:07:55 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <CAKh93wOxWcF5tYNzWFaKvCUUVsJ5oDRVhyu93X=1GPfcBcuZBA@mail.gmail.com>
References: <1375487953644-4660437.post@n4.nabble.com>	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>	<1375710620712-4660457.post@n4.nabble.com>	<1375711420502-4660459.post@n4.nabble.com>	<1375713411619-4660462.post@n4.nabble.com>	<1375716335502-4660464.post@n4.nabble.com>	<1375717043247-4660465.post@n4.nabble.com>
	<CAKh93wOxWcF5tYNzWFaKvCUUVsJ5oDRVhyu93X=1GPfcBcuZBA@mail.gmail.com>
Message-ID: <1375718875.86856.YahooMailNeo@web164704.mail.gq1.yahoo.com>

I'm able to do frame-exact editing with ffmpeg.? It's described in this thread: 


??? http://lists.ffmpeg.org/pipermail/ffmpeg-user/2013-July/016291.html

The command is something like this (thanks to Carl Lindqvist)
ffmpeg -i SOURCEFILE.avi -vf
select='-between(n,0,100)-between(n,130,200)',setpts=N/FRAME_RATE/TB
-c:v libx264 -preset veryfast -b:v 700k -profile:v main -af
aselect='-between(n,0,100)-between(n,130,200)',asetpts=N/SR/TB -c:a
libvo_aacenc -b:a 96k -ac 2 OUT.mp4
I verified that the frames are exactly what I wanted by checking the md5sums of each frame.

My input avi file used a raw codec with no inter-frame compression.? I'm not sure if frame-exact editing will work when the input file is compressed with mpeg-like codecs.

From cmharwoo at umich.edu  Mon Aug  5 17:25:35 2013
From: cmharwoo at umich.edu (cmharwood89)
Date: Mon, 5 Aug 2013 08:25:35 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375713411619-4660462.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
Message-ID: <1375716335502-4660464.post@n4.nabble.com>

I'm working with a 60 FPS (or 59.94) video, though the camera is ostensibly
able to capture up to 240 FPS (under which codec I'm not sure). Later on, I
may be capturing video at 1000+ FPS, but I'll be using machine-vision
cameras with their own controllers; I doubt I'll need to do the same kind of
post-processing with those.

I've been using AVIdemux to seek the exact frame I want, and then pulling
the time marker in HH:MM:SS.ms format from the AVIdemux GUI, which I then
use with '-ss' command in ffmpeg to get trimming. Mark's tip to remove the
square brackets seemed to do the trick, because my output begins with the
exact frame I picked in AVIdemux in my test case.

The duration of the cut video is less important; I do not need the trimmed
video to be a specific number of frames. I do need it to begin at a specific
frame in the master video. My method for doing this is a bit convoluted, so
if specifying a frame number to begin cutting at is more accurate, I'd
really appreciate any tips on how to do that.

I think that in the future, I'll buck up and get nicer cameras that have
millisecond-precision triggering so I don't have to trim the extra dead time
from each video.

Thanks so much!



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From jpboard2 at yahoo.com  Mon Aug  5 18:29:58 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 5 Aug 2013 09:29:58 -0700 (PDT)
Subject: [FFmpeg-user] OpenCL?  What Effect Does It Have
Message-ID: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>

If I were to install a more powerful Nvidia graphics card on my
machine, and then configure and build ffmpeg with OpenCL
enabled, what effect would it have?? The main things I do
are

	* Inverse telecine, or deinterlace filters
	* Convert to a lossless format: ffvhuff, ljpeg, etc (I currently do this with mencoder, but would switch to ffmpeg if it's faster).? This takes a large amount of time.

	* Cut edits with the Select filter
	* Brightness filters
	* Encode to mpeg4.Since my files are all stored in a lossless format like ffvhuff, much of the 
time is spent compressing and decompressing that format.? Would 
OpenGL help withthis or any of the other steps above? ? What are the
types of things that OpenCL improves in ffmpeg?

From jiachielee at live.com  Mon Aug  5 18:02:52 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 5 Aug 2013 09:02:52 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <CAKh93wOxWcF5tYNzWFaKvCUUVsJ5oDRVhyu93X=1GPfcBcuZBA@mail.gmail.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
	<1375717043247-4660465.post@n4.nabble.com>
	<CAKh93wOxWcF5tYNzWFaKvCUUVsJ5oDRVhyu93X=1GPfcBcuZBA@mail.gmail.com>
Message-ID: <1375718572088-4660467.post@n4.nabble.com>

Does your source have one or more audio tracks? You do not know of the codec
which is the source is encoded in?



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Inaccurate-trimming-with-re-encoding-tp4660437p4660467.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jan at iwebwarestudio.com  Mon Aug  5 12:50:59 2013
From: jan at iwebwarestudio.com (Jan Neil Gador)
Date: Mon, 5 Aug 2013 18:50:59 +0800
Subject: [FFmpeg-user] FFmpeg Question
Message-ID: <CAAdiqJWbTSEnDx2M6wzxbBAnJayYSypRMt1yByzqTSybbMy1JQ@mail.gmail.com>

Hello! I'm Jan and I was hoping if I could consult/ask a question regarding
what I am working on. It's like an online moviemaker and one of the
features is to be able to merge photo with video.

Now what I did is I followed the codes for converting a single image
into a video
(mp4 file)

 "ffmpeg -y -loop 1 -f image2 -i img.jpg -s 720x480 -r 25 -c:v libx264 -t 5
 out.mp4"

(which gives me a 5 second mp4 video only output) so that I could merge it
with a video file (which is also an mp4 file) using concat.

"ffmpeg -y -i out.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts out.mp4.ts"

"ffmpeg -y -i out2.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts
out2.mp4.ts");

(out2.mp4 is an audio-video file.)

"ffmpeg -y -i concat:'out.mp4.ts|out2.mp4.ts' -c copy -bsf:a aac_adtstoasc
final_output.mp4"

I get an output with the right sequence but there is no audio that used to
be in the second file. I don't know if I'm doing it right or what. My
question is can a video only file be merged with a audio-video file? Is
this possible? If so what should I do or how can it be done. I haven't seen any
reference of it. Hoping for your reply. Thanks!

From soho123.2012 at gmail.com  Mon Aug  5 19:13:49 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 01:13:49 +0800
Subject: [FFmpeg-user] [HELP]how to avoid "ALSA buffer xrun."
Message-ID: <CAFd2cZXxzZ-qj4fiKUDOgVXc_Sswzn87XKj=E=aL9P8ZuYPtKw@mail.gmail.com>

Hi All,


When I use ffmpeg to capture video and sound (USB device), I can see the
message from alsa-audio-common.c "ALSA buffer xrun."

the command line I use :

ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy
http://localhost:8090/feed1.ffm


In record case, it seems ffmpeg get audio slower than usb device capture
the sound, such that the buffer overflow, right?
Does anyone have idea how to avoid the issue about "ALSA buffer xrun" when
ffmpeg capture sound from usb sound card?


thanks!

From cehoyos at ag.or.at  Mon Aug  5 19:41:11 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 5 Aug 2013 17:41:11 +0000 (UTC)
Subject: [FFmpeg-user] FFmpeg Question
References: <CAAdiqJWbTSEnDx2M6wzxbBAnJayYSypRMt1yByzqTSybbMy1JQ@mail.gmail.com>
Message-ID: <loom.20130805T193951-216@post.gmane.org>

Jan Neil Gador <jan <at> iwebwarestudio.com> writes:

> "ffmpeg -y -i concat:'out.mp4.ts|out2.mp4.ts' -c copy 
> -bsf:a aac_adtstoasc final_output.mp4"

Complete, uncut console output missing.
Does it work with -probesize 2G -analyzeduration 2G ?
If not, you will have to add silence to out.mp4 which 
should be trivial:
$ ffmpeg -loop 1 -i img.jpg -f pcm_s16le -i /dev/zero

Carl Eugen


From cehoyos at ag.or.at  Mon Aug  5 19:47:25 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 5 Aug 2013 17:47:25 +0000 (UTC)
Subject: [FFmpeg-user] OpenCL?  What Effect Does It Have
References: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <loom.20130805T194218-441@post.gmane.org>

James Board <jpboard2 <at> yahoo.com> writes:

> If I were to install a more powerful Nvidia graphics 
> card on my machine, and then configure and build 
> ffmpeg with OpenCL enabled, what effect would it have?

In theory, you could use opencl for the deshake and 
the unsharp filters but I don't know if sufficiently 
new OpenCL drivers exist.

> * Inverse telecine, or deinterlace filters

Unrelated to your question:
I suspect that deinterlacing (yadif) is faster with 
FFmpeg than MEncoder.

> * Convert to a lossless format: ffvhuff, ljpeg, etc 
> (I currently do this with mencoder, but would switch 
> to ffmpeg if it's faster).

Given that MEncoder uses the exact same source code 
to encode ffvhuff and ljpeg (and most probably "etc") 
how do you expect that FFmpeg could be faster?
Allow me to repeat that since MEncoder is not 
actively maintained, it makes no sense to use it if 
you don't have very special needs (like binary 
codecs but you should be able to use them with 
mplayer -vo yuv4mpeg and FFmpeg).

Carl Eugen


From feanor2112 at covad.net  Mon Aug  5 19:50:11 2013
From: feanor2112 at covad.net (Ned Haskin)
Date: Mon, 05 Aug 2013 13:50:11 -0400
Subject: [FFmpeg-user] ffmpeg Unknown encoder 'libx264' error
In-Reply-To: <loom.20130805T141926-122@post.gmane.org>
References: <1375666070.5067.41.camel@videoathlon>
	<loom.20130805T141926-122@post.gmane.org>
Message-ID: <1375725011.3919.8.camel@videoathlon>

On Mon, 2013-08-05 at 12:31 +0000, Carl Eugen Hoyos wrote:
> Ned Haskin <feanor2112 <at> covad.net> writes:
> 
> > ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
> 
> If you are a user (and not a distributor) please use git head.
> 
> >   built on Jul 29 2013 13:49:23 with gcc 4.7 (Debian 4.7.2-5)
> >   configuration: 
> 
> This means you compiled without x264 support ...
> 
> > Unknown encoder 'libx264'
> 
> ... so it cannot be used.
> 
> Carl Eugen Hoyos
> 
> 
Carl, my ./configure for ffmpeg has --enable-libx264 as show in my 1st post. Also in the 1st post libmp3lame, libtwolame 
and libxvid gave the same error when I tried to use them. All are installed with -dev or were compiled --enable-shared. 

ned337 at vid:~/ffmpeg$ make clean
ned17 at videoathlon:~/ffmpeg$ ./configure --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --enable-gpl 
--enable-shared --enable-nonfree --enable-libx264  --enable-encoder=libx264 --enable-libmp3lame --enable-libvorbis 
--enable-libtwolame --enable-libv4l2 --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc 
--enable-x11grab --enable-libtheora --disable-stripping --enable-avfilter --enable-libfreetype --enable-opencl 
--enable-version3 --enable-libopenjpeg --disable-altivec --disable-armv5te --disable-armv6 --disable-vis
install prefix            /usr
source path               .
C compiler                gcc
ARCH                      x86 (generic)
big-endian                no
runtime cpu detection     yes
yasm                      yes
MMX enabled               yes
MMXEXT enabled            yes
3DNow! enabled            yes
3DNow! extended enabled   yes
SSE enabled               yes
SSSE3 enabled             yes
AVX enabled               yes
FMA4 enabled              yes
i686 features enabled     yes
CMOV is fast              yes
EBX available             yes
EBP available             yes
debug symbols             yes
strip symbols             no
optimize for size         no
optimizations             yes
static                    yes
shared                    yes
postprocessing support    yes
new filter support        yes
network support           yes
threading support         pthreads
safe bitstream reader     yes
SDL support               yes
opencl enabled            yes
texi2html enabled         yes
perl enabled              yes
pod2man enabled           yes
makeinfo enabled          yes

External libraries:
iconv			libopenjpeg		libx264
libdc1394		libtheora		libxvid
libfaac			libtwolame		opencl
libfreetype		libv4l2			x11grab
libmp3lame		libvorbis		zlib

Enabled decoders:
aac			bmv_audio		idf
aac_latm		bmv_video		iff_byterun1
aasc			brender_pix		iff_ilbm
ac3			c93			imc
adpcm_4xm		cavs			indeo2
adpcm_adx		cdgraphics		indeo3
adpcm_afc		cdxl			indeo4
adpcm_ct		cinepak			indeo5
adpcm_dtk		cljr			interplay_dpcm
adpcm_ea		cllc			interplay_video
adpcm_ea_maxis_xa	comfortnoise		jacosub
adpcm_ea_r1		cook			jpeg2000
adpcm_ea_r2		cpia			jpegls
adpcm_ea_r3		cscd			jv
adpcm_ea_xas		cyuv			kgv1
adpcm_g722		dca			kmvc
adpcm_g726		dfa			lagarith
adpcm_ima_amv		dirac			libopenjpeg
adpcm_ima_apc		dnxhd			libvorbis
adpcm_ima_dk3		dpx			loco
adpcm_ima_dk4		dsicinaudio		mace3
adpcm_ima_ea_eacs	dsicinvideo		mace6
adpcm_ima_ea_sead	dvbsub			mdec
adpcm_ima_iss		dvdsub			microdvd
adpcm_ima_oki		dvvideo			mimic
adpcm_ima_qt		dxa			mjpeg
adpcm_ima_rad		dxtory			mjpegb
adpcm_ima_smjpeg	eac3			mlp
adpcm_ima_wav		eacmv			mmvideo
adpcm_ima_ws		eamad			motionpixels
adpcm_ms		eatgq			movtext
adpcm_sbpro_2		eatgv			mp1
adpcm_sbpro_3		eatqi			mp1float
adpcm_sbpro_4		eightbps		mp2
adpcm_swf		eightsvx_exp		mp2float
adpcm_thp		eightsvx_fib		mp3
adpcm_xa		escape124		mp3adu
adpcm_yamaha		escape130		mp3adufloat
aic			evrc			mp3float
alac			exr			mp3on4
als			ffv1			mp3on4float
amrnb			ffvhuff			mpc7
amrwb			ffwavesynth		mpc8
amv			flac			mpeg1_vdpau
anm			flashsv			mpeg1video
ansi			flashsv2		mpeg2video
ape			flic			mpeg4
ass			flv			mpeg4_vdpau
asv1			fourxm			mpeg_vdpau
asv2			fraps			mpegvideo
atrac1			frwu			mpl2
atrac3			g2m			msa1
aura			g723_1			msmpeg4v1
aura2			g729			msmpeg4v2
avrn			gif			msmpeg4v3
avrp			gsm			msrle
avs			gsm_ms			mss1
avui			h261			mss2
ayuv			h263			msvideo1
bethsoftvid		h263i			mszh
bfi			h263p			mts2
bink			h264			mvc1
binkaudio_dct		h264_vdpau		mvc2
binkaudio_rdft		huffyuv			mxpeg
bintext			iac			nellymoser
bmp			idcin			nuv
paf_audio		rawvideo		v210x
paf_video		realtext		v308
pam			rl2			v408
pbm			roq			v410
pcm_alaw		roq_dpcm		vb
pcm_bluray		rpza			vble
pcm_dvd			rv10			vc1
pcm_f32be		rv20			vc1_vdpau
pcm_f32le		rv30			vc1image
pcm_f64be		rv40			vcr1
pcm_f64le		s302m			vima
pcm_lxf			sami			vmdaudio
pcm_mulaw		sanm			vmdvideo
pcm_s16be		sgi			vmnc
pcm_s16be_planar	sgirle			vorbis
pcm_s16le		shorten			vp3
pcm_s16le_planar	sipr			vp5
pcm_s24be		smackaud		vp6
pcm_s24daud		smacker			vp6a
pcm_s24le		smc			vp6f
pcm_s24le_planar	smvjpeg			vp8
pcm_s32be		snow			vplayer
pcm_s32le		sol_dpcm		vqa
pcm_s32le_planar	sonic			wavpack
pcm_s8			sp5x			webp
pcm_s8_planar		srt			webvtt
pcm_u16be		ssa			wmalossless
pcm_u16le		subrip			wmapro
pcm_u24be		subviewer		wmav1
pcm_u24le		subviewer1		wmav2
pcm_u32be		sunrast			wmavoice
pcm_u32le		svq1			wmv1
pcm_u8			svq3			wmv2
pcm_zork		tak			wmv3
pcx			targa			wmv3_vdpau
pgm			targa_y216		wmv3image
pgmyuv			text			wnv1
pgssub			theora			ws_snd1
pictor			thp			xan_dpcm
pjs			tiertexseqvideo		xan_wc3
png			tiff			xan_wc4
ppm			tmv			xbin
prores			truehd			xbm
prores_lgpl		truemotion1		xface
ptx			truemotion2		xl
qcelp			truespeech		xsub
qdm2			tscc			xwd
qdraw			tscc2			y41p
qpeg			tta			yop
qtrle			twinvq			yuv4
r10k			txd			zero12v
r210			ulti			zerocodec
ra_144			utvideo			zlib
ra_288			v210			zmbv
ralf

Enabled encoders:
a64multi		libopenjpeg		pgm
a64multi5		libtheora		pgmyuv
aac			libtwolame		png
ac3			libvorbis		ppm
ac3_fixed		libx264			prores
adpcm_adx		libx264rgb		prores_aw
adpcm_g722		libxvid			prores_ks
adpcm_g726		ljpeg			qtrle
adpcm_ima_qt		mjpeg			r10k
adpcm_ima_wav		movtext			r210
adpcm_ms		mp2			ra_144
adpcm_swf		mpeg1video		rawvideo
adpcm_yamaha		mpeg2video		roq
alac			mpeg4			roq_dpcm
amv			msmpeg4v2		rv10
ass			msmpeg4v3		rv20
asv1			msvideo1		s302m
asv2			nellymoser		sgi
avrp			pam			snow
avui			pbm			sonic
ayuv			pcm_alaw		sonic_ls
bmp			pcm_f32be		srt
cljr			pcm_f32le		ssa
comfortnoise		pcm_f64be		subrip
dca			pcm_f64le		sunrast
dnxhd			pcm_mulaw		svq1
dpx			pcm_s16be		targa
dvbsub			pcm_s16be_planar	tiff
dvdsub			pcm_s16le		tta
dvvideo			pcm_s16le_planar	utvideo
eac3			pcm_s24be		v210
ffv1			pcm_s24daud		v308
ffvhuff			pcm_s24le		v408
flac			pcm_s24le_planar	v410
flashsv			pcm_s32be		vorbis
flashsv2		pcm_s32le		wmav1
flv			pcm_s32le_planar	wmav2
g723_1			pcm_s8			wmv1
gif			pcm_s8_planar		wmv2
h261			pcm_u16be		xbm
h263			pcm_u16le		xface
h263p			pcm_u24be		xsub
huffyuv			pcm_u24le		xwd
jpeg2000		pcm_u32be		y41p
jpegls			pcm_u32le		yuv4
libfaac			pcm_u8			zlib
libmp3lame		pcx			zmbv

Enabled hwaccels:
h263_vaapi		mpeg2_vaapi		vc1_vaapi
h263_vdpau		mpeg2_vdpau		vc1_vdpau
h264_vaapi		mpeg4_vaapi		wmv3_vaapi
h264_vdpau		mpeg4_vdpau		wmv3_vdpau
mpeg1_vdpau

Enabled parsers:
aac			dvd_nav			mpegvideo
aac_latm		dvdsub			png
ac3			flac			pnm
adx			gsm			rv30
bmp			h261			rv40
cavsvideo		h263			tak
cook			h264			vc1
dca			mjpeg			vorbis
dirac			mlp			vp3
dnxhd			mpeg4video		vp8
dvbsub			mpegaudio

Enabled demuxers:
aac			hls			pcm_u16be
ac3			ico			pcm_u16le
act			idcin			pcm_u24be
adf			idf			pcm_u24le
adp			iff			pcm_u32be
adx			ilbc			pcm_u32le
aea			image2			pcm_u8
afc			image2pipe		pjs
aiff			ingenient		pmp
amr			ipmovie			pva
anm			ircam			pvf
apc			iss			qcp
ape			iv8			r3d
aqtitle			ivf			rawvideo
asf			jacosub			realtext
ass			jv			redspark
ast			latm			rl2
au			lmlm4			rm
avi			loas			roq
avr			lvf			rpl
avs			lxf			rsd
bethsoftvid		m4v			rso
bfi			matroska		rtp
bink			mgsts			rtsp
bintext			microdvd		sami
bit			mjpeg			sap
bmv			mlp			sbg
boa			mm			sdp
brstm			mmf			segafilm
c93			mov			shorten
caf			mp3			siff
cavsvideo		mpc			smacker
cdg			mpc8			smjpeg
cdxl			mpegps			smush
concat			mpegts			sol
data			mpegtsraw		sox
daud			mpegvideo		spdif
dfa			mpl2			srt
dirac			mpsub			str
dnxhd			msnwc_tcp		subviewer
dsicin			mtv			subviewer1
dts			mv			swf
dtshd			mvi			tak
dv			mxf			tedcaptions
dxa			mxg			thp
ea			nc			tiertexseq
ea_cdata		nistsphere		tmv
eac3			nsv			truehd
epaf			nut			tta
ffm			nuv			tty
ffmetadata		ogg			txd
filmstrip		oma			vc1
flac			paf			vc1t
flic			pcm_alaw		vivo
flv			pcm_f32be		vmd
fourxm			pcm_f32le		vobsub
frm			pcm_f64be		voc
g722			pcm_f64le		vplayer
g723_1			pcm_mulaw		vqf
g729			pcm_s16be		w64
gif			pcm_s16le		wav
gsm			pcm_s24be		wc3
gxf			pcm_s24le		webvtt
h261			pcm_s32be		wsaud
h263			pcm_s32le		wsvqa
h264			pcm_s8			wtv
wv			xmv			yop
xa			xwma			yuv4mpegpipe
xbin

Enabled muxers:
a64			image2pipe		pcm_s24be
ac3			ipod			pcm_s24le
adts			ircam			pcm_s32be
adx			ismv			pcm_s32le
aiff			ivf			pcm_s8
amr			jacosub			pcm_u16be
asf			latm			pcm_u16le
asf_stream		m4v			pcm_u24be
ass			matroska		pcm_u24le
ast			matroska_audio		pcm_u32be
au			md5			pcm_u32le
avi			microdvd		pcm_u8
avm2			mjpeg			psp
bit			mkvtimestamp_v2		rawvideo
caf			mlp			rm
cavsvideo		mmf			roq
crc			mov			rso
data			mp2			rtp
daud			mp3			rtsp
dirac			mp4			sap
dnxhd			mpeg1system		segment
dts			mpeg1vcd		smjpeg
dv			mpeg1video		smoothstreaming
eac3			mpeg2dvd		sox
f4v			mpeg2svcd		spdif
ffm			mpeg2video		srt
ffmetadata		mpeg2vob		stream_segment
filmstrip		mpegts			swf
flac			mpjpeg			tee
flv			mxf			tg2
framecrc		mxf_d10			tgp
framemd5		null			truehd
g722			nut			vc1
g723_1			ogg			vc1t
gif			oma			voc
gxf			pcm_alaw		w64
h261			pcm_f32be		wav
h263			pcm_f32le		webm
h264			pcm_f64be		webvtt
hls			pcm_f64le		wtv
ico			pcm_mulaw		wv
ilbc			pcm_s16be		yuv4mpegpipe
image2			pcm_s16le

Enabled protocols:
cache			gopher			pipe
concat			hls			rtmp
crypto			http			rtmpt
data			httpproxy		rtp
ffrtmphttp		md5			srtp
file			mmsh			tcp
ftp			mmst			udp

Enabled filters:
aconvert		curves			nullsink
aecho			dctdnoiz		nullsrc
aevalsrc		decimate		overlay
afade			delogo			owdenoise
aformat			deshake			pad
ainterleave		drawbox			pan
allpass			drawgrid		perms
alphaextract		drawtext		perspective
alphamerge		earwax			pixdesctest
amerge			ebur128			pp
amix			edgedetect		psnr
amovie			equalizer		removelogo
anull			extractplanes		rgbtestsrc
anullsink		fade			rotate
anullsrc		field			sab
apad			fieldmatch		scale
aperms			fieldorder		select
aphaser			format			sendcmd
aresample		fps			separatefields
aselect			framestep		setdar
asendcmd		geq			setfield
asetnsamples		gradfun			setpts
asetpts			haldclut		setsar
asetrate		haldclutsrc		settb
asettb			hflip			showinfo
ashowinfo		highpass		showspectrum
asplit			histeq			showwaves
astats			histogram		silencedetect
astreamsync		hqdn3d			sine
atempo			hue			smartblur
atrim			idet			smptebars
avectorscope		il			smptehdbars
bandpass		interlace		split
bandreject		interleave		spp
bass			join			stereo3d
bbox			kerndeint		super2xsai
biquad			life			swapuv
blackdetect		lowpass			telecine
blackframe		lut			testsrc
blend			lut3d			thumbnail
boxblur			lutrgb			tile
cellauto		lutyuv			tinterlace
channelmap		mandelbrot		transpose
channelsplit		mcdeint			treble
color			movie			trim
colorbalance		mp			unsharp
colorchannelmixer	mpdecimate		vflip
colormatrix		mptestsrc		vignette
concat			negate			volume
copy			noformat		volumedetect
crop			noise			yadif
cropdetect		null

Enabled bsfs:
aac_adtstoasc		mjpeg2jpeg		mp3_header_decompress
chomp			mjpega_dump_header	noise
dump_extradata		mov2textsub		remove_extradata
h264_mp4toannexb	mp3_header_compress	text2movsub
imx_dump_header

Enabled indevs:
alsa			lavfi			v4l2
dv1394			libdc1394		x11grab
fbdev			oss

Enabled outdevs:
alsa			sdl			v4l2
oss

License: nonfree and unredistributable
Creating config.mak, config.h, and doc/config.texi...
libavutil/avconfig.h is unchanged
ned337 at vid:~/ffmpeg$ 

It looks enabled to me and I will install git. 
I have the entire make cl output too, if you need it. 
Thanks, 
Ned 



From cehoyos at ag.or.at  Mon Aug  5 19:57:29 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 5 Aug 2013 17:57:29 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg Unknown encoder 'libx264' error
References: <1375666070.5067.41.camel@videoathlon>
Message-ID: <loom.20130805T194808-897@post.gmane.org>

Ned Haskin <feanor2112 <at> covad.net> writes:

> $ ./configure --enable-libdc1394 --prefix=/usr 

Mostly unrelated to your original question:

> --extra-cflags='-Wall -g '

Please remove this.

> --enable-gpl

> --enable-shared 

(This is of course 100% ok, but if you are unexperienced 
I would strongly recommend not to use it, it generally 
has theoretical problems and the gain for you as a 
user is not completely clear to me, in combination with 
"--prefix=/usr", I personally consider it highly 
dangerous and would not use it on my system.)

> --enable-nonfree --enable-libx264 

> --enable-encoder=libx264

This makes no sense in the configure line you posted.

> --enable-libmp3lame --enable-libvorbis 

> --enable-libtwolame --enable-libv4l2

(I would even recommend to remove these two.)

> --enable-pthreads

This is the default, please remove it.

> --enable-libfaac --enable-libxvid

> --enable-postproc

This also makes no sense.

> --enable-x11grab --enable-libtheora --disable-stripping
> --enable-avfilter --enable-libfreetype --enable-opencl 

> --enable-version3

I suspect you do not need / want this.

> --enable-libopenjpeg 

> --disable-altivec --disable-armv5te 
> --disable-armv6 --disable-vis 

I may miss something but I suspect you should remove 
those (at least some of them).

Carl Eugen


From cehoyos at ag.or.at  Mon Aug  5 20:00:03 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 5 Aug 2013 18:00:03 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg Unknown encoder 'libx264' error
References: <1375666070.5067.41.camel@videoathlon>
	<loom.20130805T141926-122@post.gmane.org>
	<1375725011.3919.8.camel@videoathlon>
Message-ID: <loom.20130805T195757-357@post.gmane.org>

Ned Haskin <feanor2112 <at> covad.net> writes:

> > >   built on Jul 29 2013 13:49:23 with gcc 4.7 (Debian 4.7.2-5)
> > >   configuration: 
> > 
> > This means you compiled without x264 support ...
> > 
> > > Unknown encoder 'libx264'
> > 
> > ... so it cannot be used.
> > 
> > Carl Eugen Hoyos
> > 
> > 
> Carl, my ./configure for ffmpeg has --enable-libx264

Not for the executable that you posted output for, just 
read the line above beyond "built on".

As said, please update to current git head and please 
read my other comment about your configure line, 
regarding my last paragraph: I did not miss anything, 
please remove the last options.

Carl Eugen


From dave at dericed.com  Mon Aug  5 20:13:21 2013
From: dave at dericed.com (Dave Rice)
Date: Mon, 5 Aug 2013 14:13:21 -0400
Subject: [FFmpeg-user] 50 Mbps Ifram mpg2 with scaling and pad --sync
	problem
In-Reply-To: <51FFA44D.1010104@pps-inc.com>
References: <99b707c624bfd985e2248ec04e42b10b.squirrel@webmail.theppsgroup.com>
	<51FFA44D.1010104@pps-inc.com>
Message-ID: <6C9B9680-AF1D-45A9-9D36-E66521625A5E@dericed.com>

Hi Jim,

On Aug 5, 2013, at 9:10 AM, Jim Shupert <jshupert at pps-inc.com> wrote:
> On 8/2/2013 12:46 PM, Jim Shupert, Jr wrote:
>> Friends,
>> I wish to take 720x486 Unc 8 bit rawVideo with 2 chn  pcm audio
>> and make 50 mbps Iframe mpg2 422 1920x1080i with mpg2 audio
>> 
>> I realize that upScaling is not best - but I do need to in this situation.
>> 
>> I have something that I think is close but the audio goes out of sync.
>> I wonder why?
>> 
>> You will see I have my buffer set rather high - i did that to avoid
>> ...
>> [mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
>> [mpeg @ 0x307afa0] packet too large, ignoring buffer limits to mux it
>> [mpeg @ 0x307afa0] buffer underflow i=0 bufi=354801 size=358510
>> ...
>> 
>> here is my command
>> 
>> jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
>> "scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
>> mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
>> +ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
>> 5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
>> -async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg

This probably isn't the answer you need, but with your source material I'd suggest using an interlace-aware scaler, i.e. try scale=1620:1080:interl=1 instead of scale=1620:1080.

Also video that is 720x486 and captured from analog source often can be cropped down to 720x480 at 4/3 by removing the first 4 and last two lines. You may have a better looking result if you start your filterchain with crop=720:480:0:4 before scaling.

Dave Rice

>> jj at gagarin:~$ ffmpeg -i /media/data/z_parts/dog.avi -vf
>> "scale=1620:1080,pad=1920:1080:150:0:black"  -pix_fmt yuv422p -c:v
>> mpeg2video  -minrate 50000k -maxrate 50000k -b:v 50000k -intra -flags
>> +ildct+low_delay -g 1  -dc 10  -ps 1 -qmin 1 -qmax 3 -top 1  -bufsize
>> 5000000 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25  -c:a mp2
>> -async 1 -b:a 224k -ar 48000 -ac 2    /media/data/jtest/dog-3a_1080i.mpg
>> ffmpeg version git-2012-09-24-fd63c2f Copyright (c) 2000-2012 the FFmpeg
>> developers
>> built on Sep 24 2012 14:09:13 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
>> configuration: --enable-gpl --enable-libfaac --enable-libmp3lame
>> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora
>> --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
>> --enable-version3 --enable-x11grab
>> libavutil      51. 73.101 / 51. 73.101
>> libavcodec     54. 58.100 / 54. 58.100
>> libavformat    54. 28.101 / 54. 28.101
>> libavdevice    54.  2.101 / 54.  2.101
>> libavfilter     3. 17.100 /  3. 17.100
>> libswscale      2.  1.101 /  2.  1.101
>> libswresample   0. 15.100 /  0. 15.100
>> libpostproc    52.  0.100 / 52.  0.100
>> [avi @ 0x28b5240] non-interleaved AVI
>> Guessed Channel Layout for  Input Stream #0.1 : stereo
>> Input #0, avi, from '/media/data/z_parts/dog.avi':
>> Duration: 00:00:05.03, start: 0.000000, bitrate: 170941 kb/s
>>   Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x486,
>> 29.97 tbr, 29.97 tbn, 29.97 tbc
>>   Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
>> stereo, s16, 1536 kb/s
>> File '/media/data/jtest/dog-3a_1080i.mpg' already exists. Overwrite ? [y/N] y
>> -async is forwarded to lavfi similarly to -af
>> aresample=min_comp=0.001:min_hard_comp=0.100000.
>> Output #0, mpeg, to '/media/data/jtest/dog-3a_1080i.mpg':
>> Metadata:
>>   encoder         : Lavf54.28.101
>>   Stream #0:0: Video: mpeg2video, yuv422p, 1920x1080, q=1-3, 50000 kb/s,
>> 90k tbn, 29.97 tbc
>>   Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 224 kb/s
>> Stream mapping:
>> Stream #0:0 -> #0:0 (rawvideo -> mpeg2video)
>> Stream #0:1 -> #0:1 (pcm_s16le -> mp2)
>> Press [q] to stop, [?] for help
>> [mpeg2video @ 0x28b6800] rc buffer underflow
>>   Last message repeated 24 times
>> frame=   25 fps=0.0 q=24.8 size=    6190kB time=00:00:00.83
>> bitrate=60789.4kbitsrc buffer underflow
>> [mpeg2video @ 0x28b6800] rc buffer underflow
>> 
> Maybe I could ask another way..
> what might be the suggested means to scale raw video , uncompressed d1 720x486 ntsc w pcm audio to 50 Mbps I fram mpg2.
> I am scaling and padding . My audio comes out of sync.
> Likely this is unrelated to the scaling and padding - but maybe not.
> and
> Does anyone have an opinion on buffer setting?
> 
> Thank you
> 
> best regards
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From cehoyos at ag.or.at  Mon Aug  5 20:21:43 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 5 Aug 2013 18:21:43 +0000 (UTC)
Subject: [FFmpeg-user] 50 Mbps Ifram mpg2 with scaling and pad --sync
	problem
References: <99b707c624bfd985e2248ec04e42b10b.squirrel@webmail.theppsgroup.com>
	<51FFA44D.1010104@pps-inc.com>
Message-ID: <loom.20130805T202033-596@post.gmane.org>

Jim Shupert <jshupert <at> pps-inc.com> writes:

> what might be the suggested means to scale raw video , 
> uncompressed d1 720x486 ntsc w pcm audio to 50 Mbps I 
> fram mpg2. I am scaling and padding . My audio comes 
> out of sync.

Please provide an input sample that allows to 
reproduce the A/V desync with current git head (!).

Carl Eugen


From jshupert at pps-inc.com  Mon Aug  5 20:24:29 2013
From: jshupert at pps-inc.com (Jim Shupert)
Date: Mon, 05 Aug 2013 14:24:29 -0400
Subject: [FFmpeg-user] 50 Mbps Ifram mpg2 with scaling and pad --sync
 problem
In-Reply-To: <6C9B9680-AF1D-45A9-9D36-E66521625A5E@dericed.com>
References: <99b707c624bfd985e2248ec04e42b10b.squirrel@webmail.theppsgroup.com>
	<51FFA44D.1010104@pps-inc.com>
	<6C9B9680-AF1D-45A9-9D36-E66521625A5E@dericed.com>
Message-ID: <51FFEDDD.3030602@pps-inc.com>


On 8/5/2013 2:13 PM, Dave Rice wrote:
> Hi Jim,
>
>
> This probably isn't the answer you need, but with your source material I'd suggest using an interlace-aware scaler, i.e. try scale=1620:1080:interl=1 instead of scale=1620:1080.
>
> Also video that is 720x486 and captured from analog source often can be cropped down to 720x480 at 4/3 by removing the first 4 and last two lines. You may have a better looking result if you start your filterchain with crop=720:480:0:4 before scaling.
>
Thanks Dave
I have not done the

crop=720:480:0:4 before scaling before



From jpboard2 at yahoo.com  Mon Aug  5 20:32:11 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 5 Aug 2013 11:32:11 -0700 (PDT)
Subject: [FFmpeg-user] OpenCL?  What Effect Does It Have
In-Reply-To: <loom.20130805T194218-441@post.gmane.org>
References: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<loom.20130805T194218-441@post.gmane.org>
Message-ID: <1375727531.54700.YahooMailNeo@web164705.mail.gq1.yahoo.com>

>In theory, you could use opencl for the deshake and 
>the unsharp filters but I don't know if sufficiently 
>new OpenCL drivers exist.

Unrelated to my question.? I don't use deshake and I don't
use unsharp.? That's why I listed the filters I do use: so that
someone can tell me whether or not OpenCL will improve
performance for those filters (and codecs).? That was my 
original question.
?
>> * Inverse telecine, or deinterlace filters
>
>Unrelated to your question:

?
No.? Not unrelated to my question.? In fact, that?is my question:
Will ffmpeg+OpenCL improve the performance of the filters
and codecs I listed.
?
>Given that MEncoder uses the exact same source code 
>to encode ffvhuff and ljpeg (and most probably "etc") 
> how do you expect that FFmpeg could be faster?

?
But I don't expect ffmpeg is faster then mencoder.? Possibly,
ffmpeg+OpenCL is faster.? That's the original question.
My original question was whether ot not ffmpeg+OpenCL
will improve my performance.? Will it?
?
>Allow me to repeat that since MEncoder is not 
>actively maintained, it makes no sense to use it if 
>you don't have very special needs 
?
I already know how to use mencoder.? Why learn another
tool which uses, as you claim, the exact same source code?
?
But that's besides the point.? The?original question was whether 
or not ffmpeg+OpenCL will improve my performance.? Not whether or
not it makes sense to switch from mencoder to ffmpeg.

From cehoyos at ag.or.at  Mon Aug  5 20:44:30 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 5 Aug 2013 18:44:30 +0000 (UTC)
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
Message-ID: <loom.20130805T204233-571@post.gmane.org>

Francois Visagie <francois.visagie <at> gmail.com> writes:

> > As expected, the three output files are bit-identical.
> 
> Thanks for doing this testing, which I repeated here.
> 
> With
> [ff130d7] which I was using up to now, out2.wav differs 
> from the other two.

This changed in fe328f0

> However, it's the only one equally loud as the input file.

I suspect that this was unintended (they should be identical), 
but I may miss something.

Carl Eugen


From leo.currie at gmail.com  Mon Aug  5 21:23:13 2013
From: leo.currie at gmail.com (Leo Currie)
Date: Mon, 5 Aug 2013 20:23:13 +0100
Subject: [FFmpeg-user] audio-only flv over rtmp
In-Reply-To: <20130805105008.GF1508@krille.blacktrash.org>
References: <CANBFD3oOjoiHKWFWLyc0EKM9hzqZhqDFUYXF_q80h=-BpaXhEQ@mail.gmail.com>
	<20130805105008.GF1508@krille.blacktrash.org>
Message-ID: <CANBFD3rKFoYe-HY36sJ2tfqkBSNcx6vOBNzD7H49T3cGdfggvg@mail.gmail.com>

On 5 August 2013 11:50, Christian Ebert <blacktrash at gmx.net> wrote:

> * Leo Currie on Monday, August 05, 2013 at 11:30:52 +0100
> > I'm trying to generate an RTMP live stream for Flash clients that
> contains
> > audio only.
>
> Why don't you stream just MP3?
>

My source is actually a 32kbps AAC+ stream coming from an Icecast server,
so I should really be doing '-acodec copy'.


>
> > Is there a special option to prevent the flv muxer generating a video
> > stream?
> >
> > Here's an example of the kind of stream I'd like to generate:
> > rtmp://87.117.250.4/radiosix3/
> >
> > I'm trying this:
> > ffmpeg -re -i (my input stream) -acodec aac -strict experimental -ab 96k
> > -ar 44100 -vn -f flv rtmp://127.0.0.1/live/test
> > ..which connects to my local nginx instance running the rtmp module.
> >
> > When I try to play the stream in VLC, it complains about an 'undf' video
> > format, but audio plays fine. Looking at the stream properties in VLC I
> can
> > see there is indeed an 'undf' video stream 0, with the audio being
> stream 1.
> >
> > Is there an option for the flv muxer to prevent this video stream being
> > generated? I tried -vn but this doesn't do what I expect.
> >
> > Or am I going about this the wrong way?
>
> FLV is Flash Video, so a missing video track does not make sense
> imho. Furthermore I don't think you need this contortion for RTMP
> (see above). I packed audio into a FLV container for
> *pseudo*streaming, but again then you need a _minimal_ video
> track to be able to inject metadata.
>


Indeed, and MPEG is Moving Picture Experts Group, but we can have MP3
audio. So perhaps none of these formats have very good names!

I didn't know that a video track was needed for metadata, so perhaps all I
need to do is be able to generate the "right kind" of dummy video track for
my FLV.

Or is there another way to stream aac audio over rtmp in a way that Flash
player (e.g. JWPlayer) will understand?
Perhaps a different mux? Is this what you meant by streaming in MP3 - can I
stream without a container?

Thanks -

Leo

From blacktrash at gmx.net  Mon Aug  5 21:58:40 2013
From: blacktrash at gmx.net (Christian Ebert)
Date: Mon, 5 Aug 2013 20:58:40 +0100
Subject: [FFmpeg-user] audio-only flv over rtmp
In-Reply-To: <CANBFD3rKFoYe-HY36sJ2tfqkBSNcx6vOBNzD7H49T3cGdfggvg@mail.gmail.com>
References: <CANBFD3oOjoiHKWFWLyc0EKM9hzqZhqDFUYXF_q80h=-BpaXhEQ@mail.gmail.com>
	<20130805105008.GF1508@krille.blacktrash.org>
	<CANBFD3rKFoYe-HY36sJ2tfqkBSNcx6vOBNzD7H49T3cGdfggvg@mail.gmail.com>
Message-ID: <20130805195840.GG1508@krille.blacktrash.org>

* Leo Currie on Monday, August 05, 2013 at 20:23:13 +0100
> On 5 August 2013 11:50, Christian Ebert <blacktrash at gmx.net> wrote:
>> * Leo Currie on Monday, August 05, 2013 at 11:30:52 +0100
>>> I'm trying to generate an RTMP live stream for Flash clients that
>>> contains audio only.
>> 
>> Why don't you stream just MP3?
> 
> My source is actually a 32kbps AAC+ stream coming from an Icecast server,
> so I should really be doing '-acodec copy'.

Problem might be the live packaging. I only know about on-demand.

You can certainly stream a aac audio in a proper container like
M4A. But I would not know how to do this kind of "muxing" for
live streams.

> Indeed, and MPEG is Moving Picture Experts Group, but we can have MP3
> audio. So perhaps none of these formats have very good names!
> 
> I didn't know that a video track was needed for metadata, so perhaps all I
> need to do is be able to generate the "right kind" of dummy video track for
> my FLV.
> 
> Or is there another way to stream aac audio over rtmp in a way that Flash
> player (e.g. JWPlayer) will understand?
> Perhaps a different mux? Is this what you meant by streaming in MP3 - can I
> stream without a container?

See above. An M4A on demand stream can certainly be played in a
Flash player - at least Flowplayer can do it. But again, for live
streams I don't know. Sorry.

-- 
theatre - books - texts - movies
Black Trash Productions at home: http://www.blacktrash.org
Black Trash Productions on Facebook:
http://www.facebook.com/blacktrashproductions

From barsnick at gmx.net  Mon Aug  5 23:41:58 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Mon, 5 Aug 2013 23:41:58 +0200
Subject: [FFmpeg-user] audio-only flv over rtmp
In-Reply-To: <20130805105008.GF1508@krille.blacktrash.org>
References: <CANBFD3oOjoiHKWFWLyc0EKM9hzqZhqDFUYXF_q80h=-BpaXhEQ@mail.gmail.com>
	<20130805105008.GF1508@krille.blacktrash.org>
Message-ID: <20130805214158.GC20067@sunshine.barsnick.net>

On Mon, Aug 05, 2013 at 11:50:09 +0100, Christian Ebert wrote:
> FLV is Flash Video, so a missing video track does not make sense
> imho.

Are you sure? It's just "another" container format. If I use the tool myspace-dl
to dump MySpace's music tracks to disks (an RTMP "stream", dumped with
rtmpdump), it gives me something like this:

> ffprobe ~/Downloads/Upstrokes/Upstrokes\ -\ 15\ Punx.flv
ffprobe version 1.1.git Copyright (c) 2007-2013 the FFmpeg developers
  built on Jul  2 2013 23:02:05 with icc (ICC) 13.1.1 20130313
  configuration: --prefix=/usr/new/tools/video/install/ffmpeg/20130702 --enable-gpl --enable-version3 --enable-nonfree --disable-shared --enable-x11grab --enable-gnutls --enable-libcdio --enable-libfreetype --enable-libx264 --enable-libmp3lame --enable-openal --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libass --enable-libv4l2 --enable-libvidstab --enable-libfdk-aac --cc=icc --cxx=icpc --extra-ldflags='-L /usr/new/tools/video/install/x264/current/lib -L /usr/new/tools/video/install/fdk-aac/current/lib' --extra-cflags='-I /usr/new/tools/video/install/x264/current/include -I /usr/new/tools/video/install/fdk-aac/current/include'
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.100 / 55. 18.100
  libavformat    55. 11.100 / 55. 11.100
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 77.101 /  3. 77.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Input #0, flv, from '/usr/home/barsnick/Downloads/Upstrokes/Upstrokes - 15 Punx.flv':
  Metadata:
    moovPosition    : 1433058
    aacaot          : 2
    audiochannels   : 2
  Duration: 00:01:48.80, start: 0.000000, bitrate: 111 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp

That indicates that that container contains only audio. (Other tools
confirm this.) I have no idea whether that's considered a "live stream"
though, and whether that would change anything.

I'm not very firm with the simplicity of containers, but this header
description nicely hints that FLV can contain either video, audio, or
both (max one of each):
http://en.wikipedia.org/wiki/FLV#Header

Leo, have you tried pointing ffprobe both at your Radio Six stream and
your self-generated stream, to see what the difference is? Don't give
up! :-)

Moritz

From jiachielee at live.com  Tue Aug  6 00:04:32 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 5 Aug 2013 15:04:32 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375716335502-4660464.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
Message-ID: <1375740272587-4660485.post@n4.nabble.com>

<iframe width="560" height="315" src="//www.youtube.com/embed/udrxmrLIDGU"
frameborder="0" allowfullscreen></iframe>


Well, I hope the above slideshow I created can help you.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Inaccurate-trimming-with-re-encoding-tp4660437p4660485.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From Eric at ebbtech.net  Tue Aug  6 01:50:14 2013
From: Eric at ebbtech.net (Eric)
Date: Mon, 05 Aug 2013 18:50:14 -0500
Subject: [FFmpeg-user] Compositing more than 2 videos side-by-side
Message-ID: <6x770gkui6037p8h7tjukhr5.1375746614612@email.android.com>

I'm trying to figure out how to combine 3-4 videos with ffmpeg so each video is in its own quadrant of the output video file. I have had success combining two videos side-by-side. Perhaps someone with a better working knowledge of ffmpeg could help me out??

Thanks,?
Eric


From jpboard2 at yahoo.com  Tue Aug  6 01:56:18 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 5 Aug 2013 16:56:18 -0700 (PDT)
Subject: [FFmpeg-user] ffmpeg build fail with libx264
Message-ID: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>

I'm trying to switch from mencoder to ffmpeg as Carl suggested.? In
order to do this, I have to rebuild ffmpeg with libx264.? After my first
attempt failed, I got the latest version of x264, built it, and installed

??? git clone git:://git.videolan.org/x264.git
??? make
??? make install

There were no errors or warnings.? However, the /usr/include/264.h file wasn't updated.
That version is still 107.


Then I tried to configure ffmpeg
??? ./configure --enable-gpl --enable-libx264
and received this error
??? ERROR: libx264 must be installed and version must be >= 0.118

How to proceed?

Thank You.

From lou at lrcd.com  Tue Aug  6 02:04:34 2013
From: lou at lrcd.com (Lou)
Date: Mon, 5 Aug 2013 16:04:34 -0800
Subject: [FFmpeg-user] Compositing more than 2 videos side-by-side
In-Reply-To: <6x770gkui6037p8h7tjukhr5.1375746614612@email.android.com>
References: <6x770gkui6037p8h7tjukhr5.1375746614612@email.android.com>
Message-ID: <20130805160434.354b21a3@lrcd.com>

On Mon, 05 Aug 2013 18:50:14 -0500
Eric <Eric at ebbtech.net> wrote:

> I'm trying to figure out how to combine 3-4 videos with ffmpeg so each video is in its own quadrant of the output video file. I have had success combining two videos side-by-side. Perhaps someone with a better working knowledge of ffmpeg could help me out??
> 
> Thanks,?
> Eric

See this example:
<https://trac.ffmpeg.org/wiki/FilteringGuide#multipleinputoverlayin2x2grid>

It's not as complicated as it looks. The example just applies some
extra filters to differentiate the inputs.

Basically it uses the pad [1] filter to create a "canvas" and then uses
overlay [2] filters to place each quadrant.

[1] http://ffmpeg.org/ffmpeg-filters.html#pad
[2] http://ffmpeg.org/ffmpeg-filters.html#overlay-1

From cehoyos at ag.or.at  Tue Aug  6 02:10:25 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 6 Aug 2013 00:10:25 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg build fail with libx264
References: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <loom.20130806T020742-383@post.gmane.org>

James Board <jpboard2 <at> yahoo.com> writes:

> ??? git clone git:://git.videolan.org/x264.git
> ??? make

This fails here (and makes it practically impossible 
to help you / answer your question).

[...]

> Then I tried to configure ffmpeg
> ??? ./configure --enable-gpl --enable-libx264

You probably need something like: 
--extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib

> and received this error
> ERROR: libx264 must be installed and version must be >= 0.118

Doesn't your distribution offer a newer version?
(And how does this work with MEncoder which uses 
the exact same configure check?)

Carl Eugen


From parshap at gmail.com  Mon Aug  5 22:37:09 2013
From: parshap at gmail.com (Parsha Pourkhomami)
Date: Mon, 5 Aug 2013 13:37:09 -0700
Subject: [FFmpeg-user] Writing an MP3 file fails only when piping output
Message-ID: <CAHjQg2ogcM6ATWQwuKoR7uoRc3c5ZXZKeVbstMxyPGZ63HFhkg@mail.gmail.com>

While updating MP3 ID3 tags using the `-metadata` option, I have come
across some files that seem to break ffmpeg but only when writing out
to pipe:1. By break, I mean that when reading back the output file,
ffprobe reports an error, some players refuse to play it, and all the
original metadata (ID3 tags) seem to have been stripped.

If I write out to a file path, then it works fine! Here is a demonstration:

> ffmpeg -i test.mp3 -codec copy -f mp3 pipe:1 > piped.mp3
> ffprobe -show_format piped.mp3
[mp3 @ 0x9c4f0e0] Header missing
... more ffprobe output

> ffmpeg -i test.mp3 -codec copy -f mp3 saved.mp3
> ffprobe -show_format saved.mp3
... normal ffprobe output

I've tested multiple files and have found several failing examples
from different sources, so it does not seem to a one-off encoding
error. It also seems like if one MP3 file from a collection (e.g.,
album) fails then the rest from that group will fail too.

Here is an example of a failing file: https://s3.amazonaws.com/parshap/test.mp3

From jpboard2 at yahoo.com  Tue Aug  6 02:41:10 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 5 Aug 2013 17:41:10 -0700 (PDT)
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from mencoder)
Message-ID: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>

I'm trying to convert from mencoder to ffmpeg, at Carl's suggestion.? I want
to do 2-pass encoding.? I saw this command (for pass 1) in an ffmpeg user
guide and tried this:

??? ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -an -vcodec libx264 -vpre fast_firstpass -b 2000k -bt 2000k -vf "crop=1440:1080:240:0,scale=640:480" -pass 1 -threads 0 /dev/null

This was the error

??? /usr/local/share/ffmpeg/libx264-fast_firstpass.ffpreset: Invalid option or argument: 'directpred=1', parsed as 'directpred' = '1'


----------------------------------------
I saw this in another post on how to do 2-pass encoding:

??? /usr/local/bin/ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -c:v libx264 -preset medium -b:v 10m -pass 1 -an -f mp4 /dev/null
??? /usr/local/bin/ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -c:v libx264 -preset medium -b:v 10m -pass 2 -an OUT.mp4

There was no error on the first command.? The error on the second was
??? Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

----------------------------------------

So, I want to use ffmpeg instead of mencoder for doing 2-pass mpeg4 encoding.
What's wrong with the above commands, and what's the correct way to do this?

Thank You

From jpboard2 at yahoo.com  Tue Aug  6 02:44:40 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 5 Aug 2013 17:44:40 -0700 (PDT)
Subject: [FFmpeg-user] ffmpeg build fail with libx264
In-Reply-To: <loom.20130806T020742-383@post.gmane.org>
References: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<loom.20130806T020742-383@post.gmane.org>
Message-ID: <1375749880.42456.YahooMailNeo@web164701.mail.gq1.yahoo.com>

>> ??? git clone git:://git.videolan.org/x264.git

>> ??? make
>
>This fails here (and makes it practically impossible 
>to help you / answer your question).

What fails where?? I really don't understand you.? That command didn't
fail.? In fact I explicitly stated there were no errors or warnings for
that command.? ????

>> and received this error
>> ERROR: libx264 must be installed and version must be >= 0.118
>
>Doesn't your distribution offer a newer version?
>(And how does this work with MEncoder which uses 
>the exact same configure check?)

No, Centos 6.4.? Version 107 of x264.? Mencoder had no problems
with it.

From soho123.2012 at gmail.com  Tue Aug  6 08:16:22 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 14:16:22 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
	audio to the specific stream
Message-ID: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>

hi All,


the question is :
how to set video to the specific stream
how to set audio to the specific stream
if I use ffmpeg to capture both video and audio simultaneously?

for example :
ffmpeg command:
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy
http://localhost:8090/feed1.ffm

I get the result :
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
  Metadata:
    creation_time   : now
    encoder         : Lavf55.12.100
    Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
    Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30, q=2-31,
-5 kb/s, 30 fps, 1000k tbn, 30 tbc
Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)

how to modify the stream layout when capture video + audio?

From francois.visagie at gmail.com  Tue Aug  6 08:39:43 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Tue, 6 Aug 2013 08:39:43 +0200
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
	audio to the specific stream
In-Reply-To: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>
References: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>
Message-ID: <003e01ce926f$c2758570$47609050$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Soho Soho123
> Sent: 06 August 2013 08:16
> To: FFmpeg user questions and RTFMs
> Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video , audio to the
> specific stream
> 
> hi All,
> 
> 
> the question is :
> how to set video to the specific stream
> how to set audio to the specific stream
> if I use ffmpeg to capture both video and audio simultaneously?
> 
> for example :
> ffmpeg command:
> ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i
> /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy
> http://localhost:8090/feed1.ffm
> 
> I get the result :
> Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
>   Metadata:
>     creation_time   : now
>     encoder         : Lavf55.12.100
>     Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536
kb/s
>     Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30,
q=2-31,
> -5 kb/s, 30 fps, 1000k tbn, 30 tbc
> Stream mapping:
>   Stream #1:0 -> #0:0 (copy)
>   Stream #0:0 -> #0:1 (copy)
> 
> how to modify the stream layout when capture video + audio?

Do you mean you want to reverse the stream order in this instance? In that
case, use the -map option to enumerate streams in the order you want them to
appear in the output:

-map 0:1 -map 0:0

will put the 2nd stream (video in this case) first.

> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From soho123.2012 at gmail.com  Tue Aug  6 08:43:57 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 14:43:57 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
 audio to the specific stream
In-Reply-To: <003e01ce926f$c2758570$47609050$@gmail.com>
References: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>
	<003e01ce926f$c2758570$47609050$@gmail.com>
Message-ID: <CAFd2cZWjehypM1KsMMKO0xKWCscLZn-PjxGsRipV+Xv0iAMH+w@mail.gmail.com>

Hi ,

if I want to reverse the stream order ,
where I should put the "-map 0:1 -map 0:0"?

for exampe :

ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy
-map 0:1 -map 0:0 http://localhost:8090/feed1.ffm

is the order correct if I put the "map" option ?


2013/8/6 Francois Visagie <francois.visagie at gmail.com>

> > -----Original Message-----
> > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> > bounces at ffmpeg.org] On Behalf Of Soho Soho123
> > Sent: 06 August 2013 08:16
> > To: FFmpeg user questions and RTFMs
> > Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video , audio to the
> > specific stream
> >
> > hi All,
> >
> >
> > the question is :
> > how to set video to the specific stream
> > how to set audio to the specific stream
> > if I use ffmpeg to capture both video and audio simultaneously?
> >
> > for example :
> > ffmpeg command:
> > ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format
> h264
> -i
> > /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy
> > http://localhost:8090/feed1.ffm
> >
> > I get the result :
> > Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
> >   Metadata:
> >     creation_time   : now
> >     encoder         : Lavf55.12.100
> >     Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536
> kb/s
> >     Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30,
> q=2-31,
> > -5 kb/s, 30 fps, 1000k tbn, 30 tbc
> > Stream mapping:
> >   Stream #1:0 -> #0:0 (copy)
> >   Stream #0:0 -> #0:1 (copy)
> >
> > how to modify the stream layout when capture video + audio?
>
> Do you mean you want to reverse the stream order in this instance? In that
> case, use the -map option to enumerate streams in the order you want them
> to
> appear in the output:
>
> -map 0:1 -map 0:0
>
> will put the 2nd stream (video in this case) first.
>
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From francois.visagie at gmail.com  Tue Aug  6 08:56:53 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Tue, 6 Aug 2013 08:56:53 +0200
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from
	mencoder)
In-Reply-To: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <003f01ce9272$25a32920$70e97b60$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of James Board
> Sent: 06 August 2013 02:41
> To: ffmpeg-user at ffmpeg.org
> Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from
> mencoder)
> 
> I'm trying to convert from mencoder to ffmpeg, at Carl's suggestion.? I
want
> to do 2-pass encoding.? I saw this command (for pass 1) in an ffmpeg user
> guide and tried this:
> 
> ??? ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -an -vcodec libx264 -vpre
> fast_firstpass -b 2000k -bt 2000k -vf "crop=1440:1080:240:0,scale=640:480"
-
> pass 1 -threads 0 /dev/null
> 
> This was the error
> 
> ??? /usr/local/share/ffmpeg/libx264-fast_firstpass.ffpreset: Invalid
option or
> argument: 'directpred=1', parsed as 'directpred' = '1'

This means ffmpeg does not recognise the 'directpred' option specified in
the preset file libx264-fast_firstpass.ffpreset (in turn specified by the
command line option '-vpre fast_firstpass'). This was probably caused by the
ffmpeg syntax changing since the preset file was changed.

According to http://www.ffmpeg.org/ffmpeg-all.html the current option name
seems to be 'direct-pred'. Alternatively, libx264 supports the
'-fastfirstpass' option, which you can use instead of '-vpre
fast_firstpass'.

> 
> 
> ----------------------------------------
> I saw this in another post on how to do 2-pass encoding:
> 
> ??? /usr/local/bin/ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -c:v libx264
-
> preset medium -b:v 10m -pass 1 -an -f mp4 /dev/null
> ??? /usr/local/bin/ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -c:v libx264
-
> preset medium -b:v 10m -pass 2 -an OUT.mp4
> 
> There was no error on the first command.? The error on the second was
> ??? Error while opening encoder for output stream #0:0 - maybe incorrect
> parameters such as bit_rate, rate, width or height

There probably is something wrong with the format you're passing to the
encoder. Normally I get this error when trying to pass 5.1-channel audio to
the libvo-aacenc encoder which can't handle it, but in this case you've
disabled audio with '-an' (do you really want that?). The problem therefore
is with video, and it only surfaced on the 2nd pass because the 1st pass
doesn't encode video. You'd have to inspect ffmpeg's console output for
video information it reports to try work out what video parameter may have
caused the error.

Or post the full command and uncut console output on the list.

> 
> ----------------------------------------
> 
> So, I want to use ffmpeg instead of mencoder for doing 2-pass mpeg4
> encoding.
> What's wrong with the above commands, and what's the correct way to do
> this?

All that there is to it is using '-pass 1' for the first command line and
'-pass 2' for the second. You're doing things right but just getting tripped
up by detail at this point.

> 
> Thank You
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From francois.visagie at gmail.com  Tue Aug  6 09:01:34 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Tue, 6 Aug 2013 09:01:34 +0200
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
	audio to the specific stream
In-Reply-To: <CAFd2cZWjehypM1KsMMKO0xKWCscLZn-PjxGsRipV+Xv0iAMH+w@mail.gmail.com>
References: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>	<003e01ce926f$c2758570$47609050$@gmail.com>
	<CAFd2cZWjehypM1KsMMKO0xKWCscLZn-PjxGsRipV+Xv0iAMH+w@mail.gmail.com>
Message-ID: <004001ce9272$ccd6f690$6684e3b0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Soho Soho123
> Sent: 06 August 2013 08:44
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] [HELP] ffmpeg how to arrange video , audio to
> the specific stream
> 
> Hi ,
> 
> if I want to reverse the stream order ,
> where I should put the "-map 0:1 -map 0:0"?
> 
> for exampe :
> 
> ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i
> /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map
0:1
> -map 0:0 http://localhost:8090/feed1.ffm
> 
> is the order correct if I put the "map" option ?
> 
> 
> 2013/8/6 Francois Visagie <francois.visagie at gmail.com>
> 
> > > -----Original Message-----
> > > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> > > bounces at ffmpeg.org] On Behalf Of Soho Soho123
> > > Sent: 06 August 2013 08:16
> > > To: FFmpeg user questions and RTFMs
> > > Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video , audio to
> > > the specific stream
> > >
> > > hi All,
> > >
> > >
> > > the question is :
> > > how to set video to the specific stream how to set audio to the
> > > specific stream if I use ffmpeg to capture both video and audio
> > > simultaneously?
> > >
> > > for example :
> > > ffmpeg command:
> > > ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480
> > > -input_format
> > h264
> > -i
> > > /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec
> > > copy http://localhost:8090/feed1.ffm
> > >
> > > I get the result :
> > > Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
> > >   Metadata:
> > >     creation_time   : now
> > >     encoder         : Lavf55.12.100
> > >     Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo,
> > > 1536
> > kb/s
> > >     Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30,
> > q=2-31,
> > > -5 kb/s, 30 fps, 1000k tbn, 30 tbc
> > > Stream mapping:
> > >   Stream #1:0 -> #0:0 (copy)
> > >   Stream #0:0 -> #0:1 (copy)

When this changes to the order you want, you've done it right :-). The
streams are identified under "Metadata" above.

> > >
> > > how to modify the stream layout when capture video + audio?
> >
> > Do you mean you want to reverse the stream order in this instance? In
> > that case, use the -map option to enumerate streams in the order you
> > want them to appear in the output:
> >
> > -map 0:1 -map 0:0
> >
> > will put the 2nd stream (video in this case) first.
> >
> > > _______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From soho123.2012 at gmail.com  Tue Aug  6 09:09:34 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 15:09:34 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
 audio to the specific stream
In-Reply-To: <004001ce9272$ccd6f690$6684e3b0$@gmail.com>
References: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>
	<003e01ce926f$c2758570$47609050$@gmail.com>
	<CAFd2cZWjehypM1KsMMKO0xKWCscLZn-PjxGsRipV+Xv0iAMH+w@mail.gmail.com>
	<004001ce9272$ccd6f690$6684e3b0$@gmail.com>
Message-ID: <CAFd2cZXMWryR42eO9ja+7JyEDkjU5S730tNPjm+8UVEW=B5m2Q@mail.gmail.com>

Hi ,

I still get the error when I use "map" option.

# ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format
h264 -i
/dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map
0:0 -
map 0:1 http://localhost:8090/feed1.ffm
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  6 2013 13:22:29 with gcc 4.4.5-1.5.5p4 (GCC)
  configuration: --enable-cross-compile --cross-prefix=sdk-linux-
--arch=mips --target-os=linux --disable-doc --disable-htmlpages
--disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2
--disable-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --disable-mipsdspr2
--disable-mipsfpu --enable-small --prefix=/ffmpeg-2.0/romfs
--bindir=/ffmpeg-2.0/romfs --disable-bsfs --disable-filters --enable-gpl
--enable-libfaac --enable-nonfree --enable-libx264
--extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,anull,copy,format,fps,framestep,resample'
libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 79.101 /  3. 79.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging
level) with argument 'debug'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'video4linux2'.
Reading option '-r' ... matched as option 'r' (set frame rate (Hz value,
fraction or abbreviation)) with argument '30'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '640x480'.
Reading option '-input_format' ... matched as AVOption 'input_format' with
argument 'h264'.
Reading option '-i' ... matched as input file with argument '/dev/video1'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'alsa'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz)) with argument '48000'.
Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.
Reading option '-i' ... matched as input file with argument 'hw:0'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '0:0'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '0:1'.
Reading option 'http://localhost:8090/feed1.ffm' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file /dev/video1.
Applying option f (force format) with argument video4linux2.
Applying option r (set frame rate (Hz value, fraction or abbreviation))
with argument 30.
Applying option s (set frame size (WxH or abbreviation)) with argument
640x480.
Successfully parsed a group of options.
Opening an input file: /dev/video1.
[video4linux2,v4l2 @ 0x71ed40] fd:3 capabilities:4000001
[video4linux2,v4l2 @ 0x71ed40] Current input_channel: 0, input_name: Camera
1
[video4linux2,v4l2 @ 0x71ed40] Setting time per frame to 1/30
[video4linux2,v4l2 @ 0x71ed40] All info found
Input #0, video4linux2,v4l2, from '/dev/video1':
  Duration: N/A, start: 1193.780000, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 640x480, 1/2000000, -5
kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc
Successfully opened the file.
Parsing a group of options: input file hw:0.
Applying option f (force format) with argument alsa.
Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
Applying option ac (set number of audio channels) with argument 2.
Successfully parsed a group of options.
Opening an input file: hw:0.
[alsa @ 0x79ca50] All info found
Guessed Channel Layout for  Input Stream #1.0 : stereo
Input #1, alsa, from 'hw:0':
  Duration: N/A, start: 1375801320.208875, bitrate: N/A
    Stream #1:0, 1, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file http://localhost:8090/feed1.ffm.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Applying option acodec (force audio codec ('copy' to copy stream)) with
argument copy.
Applying option map (set input stream mapping) with argument 0:0.
Applying option map (set input stream mapping) with argument 0:1.
Stream map '0:1' matches no streams.



2013/8/6 Francois Visagie <francois.visagie at gmail.com>

> > -----Original Message-----
> > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> > bounces at ffmpeg.org] On Behalf Of Soho Soho123
> > Sent: 06 August 2013 08:44
> > To: FFmpeg user questions
> > Subject: Re: [FFmpeg-user] [HELP] ffmpeg how to arrange video , audio to
> > the specific stream
> >
> > Hi ,
> >
> > if I want to reverse the stream order ,
> > where I should put the "-map 0:1 -map 0:0"?
> >
> > for exampe :
> >
> > ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format
> h264
> -i
> > /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy
> -map
> 0:1
> > -map 0:0 http://localhost:8090/feed1.ffm
> >
> > is the order correct if I put the "map" option ?
> >
> >
> > 2013/8/6 Francois Visagie <francois.visagie at gmail.com>
> >
> > > > -----Original Message-----
> > > > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> > > > bounces at ffmpeg.org] On Behalf Of Soho Soho123
> > > > Sent: 06 August 2013 08:16
> > > > To: FFmpeg user questions and RTFMs
> > > > Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video , audio to
> > > > the specific stream
> > > >
> > > > hi All,
> > > >
> > > >
> > > > the question is :
> > > > how to set video to the specific stream how to set audio to the
> > > > specific stream if I use ffmpeg to capture both video and audio
> > > > simultaneously?
> > > >
> > > > for example :
> > > > ffmpeg command:
> > > > ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480
> > > > -input_format
> > > h264
> > > -i
> > > > /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec
> > > > copy http://localhost:8090/feed1.ffm
> > > >
> > > > I get the result :
> > > > Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
> > > >   Metadata:
> > > >     creation_time   : now
> > > >     encoder         : Lavf55.12.100
> > > >     Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo,
> > > > 1536
> > > kb/s
> > > >     Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30,
> > > q=2-31,
> > > > -5 kb/s, 30 fps, 1000k tbn, 30 tbc
> > > > Stream mapping:
> > > >   Stream #1:0 -> #0:0 (copy)
> > > >   Stream #0:0 -> #0:1 (copy)
>
> When this changes to the order you want, you've done it right :-). The
> streams are identified under "Metadata" above.
>
> > > >
> > > > how to modify the stream layout when capture video + audio?
> > >
> > > Do you mean you want to reverse the stream order in this instance? In
> > > that case, use the -map option to enumerate streams in the order you
> > > want them to appear in the output:
> > >
> > > -map 0:1 -map 0:0
> > >
> > > will put the 2nd stream (video in this case) first.
> > >
> > > > _______________________________________________
> > > > ffmpeg-user mailing list
> > > > ffmpeg-user at ffmpeg.org
> > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > > _______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Tue Aug  6 09:17:07 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 6 Aug 2013 07:17:07 +0000 (UTC)
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
	audio to the specific stream
References: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>
	<003e01ce926f$c2758570$47609050$@gmail.com>
	<CAFd2cZWjehypM1KsMMKO0xKWCscLZn-PjxGsRipV+Xv0iAMH+w@mail.gmail.com>
	<004001ce9272$ccd6f690$6684e3b0$@gmail.com>
	<CAFd2cZXMWryR42eO9ja+7JyEDkjU5S730tNPjm+8UVEW=B5m2Q@mail.gmail.com>
Message-ID: <loom.20130806T091541-728@post.gmane.org>

Soho Soho123 <soho123.2012 <at> gmail.com> writes:

> # ffmpeg -loglevel debug -f video4linux2 -r 30 
> -s 640x480 -input_format h264 -i /dev/video1 
> -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy 
> -acodec copy -map 0:0 -> map 0:1 
> http://localhost:8090/feed1.ffm
> ffmpeg version 2.0 

Please update to current git head.

[...]

> Applying option map (set input stream mapping) with argument 0:1.
> Stream map '0:1' matches no streams.

Use "-map 1:0" instead of "-map 0:1"

Please stop top-posting, Carl Eugen


From soho123.2012 at gmail.com  Tue Aug  6 09:52:55 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 15:52:55 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
 audio to the specific stream
In-Reply-To: <loom.20130806T091541-728@post.gmane.org>
References: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>
	<003e01ce926f$c2758570$47609050$@gmail.com>
	<CAFd2cZWjehypM1KsMMKO0xKWCscLZn-PjxGsRipV+Xv0iAMH+w@mail.gmail.com>
	<004001ce9272$ccd6f690$6684e3b0$@gmail.com>
	<CAFd2cZXMWryR42eO9ja+7JyEDkjU5S730tNPjm+8UVEW=B5m2Q@mail.gmail.com>
	<loom.20130806T091541-728@post.gmane.org>
Message-ID: <CAFd2cZX4g0yiSEmZRb_-b-NqKbJbjPJ_rS8OBT85teq=qGLuDA@mail.gmail.com>

2013/8/6 Carl Eugen Hoyos <cehoyos at ag.or.at>

> Soho Soho123 <soho123.2012 <at> gmail.com> writes:
>
> > # ffmpeg -loglevel debug -f video4linux2 -r 30
> > -s 640x480 -input_format h264 -i /dev/video1
> > -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy
> > -acodec copy -map 0:0 -> map 0:1
> > http://localhost:8090/feed1.ffm
> > ffmpeg version 2.0
>
> Please update to current git head.
>
>
Why you recommend git head ?
Is "map" not function in version 2.0 not?
the git head version  is cheanged frequently.
it is not convenient for me to do verification.




>
> > Applying option map (set input stream mapping) with argument 0:1.
> > Stream map '0:1' matches no streams.
>
> Use "-map 1:0" instead of "-map 0:1"
>

the result seems does not get any change about stream arrangement.
the log is :

ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i
/dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map
1:0 -
map 0:0 http://localhost:8090/feed1.ffm
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  6 2013 13:22:29 with gcc 4.4.5-1.5.5p4 (GCC)
  configuration: --enable-cross-compile --cross-prefix=sdk-linux-
--arch=mips --target-os=linux --disable-doc --disable-htmlpages
--disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2
--disable-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --disable-mipsdspr2
--disable-mipsfpu --enable-small --prefix=/ffmpeg-2.0/romfs
--bindir=/ffmpeg-2.0/romfs --disable-bsfs --disable-filters --enable-gpl
--enable-libfaac --enable-nonfree --enable-libx264
--extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,anull,copy,format,fps,framestep,resample'
libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 79.101 /  3. 79.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging
level) with argument 'debug'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'video4linux2'.
Reading option '-r' ... matched as option 'r' (set frame rate (Hz value,
fraction or abbreviation)) with argument '30'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '640x480'.
Reading option '-input_format' ... matched as AVOption 'input_format' with
argument 'h264'.
Reading option '-i' ... matched as input file with argument '/dev/video1'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'alsa'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz)) with argument '48000'.
Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.
Reading option '-i' ... matched as input file with argument 'hw:0'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '1:0'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '0:0'.
Reading option 'http://localhost:8090/feed1.ffm' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file /dev/video1.
Applying option f (force format) with argument video4linux2.
Applying option r (set frame rate (Hz value, fraction or abbreviation))
with argument 30.
Applying option s (set frame size (WxH or abbreviation)) with argument
640x480.
Successfully parsed a group of options.
Opening an input file: /dev/video1.
[video4linux2,v4l2 @ 0x71ed40] fd:3 capabilities:4000001
[video4linux2,v4l2 @ 0x71ed40] Current input_channel: 0, input_name: Camera
1
[video4linux2,v4l2 @ 0x71ed40] Setting time per frame to 1/30
[h264 @ 0x71f9a0] Frame num gap 8 6
[video4linux2,v4l2 @ 0x71ed40] All info found
Input #0, video4linux2,v4l2, from '/dev/video1':
  Duration: N/A, start: 40.820000, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 640x480, 1/2000000, -5
kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc
Successfully opened the file.
Parsing a group of options: input file hw:0.
Applying option f (force format) with argument alsa.
Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
Applying option ac (set number of audio channels) with argument 2.
Successfully parsed a group of options.
Opening an input file: hw:0.
[alsa @ 0x816250] All info found
Guessed Channel Layout for  Input Stream #1.0 : stereo
Input #1, alsa, from 'hw:0':
  Duration: N/A, start: 1375803007.838875, bitrate: N/A
    Stream #1:0, 1, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file http://localhost:8090/feed1.ffm.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Applying option acodec (force audio codec ('copy' to copy stream)) with
argument copy.
Applying option map (set input stream mapping) with argument 1:0.
Applying option map (set input stream mapping) with argument 0:0.
Successfully parsed a group of options.
Opening an output file: http://localhost:8090/feed1.ffm.
[ffm @ 0x7a43f0] Format ffm probed with size=2048 and score=101
[AVIOContext @ 0x7a4900] Statistics: 4096 bytes read, 0 seeks
Tue Aug  6 15:30:07 2013 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 200 4175
Successfully opened the file.
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
  Metadata:
    creation_time   : now
    encoder         : Lavf55.12.100
    Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
    Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30, q=2-31,
-5 kb/s, 30 fps, 1000k tbn, 30 tbc
Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help

================================================================
and
# ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format
h264 -i
/dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy -map
0:0 -
map 1:0 http://localhost:8090/feed1.ffm
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  6 2013 13:22:29 with gcc 4.4.5-1.5.5p4 (GCC)
  configuration: --enable-cross-compile --cross-prefix=sdk-linux-
--arch=mips --target-os=linux --disable-doc --disable-htmlpages
--disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2
--disable-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --disable-mipsdspr2
--disable-mipsfpu --enable-small --prefix=/ffmpeg-2.0/romfs
--bindir=/ffmpeg-2.0/romfs --disable-bsfs --disable-filters --enable-gpl
--enable-libfaac --enable-nonfree --enable-libx264
--extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,anull,copy,format,fps,framestep,resample'
libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 79.101 /  3. 79.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging
level) with argument 'debug'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'video4linux2'.
Reading option '-r' ... matched as option 'r' (set frame rate (Hz value,
fraction or abbreviation)) with argument '30'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '640x480'.
Reading option '-input_format' ... matched as AVOption 'input_format' with
argument 'h264'.
Reading option '-i' ... matched as input file with argument '/dev/video1'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'alsa'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz)) with argument '48000'.
Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.
Reading option '-i' ... matched as input file with argument 'hw:0'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '0:0'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '1:0'.
Reading option 'http://localhost:8090/feed1.ffm' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file /dev/video1.
Applying option f (force format) with argument video4linux2.
Applying option r (set frame rate (Hz value, fraction or abbreviation))
with argument 30.
Applying option s (set frame size (WxH or abbreviation)) with argument
640x480.
Successfully parsed a group of options.
Opening an input file: /dev/video1.
[video4linux2,v4l2 @ 0x71ed40] fd:3 capabilities:4000001
[video4linux2,v4l2 @ 0x71ed40] Current input_channel: 0, input_name: Camera
1
[video4linux2,v4l2 @ 0x71ed40] Setting time per frame to 1/30
[video4linux2,v4l2 @ 0x71ed40] All info found
Input #0, video4linux2,v4l2, from '/dev/video1':
  Duration: N/A, start: 497.140000, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 640x480, 1/2000000, -5
kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc
Successfully opened the file.
Parsing a group of options: input file hw:0.
Applying option f (force format) with argument alsa.
Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
Applying option ac (set number of audio channels) with argument 2.
Successfully parsed a group of options.
Opening an input file: hw:0.
[alsa @ 0x7fe4d0] audio_device=hw:0
[alsa @ 0x7fe4d0] codec_id ==NONE
[alsa @ 0x7fe4d0] Do RECORD,1
[alsa @ 0x7fe4d0] All info found
Guessed Channel Layout for  Input Stream #1.0 : stereo
Input #1, alsa, from 'hw:0':
  Duration: N/A, start: 1375803463.588875, bitrate: N/A
    Stream #1:0, 1, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file http://localhost:8090/feed1.ffm.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Applying option acodec (force audio codec ('copy' to copy stream)) with
argument copy.
Applying option map (set input stream mapping) with argument 0:0.
Applying option map (set input stream mapping) with argument 1:0.
Successfully parsed a group of options.
Opening an output file: http://localhost:8090/feed1.ffm.
[ffm @ 0x808ea0] Format ffm probed with size=2048 and score=101
[AVIOContext @ 0x809400] Statistics: 4096 bytes read, 0 seeks
Tue Aug  6 15:37:43 2013 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 200 4175
Successfully opened the file.
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
  Metadata:
    creation_time   : now
    encoder         : Lavf55.12.100
    Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
    Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30, q=2-31,
-5 kb/s, 30 fps, 1000k tbn, 30 tbc
Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help

From soho123.2012 at gmail.com  Tue Aug  6 10:15:55 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 16:15:55 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg how to arrange video ,
 audio to the specific stream
In-Reply-To: <loom.20130806T091541-728@post.gmane.org>
References: <CAFd2cZXPkuChWVgYBDjPtsH5ciFefg7qmTUn=aSz92JUbt-SnQ@mail.gmail.com>
	<003e01ce926f$c2758570$47609050$@gmail.com>
	<CAFd2cZWjehypM1KsMMKO0xKWCscLZn-PjxGsRipV+Xv0iAMH+w@mail.gmail.com>
	<004001ce9272$ccd6f690$6684e3b0$@gmail.com>
	<CAFd2cZXMWryR42eO9ja+7JyEDkjU5S730tNPjm+8UVEW=B5m2Q@mail.gmail.com>
	<loom.20130806T091541-728@post.gmane.org>
Message-ID: <CAFd2cZVqgN46wTmpKCnP8F_a2HEjWtwYapgNeE2GocnuQ=xuoA@mail.gmail.com>

> > Applying option map (set input stream mapping) with argument 0:1.
> > Stream map '0:1' matches no streams.
>
> Use "-map 1:0" instead of "-map 0:1"
> I use the map option when I start ffmepg,
>
I got the same stream arrangement.
the command :
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -map 0:0 -map 1:0 -vcodec
copy -acodec copy http://localhost:8090/feed1.ffm

OR

ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -map 1:0 -map 0:0 -vcodec
copy -acodec copy http://localhost:8090/feed1.ffm

All get the stream mapping:

Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)


Do you know where is wrong?

From onemda at gmail.com  Tue Aug  6 11:56:55 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 6 Aug 2013 09:56:55 +0000
Subject: [FFmpeg-user] OpenCL? What Effect Does It Have
In-Reply-To: <loom.20130805T194218-441@post.gmane.org>
References: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<loom.20130805T194218-441@post.gmane.org>
Message-ID: <CAPYw7P7iO=EOQi-Xy+1MkosmbNp7R0kEjSqCs4tSTf2DsMLAWg@mail.gmail.com>

On 8/5/13, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> James Board <jpboard2 <at> yahoo.com> writes:
>
>> If I were to install a more powerful Nvidia graphics
>> card on my machine, and then configure and build
>> ffmpeg with OpenCL enabled, what effect would it have?
>
> In theory, you could use opencl for the deshake and
> the unsharp filters but I don't know if sufficiently
> new OpenCL drivers exist.
>
>> * Inverse telecine, or deinterlace filters
>
> Unrelated to your question:
> I suspect that deinterlacing (yadif) is faster with
> FFmpeg than MEncoder.

Yadif from libavfilter can also use multiple threads.

>
>> * Convert to a lossless format: ffvhuff, ljpeg, etc
>> (I currently do this with mencoder, but would switch
>> to ffmpeg if it's faster).

I suppose those could be made multithread too.

>
> Given that MEncoder uses the exact same source code
> to encode ffvhuff and ljpeg (and most probably "etc")
> how do you expect that FFmpeg could be faster?
> Allow me to repeat that since MEncoder is not
> actively maintained, it makes no sense to use it if
> you don't have very special needs (like binary
> codecs but you should be able to use them with
> mplayer -vo yuv4mpeg and FFmpeg).
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Tue Aug  6 12:40:20 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 6 Aug 2013 10:40:20 +0000 (UTC)
Subject: [FFmpeg-user] Writing an MP3 file fails only when piping output
References: <CAHjQg2ogcM6ATWQwuKoR7uoRc3c5ZXZKeVbstMxyPGZ63HFhkg@mail.gmail.com>
Message-ID: <loom.20130806T123938-12@post.gmane.org>

Parsha Pourkhomami <parshap <at> gmail.com> writes:

> ffmpeg -i test.mp3 -codec copy -f mp3 pipe:1 > piped.mp3

Complete, uncut console output missing.

Is the problem only reproducible with -acodec copy or 
also if you reencode?

Carl Eugen


From cehoyos at ag.or.at  Tue Aug  6 12:42:17 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 6 Aug 2013 10:42:17 +0000 (UTC)
Subject: [FFmpeg-user] FFmpeg Question
References: <CAAdiqJWbTSEnDx2M6wzxbBAnJayYSypRMt1yByzqTSybbMy1JQ@mail.gmail.com>
	<loom.20130805T193951-216@post.gmane.org>
Message-ID: <loom.20130806T124151-290@post.gmane.org>

Carl Eugen Hoyos <cehoyos <at> ag.or.at> writes:

> you will have to add silence to out.mp4 which 
> should be trivial:
> $ ffmpeg -loop 1 -i img.jpg -f pcm_s16le -i /dev/zero

This should have been:
$ ffmpeg -loop 1 -i img.jpg -f s16le -i /dev/zero

Sorry, Carl Eugen


From cehoyos at ag.or.at  Tue Aug  6 12:48:05 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 6 Aug 2013 10:48:05 +0000 (UTC)
Subject: [FFmpeg-user] 2nd Attempt: error enabling libx264
References: <51F69EA4.8080609@gmx.com>
Message-ID: <loom.20130806T124701-875@post.gmane.org>

Sean Shaffer <d4rkw0lf <at> gmx.com> writes:

> libx264 (unknown, sourced from git 2013-07-26)
> 
> configure fails; indicates libx264 is not found or 
> wrong version.

It appears that you have several x264.h files on your 
system, try using --extra-cflags=-I/usr/local/include 
to choose the right one.

Carl Eugen


From cehoyos at ag.or.at  Tue Aug  6 12:53:41 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 6 Aug 2013 10:53:41 +0000 (UTC)
Subject: [FFmpeg-user] Cross-compiling FFMPEG for Blackberry 10 platform
References: <BLU0-SMTP238F7DCAF4AEB3C6A89F17A93550@phx.gbl>
Message-ID: <loom.20130806T124907-693@post.gmane.org>

emanuele <e.sansone <at> hotmail.it> writes:

> I'm trying to cross-compiling FFMPEG for Blackberry 10 platform. 
> Currently, I'm having problems in configuring the build of the 
> source code.
> 
> As suggested, I attach my 'config.log' file.

An alternative would have been to paste the last lines:

BEGIN /var/folders/0h/jp3pch4d5bz1lhn1dyrz5qkh0000gn/T//ffconf.JYkI0xun.c
    1   int main(void){ return 0; }
END /var/folders/0h/jp3pch4d5bz1lhn1dyrz5qkh0000gn/T//ffconf.JYkI0xun.c
.../usr/bin/arm-unknown-nto-qnx8.0.0eabi-gcc -mcpu= -c -o 
/var/folders/0h/jp3pch4d5bz1lhn1dyrz5qkh0000gn/T//ffconf.B3x1HBq1.o 
/var/folders/0h/jp3pch4d5bz1lhn1dyrz5qkh0000gn/T//ffconf.JYkI0xun.c 
arm-unknown-nto-qnx8.0.0eabi-gcc: fatal error: environment variables
QNX_HOST and QNX_TARGET not defined
compilation terminated.
C compiler test failed.

Hope this helps, Carl Eugen


From soho123.2012 at gmail.com  Tue Aug  6 13:03:42 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 19:03:42 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg "-map" option question
Message-ID: <CAFd2cZV920R7ifrKnBYOOoXZeJ99Bin_2zzjYzBwaEC9V50xyQ@mail.gmail.com>

Hi All,


I use the git head version to verification ttest.
When I add "-map" option to specify the stream id that I want.
But it seems does not work.
I can not modify stream index that I want.
does anyone have ideas?

the stream mapping is always:

Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)


I can not modify the order for output stream.



the log delow:

ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i
/dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -map 0:0 -map 1:0 -vcodec copy
-acod
ec copy http://localhost:8090/feed1.ffm
ffmpeg version N-55257-g84bc317 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug  6 2013 17:52:35 with gcc 4.4.5-1.5.5p4 (GCC)
  configuration: --enable-cross-compile --cross-prefix=sdk-linux-
--arch=mips --target-os=linux --disable-doc --disable-htmlpages
--disable-manpages --disable-podpages --disable-txtpages --disable-mips32r2
--disable-ffprobe --disable-ffplay --disable-postproc
--disable-runtime-cpudetect --disable-swscale-alpha --disable-mipsdspr1
--disable-mipsdspr2 --disable-mipsfpu --enable-small
--prefix=ffmpeg_head/romfs --bindir=/ffmpeg_head/romfs --disable-bsfs
--disable-filters --enable-gpl --enable-libfaac --enable-nonfree
--enable-libx264 --extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,anull,copy,format,fps,  libavutil
52. 40.100 / 52. 40.100
  libavcodec     55. 20.100 / 55. 20.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging
level) with argument 'debug'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'video4linux2'.
Reading option '-r' ... matched as option 'r' (set frame rate (Hz value,
fraction or abbreviation)) with argument '30'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '640x480'.
Reading option '-input_format' ... matched as AVOption 'input_format' with
argument 'h264'.
Reading option '-i' ... matched as input file with argument '/dev/video1'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'alsa'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz)) with argument '48000'.
Reading option '-ac' ... matched as option 'ac' (set number of audio
channels) with argument '2'.
Reading option '-i' ... matched as input file with argument 'hw:0'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '0:0'.
Reading option '-map' ... matched as option 'map' (set input stream
mapping) with argument '1:0'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'copy'.
Reading option 'http://localhost:8090/feed1.ffm' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file /dev/video1.
Applying option f (force format) with argument video4linux2.
Applying option r (set frame rate (Hz value, fraction or abbreviation))
with argument 30.
Applying option s (set frame size (WxH or abbreviation)) with argument
640x480.
Successfully parsed a group of options.
Opening an input file: /dev/video1.
[video4linux2,v4l2 @ 0x728e00] fd:3 capabilities:4000001
[video4linux2,v4l2 @ 0x728e00] Current input_channel: 0, input_name: Camera
1
[video4linux2,v4l2 @ 0x728e00] Setting time per frame to 1/30
[video4linux2,v4l2 @ 0x728e00] All info found
Input #0, video4linux2,v4l2, from '/dev/video1':
  Duration: N/A, start: 497.720000, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 640x480, 1/2000000, -5
kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc
Successfully opened the file.
Parsing a group of options: input file hw:0.
Applying option f (force format) with argument alsa.
Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
Applying option ac (set number of audio channels) with argument 2.
Successfully parsed a group of options.
Opening an input file: hw:0.
Guessed Channel Layout for  Input Stream #1.0 : stereo
Input #1, alsa, from 'hw:0':
  Duration: N/A, start: 1375814596.248875, bitrate: N/A
    Stream #1:0, 1, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file http://localhost:8090/feed1.ffm.
Applying option map (set input stream mapping) with argument 0:0.
Applying option map (set input stream mapping) with argument 1:0.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Applying option acodec (force audio codec ('copy' to copy stream)) with
argument copy.
Successfully parsed a group of options.
Opening an output file: http://localhost:8090/feed1.ffm.
[ffm @ 0x8001f0] Format ffm probed with size=2048 and score=101
[AVIOContext @ 0x7a6040] Statistics: 4096 bytes read, 0 seeks
Tue Aug  6 18:43:16 2013 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 200 4175
Successfully opened the file.
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
  Metadata:
    creation_time   : now
    encoder         : Lavf55.13.102
    Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
    Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30, q=2-31,
-5 kb/s, 30 fps, 1000k tbn, 30 tbc
Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help

From krueger at lesspain.de  Tue Aug  6 13:09:00 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Tue, 6 Aug 2013 13:09:00 +0200
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
Message-ID: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>

Hi,

it seems the ffmpeg always decodes the listed types of images to
determine their properties. What I did to check this, was to run
ffmpeg -i <imagefile> as well as ffprobe -show_streams <imagefile>
with each of these formats and set a breakpoint in
avcodec_decode_video2.

This is not a problem in many cases, however when using ffmpeg,
ffprobe or its libraries to extract metadata such as codec, size and
pixel format from thousands of image files (imagine analyzing a full
SD card of Hires JPG images), the difference between header parsing
and decoding becomes very significant (especially with large images
for obvious reasons).

Is this (extracting metadata for those formats just by header parsing)
currently implemented and I am just missing a command line argument?

If not, shall I submit this as an enhancement request in trac?

Best regards,

Robert

From cehoyos at ag.or.at  Tue Aug  6 13:23:33 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 6 Aug 2013 11:23:33 +0000 (UTC)
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
Message-ID: <loom.20130806T131859-871@post.gmane.org>

Robert Kr?ger <krueger <at> lesspain.de> writes:

> it seems the ffmpeg always decodes the listed 
> types of images to determine their properties.

In your email's subject, you used the term "probing" 
which is defined for FFmpeg / libavformat.
For all listed image formats, FFmpeg does no probing, 
it guesses the format reading the file suffix.
(Something FFmpeg does not do for nearly all formats 
it supports.)

> What I did to check this, was to run
> ffmpeg -i <imagefile> as well as 
> ffprobe -show_streams <imagefile> with each of 
> these formats and set a breakpoint in
> avcodec_decode_video2.

I don't think it is generally possible to find out 
properties of image files (that is apparently what 
you need) without calling this function.

Carl Eugen


From jpboard2 at yahoo.com  Tue Aug  6 14:40:25 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 6 Aug 2013 05:40:25 -0700 (PDT)
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching
	from	mencoder)
In-Reply-To: <003f01ce9272$25a32920$70e97b60$@gmail.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<003f01ce9272$25a32920$70e97b60$@gmail.com>
Message-ID: <1375792825.82501.YahooMailNeo@web164704.mail.gq1.yahoo.com>


>Normally I get this error when trying to pass 5.1-channel audio to

>the libvo-aacenc encoder which can't handle it, but in this case you've
>disabled audio with '-an' (do you really want that?). 

Actually, I'd rather pass the audio through in the same exact format.? I
can do this in mencoder with '-avc copy'.? What's the equivalent way
to pass audio through in ffmpeg?

From jpboard2 at yahoo.com  Tue Aug  6 14:42:49 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 6 Aug 2013 05:42:49 -0700 (PDT)
Subject: [FFmpeg-user] Compress Raw AVI file to ffvhuff,
	and pass audio through as is
Message-ID: <1375792969.47365.YahooMailNeo@web164703.mail.gq1.yahoo.com>

I've been using mencoder to convert an AVI file with a raw pixel code to
ffvhuff.? I used the 'avc copy' command-line option to pass the audio samples
through without encoding.

What is an equivalent command in ffmpeg to convert a raw AVI file to an AVI
file where the video is compressed with ffvhuff, and the audio is merely copied
to the output file?

Thank you.

From jpboard2 at yahoo.com  Tue Aug  6 14:56:02 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 6 Aug 2013 05:56:02 -0700 (PDT)
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and produce
	JPG images
Message-ID: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>

Is it possible, within ffmpeg, to do both of these at the same time:
??? Convert a raw AVI file to a lossless compressed ffvhuff file.
??? Convert each frame to an individual image (JPEG or TIFF, or whatever)?
Concurrently I'm doing both of the above in mencoder with two separate
commands.? But that is inefficient since the video frames have to be decoded
again for the image extraction step.? Also, I'm trying to convert from
mencoder to ffmpeg, as Carl has repeatedly reprimanded me for not doing
so, and I'm not that familiar with the ffmpeg command syntax.

Can I do this?

From jpboard2 at yahoo.com  Tue Aug  6 15:02:21 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 6 Aug 2013 06:02:21 -0700 (PDT)
Subject: [FFmpeg-user] OpenCL? What Effect Does It Have
In-Reply-To: <CAPYw7P7iO=EOQi-Xy+1MkosmbNp7R0kEjSqCs4tSTf2DsMLAWg@mail.gmail.com>
References: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<loom.20130805T194218-441@post.gmane.org>
	<CAPYw7P7iO=EOQi-Xy+1MkosmbNp7R0kEjSqCs4tSTf2DsMLAWg@mail.gmail.com>
Message-ID: <1375794141.79897.YahooMailNeo@web164706.mail.gq1.yahoo.com>



>Yadif from libavfilter can also use multiple threads.

Okay.? But are they accelerated with OpenCL?

>> * Convert to a lossless format: ffvhuff, ljpeg, etc
>> (I currently do this with mencoder, but would switch
>> to ffmpeg if it's faster).
>
>I suppose those could be made multithread too.

Multi-threaded is one thing.? OpenCL is another.? Are the
above converted to run with OpenCL?

> Given that MEncoder uses the exact same source code
> to encode ffvhuff and ljpeg (and most probably "etc")
> how do you expect that FFmpeg could be faster?

Maybe my original question, and reply to Carl wasn't clear enough.

OpenCL is an API for using GPUs to do computationally
intensive tasks.? See this URL
??? http://en.wikipedia.org/wiki/OpenCL
By coding expensiev tasks in OpenCL, especially image processing operations
which ffmpeg does all the time, you can achieve 10x to 100x performance
improvement.? I'd like to know what parts of ffmpeg have been
accelerated with OpenCL.? This could potentially improve the
performance and prestige of ffmpeg.? But it doesn't sound like much
work has been done in this area.? Is that true?

From francois.visagie at gmail.com  Tue Aug  6 15:10:54 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Tue, 6 Aug 2013 15:10:54 +0200
Subject: [FFmpeg-user] [HELP] ffmpeg "-map" option question
In-Reply-To: <CAFd2cZV920R7ifrKnBYOOoXZeJ99Bin_2zzjYzBwaEC9V50xyQ@mail.gmail.com>
References: <CAFd2cZV920R7ifrKnBYOOoXZeJ99Bin_2zzjYzBwaEC9V50xyQ@mail.gmail.com>
Message-ID: <005501ce92a6$644c46f0$2ce4d4d0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Soho Soho123
> Sent: 06 August 2013 13:04
> To: FFmpeg user questions and RTFMs
> Subject: [FFmpeg-user] [HELP] ffmpeg "-map" option question
> 
> Hi All,
> 
> 
> I use the git head version to verification ttest.
> When I add "-map" option to specify the stream id that I want.
> But it seems does not work.
> I can not modify stream index that I want.
> does anyone have ideas?

I haven't tried it, but as a work-around what about omitting the '-map'
options and specifying your input files in reverse order?

> 
> the stream mapping is always:
> 
> Stream mapping:
>   Stream #1:0 -> #0:0 (copy)
>   Stream #0:0 -> #0:1 (copy)
> 
> 
> I can not modify the order for output stream.
> 
> 
> 
> the log delow:
> 
> ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i
> /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -map 0:0 -map 1:0 -vcodec copy
-
> acod ec copy http://localhost:8090/feed1.ffm ffmpeg version N-55257-
> g84bc317 Copyright (c) 2000-2013 the FFmpeg developers
>   built on Aug  6 2013 17:52:35 with gcc 4.4.5-1.5.5p4 (GCC)
>   configuration: --enable-cross-compile --cross-prefix=sdk-linux-
--arch=mips
> --target-os=linux --disable-doc --disable-htmlpages --disable-manpages --
> disable-podpages --disable-txtpages --disable-mips32r2 --disable-ffprobe
--
> disable-ffplay --disable-postproc --disable-runtime-cpudetect --disable-
> swscale-alpha --disable-mipsdspr1
> --disable-mipsdspr2 --disable-mipsfpu --enable-small --
> prefix=ffmpeg_head/romfs --bindir=/ffmpeg_head/romfs --disable-bsfs --
> disable-filters --enable-gpl --enable-libfaac --enable-nonfree
> --enable-libx264 --extra-cflags=-I/ShareLib_Install/bin/../include
> --extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
> --enable-filter='aformat,aresample,anull,copy,format,fps,  libavutil 52.
40.100
> / 52. 40.100
>   libavcodec     55. 20.100 / 55. 20.100
>   libavformat    55. 13.102 / 55. 13.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 82.100 /  3. 82.100
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.103 /  0. 17.103
> Splitting the commandline.
> Reading option '-loglevel' ... matched as option 'loglevel' (set logging
> level) with argument 'debug'.
> Reading option '-f' ... matched as option 'f' (force format) with argument
> 'video4linux2'.
> Reading option '-r' ... matched as option 'r' (set frame rate (Hz value,
fraction
> or abbreviation)) with argument '30'.
> Reading option '-s' ... matched as option 's' (set frame size (WxH or
> abbreviation)) with argument '640x480'.
> Reading option '-input_format' ... matched as AVOption 'input_format' with
> argument 'h264'.
> Reading option '-i' ... matched as input file with argument '/dev/video1'.
> Reading option '-f' ... matched as option 'f' (force format) with argument
> 'alsa'.
> Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz))
> with argument '48000'.
> Reading option '-ac' ... matched as option 'ac' (set number of audio
> channels) with argument '2'.
> Reading option '-i' ... matched as input file with argument 'hw:0'.
> Reading option '-map' ... matched as option 'map' (set input stream
> mapping) with argument '0:0'.
> Reading option '-map' ... matched as option 'map' (set input stream
> mapping) with argument '1:0'.
> Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
> ('copy' to copy stream)) with argument 'copy'.
> Reading option '-acodec' ... matched as option 'acodec' (force audio codec
> ('copy' to copy stream)) with argument 'copy'.
> Reading option 'http://localhost:8090/feed1.ffm' ... matched as output
file.
> Finished splitting the commandline.
> Parsing a group of options: global .
> Applying option loglevel (set logging level) with argument debug.
> Successfully parsed a group of options.
> Parsing a group of options: input file /dev/video1.
> Applying option f (force format) with argument video4linux2.
> Applying option r (set frame rate (Hz value, fraction or abbreviation))
with
> argument 30.
> Applying option s (set frame size (WxH or abbreviation)) with argument
> 640x480.
> Successfully parsed a group of options.
> Opening an input file: /dev/video1.
> [video4linux2,v4l2 @ 0x728e00] fd:3 capabilities:4000001
> [video4linux2,v4l2 @ 0x728e00] Current input_channel: 0, input_name:
> Camera
> 1
> [video4linux2,v4l2 @ 0x728e00] Setting time per frame to 1/30
> [video4linux2,v4l2 @ 0x728e00] All info found Input #0, video4linux2,v4l2,
> from '/dev/video1':
>   Duration: N/A, start: 497.720000, bitrate: N/A
>     Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 640x480, 1/2000000,
-5
> kb/s, 30 fps, 30 tbr, 1000k tbn, 2000k tbc Successfully opened the file.
> Parsing a group of options: input file hw:0.
> Applying option f (force format) with argument alsa.
> Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
> Applying option ac (set number of audio channels) with argument 2.
> Successfully parsed a group of options.
> Opening an input file: hw:0.
> Guessed Channel Layout for  Input Stream #1.0 : stereo Input #1, alsa,
from
> 'hw:0':
>   Duration: N/A, start: 1375814596.248875, bitrate: N/A
>     Stream #1:0, 1, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536
kb/s
> Successfully opened the file.
> Parsing a group of options: output file http://localhost:8090/feed1.ffm.
> Applying option map (set input stream mapping) with argument 0:0.
> Applying option map (set input stream mapping) with argument 1:0.
> Applying option vcodec (force video codec ('copy' to copy stream)) with
> argument copy.
> Applying option acodec (force audio codec ('copy' to copy stream)) with
> argument copy.
> Successfully parsed a group of options.
> Opening an output file: http://localhost:8090/feed1.ffm.
> [ffm @ 0x8001f0] Format ffm probed with size=2048 and score=101
> [AVIOContext @ 0x7a6040] Statistics: 4096 bytes read, 0 seeks Tue Aug  6
> 18:43:16 2013 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 200 4175
Successfully
> opened the file.
> Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
>   Metadata:
>     creation_time   : now
>     encoder         : Lavf55.13.102
>     Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 48000 Hz, stereo, 1536
kb/s
>     Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 640x480, 1/30,
q=2-31,
> -5 kb/s, 30 fps, 1000k tbn, 30 tbc
> Stream mapping:
>   Stream #1:0 -> #0:0 (copy)
>   Stream #0:0 -> #0:1 (copy)
> Press [q] to stop, [?] for help
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From francois.visagie at gmail.com  Tue Aug  6 15:11:51 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Tue, 6 Aug 2013 15:11:51 +0200
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg
	(switching	from	mencoder)
In-Reply-To: <1375792825.82501.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<003f01ce9272$25a32920$70e97b60$@gmail.com>
	<1375792825.82501.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <005601ce92a6$8622b110$92681330$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of James Board
> Sent: 06 August 2013 14:40
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from
> mencoder)
> 
> 
> >Normally I get this error when trying to pass 5.1-channel audio to
> 
> >the libvo-aacenc encoder which can't handle it, but in this case you've
> >disabled audio with '-an' (do you really want that?).
> 
> Actually, I'd rather pass the audio through in the same exact format.? I
can do
> this in mencoder with '-avc copy'.? What's the equivalent way to pass
audio
> through in ffmpeg?

'-c[odec]:a copy'

> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From soho123.2012 at gmail.com  Tue Aug  6 15:13:31 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 21:13:31 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg post to local loopback stuck
Message-ID: <CAFd2cZWwRyS2AEMTm=-DZJLDXVBTMaL-Mp=eK3T3dEimyOHPwQ@mail.gmail.com>

Hi all,


when I setup a stream server with ffserver + ffmpeg,
the command :
ffmpeg -sn -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format
h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm

it can work for a few minutes only .
because ffserver can not get new video data  from ffmpeg local post.
ffmpeg get stuck when it send video to localhost.

Does anyone have idea about this issue?

From soho123.2012 at gmail.com  Tue Aug  6 15:15:39 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 6 Aug 2013 21:15:39 +0800
Subject: [FFmpeg-user] [HELP] ffmpeg "-map" option question
In-Reply-To: <005501ce92a6$644c46f0$2ce4d4d0$@gmail.com>
References: <CAFd2cZV920R7ifrKnBYOOoXZeJ99Bin_2zzjYzBwaEC9V50xyQ@mail.gmail.com>
	<005501ce92a6$644c46f0$2ce4d4d0$@gmail.com>
Message-ID: <CAFd2cZW1d6SHZ+b3f3UCjWr5fruiHK0N5k2EWB_BMaoa_-bz1A@mail.gmail.com>

> >
> > I use the git head version to verification ttest.
> > When I add "-map" option to specify the stream id that I want.
> > But it seems does not work.
> > I can not modify stream index that I want.
> > does anyone have ideas?
>
> I haven't tried it, but as a work-around what about omitting the '-map'
> options and specifying your input files in reverse order?
>
> Hi,
I have tried.
I got the same result!
The order can not be modified!
Do you have any idea?

From francois.visagie at gmail.com  Tue Aug  6 15:17:48 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Tue, 6 Aug 2013 15:17:48 +0200
Subject: [FFmpeg-user] [HELP] ffmpeg "-map" option question
In-Reply-To: <CAFd2cZW1d6SHZ+b3f3UCjWr5fruiHK0N5k2EWB_BMaoa_-bz1A@mail.gmail.com>
References: <CAFd2cZV920R7ifrKnBYOOoXZeJ99Bin_2zzjYzBwaEC9V50xyQ@mail.gmail.com>	<005501ce92a6$644c46f0$2ce4d4d0$@gmail.com>
	<CAFd2cZW1d6SHZ+b3f3UCjWr5fruiHK0N5k2EWB_BMaoa_-bz1A@mail.gmail.com>
Message-ID: <005701ce92a7$5bd48090$137d81b0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Soho Soho123
> Sent: 06 August 2013 15:16
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] [HELP] ffmpeg "-map" option question
> 
> > >
> > > I use the git head version to verification ttest.
> > > When I add "-map" option to specify the stream id that I want.
> > > But it seems does not work.
> > > I can not modify stream index that I want.
> > > does anyone have ideas?
> >
> > I haven't tried it, but as a work-around what about omitting the '-map'
> > options and specifying your input files in reverse order?
> >
> > Hi,
> I have tried.
> I got the same result!
> The order can not be modified!
> Do you have any idea?

Not me :-(.

> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From feanor2112 at covad.net  Tue Aug  6 15:22:04 2013
From: feanor2112 at covad.net (Ned Haskin)
Date: Tue, 06 Aug 2013 09:22:04 -0400
Subject: [FFmpeg-user] ffmpeg Unknown encoder 'libx264' error
In-Reply-To: <loom.20130805T194808-897@post.gmane.org>
References: <1375666070.5067.41.camel@videoathlon>
	<loom.20130805T194808-897@post.gmane.org>
Message-ID: <1375795324.7275.25.camel@videoathlon>

On Mon, 2013-08-05 at 17:57 +0000, Carl Eugen Hoyos wrote:
> Ned Haskin <feanor2112 <at> covad.net> writes:
> 
> > $ ./configure --enable-libdc1394 --prefix=/usr 
> 
> Mostly unrelated to your original question:
> 
> > --extra-cflags='-Wall -g '
> 
> Please remove this.
> 
> > --enable-gpl
> 
> > --enable-shared 
> 
> (This is of course 100% ok, but if you are unexperienced 
> I would strongly recommend not to use it, it generally 
> has theoretical problems and the gain for you as a 
> user is not completely clear to me, in combination with 
> "--prefix=/usr", I personally consider it highly 
> dangerous and would not use it on my system.)
> 
> > --enable-nonfree --enable-libx264 
> 
> > --enable-encoder=libx264
> 
> This makes no sense in the configure line you posted. 
I was trying to get the encoder to compile, not sure why it wouldn't. I did compile ffmpeg many times before posting. 
> 
> > --enable-libmp3lame --enable-libvorbis 
> 
> > --enable-libtwolame --enable-libv4l2
I use them both on other projects. Although I see now that a mp2 encoder is in ffmpeg. 
> 
> (I would even recommend to remove these two.)
> 
> > --enable-pthreads
> 
> This is the default, please remove it.
> 
> > --enable-libfaac --enable-libxvid
> 
> > --enable-postproc
> 
> This also makes no sense. 
It is in the debian ffmpeg configure line. 
> 
Got ffmpeg to compile and libx264 to work with yesterdays git. I noticed that --arch=amd64 didn't change ARCH. I also use 
ffmpeg to transcode input files for cinelerra-cv & mjpeg is quite slow. Or is that just mjpeg code? 

ned337 at vid:~/ffmpeg$ ./configure --arch=amd64 --enable-opencl --enable-gpl --enable-nonfree --enable-libx264 
--enable-libxvid --enable-libmp3lame --enable-libfaac
install prefix            /usr/local
source path               .
C compiler                gcc
ARCH                      x86 (generic)
big-endian                no
runtime cpu detection     yes
yasm                      yes
MMX enabled               yes
MMXEXT enabled            yes
3DNow! enabled            yes
3DNow! extended enabled   yes
SSE enabled               yes
SSSE3 enabled             yes
AVX enabled               yes
FMA4 enabled              yes
i686 features enabled     yes
CMOV is fast              yes
EBX available             yes
EBP available             yes
debug symbols             yes
strip symbols             yes
optimize for size         no
optimizations             yes
static                    yes
shared                    no
postprocessing support    yes
new filter support        yes
network support           yes
threading support         pthreads
safe bitstream reader     yes
SDL support               yes
opencl enabled            yes
texi2html enabled         yes
perl enabled              yes
pod2man enabled           yes
makeinfo enabled          yes

External libraries:
iconv			libx264			opencl
libfaac			libxvid			zlib
libmp3lame

skip stuff: no errors 

License: nonfree and unredistributable
Creating config.mak, config.h, and doc/config.texi...
libavutil/avconfig.h is unchanged

Thanks, 
Ned 


From rodney.baker at iinet.net.au  Tue Aug  6 15:22:46 2013
From: rodney.baker at iinet.net.au (Rodney Baker)
Date: Tue, 06 Aug 2013 22:52:46 +0930
Subject: [FFmpeg-user] ffmpeg build fail with libx264
In-Reply-To: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <5021193.W6HfUystka@mako>

On Mon, 5 Aug 2013 16:56:18 James Board wrote:
> I'm trying to switch from mencoder to ffmpeg as Carl suggested.  In
> order to do this, I have to rebuild ffmpeg with libx264.  After my first
> attempt failed, I got the latest version of x264, built it, and installed
> 
>     git clone git:://git.videolan.org/x264.git
>     make
>     make install

You forgot ./configure --enable-shared before make.

-- 
==============================================================
Rodney Baker VK5ZTV
rodney.baker at iinet.net.au
==============================================================

From tevans.uk at googlemail.com  Tue Aug  6 15:37:25 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Tue, 6 Aug 2013 14:37:25 +0100
Subject: [FFmpeg-user] OpenCL? What Effect Does It Have
In-Reply-To: <1375794141.79897.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<loom.20130805T194218-441@post.gmane.org>
	<CAPYw7P7iO=EOQi-Xy+1MkosmbNp7R0kEjSqCs4tSTf2DsMLAWg@mail.gmail.com>
	<1375794141.79897.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1KxPyHJcsCBLAGXSnXRpsUvKECir8n-vuAdEw1sH2ssdw@mail.gmail.com>

On Tue, Aug 6, 2013 at 2:02 PM, James Board <jpboard2 at yahoo.com> wrote:
> OpenCL is an API for using GPUs to do computationally
> intensive tasks.  See this URL
>     http://en.wikipedia.org/wiki/OpenCL
> By coding expensiev tasks in OpenCL, especially image processing operations
> which ffmpeg does all the time, you can achieve 10x to 100x performance
> improvement.  I'd like to know what parts of ffmpeg have been
> accelerated with OpenCL.  This could potentially improve the
> performance and prestige of ffmpeg.  But it doesn't sound like much
> work has been done in this area.  Is that true?

OpenCL can be used to accelerate the unsharp and deshake filters, as
documented in the filters documentation.

https://www.ffmpeg.org/ffmpeg-filters.html

Sometimes it gives worse results than using the CPU:

http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=10&t=1284

Generally the consensus (at least on here, as far as I can tell) is
that GPU assisted encoding is slow and gives crap results. YMMV.

Cheers

Tom

From tevans.uk at googlemail.com  Tue Aug  6 15:48:18 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Tue, 6 Aug 2013 14:48:18 +0100
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
 produce JPG images
In-Reply-To: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>

On Tue, Aug 6, 2013 at 1:56 PM, James Board <jpboard2 at yahoo.com> wrote:
> Is it possible, within ffmpeg, to do both of these at the same time:
>     Convert a raw AVI file to a lossless compressed ffvhuff file.
>     Convert each frame to an individual image (JPEG or TIFF, or whatever)?
> Concurrently I'm doing both of the above in mencoder with two separate
> commands.  But that is inefficient since the video frames have to be decoded
> again for the image extraction step.  Also, I'm trying to convert from
> mencoder to ffmpeg, as Carl has repeatedly reprimanded me for not doing
> so, and I'm not that familiar with the ffmpeg command syntax.
>
> Can I do this?

Yes.

ffmpeg -i in.avi -c:v ffvhuff -c:a copy out.nut -f image2 out%d.png

You can even do super-cool things like only take one frame per second:

ffmpeg -i in.avi -c:v ffvhuff -c:a copy out.nut -f image2 -vf
fps=fps=1 out%d.png

There are also extensive docs one can use to get familiar with the
ffmpeg command syntax and/or discern what ffmpeg can do.

Cheers

Tom

From krueger at lesspain.de  Tue Aug  6 15:52:16 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Tue, 6 Aug 2013 15:52:16 +0200
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
In-Reply-To: <loom.20130806T131859-871@post.gmane.org>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
	<loom.20130806T131859-871@post.gmane.org>
Message-ID: <CAEnqZEWX=b0qeJ3Gh_QRFKh-2cYVWYYFQ-3p_WwwjS=-w_1zsA@mail.gmail.com>

On Tue, Aug 6, 2013 at 1:23 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> it seems the ffmpeg always decodes the listed
>> types of images to determine their properties.
>
> In your email's subject, you used the term "probing"
> which is defined for FFmpeg / libavformat.
> For all listed image formats, FFmpeg does no probing,
> it guesses the format reading the file suffix.
> (Something FFmpeg does not do for nearly all formats
> it supports.)
>
>> What I did to check this, was to run
>> ffmpeg -i <imagefile> as well as
>> ffprobe -show_streams <imagefile> with each of
>> these formats and set a breakpoint in
>> avcodec_decode_video2.
>
> I don't think it is generally possible to find out
> properties of image files (that is apparently what
> you need) without calling this function.

apparently my writing is too imprecise. So let me try another time to
avoid misunderstandings.

So are you saying that for the given file formats it is technically
not possible to find out resolution and pixel format without decoding
an entire frame or that it is currently not possible with
ffmpeg/libavformat/libavcodec?

Or formulated in another way: I am looking for a way to extract this
type of metadata from these file types with a lot fewer system
resources than are currently used when calling
avformat_find_stream_info, the way it is done when using the command
line tools. Is there a way to do this with the current code or is this
generally not possible?

Is it clear now what my goal is and what the actual question is? If
there is a solution that requires using the libraries in a certain
way, this thread should probably be moved to libav-user.

From mythtv at american-hero.com  Tue Aug  6 16:35:09 2013
From: mythtv at american-hero.com (Larry Roberts)
Date: Tue, 06 Aug 2013 10:35:09 -0400
Subject: [FFmpeg-user] getting latest version of ffmpeg to work with Roku
In-Reply-To: <CAOeAR15Q8tKdo-y5JPPoUon47qNmO9+31uz-Ka4J3FH0XTpSog@mail.gmail.com>
References: <51FC5683.9090405@american-hero.com>
	<51FCC130.3050001@thelounge.net>
	<51FD7773.6040609@american-hero.com>
	<51FD89EC.5000101@american-hero.com>
	<CAOeAR15Q8tKdo-y5JPPoUon47qNmO9+31uz-Ka4J3FH0XTpSog@mail.gmail.com>
Message-ID: <5201099D.1040900@american-hero.com>

On 8/5/2013 2:54 AM, Andrew Sinclair wrote:
> Not definitive but constrained baseline is an unusual profile for a HD
> encode and stands out in your "DOESN'T WORK" option, should try main or high
>
> If you want HD you should also drop the -s 640x360 bit
>
>>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Thanks.  So I have tried multiple different baseline, tried scaling 
different sizes, etc.  Tried even grabbing the latest and greatest 
ffmpeg version (and x264) and the problem still existed.

In the end I found a working solution, albiet not with ffmpeg.  I 
downloaded HandBrakeCLI and built it.  It encodes the videos just fine 
and they play on all systems.  It appears that HB includes its own build 
of h264 so I dont know if the problem is an ffmpeg problem or an h264 
problem.  I dont have need the options with HB, but right now its the 
only working solution I could come up with given my limited knowledge of 
video encoding.



From jpboard2 at yahoo.com  Tue Aug  6 18:09:14 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 6 Aug 2013 09:09:14 -0700 (PDT)
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
	produce JPG images
In-Reply-To: <CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>
Message-ID: <1375805354.99573.YahooMailNeo@web164705.mail.gq1.yahoo.com>

>ffmpeg -i in.avi -c:v ffvhuff -c:a copy out.nut -f image2 out%d.png


Thanks.? I will try that.

>There are also extensive docs one can use to get familiar with the
>ffmpeg command syntax and/or discern what ffmpeg can do.

Yes, I spent 2-3 hours reading the online documentation.? The problem
is that every example they gave didn't work for me.? And I could not
get ffmpeg to do what I wanted.? So, maybe it's easier to simply ask
people that already know.

From cmharwoo at umich.edu  Tue Aug  6 19:15:46 2013
From: cmharwoo at umich.edu (cmharwood89)
Date: Tue, 6 Aug 2013 10:15:46 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375740272587-4660485.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
	<1375740272587-4660485.post@n4.nabble.com>
Message-ID: <1375809346408-4660526.post@n4.nabble.com>

@ James,

Thanks for the referral. I checked out the link and I'll probably use the
filter eventually. For the time being, I'm making-do with time-indexed
trimming.

@thljcl,

Wow, thanks for putting together those slides! That looks like a very
helpful solution. I see the value of changing to 1fps, because I could
essentially use time as a dummy variable for frame number, but I don't
understand how that would allow me to trim at non keyframes. 

>From my (limited) understanding, there simply is no way to trim at a P-frame
without decompressing into a raw format first, trimming at the shiny new
keyframe, and then recompressing back into my desired. Right now, I'm having
pretty good results doing it this way. If I change the framerate to 1fps,
it's not going to redistribute keyframes, right? If I choose a cutting
point, say at frame 300, but the nearest I-frame is at 295, then my output
video will be trimmed at 5 frames/5 seconds from the desired cut-point. 

For reference, my input and desired output are:
INPUT:
h264 codec @59.94 fps, 1920x1080 res; AAC audio codec; MP4 container

OUTPUT
h264 codec @59.94 fps, 1920x1080 res; NO AUDIO; AVI container

I want the output video to be encoded with no subjectively observable
losses.  Computing time and disk space are not a concern, but video quality
and accurate trimming are absolutely necessary.

My original problem has been solved (by removing the square brackets
surrounding my millisecond count in the -ss argument), but I'm definitely
interested in making my approach smarter- i.e. avoiding a brute-force
complete encoding of the video. I like the speed and losslessness of
container-only manipulation and cutting at keyframes, and I like the
accuracy of a decompress-then-cut approach. My new challenge is finding a
way to combine the two.




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From jiachielee at live.com  Tue Aug  6 19:38:54 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 6 Aug 2013 10:38:54 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375809346408-4660526.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
	<1375740272587-4660485.post@n4.nabble.com>
	<1375809346408-4660526.post@n4.nabble.com>
Message-ID: <1375810734377-4660527.post@n4.nabble.com>

cmharwood89 wrote
> @thljcl,
> 
> Wow, thanks for putting together those slides! That looks like a very
> helpful solution. I see the value of changing to 1fps, because I could
> essentially use time as a dummy variable for frame number, but I don't
> understand how that would allow me to trim at non keyframes. 
> 
> From my (limited) understanding, there simply is no way to trim at a
> P-frame without decompressing into a raw format first, trimming at the
> shiny new keyframe, and then recompressing back into my desired. Right
> now, I'm having pretty good results doing it this way. If I change the
> framerate to 1fps, it's not going to redistribute keyframes, right? If I
> choose a cutting point, say at frame 300, but the nearest I-frame is at
> 295, then my output video will be trimmed at 5 frames/5 seconds from the
> desired cut-point. 
> 
> For reference, my input and desired output are:
> INPUT:
> h264 codec @59.94 fps, 1920x1080 res; AAC audio codec; MP4 container
> 
> OUTPUT
> h264 codec @59.94 fps, 1920x1080 res; NO AUDIO; AVI container
> 
> I want the output video to be encoded with no subjectively observable
> losses.  Computing time and disk space are not a concern, but video
> quality and accurate trimming are absolutely necessary.
> 
> My original problem has been solved (by removing the square brackets
> surrounding my millisecond count in the -ss argument), but I'm definitely
> interested in making my approach smarter- i.e. avoiding a brute-force
> complete encoding of the video. I like the speed and losslessness of
> container-only manipulation and cutting at keyframes, and I like the
> accuracy of a decompress-then-cut approach. My new challenge is finding a
> way to combine the two.

In H.264, or any other codecs, there is a compression technique called
inter-frame compression. Because of inter-frame compression, typically you
cannot cut H.264 video stream based on frame numbers without round-off
error. You?re right, round-off error ?usually? resulted in inaccurate
trimming; instead of cutting at exactly the frame number you specified, what
you got was usually with round-off error. In fact, that?s exactly what
happened if you use mkvtoolnix directly to cut the video into pieces.
However, the method I described in the slideshow did not have such issue.
The round-off error problem was put away because of the constant frame rate
of 1 FPS. No, it would not get to the nearest key frame; it would get to
exactly the frame or ?time? you specified. If you think about it, it?s
pretty much like how to extract all frames (no matter whether or not they
are key frames) to individual images; to be in more details, it actually
involves both decoding and encoding process.
The typical lossy encoding process would result in the loss of quality.
Strictly speaking, it?s not possible for absolutely accurate trimming to
happen without re-encoding process for the source with inter-frame
compression. However, what really happens here is can be thought of
?lossless encoding?; but it?s not to create a video file of ?lossless
compression? but just cut a portion of video out. As you may already know
-codec ?copy? is to instruct the encoder the copy the stream without
re-encoding. But you can also say re-encoding is also happening; it?s just
that no change of information would occur in direct stream copying. To be
more specific, you just copy a single frame from the H.264 source; if that
happens to be non-key frame from the source, it would be the key frame for
the output file; but this is strictly stream-copying process, thus no
mathematical loss of information would occur in this process; it?s copying
data from multiple frames from the source to be a single key frame in the
output file.



--
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From cmharwoo at umich.edu  Tue Aug  6 20:05:13 2013
From: cmharwoo at umich.edu (cmharwood89)
Date: Tue, 6 Aug 2013 11:05:13 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375810734377-4660527.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
	<1375740272587-4660485.post@n4.nabble.com>
	<1375809346408-4660526.post@n4.nabble.com>
	<1375810734377-4660527.post@n4.nabble.com>
Message-ID: <1375812313667-4660528.post@n4.nabble.com>

thljcl wrote
> In H.264, or any other codecs, there is a compression technique called
> inter-frame compression. Because of inter-frame compression, typically you
> cannot cut H.264 video stream based on frame numbers without round-off
> error. You?re right, round-off error ?usually? resulted in inaccurate
> trimming; instead of cutting at exactly the frame number you specified,
> what you got was usually with round-off error. In fact, that?s exactly
> what happened if you use mkvtoolnix directly to cut the video into pieces.
> However, the method I described in the slideshow did not have such issue.
> The round-off error problem was put away because of the constant frame
> rate of 1 FPS. No, it would not get to the nearest key frame; it would get
> to exactly the frame or ?time? you specified. If you think about it, it?s
> pretty much like how to extract all frames (no matter whether or not they
> are key frames) to individual images; to be in more details, it actually
> involves both decoding and encoding process.
> The typical lossy encoding process would result in the loss of quality.
> Strictly speaking, it?s not possible for absolutely accurate trimming to
> happen without re-encoding process for the source with inter-frame
> compression. However, what really happens here is can be thought to be
> ?lossless encoding?; but it?s not to create a video file of ?lossless
> compression? but just cut a portion of video out. As you may already know
> -codec ?copy? is to instruct the encoder the copy the stream without
> re-encoding. But you can also say re-encoding is also happening; it?s just
> that no change of information would occur in direct stream copying. To be
> more specific, you just copy a single frame from the H.264 source; if that
> happens to be non-key frame from the source, it would be the key frame for
> the output file; but this is strictly stream-copying process, thus no
> mathematical loss of information would occur in this process; it?s copying
> data from multiple frames from the source to be a single key frame in the
> output file.

I stand corrected. Thanks a ton for taking the time to explain that. I'm
definitely going to try this out. It looks like mkvtoolnix has a command
line execution, so I should be able to execute everything programmatically. 



--
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From jiachielee at live.com  Tue Aug  6 20:11:03 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 6 Aug 2013 11:11:03 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375810734377-4660527.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
	<1375740272587-4660485.post@n4.nabble.com>
	<1375809346408-4660526.post@n4.nabble.com>
	<1375810734377-4660527.post@n4.nabble.com>
Message-ID: <1375812663183-4660529.post@n4.nabble.com>

thljcl wrote
> In H.264, or any other codecs, there is a compression technique called
> inter-frame compression. Because of inter-frame compression, typically you
> cannot cut H.264 video stream based on frame numbers without round-off
> error. You?re right, round-off error ?usually? resulted in inaccurate
> trimming; instead of cutting at exactly the frame number you specified,
> what you got was usually with round-off error. In fact, that?s exactly
> what happened if you use mkvtoolnix directly to cut the video into pieces.
> However, the method I described in the slideshow did not have such issue.
> The round-off error problem was put away because of the constant frame
> rate of 1 FPS. No, it would not get to the nearest key frame; it would get
> to exactly the frame or ?time? you specified. If you think about it, it?s
> pretty much like how to extract all frames (no matter whether or not they
> are key frames) to individual images; to be in more details, it actually
> involves both decoding and encoding process.
> The typical lossy encoding process would result in the loss of quality.
> Strictly speaking, it?s not possible for absolutely accurate trimming to
> happen without re-encoding process for the source with inter-frame
> compression. However, what really happens here is can be thought to be
> ?lossless encoding?; but it?s not to create a video file of ?lossless
> compression? but just cut a portion of video out. As you may already know
> -codec ?copy? is to instruct the encoder the copy the stream without
> re-encoding. But you can also say re-encoding is also happening; it?s just
> that no change of information would occur in direct stream copying. To be
> more specific, you just copy a single frame from the H.264 source; if that
> happens to be non-key frame from the source, it would be the key frame for
> the output file; but this is strictly stream-copying process, thus no
> mathematical loss of information would occur in this process; it?s copying
> data from multiple frames from the source to be a single key frame in the
> output file.

While I use mkvtoolnix to manipulate the MKV container, if you like your
video/audio track to be stored in other containers, if you use the -codec
?copy? to do that; that?s subject to limitation of the containers; other
containers may not have supported the as many codecs as MKV does. Not only
the video/audio streams would be copied, the stored container information
would be copied as well, subject to the container?s limitation. If not, as I
said earlier, the output file cannot be played correctly because h.264
elementary stream does not store frame rate information.



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From jiachielee at live.com  Tue Aug  6 20:33:09 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 6 Aug 2013 11:33:09 -0700 (PDT)
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375812313667-4660528.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
	<1375713411619-4660462.post@n4.nabble.com>
	<1375716335502-4660464.post@n4.nabble.com>
	<1375740272587-4660485.post@n4.nabble.com>
	<1375809346408-4660526.post@n4.nabble.com>
	<1375810734377-4660527.post@n4.nabble.com>
	<1375812313667-4660528.post@n4.nabble.com>
Message-ID: <1375813989086-4660530.post@n4.nabble.com>

cmharwood89 wrote
> I stand corrected. Thanks a ton for taking the time to explain that. I'm
> definitely going to try this out. It looks like mkvtoolnix has a command
> line execution, so I should be able to execute everything
> programmatically.

I typically write batch file to let ffmpeg execute a list of encoding jobs,
instead of writing a line of command directly in command processor. In my
previous posts of ?Create slideshow with ffmpeg from still images? at
http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-td4660277.html,
I even published the batch files I created. As promised, I did remove the
slideshow with slides of varying length together with the batch file from
SkyDrive.
I did try to include mkvtoolnix command in my batch file. Theoretically, it
should work. In practice, when it comes to creating batch file, I did make
careless mistakes quite often. The more complex the batch file got, the
harder for me to iron out the problems. I use the GUI front end because of
the ease of use. The visual feedback does help to reduce careless mistakes.
On the other hand, I do appreciate automated processes, if not I would not
prefer creating batch files for different tasks. So, at the end, I use the
combination of command-line UI and GUI. If you are a person who is extremely
careful and rarely make careless mistakes, I would admit that doing job
programmatically sounds a lot better.
GUI front end for x264 does exist, such as HandBrake. HandBrake even has a
command-line front end. But I eventually choose ffmpeg (as x264 front end)
instead of HandBrake because of the number of options I got from ffmpeg. I
do like more functionalities, features, or options, whatever you might call
them. I do not necessarily prefer one over another; it really depends on the
usage scenarios. I still keep HandBrake around.




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From parshap at gmail.com  Tue Aug  6 19:46:30 2013
From: parshap at gmail.com (Parsha Pourkhomami)
Date: Tue, 6 Aug 2013 10:46:30 -0700
Subject: [FFmpeg-user] Writing an MP3 file fails only when piping output
Message-ID: <CAHjQg2rMbGz7_W1qCS4obsmtPDmd_1thHL4iMuZyV_RySBpfVw@mail.gmail.com>

Tue Aug 6 12:40:20 CEST 2013, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Complete, uncut console output missing.

You're right. Sorry. See [1].

> Is the problem only reproducible with -acodec copy or also if you reencode?

The same issue exists when re-encoding. [2] Though, through this
exercise I've discovered that the problem is related to embedded
images in the mp3s. The failing files all have one or more mjpeg
streams in addition to the audio stream. If strip these out and keep
only the audio stream, it works. [3] I want to keep the embedded
images too though.

Thanks for the help.

[1]: https://gist.github.com/parshap/7e7d174db24ed061ae3b
[2]: https://gist.github.com/parshap/ba37d7ad1a61f2664846
[3]: https://gist.github.com/parshap/5de031eadbb98812bb17

From xuej at bokecc.com  Tue Aug  6 11:53:03 2013
From: xuej at bokecc.com (=?utf-8?B?6Jab5p2w?=)
Date: Tue, 6 Aug 2013 17:53:03 +0800
Subject: [FFmpeg-user] May I ask a question?
Message-ID: <tencent_10EE985A29C7CB276C92DA8F@qq.com>

Video transcoding time is related with the size of the video and the time of video,Whether a specific proportion?

From adf.lists at gmail.com  Wed Aug  7 01:45:34 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Wed, 07 Aug 2013 00:45:34 +0100
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
Message-ID: <52018A9E.4050604@gmail.com>

Francois Visagie wrote:

> Therein lies part of the problem, not all input files are AC3. Up to at
> least 30 June -filter:a aformat=channel_layouts=stereo could be used in a
> standard command line to produce stereo from multi-channel inputs with input
> and output volumes perceivably equal. Now each encode needs to be inspected
> individually for input/output differences, and the remedy will in each case
> also differ according to input type and/or volume differences. Really
> sub-optimal in my view, one which I expect to be more widely shared once
> these implications are more widely understood.

I had a look at the old behavior and it clipped, which is not good.

It was also inconsistent - wav and 7ch thd behaved like -ac 2.

I don't know what it did as such - maybe there is a way to explicitly 
recreate it, or perhaps just blindly boost the levels by xDb as part of 
the processing if you don't care about clipping.

I don't know about your use case, but if I were mixing for my self I 
would take care to process individually because that's what's needed to 
get the correct results.

> I sincerely appreciate the trouble you took with outlining various
> principles involved, but, on a more practical level: rather than making
> -filter:a aformat=channel_layouts=stereo now share the mechanism of -ac 2
> and -filter:a aresample=ocl=3 (incorrectly so wrt. volume levels in my
> view), what is the feasibility of making the other two behave like -filter:a
> aformat=channel_layouts=stereo instead?

I am not a developer - but IMHO the old behavior was wrong, but I 
haven't tested enough to work out what/why it did.

It's possible that it was intended by someone - it does seem to down mix 
in the sense it's not just blindly putting 100% in, but then it's not 
normalised enough to prevent clipping.

I must admit I saw a little bit of clipping on some of the 6ch masters I 
looked at - but there was even more after the "old" down mix.

FWIW I also consider the new behavior wrong in that the description of 
aformat says -

"Set output format constraints for the input audio. The framework will 
negotiate the most appropriate format to minimize conversions"

I think it should use -request_channels (where possible) and it doesn't, 
so anyone using -

aformat=channel_layouts=stereo

on say a 7.1 thd stream will not get the best result = a proper studio 
stereo mix, but instead a 7 -> 2 conversion and very low levels.




From jpboard2 at yahoo.com  Wed Aug  7 02:27:14 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 6 Aug 2013 17:27:14 -0700 (PDT)
Subject: [FFmpeg-user] OpenCL? What Effect Does It Have
In-Reply-To: <CAFHbX1KxPyHJcsCBLAGXSnXRpsUvKECir8n-vuAdEw1sH2ssdw@mail.gmail.com>
References: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<loom.20130805T194218-441@post.gmane.org>	<CAPYw7P7iO=EOQi-Xy+1MkosmbNp7R0kEjSqCs4tSTf2DsMLAWg@mail.gmail.com>	<1375794141.79897.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAFHbX1KxPyHJcsCBLAGXSnXRpsUvKECir8n-vuAdEw1sH2ssdw@mail.gmail.com>
Message-ID: <1375835234.77440.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>OpenCL can be used to accelerate the unsharp and deshake filters, as

>documented in the filters documentation.

Okay.? I saw that.? But are you saying that only those components of 
ffmpeg that are OpenCL accelerated?

>Generally the consensus (at least on here, as far as I can tell) is
>that GPU assisted encoding is slow and gives crap results. YMMV.

If optimized well, GPU assisted encoding would be extremely fast.
I bet the people that coded the OpenCL encoders didn't have much
GPU optimization experience.

From jpboard2 at yahoo.com  Wed Aug  7 02:30:18 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 6 Aug 2013 17:30:18 -0700 (PDT)
Subject: [FFmpeg-user] ffmpeg build fail with libx264
In-Reply-To: <5021193.W6HfUystka@mako>
References: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<5021193.W6HfUystka@mako>
Message-ID: <1375835418.42547.YahooMailNeo@web164705.mail.gq1.yahoo.com>

>>? ?  git clone git:://git.videolan.org/x264.git

>>? ?  make
>>? ?  make install
>
>You forgot ./configure --enable-shared before make.

Yes, I figured that out.? Is this documented anywhere?

From dashing.meng at gmail.com  Wed Aug  7 02:35:14 2013
From: dashing.meng at gmail.com (littlebat)
Date: Wed, 7 Aug 2013 08:35:14 +0800
Subject: [FFmpeg-user] May I ask a question?
In-Reply-To: <tencent_10EE985A29C7CB276C92DA8F@qq.com>
References: <tencent_10EE985A29C7CB276C92DA8F@qq.com>
Message-ID: <20130807083514.edf37a2c.dashing.meng@gmail.com>

On Tue, 6 Aug 2013 17:53:03 +0800
"??" <xuej at bokecc.com> wrote:

> Video transcoding time is related with the size of the video and the
> time of video,Whether a specific proportion?

Should be both of them, and more, e.g., source and destination video
bitrate, decoder, encoder, etc..

More detail you should go to google.

From francois.visagie at gmail.com  Wed Aug  7 11:03:15 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Wed, 7 Aug 2013 11:03:15 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
In-Reply-To: <52018A9E.4050604@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
Message-ID: <002101ce934c$f6c07a50$e4416ef0$@gmail.com>

> -----Original Message-----
> From: Andy Furniss [mailto:adf.lists at gmail.com]
> Sent: 07 August 2013 01:46
> To: FFmpeg user questions
> Cc: Francois Visagie
> Subject: Re: [FFmpeg-user] Preserving perceived loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Francois Visagie wrote:
> 
> > Therein lies part of the problem, not all input files are AC3. Up to
> > at least 30 June -filter:a aformat=channel_layouts=stereo could be
> > used in a standard command line to produce stereo from multi-channel
> > inputs with input and output volumes perceivably equal. Now each
> > encode needs to be inspected individually for input/output
> > differences, and the remedy will in each case also differ according to
> > input type and/or volume differences. Really sub-optimal in my view,
> > one which I expect to be more widely shared once these implications are
> more widely understood.
> 
> I had a look at the old behavior and it clipped, which is not good.
> 
> It was also inconsistent - wav and 7ch thd behaved like -ac 2.
> 
> I don't know what it did as such - maybe there is a way to explicitly
recreate
> it, or perhaps just blindly boost the levels by xDb as part of the
processing if
> you don't care about clipping.
> 
> I don't know about your use case, but if I were mixing for my self I would
> take care to process individually because that's what's needed to get the
> correct results.
> 
> > I sincerely appreciate the trouble you took with outlining various
> > principles involved, but, on a more practical level: rather than
> > making -filter:a aformat=channel_layouts=stereo now share the
> > mechanism of -ac 2 and -filter:a aresample=ocl=3 (incorrectly so wrt.
> > volume levels in my view), what is the feasibility of making the other
> > two behave like -filter:a aformat=channel_layouts=stereo instead?
> 
> I am not a developer - but IMHO the old behavior was wrong, but I haven't
> tested enough to work out what/why it did.
> 
> It's possible that it was intended by someone - it does seem to down mix
in
> the sense it's not just blindly putting 100% in, but then it's not
normalised
> enough to prevent clipping.
> 
> I must admit I saw a little bit of clipping on some of the 6ch masters I
looked
> at - but there was even more after the "old" down mix.
> 
> FWIW I also consider the new behavior wrong in that the description of
> aformat says -
> 
> "Set output format constraints for the input audio. The framework will
> negotiate the most appropriate format to minimize conversions"
> 
> I think it should use -request_channels (where possible) and it doesn't,
so
> anyone using -
> 
> aformat=channel_layouts=stereo
> 
> on say a 7.1 thd stream will not get the best result = a proper studio
stereo
> mix, but instead a 7 -> 2 conversion and very low levels.

I'm not sure even -request_channels produces the expected result. It merely
seems to influence the number of input channels guessed:
 
C:\Users\fvisagie\Videos\Home Videos\Testing\x264\Downmixing>ffmpeg -y -i
in.ac3 out.wav
ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug  1 2013 18:01:57 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
[ac3 @ 0275bc80] Estimating duration from bitrate, this may be inaccurate
Input #0, ac3, from 'in.ac3':
  Duration: 00:00:09.02, start: 0.000000, bitrate: 448 kb/s
    Stream #0:0: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
5.1(side), s16, 4608 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (ac3 -> pcm_s16le)
Press [q] to stop, [?] for help
size=    5076kB time=00:00:09.02 bitrate=4608.1kbits/s
video:0kB audio:5076kB subtitle:0 global headers:0kB muxing overhead
0.001962%

C:\Users\fvisagie\Videos\Home Videos\Testing\x264\Downmixing>ffmpeg -y
-request_channels 2 -i in.ac3 out.wav
ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug  1 2013 18:01:57 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
[ac3 @ 0365ea60] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, ac3, from 'in.ac3':
  Duration: 00:00:09.02, start: 0.000000, bitrate: 448 kb/s
    Stream #0:0: Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s
Output #0, wav, to 'out.wav':
  Metadata:
    ISFT            : Lavf55.12.102
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo,
s16, 1536 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (ac3 -> pcm_s16le)
Press [q] to stop, [?] for help
size=    1692kB time=00:00:09.02 bitrate=1536.1kbits/s
video:0kB audio:1692kB subtitle:0 global headers:0kB muxing overhead
0.004617%

Would it therefore be correct to assume that -request_channels leads to only
that number of channels being extracted, hence no down-mix?

I'm now thoroughly confused by the various "down-mixing" possibilities and
their potentially differing behaviour, but let me try to consolidate:

* you suggest processing individually which of course is the best approach
in principle
* once intended down-mixing and perhaps level adjustment have been decided
upon, which ffmpeg mechanism:
	* produces technically correct down-mixing?
	* works for most common audio input formats (e.g. according to Carl
Eugen aac does not support -request_channels?);
* or, can these two only be satisfied by down-mixing externally?



From tevans.uk at googlemail.com  Wed Aug  7 11:05:11 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Wed, 7 Aug 2013 10:05:11 +0100
Subject: [FFmpeg-user] OpenCL? What Effect Does It Have
In-Reply-To: <1375835234.77440.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1375720198.6431.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<loom.20130805T194218-441@post.gmane.org>
	<CAPYw7P7iO=EOQi-Xy+1MkosmbNp7R0kEjSqCs4tSTf2DsMLAWg@mail.gmail.com>
	<1375794141.79897.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAFHbX1KxPyHJcsCBLAGXSnXRpsUvKECir8n-vuAdEw1sH2ssdw@mail.gmail.com>
	<1375835234.77440.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1J_0nE01UABkXeC=iCPnSOYjDyBrqz2bkfcrmat_QU+aw@mail.gmail.com>

On Wed, Aug 7, 2013 at 1:27 AM, James Board <jpboard2 at yahoo.com> wrote:
>>OpenCL can be used to accelerate the unsharp and deshake filters, as
>
>>documented in the filters documentation.
>
> Okay.  I saw that.  But are you saying that only those components of
> ffmpeg that are OpenCL accelerated?

Yes. Carl said it earlier too.

>>Generally the consensus (at least on here, as far as I can tell) is
>>that GPU assisted encoding is slow and gives crap results. YMMV.
>
> If optimized well, GPU assisted encoding would be extremely fast.
> I bet the people that coded the OpenCL encoders didn't have much
> GPU optimization experience.

I'm not talking about GPU encoding in ffmpeg, but GPU encoding in
general. It's not an uncommon opinion:

"""
The primary comparison here is between Xilisoft?s Ultimate Video
Converter, Arcsoft Media Converter, and Cyberlink?s MediaEspresso.
Other potential options, including Avivo, Badaboom, and MediaCoder are
discussed at the end of the article.
?
The TL;DR version of this article is as follows: If you want a video
encoder that?ll run on virtually any system, has well-thought,
easy-to-use presets, and neatly balances quality and file size, go
download Handbrake. It?s fast, free, and efficient ? it just doesn?t
use the GPU.
"""

http://www.extremetech.com/computing/128681-the-wretched-state-of-gpu-transcoding

Handbrake, of course, being based around libav.

Cheers

Tom

From onemda at gmail.com  Wed Aug  7 14:17:28 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 7 Aug 2013 12:17:28 +0000
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
In-Reply-To: <CAEnqZEWX=b0qeJ3Gh_QRFKh-2cYVWYYFQ-3p_WwwjS=-w_1zsA@mail.gmail.com>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
	<loom.20130806T131859-871@post.gmane.org>
	<CAEnqZEWX=b0qeJ3Gh_QRFKh-2cYVWYYFQ-3p_WwwjS=-w_1zsA@mail.gmail.com>
Message-ID: <CAPYw7P5e8igYc1CKM5Eqn1h-1XfUZaRt0OuRSsyO_hszmQzZDQ@mail.gmail.com>

On 8/6/13, Robert Krueger <krueger at lesspain.de> wrote:
> On Tue, Aug 6, 2013 at 1:23 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
>> Robert Krueger <krueger <at> lesspain.de> writes:
>>
>>> it seems the ffmpeg always decodes the listed
>>> types of images to determine their properties.
>>
>> In your email's subject, you used the term "probing"
>> which is defined for FFmpeg / libavformat.
>> For all listed image formats, FFmpeg does no probing,
>> it guesses the format reading the file suffix.
>> (Something FFmpeg does not do for nearly all formats
>> it supports.)
>>
>>> What I did to check this, was to run
>>> ffmpeg -i <imagefile> as well as
>>> ffprobe -show_streams <imagefile> with each of
>>> these formats and set a breakpoint in
>>> avcodec_decode_video2.
>>
>> I don't think it is generally possible to find out
>> properties of image files (that is apparently what
>> you need) without calling this function.
>
> apparently my writing is too imprecise. So let me try another time to
> avoid misunderstandings.
>
> So are you saying that for the given file formats it is technically
> not possible to find out resolution and pixel format without decoding
> an entire frame or that it is currently not possible with
> ffmpeg/libavformat/libavcodec?

Yes, it is currently not possible, but I guess one could hack something
which will just add some private option to interested decoder(s) to just fetch
headers - without actual uncompressing of frames.

One could also write raw image demuxers in lavf which would just do that,
which would need bigger changes in img2 (de)muxer.

>
> Or formulated in another way: I am looking for a way to extract this
> type of metadata from these file types with a lot fewer system
> resources than are currently used when calling
> avformat_find_stream_info, the way it is done when using the command
> line tools. Is there a way to do this with the current code or is this
> generally not possible?
>
> Is it clear now what my goal is and what the actual question is? If
> there is a solution that requires using the libraries in a certain
> way, this thread should probably be moved to libav-user.

Everything is exceptionaly clear.

From jpboard2 at yahoo.com  Wed Aug  7 14:56:48 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 7 Aug 2013 05:56:48 -0700 (PDT)
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching
	from	mencoder)
In-Reply-To: <003f01ce9272$25a32920$70e97b60$@gmail.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<003f01ce9272$25a32920$70e97b60$@gmail.com>
Message-ID: <1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>

> ----------------------------------------
>> I saw this in another post on how to do 2-pass encoding:
>> 
>> ??? /usr/local/bin/ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -c:v libx264
-
>> preset medium -b:v 10m -pass 1 -an -f mp4 /dev/null
>> ??? /usr/local/bin/ffmpeg -y -i AI_kerndeint_ffvhuff_trim.avi -c:v libx264
-
>> preset medium -b:v 10m -pass 2 -an OUT.mp4
>> 
>> There was no error on the first command.? The error on the second was
>> ??? Error while opening encoder for output stream #0:0 - maybe incorrect
>> parameters such as bit_rate, rate, width or height

>There probably is something wrong with the format you're passing to the
>encoder. Normally I get this error when trying to pass 5.1-channel audio to
>the libvo-aacenc encoder which can't handle it, but in this case you've
>disabled audio with '-an' (do you really want that?). The problem therefore
>is with video, and it only surfaced on the 2nd pass because the 1st pass
>doesn't encode video. You'd have to inspect ffmpeg's console output for
>video information it reports to try work out what video parameter may have
>caused the error.

The only thing in the ffmpeg output from the 2ns pass that looks problematic is:
??? [libx264 @ 0x206f040] constant rate-factor is incompatible with 2pass
??? ....
??? Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

Is that a clue?
?
Thanks for your help.

From rodney.baker at iinet.net.au  Wed Aug  7 15:13:59 2013
From: rodney.baker at iinet.net.au (Rodney Baker)
Date: Wed, 07 Aug 2013 22:43:59 +0930
Subject: [FFmpeg-user] ffmpeg build fail with libx264
In-Reply-To: <1375835418.42547.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<5021193.W6HfUystka@mako>
	<1375835418.42547.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <1586135.Bssjbhb9vJ@mako>

On Tue, 6 Aug 2013 17:30:18 James Board wrote:
> >>     git clone git:://git.videolan.org/x264.git
> >>
> >>     make
> >>     make install
> >
> >You forgot ./configure --enable-shared before make.
> 
> Yes, I figured that out.  Is this documented anywhere?

If in doubt, ./configure --help usually provides configure options. It 
certainly works for both x264 and ffmpeg. The descriptions may be somewhat 
terse, but it will usually tell you what you need to know (unless its your 
first time building software from source, but I'm guessing that's not the case 
here). 

-- 
==============================================================
Rodney Baker VK5ZTV
rodney.baker at iinet.net.au
==============================================================

From tevans.uk at googlemail.com  Wed Aug  7 15:55:31 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Wed, 7 Aug 2013 14:55:31 +0100
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from
	mencoder)
In-Reply-To: <1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<003f01ce9272$25a32920$70e97b60$@gmail.com>
	<1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1+RZuJ87WcapoU5XdMG56DZ9OUaxo-wHj8Gb2urLfYn=A@mail.gmail.com>

On Wed, Aug 7, 2013 at 1:56 PM, James Board <jpboard2 at yahoo.com> wrote:
> The only thing in the ffmpeg output from the 2ns pass that looks problematic is:
>     [libx264 @ 0x206f040] constant rate-factor is incompatible with 2pass
>     ....
>     Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
>

[complete command line and uncut output missing]

CRF is one way of controlling the resulting bitrate, target bit rate
and 2-pass encoding is another. You can't use both at the same time.

Cheers

Tom

From krueger at lesspain.de  Wed Aug  7 15:56:01 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Wed, 7 Aug 2013 15:56:01 +0200
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
In-Reply-To: <CAPYw7P5e8igYc1CKM5Eqn1h-1XfUZaRt0OuRSsyO_hszmQzZDQ@mail.gmail.com>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
	<loom.20130806T131859-871@post.gmane.org>
	<CAEnqZEWX=b0qeJ3Gh_QRFKh-2cYVWYYFQ-3p_WwwjS=-w_1zsA@mail.gmail.com>
	<CAPYw7P5e8igYc1CKM5Eqn1h-1XfUZaRt0OuRSsyO_hszmQzZDQ@mail.gmail.com>
Message-ID: <CAEnqZEXsHtJYFz2Lv=cbfbHP1imG4LDtAZjMquMgafmS8dOvjQ@mail.gmail.com>

On Wed, Aug 7, 2013 at 2:17 PM, Paul B Mahol <onemda at gmail.com> wrote:
> On 8/6/13, Robert Krueger <krueger at lesspain.de> wrote:
>> On Tue, Aug 6, 2013 at 1:23 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
>>> Robert Krueger <krueger <at> lesspain.de> writes:
>>>
>>>> it seems the ffmpeg always decodes the listed
>>>> types of images to determine their properties.
>>>
>>> In your email's subject, you used the term "probing"
>>> which is defined for FFmpeg / libavformat.
>>> For all listed image formats, FFmpeg does no probing,
>>> it guesses the format reading the file suffix.
>>> (Something FFmpeg does not do for nearly all formats
>>> it supports.)
>>>
>>>> What I did to check this, was to run
>>>> ffmpeg -i <imagefile> as well as
>>>> ffprobe -show_streams <imagefile> with each of
>>>> these formats and set a breakpoint in
>>>> avcodec_decode_video2.
>>>
>>> I don't think it is generally possible to find out
>>> properties of image files (that is apparently what
>>> you need) without calling this function.
>>
>> apparently my writing is too imprecise. So let me try another time to
>> avoid misunderstandings.
>>
>> So are you saying that for the given file formats it is technically
>> not possible to find out resolution and pixel format without decoding
>> an entire frame or that it is currently not possible with
>> ffmpeg/libavformat/libavcodec?
>
> Yes, it is currently not possible, but I guess one could hack something
> which will just add some private option to interested decoder(s) to just fetch
> headers - without actual uncompressing of frames.
>
> One could also write raw image demuxers in lavf which would just do that,
> which would need bigger changes in img2 (de)muxer.

Thank you for that information!

From nicolas.george at normalesup.org  Wed Aug  7 16:15:26 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 7 Aug 2013 16:15:26 +0200
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
In-Reply-To: <CAPYw7P5e8igYc1CKM5Eqn1h-1XfUZaRt0OuRSsyO_hszmQzZDQ@mail.gmail.com>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
	<loom.20130806T131859-871@post.gmane.org>
	<CAEnqZEWX=b0qeJ3Gh_QRFKh-2cYVWYYFQ-3p_WwwjS=-w_1zsA@mail.gmail.com>
	<CAPYw7P5e8igYc1CKM5Eqn1h-1XfUZaRt0OuRSsyO_hszmQzZDQ@mail.gmail.com>
Message-ID: <20130807141526.GA11140@phare.normalesup.org>

Le decadi 20 thermidor, an CCXXI, Paul B Mahol a ?crit?:
> Yes, it is currently not possible, but I guess one could hack something
> which will just add some private option to interested decoder(s) to just fetch
> headers - without actual uncompressing of frames.

That is an interesting idea. It does not even need to be a private option,
since it makes sense for a lot of codecs, and would be of use for lavf too.

If the feature is implemented in the codec, it just returns immediately when
it has the format properties, before allocating a frame. If it is not
implemented, the frame is decoded and then discarded, that wastes time just
as it does now.

In fact, I half expected AVDISCARD_ALL to have this effect. It seems it does
not, or I did not test properly.

Regards,

-- 
  Nicolas George
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From nicolas.george at normalesup.org  Wed Aug  7 16:36:35 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 7 Aug 2013 16:36:35 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <52018A9E.4050604@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
Message-ID: <20130807143635.GA25535@phare.normalesup.org>

Le decadi 20 thermidor, an CCXXI, Andy Furniss a ?crit?:
> I had a look at the old behavior and it clipped, which is not good.
> 
> It was also inconsistent - wav and 7ch thd behaved like -ac 2.
> 
> I don't know what it did as such - maybe there is a way to
> explicitly recreate it, or perhaps just blindly boost the levels by
> xDb as part of the processing if you don't care about clipping.

The issue has been analyzed in the devel mailing list: the old downsampling
was done with samples coded in floating point, where clipping does not
happen (but can later happen if the samples are converted to integers);
because the number of conversions have been optimized it is now done with
samples coded as integers.

> I am not a developer - but IMHO the old behavior was wrong, but I
> haven't tested enough to work out what/why it did.

I tend to agree, the downsampling matrix should not depend on the format
selected to do the conversion, unless the user specifically asked for it,
because the format is selected automatically based on a rather complex
system and can change for unrelated reasons.

> FWIW I also consider the new behavior wrong in that the description
> of aformat says -
> 
> "Set output format constraints for the input audio. The framework
> will negotiate the most appropriate format to minimize conversions"

What is "wrong" in that?

> I think it should use -request_channels (where possible) and it
> doesn't, so anyone using -
> 
> aformat=channel_layouts=stereo
> 
> on say a 7.1 thd stream will not get the best result = a proper
> studio stereo mix, but instead a 7 -> 2 conversion and very low
> levels.

Not all codecs support channel layout selection like that.

Regards,

-- 
  Nicolas George
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From jpboard2 at yahoo.com  Wed Aug  7 16:37:10 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 7 Aug 2013 07:37:10 -0700 (PDT)
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching
	from	mencoder)
In-Reply-To: <CAFHbX1+RZuJ87WcapoU5XdMG56DZ9OUaxo-wHj8Gb2urLfYn=A@mail.gmail.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<003f01ce9272$25a32920$70e97b60$@gmail.com>	<1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAFHbX1+RZuJ87WcapoU5XdMG56DZ9OUaxo-wHj8Gb2urLfYn=A@mail.gmail.com>
Message-ID: <1375886230.35090.YahooMailNeo@web164704.mail.gq1.yahoo.com>

>> The only thing in the ffmpeg output from the 2ns pass that looks problematic is:

>>? ?  [libx264 @ 0x206f040] constant rate-factor is incompatible with 2pass
>>? ?  ....
>>? ?  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
>>
>
>CRF is one way of controlling the resulting bitrate, target bit rate
>and 2-pass encoding is another. You can't use both at the same time.

I'm not using both.? At least not intentionally.? Here were my commands:
?? ffmpeg -y -i IN.avi -c:v libx264 -preset medium -b:v 10m -pass 1 -an -f mp4 /dev/null
?? ffmpeg -y -i IN.avi -c:v libx264 -preset medium -b:v 10m -pass 2 -an OUT.mp4

Is the -preset option setting CRF?? I don't know.? I cut-and-pasted that command from
examples I found on the web.

What would be an example for doing 2-pass encoding, with the libx264 codec, at a video
rate of 10 million bits per second, and merely copy the audio from input file to output file, and using
an mp4 format?

From nicolas.george at normalesup.org  Wed Aug  7 16:50:29 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 7 Aug 2013 16:50:29 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <002101ce934c$f6c07a50$e4416ef0$@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>
Message-ID: <20130807145029.GB25535@phare.normalesup.org>

Le decadi 20 thermidor, an CCXXI, Francois Visagie a ?crit?:
> I'm not sure even -request_channels produces the expected result. It merely
> seems to influence the number of input channels guessed:

If it did that, the sound would be completely garbled.

> Input #0, ac3, from 'in.ac3':
>   Duration: 00:00:09.02, start: 0.000000, bitrate: 448 kb/s
>     Stream #0:0: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s

Normal.

> Guessed Channel Layout for  Input Stream #0.0 : stereo

You can safely ignore this particular message, it just means that something
somewhere set channels to 2 but neglected to set channel_layout to stereo.

> Input #0, ac3, from 'in.ac3':
>   Duration: 00:00:09.02, start: 0.000000, bitrate: 448 kb/s
>     Stream #0:0: Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s

Normal.

> Would it therefore be correct to assume that -request_channels leads to only
> that number of channels being extracted, hence no down-mix?

No. -request_channels uses codec-specific ways of extracting sound with the
specified number of channels. It only works for very few codecs that have
that feature.

Regards,

-- 
  Nicolas George
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From stefasab at gmail.com  Wed Aug  7 17:22:55 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Wed, 7 Aug 2013 17:22:55 +0200
Subject: [FFmpeg-user] Probing jpg,
 png and tiff images without decoding frames
In-Reply-To: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
Message-ID: <20130807152255.GG2578@barisone>

On date Tuesday 2013-08-06 13:09:00 +0200, Robert Kr?ger wrote:
> Hi,
> 
> it seems the ffmpeg always decodes the listed types of images to
> determine their properties. What I did to check this, was to run
> ffmpeg -i <imagefile> as well as ffprobe -show_streams <imagefile>
> with each of these formats and set a breakpoint in
> avcodec_decode_video2.
> 
> This is not a problem in many cases, however when using ffmpeg,
> ffprobe or its libraries to extract metadata such as codec, size and
> pixel format from thousands of image files (imagine analyzing a full
> SD card of Hires JPG images), the difference between header parsing
> and decoding becomes very significant (especially with large images
> for obvious reasons).
> 

> Is this (extracting metadata for those formats just by header parsing)
> currently implemented and I am just missing a command line argument?
> 
> If not, shall I submit this as an enhancement request in trac?

Makes sense, possibly mentioning this thread or a short summary.

To my knowledge, there is no way to detect the image size and other
information without decoding the frame, my bet is that this would
require some API extension.

From francois.visagie at gmail.com  Wed Aug  7 17:24:35 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Wed, 7 Aug 2013 17:24:35 +0200
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg
	(switching	from	mencoder)
In-Reply-To: <1375886230.35090.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<003f01ce9272$25a32920$70e97b60$@gmail.com>	<1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<CAFHbX1+RZuJ87WcapoU5XdMG56DZ9OUaxo-wHj8Gb2urLfYn=A@mail.gmail.com>
	<1375886230.35090.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <005101ce9382$3cab3d90$b601b8b0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of James Board
> Sent: 07 August 2013 16:37
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from
> mencoder)
> 
> >> The only thing in the ffmpeg output from the 2ns pass that looks
> problematic is:
> 
> >>? ?  [libx264 @ 0x206f040] constant rate-factor is incompatible with
> >>2pass
> >>? ?  ....
> >>? ?  Error while opening encoder for output stream #0:0 - maybe
> >>incorrect parameters such as bit_rate, rate, width or height
> >>
> >
> >CRF is one way of controlling the resulting bitrate, target bit rate
> >and 2-pass encoding is another. You can't use both at the same time.
> 
> I'm not using both.? At least not intentionally.? Here were my commands:
> ?? ffmpeg -y -i IN.avi -c:v libx264 -preset medium -b:v 10m -pass 1 -an -f
mp4
> /dev/null

'-f rawvideo' instead of '-f mp4' may be a bit faster. You can also add
'-fastfirstpass 1'.

> ?? ffmpeg -y -i IN.avi -c:v libx264 -preset medium -b:v 10m -pass 2 -an
> OUT.mp4
> 
> Is the -preset option setting CRF?? I don't know.? I cut-and-pasted that
> command from examples I found on the web.

I don't see how CRF gets specified either. For reference, here are the
command lines I use:

ffmpeg -y -i INPUT -an -c:v libx264 -preset veryslow -fastfirstpass 1 -f
rawvideo -b:v 2500000 -pass 1 NUL
ffmpeg -y -I INPUT -c:v libx264 -preset veryslow -b:v 2500000 -pass 2
OUTPUT.mp4

> What would be an example for doing 2-pass encoding, with the libx264
> codec, at a video rate of 10 million bits per second, and merely copy the
> audio from input file to output file, and using an mp4 format?

ffmpeg -y -i INPUT -an -c:v libx264 -preset veryslow -fastfirstpass 1 -f
rawvideo -b:v 10000000 -pass 1 NUL
ffmpeg -y -I INPUT -c:a copy -c:v libx264 -preset veryslow -b:v 10000000
-pass 2 OUTPUT.mp4

'-preset' according to your preferences, of course.

> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From francois.visagie at gmail.com  Wed Aug  7 18:15:22 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Wed, 7 Aug 2013 18:15:22 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
In-Reply-To: <20130807145029.GB25535@phare.normalesup.org>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>	<51FEB571.2080701@gmail.com>	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>	<52018A9E.4050604@gmail.com>	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>
	<20130807145029.GB25535@phare.normalesup.org>
Message-ID: <005501ce9389$548a5e80$fd9f1b80$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Nicolas George
> Sent: 07 August 2013 16:50
> To: FFmpeg user questions
> Cc: 'Andy Furniss'
> Subject: Re: [FFmpeg-user] Preserving perceived loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Le decadi 20 thermidor, an CCXXI, Francois Visagie a ?crit?:
> > I'm not sure even -request_channels produces the expected result. It
> > merely seems to influence the number of input channels guessed:
> 
> If it did that, the sound would be completely garbled.
> 
> > Input #0, ac3, from 'in.ac3':
> >   Duration: 00:00:09.02, start: 0.000000, bitrate: 448 kb/s
> >     Stream #0:0: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
> 
> Normal.
> 
> > Guessed Channel Layout for  Input Stream #0.0 : stereo
> 
> You can safely ignore this particular message, it just means that
something
> somewhere set channels to 2 but neglected to set channel_layout to stereo.
> 
> > Input #0, ac3, from 'in.ac3':
> >   Duration: 00:00:09.02, start: 0.000000, bitrate: 448 kb/s
> >     Stream #0:0: Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s
> 
> Normal.
> 
> > Would it therefore be correct to assume that -request_channels leads
> > to only that number of channels being extracted, hence no down-mix?
> 
> No. -request_channels uses codec-specific ways of extracting sound with
the
> specified number of channels. It only works for very few codecs that have
> that feature.

Thanks for your feedback.

Is it therefore correct to say that:
	* the only input codec-independent way of downmixing to stereo is
?-ac 2?/?-filter:a
aformat=channel_layouts=stereo?/?-filter:a_aresample=ocl=3? (which now all
behave the same?), and
	* if one wants to preserve perceived input volume one needs to
adjust gain during encoding?

Further to that, for a given energy level per input channel, does the
current down-mixing mechanism produce differing output energy levels
depending on the _number_ of input channels? I.e. is it expected that
different input layouts (with the same energy level per channel) would
require different gain factors for equally loud outputs, or will one be able
to find a suitable gain factor and use that regardless of number of input
channels?

Thanks,
Francois

> 
> Regards,
> 
> --
>   Nicolas George


From adf.lists at gmail.com  Wed Aug  7 18:16:55 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Wed, 07 Aug 2013 17:16:55 +0100
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <20130807143635.GA25535@phare.normalesup.org>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<20130807143635.GA25535@phare.normalesup.org>
Message-ID: <520272F7.4050706@gmail.com>

Nicolas George wrote:

> The issue has been analyzed in the devel mailing list: the old downsampling
> was done with samples coded in floating point, where clipping does not
> happen (but can later happen if the samples are converted to integers);
> because the number of conversions have been optimized it is now done with
> samples coded as integers.

Ok, thanks for the info.

>> FWIW I also consider the new behavior wrong in that the description
>> of aformat says -
>>
>> "Set output format constraints for the input audio. The framework
>> will negotiate the most appropriate format to minimize conversions"
>
> What is "wrong" in that?

Nothing in the statement its self and I also accept that format may mean 
more than number of channels.

What I thought was wrong was the behavior with my thd example that 
clearly doesn't

"negotiate the most appropriate format to minimize conversions"

> Not all codecs support channel layout selection like that.

Yea, but if the codec does, then maybe the code could try to do the best 
for the user that requested stereo by using it. The user may not know 
the inner workings of every codec, but the code can.

Of course dca should be exempt until it's fixed, but that should be for 
another thread/further analysis :-)






From chaitanyag at gopalsystems.com  Wed Aug  7 11:30:08 2013
From: chaitanyag at gopalsystems.com (Chaitanya Gopathi)
Date: Wed, 7 Aug 2013 15:00:08 +0530
Subject: [FFmpeg-user] FFServer config
Message-ID: <CABRdPa+FBy7s8NJvWhpe_-sb+0SyJUr=7dabeSBNJvSA=L6HfQ@mail.gmail.com>

Hi,

I am a php developer and new to work with ffserver. we have the requirement
of live stream of video storing onto the server. For this, ffserver is
installed on our server by the hosting provider and i need to configure it.
can you please guide me how can i do that?

Thanks in advance,

Chaitanya

From lionelpetit at gmail.com  Wed Aug  7 11:47:46 2013
From: lionelpetit at gmail.com (Lionel Petit)
Date: Wed, 7 Aug 2013 11:47:46 +0200
Subject: [FFmpeg-user] Using x11grab to grab two displays simultaneously
Message-ID: <CAMu2XMCiBirbuHM4YurpyfoO+pk+ozCo=KQdC2pDHGJFybUhEQ@mail.gmail.com>

Hi,

I have a computer with two screens and I would like to capture the two
displays simultaneously and put the result in a container of 10 minutes
chunks.
The idea is to have the video streams of each display at any time (playing
the video streams of the correct chunk).

Firstly, I have tested with two ffmpeg commands and I have merged the
output in one file (the result is a container file which has 2 video
streams):
The capture commands I used :

./ffmpeg -f x11grab -r 30 -s sxga -i :0.0 -vcodec libx264 -threads 0 -flags
global_header -map 0 -g 300 -f segment -segment_time 600 -reset_timestamps
1 firstScreen-%03d.mkv

./ffmpeg -f x11grab -r 30 -s sxga -i :0.1 -vcodec libx264 -threads 0 -flags
global_header -map 0 -g 300 -f segment -segment_time 600 -reset_timestamps
1 secondScreen-%03d.mkv


Later, I tried to do the same in one command :

ffmpeg -f x11grab -r 30 -s sxga -i :0.0 -f x11grab -r 30 -s sxga -i :0.1
-vcodec libx264 -map 0 -vcodec libx264 -map 1 -threads 0 -flags
global_header -g 300 -f segment -segment_time 600 -reset_timestamps 1
outputFile-%03d.mkv

I prefer this solution because I just have one command line, one container,
I'm sure to have the same 10 minutes and I have nothing to do after the
capture.
But there are some problems of I-frame at the beginning of the streams and
I have to wait few seconds before seeing a whole image.
I try to use the reference_stream option without success.

The questions I have :
- How could I force an I-frame at the beginning of a new segment ?
- Is there an easier solution to do this capture ?
- Do you have advice to help me realize this capture ?

Best regards,

Lionel

From adf.lists at gmail.com  Wed Aug  7 19:15:38 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Wed, 07 Aug 2013 18:15:38 +0100
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <002101ce934c$f6c07a50$e4416ef0$@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>
Message-ID: <520280BA.2090804@gmail.com>

Francois Visagie wrote:

> Would it therefore be correct to assume that -request_channels leads
> to only that number of channels being extracted, hence no down-mix?

I am no expert, just a little learning, which of course can be a
dangerous thing :-)

Only for TrueHD so far, dtsMA will likely be the same but it's not
currently supported.

ac3 will still down mix in the decoder, but with request_channels it can
take in to account meta data in the stream so the mix should be more
like the studio intended. In practice it's not that much different from
-ac 2 level wise because it's still normalised to avoid clipping.

> I'm now thoroughly confused by the various "down-mixing"
> possibilities and their potentially differing behaviour, but let me
> try to consolidate:
>
> * you suggest processing individually which of course is the best
> approach in principle

If you want max volume but not to clip I can't think of any other way.
The whole soundtrack needs to be analysed to assess headroom then the
volume can be boosted by whatever amount there is. Even then I guess a
real sound engineer could work out if some of the peaks are very
rare/excessive and decide to clip those in order to get more volume.

I don't know if ffmpeg can do this with sox it's as easy as

sox in.wav  out.wav  gain -n

perceived volume of course gets even more complicated and depends on the
dynamic range of the source - it can be controlled for  studio ac3 but
nothing else so far AFAIK, even though AAC and DTS use it, it's not
supported by their ffmpeg decoders. FAAD2 supports it for AAC but it's
buggy so avoid.

> * once intended down-mixing and perhaps level adjustment have been
> decided upon, which ffmpeg mechanism: * produces technically correct
> down-mixing?

Apart from getting the codec to do it where possible (excluding dts
until fixed) I think that -ac 2 should be considered correct.

> * works for most common audio input formats (e.g. according to Carl
> Eugen aac does not support -request_channels?); * or, can these two
> only be satisfied by down-mixing externally?

As I wrote elsewhere in this thread, having something that hides the
complexity of what the decoders can/can't do would be user friendly, but
I don't know the code/complexity of actually having -aformat do this.


From nicolas.george at normalesup.org  Wed Aug  7 20:00:30 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 7 Aug 2013 20:00:30 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <005501ce9389$548a5e80$fd9f1b80$@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>
	<20130807145029.GB25535@phare.normalesup.org>
	<005501ce9389$548a5e80$fd9f1b80$@gmail.com>
Message-ID: <20130807180030.GB14079@phare.normalesup.org>

Le decadi 20 thermidor, an CCXXI, Francois Visagie a ?crit?:
> Is it therefore correct to say that:
> 	* the only input codec-independent way of downmixing to stereo is
> ?-ac 2?/?-filter:a
> aformat=channel_layouts=stereo?/?-filter:a aresample=ocl=3? (which now all
> behave the same?), and

Yes. Note that it is always safe to specify both that and
"-request_channels?2": codecs that do not support it will just ignore the
option, and if the option is supported, the filters will just do nothing.

> 	* if one wants to preserve perceived input volume one needs to
> adjust gain during encoding?

Yes, but if you do that, unless your input was never at peak level, you will
get clipping, and that is probably worse than low volume.

> Further to that, for a given energy level per input channel, does the
> current down-mixing mechanism produce differing output energy levels
> depending on the _number_ of input channels? I.e. is it expected that
> different input layouts (with the same energy level per channel) would
> require different gain factors for equally loud outputs, or will one be able
> to find a suitable gain factor and use that regardless of number of input
> channels?

Well, of course, it depends on the number of input channels. If you want to
mix one channel into one, you do not need to lower the volume. If you want
to mix forty-two channels into one, you need to divide the amplitude by
forty-two to avoid clipping. What it does to energy depends on the input. If
the channels are in phase, the energy is preserved; if they are not, each
the energy of each channel is divided by forty-two squared, and then the
energies are summed, the net result is a division by forty-two.

lswr uses rather tricky coefficients to match the standard loudness and
geometry of speakers. You can get the exact matrix with -loglevel debug. For
example, here is the 5.1 -> stereo matrix:

0.414214 0.000000 0.292893 0.000000 0.292893 0.000000 
0.000000 0.414214 0.292893 0.000000 0.000000 0.292893 

That means: out_left = 0.414 front_left + 0.293 center + 0.293 back_left
and the symmetrical formula for right; note that LFE is discarded.

Regards,

-- 
  Nicolas George
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From nicolas.george at normalesup.org  Wed Aug  7 20:02:25 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 7 Aug 2013 20:02:25 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <520272F7.4050706@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<20130807143635.GA25535@phare.normalesup.org>
	<520272F7.4050706@gmail.com>
Message-ID: <20130807180225.GC14079@phare.normalesup.org>

Le decadi 20 thermidor, an CCXXI, Andy Furniss a ?crit?:
> Yea, but if the codec does, then maybe the code could try to do the
> best for the user that requested stereo by using it. The user may
> not know the inner workings of every codec, but the code can.

Oh, I see. It can not currently work: decoding and filtering are completely
separate processes. There is currently no API to query a codec to know the
channel layouts it can produce natively.

> Of course dca should be exempt until it's fixed, but that should be
> for another thread/further analysis :-)

Is there a trac ticket?

Regards,

-- 
  Nicolas George
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From mark.himsley at gmail.com  Wed Aug  7 22:14:35 2013
From: mark.himsley at gmail.com (Mark Himsley)
Date: Wed, 7 Aug 2013 21:14:35 +0100
Subject: [FFmpeg-user] Inaccurate trimming with re-encoding
In-Reply-To: <1375711420502-4660459.post@n4.nabble.com>
References: <1375487953644-4660437.post@n4.nabble.com>
	<CAPAFRRQEhCv9mNGjWjg-Min8jMvRi6YOs58n+pqN_iuXTJB9pA@mail.gmail.com>
	<1375710620712-4660457.post@n4.nabble.com>
	<1375711420502-4660459.post@n4.nabble.com>
Message-ID: <CAPAFRRTK=8=sWPsoo5JY-_J68orTArOCC-eJaS8BqHywsZeeBg@mail.gmail.com>

I said "The square brackets around the decimal parts of seconds is to
indicate to you that they are optional."

What I mean is, the format is HH:MM:SS[.ms], so you can specify 01:02:03
for hours, minutes and seconds, or 01:02:03.48 for hours minutes, seconds
and decimal parts of seconds.

You said your command was "ffmpeg -t 30 -i GOPR0103.mp4 -vcodec libx264
?threads 0 -preset ultrafast
-an -y -ss 00:03:09[.606] Run041_UW1.avi"

I said "Remove the square brackets from your command."

What I mean is, try your command as:

ffmpeg -t 30 -i GOPR0103.mp4 -vcodec libx264 ?threads 0 -preset ultrafast
-an -y -ss 00:03:09.606 Run041_UW1.avi

See. No square brackets.



On 5 August 2013 15:03, cmharwood89 <cmharwoo at umich.edu> wrote:

> Or am I misunderstanding what you're saying? Do you mean to include the
> milliseconds, but without the square brackets? I thought you were
> suggesting
> to truncate the time down to HH:MM:SS format.
>
>
>
> --
> View this message in context:
> http://ffmpeg-users.933282.n4.nabble.com/Inaccurate-trimming-with-re-encoding-tp4660437p4660459.html
> Sent from the FFmpeg-users mailing list archive at Nabble.com.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>



-- 
Mark Himsley

From adf.lists at gmail.com  Thu Aug  8 00:44:07 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Wed, 07 Aug 2013 23:44:07 +0100
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <20130807180225.GC14079@phare.normalesup.org>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<20130807143635.GA25535@phare.normalesup.org>
	<520272F7.4050706@gmail.com>
	<20130807180225.GC14079@phare.normalesup.org>
Message-ID: <5202CDB7.40707@gmail.com>

Nicolas George wrote:
> Le decadi 20 thermidor, an CCXXI, Andy Furniss a ?crit :
>> Yea, but if the codec does, then maybe the code could try to do
>> the best for the user that requested stereo by using it. The user
>> may not know the inner workings of every codec, but the code can.
>
> Oh, I see. It can not currently work: decoding and filtering are
> completely separate processes. There is currently no API to query a
> codec to know the channel layouts it can produce natively.

Ahh, OK.

>
>> Of course dca should be exempt until it's fixed, but that should
>> be for another thread/further analysis :-)
>
> Is there a trac ticket?

Not by me (yet) and I haven't searched either.

I will file a bug when I've searched out more "normal" in size and
format samples (my channel check is from MA and 1.1G) and had time to
test and look more at the code (which will take a while).

Currently I think there are three possibly separate issues - in summary

Down mix too loud/clipping - may be the same as old aformat issue
discussed here - may not.

Default matrix in libavcodec/dcadata.h looks odd assuming
dca_default_coeffs refers to dca_downmix_coeffs it could explain what I
hear on my MA channel check  - mixing L into R -6 db etc.

Why is default being used anyway - I would have expected studio stuff to
have downmix meta and so not hit that matrix anyway - again more
samples/testing/time needed.


From lou at lrcd.com  Thu Aug  8 00:56:24 2013
From: lou at lrcd.com (Lou)
Date: Wed, 7 Aug 2013 14:56:24 -0800
Subject: [FFmpeg-user] ffmpeg build fail with libx264
In-Reply-To: <1375749880.42456.YahooMailNeo@web164701.mail.gq1.yahoo.com>
References: <1375746978.84632.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<loom.20130806T020742-383@post.gmane.org>
	<1375749880.42456.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <20130807145624.4223cc82@lrcd.com>

On Mon, 5 Aug 2013 17:44:40 -0700 (PDT)
James Board <jpboard2 at yahoo.com> wrote:

> No, Centos 6.4.? Version 107 of x264.? Mencoder had no problems
> with it.

See:
http://trac.ffmpeg.org/wiki/CentosCompilationGuide

From codecomplete at free.fr  Thu Aug  8 01:20:25 2013
From: codecomplete at free.fr (Gilles)
Date: Thu, 08 Aug 2013 01:20:25 +0200
Subject: [FFmpeg-user] Repeat video?
Message-ID: <d5l509dekdpr2hs668srhp1ivimnu9m630@4ax.com>

Hello

I'd like to repeat a 2mn MP4 video so it lasts 30mn.

Has this feature been implemented since this message?

"pipe ffplay to ffmpeg - or - loop input video in ffmpeg for rtmp
broadcast stream ?"
http://ffmpeg.org/pipermail/ffmpeg-user/2012-April/006009.html

Thank you.


From jpboard2 at yahoo.com  Thu Aug  8 02:35:25 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 7 Aug 2013 17:35:25 -0700 (PDT)
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
Message-ID: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>

I'm trying to convert from mencoder to ffmpeg.? I can't
get the inverse telecine correct.? I did this command

??? ffmpeg -i IN.avi -vf fieldmatch,decimate out.avi

But the input file has 33 percent more frames then the output file.
The input file should have 25 percent more frames, right?? Seems like
ffmpeg is dropping too many frames.


By contrast, this is how I did inverse telecine with mencoder:
??? mencoder -ovc lavc -lavcopts vcodec=ffvhuff -oac copy -vf filmdint -o OUT.avi -ofps 24000/1001 IN.avi


What's the correct way, or ways, to inverse telecine with ffmpeg?

From jpboard2 at yahoo.com  Thu Aug  8 03:25:55 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 7 Aug 2013 18:25:55 -0700 (PDT)
Subject: [FFmpeg-user] 2-pass ENcoding with
	mpeg	(switching	from	mencoder)
In-Reply-To: <005101ce9382$3cab3d90$b601b8b0$@gmail.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<003f01ce9272$25a32920$70e97b60$@gmail.com>	<1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<CAFHbX1+RZuJ87WcapoU5XdMG56DZ9OUaxo-wHj8Gb2urLfYn=A@mail.gmail.com>	<1375886230.35090.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<005101ce9382$3cab3d90$b601b8b0$@gmail.com>
Message-ID: <1375925155.58796.YahooMailNeo@web164704.mail.gq1.yahoo.com>

>> What would be an example for doing 2-pass encoding, with the libx264

>> codec, at a video rate of 10 million bits per second, and merely copy the
>> audio from input file to output file, and using an mp4 format?
>
>ffmpeg -y -i INPUT -an -c:v libx264 -preset veryslow -fastfirstpass 1 -f
>rawvideo -b:v 10000000 -pass 1 NUL
>ffmpeg -y -I INPUT -c:a copy -c:v libx264 -preset veryslow -b:v 10000000
>-pass 2 OUTPUT.mp4

I tried your commands to do 2-pass encoding with ffmpeg.? The input file
was a short 6 second video clip.? Both commands and their output are listed
below.? The first command seemed seemed okay, but the second command failed.


> ffmpeg -y -i IN.avi -an -c:v libx264 -preset veryslow -fastfirstpass 1 -f rawvideo -b:v 10000000 -pass 1 NUL
ffmpeg version N-54733-gab5f581 Copyright (c) 2000-2013 the FFmpeg developers
? configuration: --enable-libfreetype --enable-gpl --enable-libx264
? libavutil????? 52. 39.100 / 52. 39.100
? libavcodec???? 55. 18.102 / 55. 18.102
? libavformat??? 55. 12.102 / 55. 12.102
? libavdevice??? 55.? 3.100 / 55.? 3.100
? libavfilter???? 3. 80.101 /? 3. 80.101
? libswscale????? 2.? 3.100 /? 2.? 3.100
? libswresample?? 0. 17.102 /? 0. 17.102
? libpostproc??? 52.? 3.100 / 52.? 3.100
[avi @ 0x2645b00] Stream #0: not enough frames to estimate rate; consider increasing probesize
[avi @ 0x2645b00] Stream #1: not enough frames to estimate rate; consider increasing probesize
Guessed Channel Layout for? Input Stream #0.1 : stereo
Input #0, avi, from 'IN.avi':
? Duration: 00:00:05.74, start: 0.000000, bitrate: 997780 kb/s
??? Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1920x1080, 29.97 tbr, 29.97 tbn, 29.97 tbc
??? Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
No pixel format specified, yuv422p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x2649ee0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x2649ee0] profile High 4:2:2, level 4.0, 4:2:2 8-bit
Output #0, rawvideo, to 'NUL':
? Metadata:
??? encoder???????? : Lavf55.12.102
??? Stream #0:0: Video: h264 (libx264), yuv422p, 1920x1080, q=-1--1, pass 1, 10000 kb/s, 90k tbn, 29.97 tbc
Stream mapping:
? Stream #0:0 -> #0:0 (rawvideo -> libx264)
Press [q] to stop, [?] for help
frame=? 172 fps= 23 q=-1.0 Lsize=??? 7709kB time=00:00:05.67 bitrate=11133.7kbits/s
video:7709kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.000000%
[libx264 @ 0x2649ee0] frame I:2???? Avg QP:11.39? size:300209
[libx264 @ 0x2649ee0] frame P:39??? Avg QP:18.47? size: 83146
[libx264 @ 0x2649ee0] frame B:131?? Avg QP:19.79? size: 30925
[libx264 @ 0x2649ee0] consecutive B-frames:? 1.2%? 0.0% 14.0% 41.9% 20.3% 17.4%? 0.0%? 0.0%? 5.2%
[libx264 @ 0x2649ee0] mb I? I16..4: 25.5%? 0.0% 74.5%
[libx264 @ 0x2649ee0] mb P? I16..4: 39.8%? 0.0%? 0.0%? P16..4: 50.6%? 0.0%? 0.0%? 0.0%? 0.0%??? skip: 9.6%
[libx264 @ 0x2649ee0] mb B? I16..4:? 6.3%? 0.0%? 0.0%? B16..8: 37.7%? 0.0%? 0.0%? direct:19.8%? skip:36.2%? L0:38.2% L1:41.8% BI:20.0%
[libx264 @ 0x2649ee0] final ratefactor: 17.47
[libx264 @ 0x2649ee0] direct mvs? spatial:90.1% temporal:9.9%
[libx264 @ 0x2649ee0] coded y,uvDC,uvAC intra: 37.2% 72.0% 37.1% inter: 17.7% 33.7% 2.2%
[libx264 @ 0x2649ee0] i16 v,h,dc,p: 29% 42% 19% 10%
[libx264 @ 0x2649ee0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 27% 10%? 6%? 6%? 7%? 5%? 7%? 5%
[libx264 @ 0x2649ee0] i8c dc,h,v,p: 42% 25% 24%? 9%
[libx264 @ 0x2649ee0] Weighted P-Frames: Y:2.6% UV:2.6%
[libx264 @ 0x2649ee0] kb/s:11004.23


> ffmpeg -y -i IN.avi -c:a copy -c:v libx264 -preset veryslow -b:v 10000000 -pass 2 OUT.mp4
ffmpeg version N-54733-gab5f581 Copyright (c) 2000-2013 the FFmpeg developers
? configuration: --enable-libfreetype --enable-gpl --enable-libx264
? libavutil????? 52. 39.100 / 52. 39.100
? libavcodec???? 55. 18.102 / 55. 18.102
? libavformat??? 55. 12.102 / 55. 12.102
? libavdevice??? 55.? 3.100 / 55.? 3.100
? libavfilter???? 3. 80.101 /? 3. 80.101
? libswscale????? 2.? 3.100 /? 2.? 3.100
? libswresample?? 0. 17.102 /? 0. 17.102
? libpostproc??? 52.? 3.100 / 52.? 3.100
[avi @ 0x1b2a9c0] Stream #0: not enough frames to estimate rate; consider increasing probesize
[avi @ 0x1b2a9c0] Stream #1: not enough frames to estimate rate; consider increasing probesize
Guessed Channel Layout for? Input Stream #0.1 : stereo
Input #0, avi, from 'IN.avi':
? Duration: 00:00:05.74, start: 0.000000, bitrate: 997780 kb/s
??? Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1920x1080, 29.97 tbr, 29.97 tbn, 29.97 tbc
??? Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
No pixel format specified, yuv422p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x1b2edc0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x1b2edc0] profile High 4:2:2, level 5.1, 4:2:2 8-bit
[libx264 @ 0x1b2edc0] 264 - core 135 r2345 f0c1c53 - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1
ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11
fast_pskip=1 chroma_qp_offset=-2 threads=18 lookahead_threads=3
sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3
weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40
intra_refresh=0 rc_lookahead=60 rc=2pass mbtree=1 bitrate=10000 ratetol=1.0
qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 cplxblur=20.0 qblur=0.5 ip_ratio=1.40
aq=1:1.00
[mp4 @ 0x1b2e500] Tag [1][0][0][0]/0x00000001 incompatible with output codec id '65536' ([0][0][0][0])
output #0, mp4, to 'OUT.mp4':
??? Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv422p, 1920x1080, q=-1--1, pass 2, 10000 kb/s, 90k tbn, 29.97 tbc
??? Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, 1536 kb/s
Stream mapping:
? Stream #0:0 -> #0:0 (rawvideo -> libx264)
? Stream #0:1 -> #0:1 (copy)
Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input

From 99191957 at qq.com  Thu Aug  8 03:32:58 2013
From: 99191957 at qq.com (99191957 at qq.com)
Date: Thu, 8 Aug 2013 09:32:58 +0800
Subject: [FFmpeg-user] How to build a Index of I frames?
Message-ID: <201308080932578907126@qq.com>

I want to make a index of key frames' location of a video, but I found in some kind of H.264 video--not all I frame are Keyframe, for example:
 type ? I , is Key: no. So, i can not judge by packet.flag.   
I tried:

int _find_head(unsigned char *buffer, int len)
{
 int i;
 BOOL isMatch=FALSE;
 for (i=0;i<len;i++){
  if (buffer[i] == 0 && buffer[i+1] == 0 && buffer[i+2] == 0 && buffer[i+3] == 1 && buffer[i+4]==0x67){
   isMatch=TRUE;
   break;
  }
 }

call _find_head(packet.date,packet.size) to find Iframe head, but it does not work either.
So, i'm wordering if anyone could help. Thx. 




99191957 at qq.com
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From 550399108 at qq.com  Thu Aug  8 04:25:19 2013
From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=)
Date: Thu, 8 Aug 2013 10:25:19 +0800
Subject: [FFmpeg-user] How about multi-pass encoding for H264?
Message-ID: <tencent_7A0415656176484179930BCE@qq.com>

Now I use 2-pass X264 encoding to transcode videos to flv format.


Someone told me that 4-pass will reduce the size of the output file.


Is that true?


I searched much material, but found nothing useful!


How can I use 4-pass encoding.


Does any one know?


Thank you !

From lou at lrcd.com  Thu Aug  8 04:40:12 2013
From: lou at lrcd.com (Lou)
Date: Wed, 7 Aug 2013 18:40:12 -0800
Subject: [FFmpeg-user] How about multi-pass encoding for H264?
In-Reply-To: <tencent_7A0415656176484179930BCE@qq.com>
References: <tencent_7A0415656176484179930BCE@qq.com>
Message-ID: <20130807184012.3e2991f0@lrcd.com>

On Thu, 8 Aug 2013 10:25:19 +0800
"???" <550399108 at qq.com> wrote:

> Now I use 2-pass X264 encoding to transcode videos to flv format.

Are you sure you need to perform more than one pass? Normal bitrate
based two-pass mode is generally used if you are targeting a specific
output file size. Otherwise you can use one pass with -crf. For
examples using both methods see:

https://trac.ffmpeg.org/wiki/x264EncodingGuide

> Someone told me that 4-pass will reduce the size of the output file.

Then wouldn't 8-passes be even better?
 
> Is that true?

Anything over two passes is a waste of time with few exceptions. See:

<http://www.mplayerhq.hu/DOCS/HTML/en/menc-feat-x264.html#menc-feat-x264-encoding-options-misc-preferences>

> How can I use 4-pass encoding.

ffmpeg -y -i input -pass 1 ? output
ffmpeg -y -i input -pass 2 ? output
ffmpeg -y -i input -pass 3 ? output
ffmpeg -y -i input -pass 4 ? output
?
ffmpeg -y -i input -pass 256 ? output

From 550399108 at qq.com  Thu Aug  8 05:17:17 2013
From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=)
Date: Thu, 8 Aug 2013 11:17:17 +0800
Subject: [FFmpeg-user] =?gb18030?b?u9i4tKO6ICBIb3cgYWJvdXQgbXVsdGktcGFz?=
 =?gb18030?q?s_encoding_for_H264=3F?=
References: <tencent_7A0415656176484179930BCE@qq.com>
	<20130807184012.3e2991f0@lrcd.com>
Message-ID: <tencent_7A74C9E56AF4A76013198FB4@qq.com>

Thank you Lou.


FFmpeg -pass only has 1 or 2 is valid.


I just want to know about the multi-pass encoding and prove that method!




------------------ ???? ------------------
???: "Lou"<lou at lrcd.com>;
????: 2013?8?8?(???) ??10:40
???: "ffmpeg-user"<ffmpeg-user at ffmpeg.org>; 

??: Re: [FFmpeg-user] How about multi-pass encoding for H264?



On Thu, 8 Aug 2013 10:25:19 +0800
"???" <550399108 at qq.com> wrote:

> Now I use 2-pass X264 encoding to transcode videos to flv format.

Are you sure you need to perform more than one pass? Normal bitrate
based two-pass mode is generally used if you are targeting a specific
output file size. Otherwise you can use one pass with -crf. For
examples using both methods see:

https://trac.ffmpeg.org/wiki/x264EncodingGuide

> Someone told me that 4-pass will reduce the size of the output file.

Then wouldn't 8-passes be even better?
 
> Is that true?

Anything over two passes is a waste of time with few exceptions. See:

<http://www.mplayerhq.hu/DOCS/HTML/en/menc-feat-x264.html#menc-feat-x264-encoding-options-misc-preferences>

> How can I use 4-pass encoding.

ffmpeg -y -i input -pass 1 ? output
ffmpeg -y -i input -pass 2 ? output
ffmpeg -y -i input -pass 3 ? output
ffmpeg -y -i input -pass 4 ? output
?
ffmpeg -y -i input -pass 256 ? output
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From soho123.2012 at gmail.com  Thu Aug  8 08:35:33 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Thu, 8 Aug 2013 14:35:33 +0800
Subject: [FFmpeg-user] how to make ffmpeg with "static link"
Message-ID: <CAFd2cZXtH0QWm-GudPg=s9t30vBCZujjyG9MqPJpE5ReT_NrOA@mail.gmail.com>

Hi All,

Does anyone have idea :
how to make ffmpeg with static link?
I do not like to link share library

Thanks!

From soho123.2012 at gmail.com  Thu Aug  8 09:18:59 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Thu, 8 Aug 2013 15:18:59 +0800
Subject: [FFmpeg-user] how to display configure procedure message when
	ffmpeg do configure
Message-ID: <CAFd2cZWv=gfaxuScx4NxvWLfu462ADQ7sodk4wbeQyrhouFj7w@mail.gmail.com>

Hi All,

Does anyone know :
how to enable display the message when the configure progress ?
I would like to check the progress.

but currently, ffmpeg does not display anything when configure progress.

From nomiya at galaxy.dti.ne.jp  Thu Aug  8 09:25:57 2013
From: nomiya at galaxy.dti.ne.jp (Masaru Nomiya)
Date: Thu, 08 Aug 2013 16:25:57 +0900
Subject: [FFmpeg-user] how to display configure procedure message when
	ffmpeg do configure
In-Reply-To: <CAFd2cZWv=gfaxuScx4NxvWLfu462ADQ7sodk4wbeQyrhouFj7w@mail.gmail.com>
References: <CAFd2cZWv=gfaxuScx4NxvWLfu462ADQ7sodk4wbeQyrhouFj7w@mail.gmail.com>
Message-ID: <871u64prsq.wl%nomiya@galaxy.dti.ne.jp>

Hello,

In the Message; 

  Subject    : [FFmpeg-user] how to display configure procedure message when ffmpeg do configure
  Message-ID : <CAFd2cZWv=gfaxuScx4NxvWLfu462ADQ7sodk4wbeQyrhouFj7w at mail.gmail.com>
  Date & Time: Thu, 8 Aug 2013 15:18:59 +0800

Soho Soho123 <soho123.2012 at gmail.com> has written:

> Hi All,

> Does anyone know :
> how to enable display the message when the configure progress ?
> I would like to check the progress.

> but currently, ffmpeg does not display anything when configure progress.

Have a look at config.log.

Regards,

---
????? Masaru Nomiya             mail-to: nomiya @ galaxy.dti.ne.jp
????
???? "Bill! You married with Computers.
          Not with Me!"
         "No..., with money."

From dashing.meng at gmail.com  Thu Aug  8 09:44:03 2013
From: dashing.meng at gmail.com (littlebat)
Date: Thu, 8 Aug 2013 15:44:03 +0800
Subject: [FFmpeg-user] how to make ffmpeg with "static link"
In-Reply-To: <CAFd2cZXtH0QWm-GudPg=s9t30vBCZujjyG9MqPJpE5ReT_NrOA@mail.gmail.com>
References: <CAFd2cZXtH0QWm-GudPg=s9t30vBCZujjyG9MqPJpE5ReT_NrOA@mail.gmail.com>
Message-ID: <20130808154403.9ebb9bdd.dashing.meng@gmail.com>

On Thu, 8 Aug 2013 14:35:33 +0800
Soho Soho123 <soho123.2012 at gmail.com> wrote:

> Hi All,
> 
> Does anyone have idea :
> how to make ffmpeg with static link?
> I do not like to link share library
> 

It seems has a --disable-shared option when configuring?

From francois.visagie at gmail.com  Thu Aug  8 09:51:33 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Thu, 8 Aug 2013 09:51:33 +0200
Subject: [FFmpeg-user] 2-pass ENcoding
	with	mpeg	(switching	from	mencoder)
In-Reply-To: <1375925155.58796.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<003f01ce9272$25a32920$70e97b60$@gmail.com>	<1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<CAFHbX1+RZuJ87WcapoU5XdMG56DZ9OUaxo-wHj8Gb2urLfYn=A@mail.gmail.com>	<1375886230.35090.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<005101ce9382$3cab3d90$b601b8b0$@gmail.com>
	<1375925155.58796.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <002c01ce940c$1e29fe30$5a7dfa90$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of James Board
> Sent: 08 August 2013 03:26
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from
> mencoder)
> 
> >> What would be an example for doing 2-pass encoding, with the libx264
> 
> >> codec, at a video rate of 10 million bits per second, and merely copy
> >> the audio from input file to output file, and using an mp4 format?
> >
> >ffmpeg -y -i INPUT -an -c:v libx264 -preset veryslow -fastfirstpass 1
> >-f rawvideo -b:v 10000000 -pass 1 NUL ffmpeg -y -I INPUT -c:a copy -c:v
> >libx264 -preset veryslow -b:v 10000000 -pass 2 OUTPUT.mp4
> 
> I tried your commands to do 2-pass encoding with ffmpeg.? The input file
was
> a short 6 second video clip.? Both commands and their output are listed
> below.? The first command seemed seemed okay, but the second command
> failed.
> 
> 
> > ffmpeg -y -i IN.avi -an -c:v libx264 -preset veryslow -fastfirstpass 1
> > -f rawvideo -b:v 10000000 -pass 1 NUL

NUL refers to the null device on Windows. However, earlier in this thread I
seem to recall that you used /dev/null or something to that effect, which
indicates that you use Unix/Linux? I.e. replace NUL in my example command
line as appropriate.

> ffmpeg version N-54733-gab5f581 Copyright (c) 2000-2013 the FFmpeg
> developers
> ? configuration: --enable-libfreetype --enable-gpl --enable-libx264
> ? libavutil????? 52. 39.100 / 52. 39.100
> ? libavcodec???? 55. 18.102 / 55. 18.102
> ? libavformat??? 55. 12.102 / 55. 12.102
> ? libavdevice??? 55.? 3.100 / 55.? 3.100
> ? libavfilter???? 3. 80.101 /? 3. 80.101
> ? libswscale????? 2.? 3.100 /? 2.? 3.100
> ? libswresample?? 0. 17.102 /? 0. 17.102
> ? libpostproc??? 52.? 3.100 / 52.? 3.100 [avi @ 0x2645b00] Stream #0: not
enough
> frames to estimate rate; consider increasing probesize [avi @ 0x2645b00]

With such warnings I'd regard any results as suspicious/unreliable - use a
longer test clip.

> Stream #1: not enough frames to estimate rate; consider increasing
> probesize

Ditto

> Guessed Channel Layout for? Input Stream #0.1 : stereo Input #0,
> avi, from 'IN.avi':
> ? Duration: 00:00:05.74, start: 0.000000, bitrate: 997780 kb/s
> ??? Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1920x1080,
> 29.97 tbr, 29.97 tbn, 29.97 tbc
> ??? Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo,
> s16, 1536 kb/s No pixel format specified, yuv422p for H.264 encoding
chosen.
> Use -pix_fmt yuv420p for compatibility with outdated media players.
> [libx264 @ 0x2649ee0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> [libx264 @ 0x2649ee0] profile High 4:2:2, level 4.0, 4:2:2 8-bit Output
#0,
> rawvideo, to 'NUL':
> ? Metadata:
> ??? encoder???????? : Lavf55.12.102
> ??? Stream #0:0: Video: h264 (libx264), yuv422p, 1920x1080, q=-1--1, pass
1,
> 10000 kb/s, 90k tbn, 29.97 tbc Stream mapping:
> ? Stream #0:0 -> #0:0 (rawvideo -> libx264) Press [q] to stop, [?] for
help
> frame=? 172 fps= 23 q=-1.0 Lsize=??? 7709kB time=00:00:05.67
> bitrate=11133.7kbits/s video:7709kB audio:0kB subtitle:0 global
headers:0kB
> muxing overhead 0.000000%
> [libx264 @ 0x2649ee0] frame I:2???? Avg QP:11.39? size:300209
> [libx264 @ 0x2649ee0] frame P:39??? Avg QP:18.47? size: 83146
> [libx264 @ 0x2649ee0] frame B:131?? Avg QP:19.79? size: 30925
> [libx264 @ 0x2649ee0] consecutive B-frames:? 1.2%? 0.0% 14.0% 41.9% 20.3%
> 17.4%? 0.0%? 0.0%? 5.2%
> [libx264 @ 0x2649ee0] mb I? I16..4: 25.5%? 0.0% 74.5%
> [libx264 @ 0x2649ee0] mb P? I16..4: 39.8%? 0.0%? 0.0%? P16..4:
> 50.6%? 0.0%? 0.0%? 0.0%? 0.0%??? skip: 9.6%
> [libx264 @ 0x2649ee0] mb B? I16..4:? 6.3%? 0.0%? 0.0%? B16..8:
> 37.7%? 0.0%? 0.0%? direct:19.8%? skip:36.2%? L0:38.2% L1:41.8% BI:20.0%
> [libx264 @ 0x2649ee0] final ratefactor: 17.47
> [libx264 @ 0x2649ee0] direct mvs? spatial:90.1% temporal:9.9%
> [libx264 @ 0x2649ee0] coded y,uvDC,uvAC intra: 37.2% 72.0% 37.1% inter:
> 17.7% 33.7% 2.2%
> [libx264 @ 0x2649ee0] i16 v,h,dc,p: 29% 42% 19% 10%
> [libx264 @ 0x2649ee0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 27%
> 10%? 6%? 6%? 7%? 5%? 7%? 5%
> [libx264 @ 0x2649ee0] i8c dc,h,v,p: 42% 25% 24%? 9%
> [libx264 @ 0x2649ee0] Weighted P-Frames: Y:2.6% UV:2.6%
> [libx264 @ 0x2649ee0] kb/s:11004.23
> 
> 
> > ffmpeg -y -i IN.avi -c:a copy -c:v libx264 -preset veryslow -b:v
> > 10000000 -pass 2 OUT.mp4
> ffmpeg version N-54733-gab5f581 Copyright (c) 2000-2013 the FFmpeg
> developers
> ? configuration: --enable-libfreetype --enable-gpl --enable-libx264
> ? libavutil????? 52. 39.100 / 52. 39.100
> ? libavcodec???? 55. 18.102 / 55. 18.102
> ? libavformat??? 55. 12.102 / 55. 12.102
> ? libavdevice??? 55.? 3.100 / 55.? 3.100
> ? libavfilter???? 3. 80.101 /? 3. 80.101
> ? libswscale????? 2.? 3.100 /? 2.? 3.100
> ? libswresample?? 0. 17.102 /? 0. 17.102
> ? libpostproc??? 52.? 3.100 / 52.? 3.100 [avi @ 0x1b2a9c0] Stream #0: not
enough
> frames to estimate rate; consider increasing probesize [avi @ 0x1b2a9c0]
> Stream #1: not enough frames to estimate rate; consider increasing
> probesize Guessed Channel Layout for? Input Stream #0.1 : stereo Input #0,
> avi, from 'IN.avi':
> ? Duration: 00:00:05.74, start: 0.000000, bitrate: 997780 kb/s
> ??? Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1920x1080,
> 29.97 tbr, 29.97 tbn, 29.97 tbc
> ??? Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo,
> s16, 1536 kb/s No pixel format specified, yuv422p for H.264 encoding
chosen.
> Use -pix_fmt yuv420p for compatibility with outdated media players.
> [libx264 @ 0x1b2edc0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> [libx264 @ 0x1b2edc0] profile High 4:2:2, level 5.1, 4:2:2 8-bit
> [libx264 @ 0x1b2edc0] 264 - core 135 r2345 f0c1c53 - H.264/MPEG-4 AVC
> codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options:
> cabac=1
> ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1
> psy_rd=1.00:0.00
> mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0
> deadzone=21,11
> fast_pskip=1 chroma_qp_offset=-2 threads=18 lookahead_threads=3
> sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
> constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3
> weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40
> intra_refresh=0 rc_lookahead=60 rc=2pass mbtree=1 bitrate=10000
> ratetol=1.0
> qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 cplxblur=20.0 qblur=0.5
> ip_ratio=1.40
> aq=1:1.00
> [mp4 @ 0x1b2e500] Tag [1][0][0][0]/0x00000001 incompatible with output
> codec id '65536' ([0][0][0][0]) output #0, mp4, to 'OUT.mp4':

See below.

> ??? Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv422p,
> 1920x1080, q=-1--1, pass 2, 10000 kb/s, 90k tbn, 29.97 tbc
> ??? Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo,
> 1536 kb/s Stream mapping:
> ? Stream #0:0 -> #0:0 (rawvideo -> libx264)
> ? Stream #0:1 -> #0:1 (copy)
> Could not write header for output file #0 (incorrect codec parameters ?):

I stand to be corrected, but I'd suspect that this is related to the message
above, and if so, that the cause might be the PCM audio you're trying to
pack into an MP4 container. I'm not sure if that's supported. See what
happens with '-c:a copy' removed from both command lines; that will
implicitly enable AAC encoding for the MP4 container.

> Invalid data found when processing input
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From francois.visagie at gmail.com  Thu Aug  8 09:59:25 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Thu, 8 Aug 2013 09:59:25 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
In-Reply-To: <20130807180030.GB14079@phare.normalesup.org>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>	<51FEB571.2080701@gmail.com>	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>	<52018A9E.4050604@gmail.com>	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>	<20130807145029.GB25535@phare.normalesup.org>	<005501ce9389$548a5e80$fd9f1b80$@gmail.com>
	<20130807180030.GB14079@phare.normalesup.org>
Message-ID: <002d01ce940d$36f93330$a4eb9990$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Nicolas George
> Sent: 07 August 2013 20:01
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] Preserving perceived loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Le decadi 20 thermidor, an CCXXI, Francois Visagie a ?crit :
> > Is it therefore correct to say that:
> > 	* the only input codec-independent way of downmixing to stereo is
> > ?-ac 2?/?-filter:a aformat=channel_layouts=stereo?/?-filter:a
> > aresample=ocl=3? (which now all behave the same?), and
> 
> Yes. Note that it is always safe to specify both that and "-
> request_channels 2": codecs that do not support it will just ignore the option,
> and if the option is supported, the filters will just do nothing.
> 
> > 	* if one wants to preserve perceived input volume one needs to
> adjust
> > gain during encoding?
> 
> Yes, but if you do that, unless your input was never at peak level, you will get
> clipping, and that is probably worse than low volume.
> 
> > Further to that, for a given energy level per input channel, does the
> > current down-mixing mechanism produce differing output energy levels
> > depending on the _number_ of input channels? I.e. is it expected that
> > different input layouts (with the same energy level per channel) would
> > require different gain factors for equally loud outputs, or will one
> > be able to find a suitable gain factor and use that regardless of
> > number of input channels?
> 
> Well, of course, it depends on the number of input channels. If you want to
> mix one channel into one, you do not need to lower the volume. If you want
> to mix forty-two channels into one, you need to divide the amplitude by
> forty-two to avoid clipping. What it does to energy depends on the input. If
> the channels are in phase, the energy is preserved; if they are not, each the
> energy of each channel is divided by forty-two squared, and then the
> energies are summed, the net result is a division by forty-two.
> 
> lswr uses rather tricky coefficients to match the standard loudness and
> geometry of speakers. You can get the exact matrix with -loglevel debug. For
> example, here is the 5.1 -> stereo matrix:
> 
> 0.414214 0.000000 0.292893 0.000000 0.292893 0.000000
> 0.000000 0.414214 0.292893 0.000000 0.000000 0.292893
> 
> That means: out_left = 0.414 front_left + 0.293 center + 0.293 back_left and
> the symmetrical formula for right; note that LFE is discarded.

Thanks for the confirmations, Nicolas, and many thanks for Andy Furniss' contributions also.

> 
> Regards,
> 
> --
>   Nicolas George


From krueger at lesspain.de  Thu Aug  8 10:04:28 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 8 Aug 2013 10:04:28 +0200
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
In-Reply-To: <20130807152255.GG2578@barisone>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
	<20130807152255.GG2578@barisone>
Message-ID: <CAEnqZEUOJTCfgrT9eFFjjwgfrrqa1sU+XwYR=8chHTT_+KcxXA@mail.gmail.com>

On Wed, Aug 7, 2013 at 5:22 PM, Stefano Sabatini <stefasab at gmail.com> wrote:
> On date Tuesday 2013-08-06 13:09:00 +0200, Robert Kr?ger wrote:
>> Hi,
>>
>> it seems the ffmpeg always decodes the listed types of images to
>> determine their properties. What I did to check this, was to run
>> ffmpeg -i <imagefile> as well as ffprobe -show_streams <imagefile>
>> with each of these formats and set a breakpoint in
>> avcodec_decode_video2.
>>
>> This is not a problem in many cases, however when using ffmpeg,
>> ffprobe or its libraries to extract metadata such as codec, size and
>> pixel format from thousands of image files (imagine analyzing a full
>> SD card of Hires JPG images), the difference between header parsing
>> and decoding becomes very significant (especially with large images
>> for obvious reasons).
>>
>
>> Is this (extracting metadata for those formats just by header parsing)
>> currently implemented and I am just missing a command line argument?
>>
>> If not, shall I submit this as an enhancement request in trac?
>
> Makes sense, possibly mentioning this thread or a short summary.

OK, I did (https://trac.ffmpeg.org/ticket/2845). I hope the
description is sufficient.

From francois.visagie at gmail.com  Thu Aug  8 10:27:42 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Thu, 8 Aug 2013 10:27:42 +0200
Subject: [FFmpeg-user] 2-pass ENcoding with mpeg (switching from
	mencoder)
In-Reply-To: <002c01ce940c$1e29fe30$5a7dfa90$@gmail.com>
References: <1375749670.23934.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<003f01ce9272$25a32920$70e97b60$@gmail.com>
	<1375880208.44635.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAFHbX1+RZuJ87WcapoU5XdMG56DZ9OUaxo-wHj8Gb2urLfYn=A@mail.gmail.com>
	<1375886230.35090.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<005101ce9382$3cab3d90$b601b8b0$@gmail.com>
	<1375925155.58796.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<002c01ce940c$1e29fe30$5a7dfa90$@gmail.com>
Message-ID: <CANp4pj8pcY+TaGQpUr-PcqLyhsV9ya8uBoSpx2nfDhvPv=VeTA@mail.gmail.com>

On 8 August 2013 09:51, Francois Visagie <francois.visagie at gmail.com> wrote:

> See what happens with '-c:a copy' removed from both command lines


That should say 'from the command line', apologies.

From phil_rhodes at rocketmail.com  Thu Aug  8 13:47:48 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Thu, 8 Aug 2013 04:47:48 -0700 (PDT)
Subject: [FFmpeg-user] How about multi-pass encoding for H264?
In-Reply-To: <20130807184012.3e2991f0@lrcd.com>
References: <tencent_7A0415656176484179930BCE@qq.com>
	<20130807184012.3e2991f0@lrcd.com>
Message-ID: <1375962468.80397.YahooMailNeo@web121101.mail.ne1.yahoo.com>

> Someone told me that 4-pass will reduce the size of the output file.


Not so much smaller, but... more accurately constrained to the target bitrate, right?


P

From jpboard2 at yahoo.com  Thu Aug  8 16:30:48 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 8 Aug 2013 07:30:48 -0700 (PDT)
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
	produce JPG images
In-Reply-To: <CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>
Message-ID: <1375972248.27925.YahooMailNeo@web164703.mail.gq1.yahoo.com>

>ffmpeg -i in.avi -c:v ffvhuff -c:a copy out.nut -f image2 out%d.png

That works, and Thank You.? However, when i used jpeg files
as output, the quality of the jpeg file is very poor.? I'd like to
have much better quaility jpeg files, but I don't wnat png files
since they consume too much disk space.
?
Is there a way to increase the quality of the jpeg files that '-f image2;
produces?

From onemda at gmail.com  Thu Aug  8 16:55:00 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 8 Aug 2013 14:55:00 +0000
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
In-Reply-To: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>

On 8/8/13, James Board <jpboard2 at yahoo.com> wrote:
> I'm trying to convert from mencoder to ffmpeg.  I can't
> get the inverse telecine correct.  I did this command
>
>     ffmpeg -i IN.avi -vf fieldmatch,decimate out.avi
>
> But the input file has 33 percent more frames then the output file.
> The input file should have 25 percent more frames, right?  Seems like
> ffmpeg is dropping too many frames.
>
>
> By contrast, this is how I did inverse telecine with mencoder:
>     mencoder -ovc lavc -lavcopts vcodec=ffvhuff -oac copy -vf filmdint -o
> OUT.avi -ofps 24000/1001 IN.avi
>
>
> What's the correct way, or ways, to inverse telecine with ffmpeg?

Perhaps you need to tune fieldmatch filter options?

It's supposed to be better than filmdint from nmplayer.

From jpboard2 at yahoo.com  Thu Aug  8 17:21:00 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 8 Aug 2013 08:21:00 -0700 (PDT)
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
In-Reply-To: <CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>
References: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>
Message-ID: <1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>

>> What's the correct way, or ways, to inverse telecine with ffmpeg?

>
>Perhaps you need to tune fieldmatch filter options?
>
>It's supposed to be better than filmdint from nmplayer.

Well, if anyone knows how to tune fieldmatch so it's better than mencoder,
or even equal to mencoder, please let me know.

Not being able to Inverse Telecine is the prime reason that keeps me from switching from mencoder to ffmpeg.

That, and the fact that I can't figure out 2-pass mpeg4 encoding, but otherwise I would switch to ffempg.

Also, I need to be able to extract higher quality JPEG images from the video stream, but that's it.? Then I'd switch.

From krueger at lesspain.de  Thu Aug  8 18:25:20 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 8 Aug 2013 18:25:20 +0200
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
In-Reply-To: <20130807141526.GA11140@phare.normalesup.org>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
	<loom.20130806T131859-871@post.gmane.org>
	<CAEnqZEWX=b0qeJ3Gh_QRFKh-2cYVWYYFQ-3p_WwwjS=-w_1zsA@mail.gmail.com>
	<CAPYw7P5e8igYc1CKM5Eqn1h-1XfUZaRt0OuRSsyO_hszmQzZDQ@mail.gmail.com>
	<20130807141526.GA11140@phare.normalesup.org>
Message-ID: <CAEnqZEXYxAuOBxdk=bk+is+nMeYngrSSeoPnrd+HFbtWVJJhAg@mail.gmail.com>

On Wed, Aug 7, 2013 at 4:15 PM, Nicolas George
<nicolas.george at normalesup.org> wrote:
> Le decadi 20 thermidor, an CCXXI, Paul B Mahol a ?crit :
>> Yes, it is currently not possible, but I guess one could hack something
>> which will just add some private option to interested decoder(s) to just fetch
>> headers - without actual uncompressing of frames.
>
> That is an interesting idea. It does not even need to be a private option,
> since it makes sense for a lot of codecs, and would be of use for lavf too.
>
> If the feature is implemented in the codec, it just returns immediately when
> it has the format properties, before allocating a frame. If it is not
> implemented, the frame is decoded and then discarded, that wastes time just
> as it does now.
>
> In fact, I half expected AVDISCARD_ALL to have this effect. It seems it does
> not, or I did not test properly.
>
global codec option sounds great. I don't know if that could also then
be used to be able to extract information like frame type without
actually decoding frames which would be super-useful but that is of
course beyond the scope of the original request.

From krueger at lesspain.de  Thu Aug  8 18:28:33 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 8 Aug 2013 18:28:33 +0200
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
In-Reply-To: <1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>
	<1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <CAEnqZEXmc4tTvBbfuPqTc+mfDR3zJLboW4te4a59Jw=5AakEdg@mail.gmail.com>

On Thu, Aug 8, 2013 at 5:21 PM, James Board <jpboard2 at yahoo.com> wrote:
>>> What's the correct way, or ways, to inverse telecine with ffmpeg?
>
>>
>>Perhaps you need to tune fieldmatch filter options?
>>
>>It's supposed to be better than filmdint from nmplayer.
>
> Well, if anyone knows how to tune fieldmatch so it's better than mencoder,
> or even equal to mencoder, please let me know.
>
> Not being able to Inverse Telecine is the prime reason that keeps me from switching from mencoder to ffmpeg.
>
> That, and the fact that I can't figure out 2-pass mpeg4 encoding, but otherwise I would switch to ffempg.
>
> Also, I need to be able to extract higher quality JPEG images from the video stream, but that's it.  Then I'd switch.

just out of curiosity, what's missing there?

From krueger at lesspain.de  Thu Aug  8 18:32:00 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 8 Aug 2013 18:32:00 +0200
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
 produce JPG images
In-Reply-To: <1375972248.27925.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>
	<1375972248.27925.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <CAEnqZEU21YBdh22uQNSsGipqy8JuVTuoEiac0g59+=whZbJS9g@mail.gmail.com>

On Thu, Aug 8, 2013 at 4:30 PM, James Board <jpboard2 at yahoo.com> wrote:
>>ffmpeg -i in.avi -c:v ffvhuff -c:a copy out.nut -f image2 out%d.png
>
> That works, and Thank You.  However, when i used jpeg files
> as output, the quality of the jpeg file is very poor.  I'd like to
> have much better quaility jpeg files, but I don't wnat png files
> since they consume too much disk space.
>
> Is there a way to increase the quality of the jpeg files that '-f image2;
> produces?

ah, sorry. responded to the other thread before reading this. Did you
try adding "-q:v 1 " to your command line to set the jpg quality?

From jpboard2 at yahoo.com  Thu Aug  8 18:44:24 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 8 Aug 2013 09:44:24 -0700 (PDT)
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
	produce JPG images
In-Reply-To: <CAEnqZEU21YBdh22uQNSsGipqy8JuVTuoEiac0g59+=whZbJS9g@mail.gmail.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>	<CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>	<1375972248.27925.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAEnqZEU21YBdh22uQNSsGipqy8JuVTuoEiac0g59+=whZbJS9g@mail.gmail.com>
Message-ID: <1375980264.68257.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>ah, sorry. responded to the other thread before reading this. Did you
>try adding "-q:v 1 " to your command line to set the jpg quality?

No, haven't tried it yet.? Will do.? Thanks (assuming it works).
?
BTW, how do you know these things?? I'd like to answer these
questions on my own, but -q:v isn't in the the ffmpeg documentation
anywhere.? And the image2 muxer section doesn't mention how to
set the JPEG image quality anywhere.

From krueger at lesspain.de  Thu Aug  8 19:21:20 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 8 Aug 2013 19:21:20 +0200
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
 produce JPG images
In-Reply-To: <1375980264.68257.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>
	<1375972248.27925.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAEnqZEU21YBdh22uQNSsGipqy8JuVTuoEiac0g59+=whZbJS9g@mail.gmail.com>
	<1375980264.68257.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAEnqZEXPVsLTpmbycft+BVvDmHeGJ-OmO5QYndMfm14P9vzafw@mail.gmail.com>

On Thu, Aug 8, 2013 at 6:44 PM, James Board <jpboard2 at yahoo.com> wrote:
>>ah, sorry. responded to the other thread before reading this. Did you
>>try adding "-q:v 1 " to your command line to set the jpg quality?
>
> No, haven't tried it yet.  Will do.  Thanks (assuming it works).
>
> BTW, how do you know these things?  I'd like to answer these
> questions on my own, but -q:v isn't in the the ffmpeg documentation
> anywhere.  And the image2 muxer section doesn't mention how to
> set the JPEG image quality anywhere.

it actually is but admittedly a bit hidden in 5.4 "Main Options"

?-q[:stream_specifier] q
(output,per-stream)??-qscale[:stream_specifier] q (output,per-stream)?

Use fixed quality scale (VBR). The meaning of q is codec-dependent.

This is something that would probably be best documented for new users
in a kind of how-to document because it is one of the more important
ones to know but as always, someone must volunteer to write and
maintain such a document. But I would guess that it is in one of the
documents that you find when googling around. Problem with those is
that many of them are outdated or plain wrong :-(. ffmpeg is such a
powerful and fast-evolving tool that writing and maintaining
comprehensive docs is a lot of work but documentation has improved _a
lot_ in the past 2 years.

From jpboard2 at yahoo.com  Thu Aug  8 20:08:30 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 8 Aug 2013 11:08:30 -0700 (PDT)
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
	produce JPG images
In-Reply-To: <CAEnqZEXPVsLTpmbycft+BVvDmHeGJ-OmO5QYndMfm14P9vzafw@mail.gmail.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>	<CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>	<1375972248.27925.YahooMailNeo@web164703.mail.gq1.yahoo.com>	<CAEnqZEU21YBdh22uQNSsGipqy8JuVTuoEiac0g59+=whZbJS9g@mail.gmail.com>	<1375980264.68257.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<CAEnqZEXPVsLTpmbycft+BVvDmHeGJ-OmO5QYndMfm14P9vzafw@mail.gmail.com>
Message-ID: <1375985310.25107.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>

>it actually is but admittedly a bit hidden in 5.4 "Main Options"
>
>?-q[:stream_specifier] q
>(output,per-stream)??-qscale[:stream_specifier] q (output,per-stream)?
>
>Use fixed quality scale (VBR). The meaning of q is codec-dependent.

I'm looking at the "documentation" right now and still don't see how
the "documentation" even remotely implies "-q:v 1" sets the quality
of the jpeg image generation.? As far as I can tell, there's only '-q 1'
in the "documentation", and it merely selects a stream for whatever reason.

From krueger at lesspain.de  Thu Aug  8 20:40:01 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 8 Aug 2013 20:40:01 +0200
Subject: [FFmpeg-user] Convert RAW AVI file to lossless ffvhuff and
 produce JPG images
In-Reply-To: <1375985310.25107.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1375793762.76308.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAFHbX1Jgq23T4TSbUm=eeejWQBxenf3YhYfwrSexrvoYKTcSOw@mail.gmail.com>
	<1375972248.27925.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAEnqZEU21YBdh22uQNSsGipqy8JuVTuoEiac0g59+=whZbJS9g@mail.gmail.com>
	<1375980264.68257.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<CAEnqZEXPVsLTpmbycft+BVvDmHeGJ-OmO5QYndMfm14P9vzafw@mail.gmail.com>
	<1375985310.25107.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAEnqZEVN1jqRHub7rjK+SJcq6m4avkLoWQHRPEwx-RpmTDv0Uw@mail.gmail.com>

On Thu, Aug 8, 2013 at 8:08 PM, James Board <jpboard2 at yahoo.com> wrote:
>>
>
>>it actually is but admittedly a bit hidden in 5.4 "Main Options"
>>
>>?-q[:stream_specifier] q
>>(output,per-stream)??-qscale[:stream_specifier] q (output,per-stream)?
>>
>>Use fixed quality scale (VBR). The meaning of q is codec-dependent.
>
> I'm looking at the "documentation" right now and still don't see how
> the "documentation" even remotely implies "-q:v 1" sets the quality
> of the jpeg image generation.  As far as I can tell, there's only '-q 1'
> in the "documentation", and it merely selects a stream for whatever reason.

the stream specifier selects which stream the option applies to (e.g.
q:v sets the video stream and q:a the audio stream quality). It's a
general concept (see
http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1) and yes, nobody
blames you for not finding the information you were looking for in
that piece of text ;-). "Use fixed quality scale (VBR). The meaning of
q is codec-dependent." does not tell you that in the case of the mjpeg
codec (which is used for writing JPEG images) 1 is the highest
quality. It's simply suboptimal/incomplete. It's done by volunteers in
their spare time and help in improving the docs is definitely greatly
appreciated.

From onemda at gmail.com  Thu Aug  8 21:17:08 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 8 Aug 2013 19:17:08 +0000
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
In-Reply-To: <1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>
	<1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P63UqNhaRWRBeLOaRUaF7myQSn+L3U0NY5cevMqU2Biug@mail.gmail.com>

On 8/8/13, James Board <jpboard2 at yahoo.com> wrote:
>>> What's the correct way, or ways, to inverse telecine with ffmpeg?
>
>>
>>Perhaps you need to tune fieldmatch filter options?
>>
>>It's supposed to be better than filmdint from nmplayer.
>
> Well, if anyone knows how to tune fieldmatch so it's better than mencoder,
> or even equal to mencoder, please let me know.
>
> Not being able to Inverse Telecine is the prime reason that keeps me from
> switching from mencoder to ffmpeg.
>
> That, and the fact that I can't figure out 2-pass mpeg4 encoding, but
> otherwise I would switch to ffempg.
>
> Also, I need to be able to extract higher quality JPEG images from the video
> stream, but that's it.  Then I'd switch.

The info you provide was not enough to provide any detailed help.
How you extract higher quality JPEG images when using nemcoder? Is it
using lavc internal encoder?

If for whatever reason nemcoder works in your specific scenario does not mean
that ffmpeg result is worse.

So far I have not found that fieldmatch + decimate gives poor results
and I assume
its working perfectly.

Otherwise there is tracker for bug reports where bug once reported
have higher chance to be fixed.

From zainmsud at gmail.com  Thu Aug  8 17:22:22 2013
From: zainmsud at gmail.com (zhezus)
Date: Thu, 8 Aug 2013 08:22:22 -0700 (PDT)
Subject: [FFmpeg-user] How to the use drawtext.c in my program
In-Reply-To: <1331033821104-4449535.post@n4.nabble.com>
References: <1331033821104-4449535.post@n4.nabble.com>
Message-ID: <1375975342362-4660578.post@n4.nabble.com>

Have a look at doc/examples/filtering_video.c ... you need to do something
similar.
In the example, they have used the scale filter (look at filter_desc).
Instead, you need to pass the drawtext command (the one you issue after -vf)
... For help on the drawtext command syntax go here:
http://ffmpeg.org/ffmpeg-filters.html#drawtext-1

I would suggest passing the drawtext command without the double quotes ...
just dont have any spaces between the args and you will not be need the
double quotes.

The rest of the code should remain the same ...

Hope this helps. I was stuck on this issue for a while but I have it working
now.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-the-use-drawtext-c-in-my-program-tp4450993p4660578.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From 550399108 at qq.com  Fri Aug  9 04:07:21 2013
From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=)
Date: Fri, 9 Aug 2013 10:07:21 +0800
Subject: [FFmpeg-user] How about multi-pass encoding for H264?
References: <tencent_7A0415656176484179930BCE@qq.com>
	<20130807184012.3e2991f0@lrcd.com>
	<1375962468.80397.YahooMailNeo@web121101.mail.ne1.yahoo.com>
Message-ID: <tencent_15A69087168E129679010E23@qq.com>

Yes?I think so.


But How can I implement the 4-pass encoding?Any one knows?
I want to prove this idea.




------------------ Original ------------------
From:  "Phil Rhodes"<phil_rhodes at rocketmail.com>;
Date:  Thu, Aug 8, 2013 07:47 PM
To:  "FFmpeg user questions"<ffmpeg-user at ffmpeg.org>; 

Subject:  Re: [FFmpeg-user] How about multi-pass encoding for H264?



> Someone told me that 4-pass will reduce the size of the output file.


Not so much smaller, but... more accurately constrained to the target bitrate, right?


P
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
.

From sjh431 at gmail.com  Fri Aug  9 12:03:11 2013
From: sjh431 at gmail.com (jhseo)
Date: Fri, 9 Aug 2013 19:03:11 +0900
Subject: [FFmpeg-user] ftp output problem
Message-ID: <028a01ce94e7$a9cb9210$fd62b630$@gmail.com>

Hi, all

I'm test about ftp output.

When I setting to AVI file format(output), file play is well..
But, mp4 is not play(using VLC).

My option is..
ffmpeg -i inputfile -pass 1 -y -strict experimental -vcodec libx264
-vprofile main -b:v 3072k -s 320x240 -ftp-write-seekable 1
ftp://128.0.0.1/1.mp4

somebody help me T.T


From onemda at gmail.com  Fri Aug  9 12:29:02 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 9 Aug 2013 10:29:02 +0000
Subject: [FFmpeg-user] ftp output problem
In-Reply-To: <028a01ce94e7$a9cb9210$fd62b630$@gmail.com>
References: <028a01ce94e7$a9cb9210$fd62b630$@gmail.com>
Message-ID: <CAPYw7P4MaHOZmb+r4tYBiqDQCVjRFbhdRazitB3d1P=-fZUkhg@mail.gmail.com>

On 8/9/13, jhseo <sjh431 at gmail.com> wrote:
> Hi, all
>
> I'm test about ftp output.
>
> When I setting to AVI file format(output), file play is well..
> But, mp4 is not play(using VLC).
>
> My option is..
> ffmpeg -i inputfile -pass 1 -y -strict experimental -vcodec libx264
> -vprofile main -b:v 3072k -s 320x240 -ftp-write-seekable 1
> ftp://128.0.0.1/1.mp4
>
> somebody help me T.T

What is error?

Perhaps ftp server you are using is not supporing seeking for write?

From onemda at gmail.com  Fri Aug  9 12:51:02 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 9 Aug 2013 10:51:02 +0000
Subject: [FFmpeg-user] Probing jpg,
	png and tiff images without decoding frames
In-Reply-To: <20130807141526.GA11140@phare.normalesup.org>
References: <CAEnqZEWpdFLk6ifMs0vT4hT16pQNGwhxS_MjSsWvr_0qmbS6qw@mail.gmail.com>
	<loom.20130806T131859-871@post.gmane.org>
	<CAEnqZEWX=b0qeJ3Gh_QRFKh-2cYVWYYFQ-3p_WwwjS=-w_1zsA@mail.gmail.com>
	<CAPYw7P5e8igYc1CKM5Eqn1h-1XfUZaRt0OuRSsyO_hszmQzZDQ@mail.gmail.com>
	<20130807141526.GA11140@phare.normalesup.org>
Message-ID: <CAPYw7P7xODxhsuscCkHV2pr2SnLwgjvCVvh5xi00we4jeuo+oQ@mail.gmail.com>

On 8/7/13, Nicolas George <nicolas.george at normalesup.org> wrote:
> Le decadi 20 thermidor, an CCXXI, Paul B Mahol a ecrit :
>> Yes, it is currently not possible, but I guess one could hack something
>> which will just add some private option to interested decoder(s) to just
>> fetch
>> headers - without actual uncompressing of frames.
>
> That is an interesting idea. It does not even need to be a private option,
> since it makes sense for a lot of codecs, and would be of use for lavf too.
>
> If the feature is implemented in the codec, it just returns immediately
> when
> it has the format properties, before allocating a frame. If it is not
> implemented, the frame is decoded and then discarded, that wastes time just
> as it does now.
>
> In fact, I half expected AVDISCARD_ALL to have this effect. It seems it
> does
> not, or I did not test properly.

It does really skip whole decoding, and only for some codecs.
It is unfortunately useless for use in this case as it does not return
frame at all.

Perhaps adding AVDISCARD_DATA or something else would be better?
It would just do what you mentioned: returning AVFrame but with
data/linesize set to 0.

>
> Regards,
>
> --
>   Nicolas George
>

From jpboard2 at yahoo.com  Fri Aug  9 14:46:30 2013
From: jpboard2 at yahoo.com (James Board)
Date: Fri, 9 Aug 2013 05:46:30 -0700 (PDT)
Subject: [FFmpeg-user] Problem with Select Filter
Message-ID: <1376052390.42913.YahooMailNeo@web164702.mail.gq1.yahoo.com>

I start with a raw AVI file that has been Inverse Telecined and single frames
were compressed with lossless codec ffvhuff.? Then I apply the Select filter
to do some cut edits on the video.? An example command is this:

??? ffmpeg -i IN.avi select='-between(n\, 10\, 10000)',setpts=N/FRAME_RATE/TB
??????? -c:v ffvhuff -af aselect='-between(n\, 10\, 10000)',asetpts=N/SR/TB
??????? -c:a wmav2 OUT.avi

Then I later mpeg4 encode it with mencoder (for Carl, I will use ffmpeg
instead of mencoder once I figure out how to do that without error).? Then I
play the file.? The problem is when I seek forward 10 seconds, or a minute,
there's often a 2 or 3 second delay once I get to the seek frame.? The video
freezes for 2 or 3 seconds.

This problem doesn't happen if I omit the above ffmpeg step (where I do Cut
Edits with the Select filter).? So, something isn't right with the way I'm
using ffmpeg to do cut edits, or with the ffmpeg select filter, or whatever.
Can anyone see anything wrong with the above ffmpeg command?? Maybe
the setpts or asetpts stuff, which I don't understand?? Does anyone know
how to figure this problem out?

Thank you.

From jpboard2 at yahoo.com  Fri Aug  9 14:52:59 2013
From: jpboard2 at yahoo.com (James Board)
Date: Fri, 9 Aug 2013 05:52:59 -0700 (PDT)
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
In-Reply-To: <CAPYw7P63UqNhaRWRBeLOaRUaF7myQSn+L3U0NY5cevMqU2Biug@mail.gmail.com>
References: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>	<CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>	<1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P63UqNhaRWRBeLOaRUaF7myQSn+L3U0NY5cevMqU2Biug@mail.gmail.com>
Message-ID: <1376052779.21726.YahooMailNeo@web164706.mail.gq1.yahoo.com>

>The info you provide was not enough to provide any detailed help.

>How you extract higher quality JPEG images when using nemcoder? Is it
>using lavc internal encoder?

Lets ignore the JPEG images for now.? The bigger problem is the ffmpeg Inverse
Telecine is getting rid of good video frames, not only the duplicated frames.

>If for whatever reason nemcoder works in your specific scenario does not mean
>that ffmpeg result is worse.
>
>So far I have not found that fieldmatch + decimate gives poor results
>and I assume
>its working perfectly.

Have you checked the number of frames in your Telecined vs Inverse
Telecined video?? In my case, the ffmpeg Inverse Telecine is dropping
a lot more video frames that the mencoder Inverse Telecine.? This is
a big problem for me, (I don't understand how anyone would think
otherwise).? Can you send me the command you use to 'tune' the Inverse
Telecine with ffmpeg?? Thanks.

From onemda at gmail.com  Fri Aug  9 14:57:09 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 9 Aug 2013 12:57:09 +0000
Subject: [FFmpeg-user] Problem with Select Filter
In-Reply-To: <1376052390.42913.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1376052390.42913.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P5Gz5EtzdgBw=9e61p6Ovsj+c1utGWaTekzdbxWjFiB-A@mail.gmail.com>

On 8/9/13, James Board <jpboard2 at yahoo.com> wrote:
> I start with a raw AVI file that has been Inverse Telecined and single
> frames
> were compressed with lossless codec ffvhuff.  Then I apply the Select
> filter
> to do some cut edits on the video.  An example command is this:
>
>     ffmpeg -i IN.avi select='-between(n\, 10\,
> 10000)',setpts=N/FRAME_RATE/TB
>         -c:v ffvhuff -af aselect='-between(n\, 10\, 10000)',asetpts=N/SR/TB
>         -c:a wmav2 OUT.avi

The above command is missing "-vf", is this true?
>
> Then I later mpeg4 encode it with mencoder (for Carl, I will use ffmpeg
> instead of mencoder once I figure out how to do that without error).  Then
> I
> play the file.  The problem is when I seek forward 10 seconds, or a minute,
> there's often a 2 or 3 second delay once I get to the seek frame.  The
> video
> freezes for 2 or 3 seconds.

Perhaps pts are wrong. For simple cut you can use trim & atrim filters.

>
> This problem doesn't happen if I omit the above ffmpeg step (where I do Cut
> Edits with the Select filter).  So, something isn't right with the way I'm
> using ffmpeg to do cut edits, or with the ffmpeg select filter, or
> whatever.
> Can anyone see anything wrong with the above ffmpeg command?  Maybe
> the setpts or asetpts stuff, which I don't understand?  Does anyone know
> how to figure this problem out?
>
> Thank you.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Fri Aug  9 14:58:54 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 9 Aug 2013 12:58:54 +0000
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
In-Reply-To: <1376052779.21726.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>
	<1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P63UqNhaRWRBeLOaRUaF7myQSn+L3U0NY5cevMqU2Biug@mail.gmail.com>
	<1376052779.21726.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P4LVPr+OzpdV5hKBpp63YE538m_V4c7FgV6wHe6rnnnow@mail.gmail.com>

On 8/9/13, James Board <jpboard2 at yahoo.com> wrote:
>>The info you provide was not enough to provide any detailed help.
>
>>How you extract higher quality JPEG images when using nemcoder? Is it
>>using lavc internal encoder?
>
> Lets ignore the JPEG images for now.  The bigger problem is the ffmpeg
> Inverse
> Telecine is getting rid of good video frames, not only the duplicated
> frames.
>
>>If for whatever reason nemcoder works in your specific scenario does not
>> mean
>>that ffmpeg result is worse.
>>
>>So far I have not found that fieldmatch + decimate gives poor results
>>and I assume
>>its working perfectly.
>
> Have you checked the number of frames in your Telecined vs Inverse
> Telecined video?  In my case, the ffmpeg Inverse Telecine is dropping
> a lot more video frames that the mencoder Inverse Telecine.  This is
> a big problem for me, (I don't understand how anyone would think
> otherwise).  Can you send me the command you use to 'tune' the Inverse
> Telecine with ffmpeg?  Thanks.

Perhaps decimate filter is too aggressive, do yet get correct output
without it?

From jpboard2 at yahoo.com  Fri Aug  9 16:36:33 2013
From: jpboard2 at yahoo.com (James Board)
Date: Fri, 9 Aug 2013 07:36:33 -0700 (PDT)
Subject: [FFmpeg-user] Problem with Select Filter
In-Reply-To: <CAPYw7P5Gz5EtzdgBw=9e61p6Ovsj+c1utGWaTekzdbxWjFiB-A@mail.gmail.com>
References: <1376052390.42913.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<CAPYw7P5Gz5EtzdgBw=9e61p6Ovsj+c1utGWaTekzdbxWjFiB-A@mail.gmail.com>
Message-ID: <1376058993.78542.YahooMailNeo@web164705.mail.gq1.yahoo.com>

>The above command is missing "-vf", is this true?


Yes.? I forgot to type -vf when writing the email.? I did use -vf when running ffmpeg.

>Perhaps pts are wrong. For simple cut you can use trim & atrim filters.

Can I do multiple cuts with trim and atrim?? Can I get rid of multiple
commercials, for example, from the same file?

Also, the problem doesn't occur if I set -noaudio in the mencoder step (again,
for Carl, I'm only using mencoder to do the mpeg4 encoding because my ffmpeg
encoder commands don't work).? When I set -noaudio when doing mencoder,
the file plays fine, and there are no pauses when I skip forward.? So, seems the
problem is with the ffmpeg aselect filter, right?? Am I setting the parameters
correct?? Thank you.

From krueger at lesspain.de  Fri Aug  9 16:37:49 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 9 Aug 2013 16:37:49 +0200
Subject: [FFmpeg-user] Inverse Telecine with ffmpeg
In-Reply-To: <1376052779.21726.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375922125.14649.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P5ET-7C6tUC3gGnQdtSx=cR+fYsKT_8Mo7vjaY4pVE3_Q@mail.gmail.com>
	<1375975260.11146.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P63UqNhaRWRBeLOaRUaF7myQSn+L3U0NY5cevMqU2Biug@mail.gmail.com>
	<1376052779.21726.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <CAEnqZEVa6AswF4X-3ZmOQsBqwWh=oPTVyrEObonyV==BVKVcRw@mail.gmail.com>

On Fri, Aug 9, 2013 at 2:52 PM, James Board <jpboard2 at yahoo.com> wrote:
>>The info you provide was not enough to provide any detailed help.
>
>>How you extract higher quality JPEG images when using nemcoder? Is it
>>using lavc internal encoder?
>
> Lets ignore the JPEG images for now.  The bigger problem is the ffmpeg Inverse
> Telecine is getting rid of good video frames, not only the duplicated frames.
>
>>If for whatever reason nemcoder works in your specific scenario does not mean
>>that ffmpeg result is worse.
>>
>>So far I have not found that fieldmatch + decimate gives poor results
>>and I assume
>>its working perfectly.
>
> Have you checked the number of frames in your Telecined vs Inverse
> Telecined video?  In my case, the ffmpeg Inverse Telecine is dropping
> a lot more video frames that the mencoder Inverse Telecine.  This is
> a big problem for me, (I don't understand how anyone would think
> otherwise).  Can you send me the command you use to 'tune' the Inverse
> Telecine with ffmpeg?  Thanks.

Do you have a small sample where you can reproduce this? That would
probably be best to get this resolved, i.e. open a ticken in trac
(https://trac.ffmpeg.org/) upload files and a description (command
line used in ffmpeg, command line used using mencoder and the
difference in the result, e.g. "frames 12, 16 and 19 are missing in
the ffmpeg output" and what you think would be correct behaviour). If
you get this resolved without this, could you post the solution here?

From jpboard2 at yahoo.com  Fri Aug  9 17:33:05 2013
From: jpboard2 at yahoo.com (James Board)
Date: Fri, 9 Aug 2013 08:33:05 -0700 (PDT)
Subject: [FFmpeg-user] Problems with Audio Codec: almost solved it
Message-ID: <1376062385.6588.YahooMailNeo@web164703.mail.gq1.yahoo.com>

I think I found the problem to both my ffmpeg encoding issues (encodes don't
work) and my playback problems (player pauses when I try to skip forward).

The original AVI files have this audio codec:
??? [pcm] afm: pcm (Uncompressed PCM)

I then run ffmpeg with the select video filters and aselect audio filters
to do cut edits.? During this step, for whatever reason I set the option
"-c:a wmav2", the audio codec becomes
??? [ffwmav2] afm: ffmpeg (DivX audio v2 (FFmpeg))


Files encode successfully and the play successfully with the PCM audio codec,
but not the wmav2 codec.? So I guess my system has problems with the
wmav2 codec, and I want to not use it.

I really don't know anything about audio codecs, and I just want to get my
[damm] app working.? If my original audio codec is uncompressed PCM, what
should I set the "-c:a " option to when doing the cut edits in order to keep
the same audio format, or as close as possible to the same format?? I can't
use 'copy' with the aselect filter.

Thank you

From krueger at lesspain.de  Fri Aug  9 17:46:05 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 9 Aug 2013 17:46:05 +0200
Subject: [FFmpeg-user] Problems with Audio Codec: almost solved it
In-Reply-To: <1376062385.6588.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1376062385.6588.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <CAEnqZEV8oMHi1CvetpVsDBaWid4EpwBAaCsu8wSX7CK-SzcxaQ@mail.gmail.com>

On Fri, Aug 9, 2013 at 5:33 PM, James Board <jpboard2 at yahoo.com> wrote:
> I think I found the problem to both my ffmpeg encoding issues (encodes don't
> work) and my playback problems (player pauses when I try to skip forward).
>
> The original AVI files have this audio codec:
>     [pcm] afm: pcm (Uncompressed PCM)
>
> I then run ffmpeg with the select video filters and aselect audio filters
> to do cut edits.  During this step, for whatever reason I set the option
> "-c:a wmav2", the audio codec becomes
>     [ffwmav2] afm: ffmpeg (DivX audio v2 (FFmpeg))
>
>
> Files encode successfully and the play successfully with the PCM audio codec,
> but not the wmav2 codec.  So I guess my system has problems with the
> wmav2 codec, and I want to not use it.
>
> I really don't know anything about audio codecs, and I just want to get my
> [damm] app working.  If my original audio codec is uncompressed PCM, what
> should I set the "-c:a " option to when doing the cut edits in order to keep
> the same audio format, or as close as possible to the same format?  I can't
> use 'copy' with the aselect filter.
>
> Thank you

What exactly does ffmpeg output when you do ffmpeg -i <your-input-file>?

You probably don't do anything wrong by using pcm_s24le as the audio
codec, which may be a bit of waste if your original pcm-encoded data
uses less than 24bit (32 is rather unlikely, I would say and no human
will hear the difference between 32 and 24), otherwise use pcm_s16le.

From pierrick.grasland at gmail.com  Fri Aug  9 18:02:14 2013
From: pierrick.grasland at gmail.com (Pierrick Grasland)
Date: Fri, 9 Aug 2013 18:02:14 +0200
Subject: [FFmpeg-user] Activating RTCP feedback with FFMpeg for VP8 rtp
Message-ID: <CAB_tPj1+q3FEHfCnAqPissfgZ1MuH07aKHiOihbEr9DosEwVEg@mail.gmail.com>

Hello,

I'm trying to receive a vp8 / G711? rtp stream with FFMpeg.

My SDP is the following :

v=0
o=- 1376063087593 1 IN IP4 127.0.0.1
s=-
t=0 0
m=audio 50008 RTP/AVP 0
c=IN IP4 192.168.2.196
a=rtcp:50009 IN IP4 192.168.2.196
a=rtpmap:0 PCMU/8000
a=sendrecv
m=video 50010 RTP/AVP 120
c=IN IP4 192.168.2.196
a=rtcp:50011 IN IP4 192.168.2.196
a=rtpmap:120 VP8/90000
a=sendrecv
a=rtcp-fb:* nack
a=rtcp-fb:* ccm fir


FFMpeg will correctly start, using the following command : ffmpeg
-reorder_queue_size 64 -analyzeduration 15000000 -probesize 50000000 -i
3.sdp -f webm {1}.webm -ss 00:00:01.000 -f image2 -vframes 1 {1}.jpg ({1}
is replaced by the correct output filename).

ffmpeg version N-51976-g8f3c440 Copyright (c) 2000-2013 the FFmpeg
developers
built on Apr 25 2013 19:24:31 with gcc 4.4.5 (Debian 4.4.5-8)
configuration: --enable-gpl --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-nonfree --enable-version3 --enable-libopus
--enable-libx264
  libavutil      52. 26.100 / 52. 26.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  2.100 / 55.  2.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 53.101 /  3. 53.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100


[sdp @ 0x218b800] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, sdp, from
'/home/pierrick/wsrtc-distribution-0.8.8-ZeWall-r8899/sdp/3.sdp':
  Metadata:
    title           : -
  Duration: 00:00:00.04, start: 0.000000, bitrate: 63 kb/s
    Stream #0:0: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
    Stream #0:1: Video: vp8, yuv420p, 480x640, 90k tbr, 90k tbn, 90k tbc
[auto-inserted scaler 0 @ 0x21f0d00] The <w>:<h>:flags=<flags> option
syntax is deprecated. Use either <w>:<h>:<flags> or
w=<w>:h=<h>:flags=<flags>.
[image2 @ 0x218f880] Frame rate very high for a muxer not efficiently
supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
[libvpx @ 0x21efd00] v1.0.0
Output #0, webm, to 'call_fx_1_3_2013-08-09_17-44-47.webm':
  Metadata:
    title           : -
    encoder         : Lavf55.2.100
    Stream #0:0: Video: vp8, yuv420p, 480x640, q=-1--1, 200 kb/s, 1k tbn,
90k tbc
    Stream #0:1: Audio: vorbis, 8000 Hz, mono, fltp
Output #1, image2, to 'call_fx_1_3_2013-08-09_17-44-47.jpg':
  Metadata:
    title           : -
    encoder         : Lavf55.2.100
    Stream #1:0: Video: mjpeg, yuvj420p, 480x640, q=2-31, 200 kb/s, 90k
tbn, 90k tbc
Stream mapping:
  Stream #0:1 -> #0:0 (vp8 -> libvpx)
  Stream #0:0 -> #0:1 (pcm_mulaw -> libvorbis)
  Stream #0:1 -> #1:0 (vp8 -> mjpeg)
Press [q] to stop, [?] for help

For some minutes (between 30 s and 3 minutes), this works fine, generating
this kind of output :

frame= 3868 fps= 27 q=0.0 q=0.0 size=    4044kB time=00:02:21.86 bitrate=
233.5kbits/s dup=0 drop=3837
frame= 3885 fps= 27 q=0.0 q=0.0 size=    4057kB time=00:02:22.44 bitrate=
233.3kbits/s dup=0 drop=3854
frame= 3900 fps= 27 q=0.0 q=0.0 size=    4067kB time=00:02:22.98 bitrate=
233.0kbits/s dup=0 drop=3869
frame= 3906 fps= 27 q=0.0 q=0.0 size=    4071kB time=00:02:23.53 bitrate=
232.4kbits/s dup=0 drop=3875


But, after a delay, I'm encountering a packet loss (due to using WiFi) and
FFMpeg generate one of the following message :
- Received no start marker; dropping frame
- Missed a picture, sequence broken.

The output will now be this :

frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:24.03 bitrate=
231.6kbits/s dup=0 drop=3876
frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:24.54 bitrate=
230.8kbits/s dup=0 drop=3876
frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:25.05 bitrate=
230.0kbits/s dup=0 drop=3876
frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:25.60 bitrate=
229.1kbits/s dup=0 drop=3876

And my webm file show a freeze at this time, and doesn't appear to restart
correctly after (freeze can be up to several minutes).


Analyzing my network with wireshark will show that my loss are limited to
one or two packets only, but the wrong one.

I found with google that I can use RTCP feedback for VP8, which seems to be
possible with FFMpeg :

http://ffmpeg.org/pipermail/ffmpeg-cvslog/2013-January/059123.html
http://ffmpeg.org/doxygen/trunk/structRTSPStream.html#a9f61d90f147f37ed608f1d151fdcdde8

How can I activate RTCP feedback ? Does this support NACK / PLI / FIR
request ?

Regards,
-- 
Pierrick Grasland

From jpboard2 at yahoo.com  Fri Aug  9 18:48:46 2013
From: jpboard2 at yahoo.com (James Board)
Date: Fri, 9 Aug 2013 09:48:46 -0700 (PDT)
Subject: [FFmpeg-user] Problems with Audio Codec: almost solved it
In-Reply-To: <CAEnqZEV8oMHi1CvetpVsDBaWid4EpwBAaCsu8wSX7CK-SzcxaQ@mail.gmail.com>
References: <1376062385.6588.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAEnqZEV8oMHi1CvetpVsDBaWid4EpwBAaCsu8wSX7CK-SzcxaQ@mail.gmail.com>
Message-ID: <1376066926.81169.YahooMailNeo@web164701.mail.gq1.yahoo.com>

>What exactly does ffmpeg output when you do ffmpeg -i <your-input-file>?


For the audio code, it displays: audio pcm_s16le 48000 Hz s1g, 1536 kb/s.

I guess that means I should use '-c:a pcm_s16le'.

Is there an ffmpeg option that tells it to use whatever audio codec
that the input file used?

Alternatively, if I want to compress the audio, what is a good suggestion
for an audio codec that is most widely supported?? I want to ensure it
works with mencoder, mplayer, and whatever tool I might use in the
future.? What's the best bet?? '-c:a mp2'?? I use Linux machines almost
exclusively.

From ricardohenrylee at gmail.com  Fri Aug  9 20:20:51 2013
From: ricardohenrylee at gmail.com (Richard H Lee)
Date: Fri, 09 Aug 2013 19:20:51 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
Message-ID: <52053303.7070107@gmail.com>

I use ffprobe -show_frames to find out the exact frame rate of videos 
that stray away from 24000/1001 fps before conversion to 25 fps so that 
I can get accurate a/v sync when I speed up the audio.

I basically use pkt_pts_time / coded_picture_number of the very last 
frame for the overall frame rate.

The problem is that on large files, it can take quite a while to dump 
all the frame data.

Is it possible to just dump the last few frames / seconds of the file?
Or would this affect coded_picture_number? (I'm not sure if 
coded_picture_number is counted by keeping track of frames/packets or is 
stored within the packet itself.)

From jiachielee at live.com  Fri Aug  9 20:48:33 2013
From: jiachielee at live.com (thljcl)
Date: Fri, 9 Aug 2013 11:48:33 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <52053303.7070107@gmail.com>
References: <52053303.7070107@gmail.com>
Message-ID: <1376074113012-4660603.post@n4.nabble.com>

The frame rate of your video source is 24000/1001 FPS. You want to convert it
to 25 FPS and still make sure that the audio is in sync with the video? Is
that what you want?



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660603.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From ricardohenrylee at gmail.com  Sat Aug 10 01:06:44 2013
From: ricardohenrylee at gmail.com (Richard H Lee)
Date: Sat, 10 Aug 2013 00:06:44 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376074113012-4660603.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
Message-ID: <52057604.30206@gmail.com>

On 09/08/2013 19:48, thljcl wrote:
> The frame rate of your video source is 24000/1001 FPS. You want to convert it
> to 25 FPS and still make sure that the audio is in sync with the video? Is
> that what you want?

No, I don't have any problems with conversion and synchronization at 
24000/1001 fps. You can just speed up the audio by 1001/960 to get it to 
match 25 fps.

The problem is that many media is not exactly 24000/1001 fps . Thus to 
find the exact frame rate run ffprobe -show_frames on the file, and 
divide the coded_picture_number by the pkt_pts_time of the last frame.

The thing is that ffprobe -show_frames can take quite a while especially 
on a slower machine as it has to go through every frame.

This leads me to my question: it is possible to save time by getting the 
coded_picture_number and the pkt_pts_time fields of the last frame 
without having to run through all the other preceding frames?

From pradhan.pushkar at gmail.com  Sat Aug 10 02:14:51 2013
From: pradhan.pushkar at gmail.com (Pushkar Pradhan)
Date: Fri, 9 Aug 2013 17:14:51 -0700
Subject: [FFmpeg-user] ffmpeg and monotonically increasing DTS
Message-ID: <CACgtWZP7O25ZTw_5NSKVEq9MVeYyXN4HYK7x97EpxHqk38au_Q@mail.gmail.com>

I have a mp4 where some of the video frames have the same DTS/PTS and
ffmpeg refuses to copy it when I try to extract the first few seconds:
$ ffmpeg -i full_file.mp4  -vcodec copy -acodec copy -t 45 short_45s.mp4
ffmpeg version git-2012-04-04-41a097a Copyright (c) 2000-2012 the FFmpeg
developers
  built on Apr  3 2012 18:08:31 with gcc 4.4.3
  configuration: --enable-gpl --enable-libfaac --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora
--enable-libvorbis --enable-libx264 --enable-nonfree --enable-version3
--enable-x11grab --enable-libvpx
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'full_file.mp4':
Stream mapping:
  Stream #0:1 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
[mp4 @ 0x24b98a0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 119974 >= 119974
av_interleaved_write_frame(): Invalid argument

However, when I take out the -acodec copy and -vcodec copy, i.e. when
ffmpeg reencodes it, it does not abort.

So my question is: Is it invalid according to the MP4 spec to have frames
with the same DTS/PTS for a video track?
The error definitely makes sense.
Thanks,

-- 
pushkar

From jiachielee at live.com  Sat Aug 10 05:43:29 2013
From: jiachielee at live.com (thljcl)
Date: Fri, 9 Aug 2013 20:43:29 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <52057604.30206@gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
Message-ID: <1376106209176-4660606.post@n4.nabble.com>

What you are saying is that some sources do not have constant frame rate of
24000/1001 FPS despite having the ?average frame rate? close to 24000/1001
FPS, thus speeding up the audio would cause the video and audio to be out of
sync.
The truth is, to me, regardless of the original frame rate of the source, I
do not need to speed up or slow down the audio in order to keep the audio
and video in sync. I do understand that speeding up the audio is the
standard practice being used by television stations in PAL regions to keep
the audio and video in sync from the source of 24 FPS. Such practice makes
sense before digital age; without re-encoding the video/audio, speeding up
the audio would keep video and audio in sync. With ffmpeg, however, there is
no need to change the playback speed to keep the audio and video in sync.
Let me briefly explains how the video filter ?fps? works.
Basically, with -vf "fps=25" as an output option, you can change the frame
rate to constant frame rate of 25 FPS; ffmpeg would duplicate frames if the
source is of lower frame rate; ffmpeg would drop frames if the source is of
higher frame rates. In fact, that?s how I create a slideshow which consists
a lot of duplicated frames (slides) as I wrote at
http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-td4660277.html.
You can read those posts as reference. By duplicating and dropping frames,
there would be minimal change of playback speed, subject to the round-off
error. In practice, there would be no need for the change of audio speed or
to actually know how the frame rate of the source varies over time.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660606.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From sjh431 at gmail.com  Sat Aug 10 07:17:33 2013
From: sjh431 at gmail.com (jhseo)
Date: Sat, 10 Aug 2013 14:17:33 +0900
Subject: [FFmpeg-user] ftp output problem
In-Reply-To: <CAPYw7P4MaHOZmb+r4tYBiqDQCVjRFbhdRazitB3d1P=-fZUkhg@mail.gmail.com>
References: <028a01ce94e7$a9cb9210$fd62b630$@gmail.com>
	<CAPYw7P4MaHOZmb+r4tYBiqDQCVjRFbhdRazitB3d1P=-fZUkhg@mail.gmail.com>
Message-ID: <029801ce9588$ecd3fdd0$c67bf970$@gmail.com>

Thanks for reply.

The output to ftp-server has less than local(hard-disk) output (in case mp4
extension)

So I compare two files using WinMerge, and I can't find out moov box in ftp
output file..

Is this case "ftp server that is not supporting seeking for write"?

And, when the "ftp-write-seekable" option is off, "[mp4 @ 0000000002887de0]
muxer does not support non seekable output" is print out..

Please comment one more.

-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org]
On Behalf Of Paul B Mahol
Sent: Friday, August 09, 2013 7:29 PM
To: FFmpeg user questions
Subject: Re: [FFmpeg-user] ftp output problem

On 8/9/13, jhseo <sjh431 at gmail.com> wrote:
> Hi, all
>
> I'm test about ftp output.
>
> When I setting to AVI file format(output), file play is well..
> But, mp4 is not play(using VLC).
>
> My option is..
> ffmpeg -i inputfile -pass 1 -y -strict experimental -vcodec libx264 
> -vprofile main -b:v 3072k -s 320x240 -ftp-write-seekable 1
> ftp://128.0.0.1/1.mp4
>
> somebody help me T.T

What is error?

Perhaps ftp server you are using is not supporing seeking for write?
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From gavr.mail at gmail.com  Sat Aug 10 09:31:56 2013
From: gavr.mail at gmail.com (Kirill Gavrilov)
Date: Sat, 10 Aug 2013 11:31:56 +0400
Subject: [FFmpeg-user] some mp3 files can not be decoded by recent FFmpeg
Message-ID: <CAJQwZb33FAjiCR5VvJC1a1LFTmqxEyu5oWhqPg0YpyU0g8BhVQ@mail.gmail.com>

Hi all!

After migration to FFmpeg 2.0 I have observed that several old mp3 files (9
from 100) can not be played anymore with the same error - no 'data' tag
found:
./ffmpeg -i anb3.mp3
ffmpeg version N-55410-gd4db7c3 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 10 2013 10:58:01 with gcc 4.2.1 (GCC) (Apple Inc. build
5666) (dot 3)
  configuration: --enable-swscale --enable-shared --disable-static
--enable-memalign-hack --enable-avfilter --enable-hardcoded-tables
--enable-pthreads --disable-libopenjpeg --enable-runtime-cpudetect
--enable-vda --disable-debug --enable-version3
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 24.100 / 55. 24.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
[wav @ 0x10180e820] Format wav probed with size=2048 and score=99
[wav @ 0x10180e820] id3v2 ver:2 flags:00 len:85
[wav @ 0x10180e820] no 'data' tag found
[AVIOContext @ 0x101111260] Statistics: 32768 bytes read, 0 seeks
anb3.mp3: Invalid data found when processing input

There no such issues with older release 0.10.2:
[wav @ 0x10100e820] max_analyze_duration 5000000 reached at 5015510
Input #0, wav, from 'anb3.mp3':
  Duration: 00:03:11.82, bitrate: 12 kb/s
    Stream #0:0: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, stereo, s16,
128 kb/s

I may suppose that these files were encoded with broken encoder but it is
weird that only recent FFmpeg can not play them now...
Could somebody tell me is it regression or new "feature"?

I have prepared 300 KB cut from one sample:
http://sview.cifro-city.ru/trash/anb3.mp3
-----------------------------------------------
Kirill Gavrilov,
Software designer.

From krueger at lesspain.de  Sat Aug 10 09:50:45 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Sat, 10 Aug 2013 09:50:45 +0200
Subject: [FFmpeg-user] Problems with Audio Codec: almost solved it
In-Reply-To: <1376066926.81169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
References: <1376062385.6588.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAEnqZEV8oMHi1CvetpVsDBaWid4EpwBAaCsu8wSX7CK-SzcxaQ@mail.gmail.com>
	<1376066926.81169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <CAEnqZEWXqeE8qJn176cyV1SGdm14v87KYeczcr99vzQRbBsVhQ@mail.gmail.com>

On Fri, Aug 9, 2013 at 6:48 PM, James Board <jpboard2 at yahoo.com> wrote:
>>What exactly does ffmpeg output when you do ffmpeg -i <your-input-file>?
>
>
> For the audio code, it displays: audio pcm_s16le 48000 Hz s1g, 1536 kb/s.
>
> I guess that means I should use '-c:a pcm_s16le'.

exactly

>
> Is there an ffmpeg option that tells it to use whatever audio codec
> that the input file used?

no, not that I know of.

>
> Alternatively, if I want to compress the audio, what is a good suggestion
> for an audio codec that is most widely supported?  I want to ensure it
> works with mencoder, mplayer, and whatever tool I might use in the
> future.  What's the best bet?  '-c:a mp2'?  I use Linux machines almost
> exclusively.

what container format will you use for the files? That might constrain
your choices although mp2 should be rather conservative.

From krueger at lesspain.de  Sat Aug 10 09:58:06 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Sat, 10 Aug 2013 09:58:06 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376106209176-4660606.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
Message-ID: <CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>

On Sat, Aug 10, 2013 at 5:43 AM, thljcl <jiachielee at live.com> wrote:
> What you are saying is that some sources do not have constant frame rate of
> 24000/1001 FPS despite having the ?average frame rate? close to 24000/1001
> FPS, thus speeding up the audio would cause the video and audio to be out of
> sync.
> The truth is, to me, regardless of the original frame rate of the source, I
> do not need to speed up or slow down the audio in order to keep the audio
> and video in sync. I do understand that speeding up the audio is the
> standard practice being used by television stations in PAL regions to keep
> the audio and video in sync from the source of 24 FPS. Such practice makes
> sense before digital age; without re-encoding the video/audio, speeding up
> the audio would keep video and audio in sync. With ffmpeg, however, there is
> no need to change the playback speed to keep the audio and video in sync.
> Let me briefly explains how the video filter ?fps? works.
> Basically, with -vf "fps=25" as an output option, you can change the frame
> rate to constant frame rate of 25 FPS; ffmpeg would duplicate frames if the
> source is of lower frame rate; ffmpeg would drop frames if the source is of
> higher frame rates. In fact, that?s how I create a slideshow which consists
> a lot of duplicated frames (slides) as I wrote at
> http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-td4660277.html.
> You can read those posts as reference. By duplicating and dropping frames,
> there would be minimal change of playback speed, subject to the round-off
> error. In practice, there would be no need for the change of audio speed or
> to actually know how the frame rate of the source varies over time.
>

this is bad advice if your input has more or less constant frame rate
and you care about smooth motion as you will have visible degradation
of perceived quality when changing frame rates that way.

From nicolas.george at normalesup.org  Sat Aug 10 10:05:25 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sat, 10 Aug 2013 10:05:25 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376106209176-4660606.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
Message-ID: <20130810080525.GA545@phare.normalesup.org>

Le duodi 22 thermidor, an CCXXI, thljcl a ?crit?:
>							 Such practice makes
> sense before digital age; without re-encoding the video/audio, speeding up
> the audio would keep video and audio in sync. With ffmpeg, however, there is
> no need to change the playback speed to keep the audio and video in sync.
> Let me briefly explains how the video filter ?fps? works.

You obviously never watched a DVD that was converted from NTSC to PAL that
way, or you would not dare make such a statement: they just look completely
ugly.

To Richard H Lee: if you follow that advice, you will get a video that has a
hiccup every second, very noticeable in regular motion scenes such as
tracking shots. A good way to give your audience a headache.

Also, this apply both to analog and digital, it makes no difference with
regard to frame rate.

Regards,

-- 
  Nicolas George
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From al-ffmpeguser at none.at  Sat Aug 10 12:44:55 2013
From: al-ffmpeguser at none.at (Aleksandar Lazic)
Date: Sat, 10 Aug 2013 12:44:55 +0200
Subject: [FFmpeg-user] Segfault with  prores and 15696x2048 input images
Message-ID: <3d3bd619754f43e21e584ceea394724b@none.at>

Dear listmembers,

today I got a segfault with the following command line.

ffmpeg -i img%03d.jpg -y -c:v prores -c:a null -r 25 -an /tmp/output.mov

The original Imagessizes are.

# identify img284.jpg
img284.jpg JPEG 15696x2048 15696x2048+0+0 8-bit DirectClass 21.98MB 
0.000u 0:00.000

I have created the sequence like that

x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" 
img"$counter".jpg; x=$(($x+1)); done

I plan to use a 30kx2K as final input size, is this with the prores 
encoder possible?
Maybe there is a better encoder for such a Movie size.

###
ffmpeg  -i img%03d.jpg -y -c:v prores -c:a null -r 25 -an 
/tmp/output.mov
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
   built on Aug 10 2013 12:33:35 with gcc 4.6 (Ubuntu/Linaro 
4.6.3-1ubuntu5)
   configuration: --prefix=/root/progs/ffmpeg --disable-stripping 
--enable-debug=2
   libavutil      52. 38.100 / 52. 38.100
   libavcodec     55. 18.102 / 55. 18.102
   libavformat    55. 12.100 / 55. 12.100
   libavdevice    55.  3.100 / 55.  3.100
   libavfilter     3. 79.101 /  3. 79.101
   libswscale      2.  3.100 /  2.  3.100
   libswresample   0. 17.102 /  0. 17.102
[image2 @ 0x2577860] Stream #0: not enough frames to estimate rate; 
consider increasing probesize
Input #0, image2, from 'img%03d.jpg':
   Duration: 00:00:27.20, start: 0.000000, bitrate: N/A
     Stream #0:0: Video: mjpeg, yuvj420p, 15696x2048 [SAR 72:72 DAR 
981:128], 25 tbr, 25 tbn, 25 tbc
[prores @ 0x2562b20] encoding with ProRes standard (apcn) profile
[prores @ 0x2607600] encoding with ProRes standard (apcn) profile
[prores @ 0x2605d80] encoding with ProRes standard (apcn) profile
[prores @ 0x26065c0] encoding with ProRes standard (apcn) profile
[prores @ 0x2713d20] encoding with ProRes standard (apcn) profile
[prores @ 0x2873e00] encoding with ProRes standard (apcn) profile
[prores @ 0x2874b20] encoding with ProRes standard (apcn) profile
[prores @ 0x28186e0] encoding with ProRes standard (apcn) profile
[prores @ 0x281af80] encoding with ProRes standard (apcn) profile
[prores @ 0x281dce0] encoding with ProRes standard (apcn) profile
[prores @ 0x2820a40] encoding with ProRes standard (apcn) profile
[prores @ 0x28b50a0] encoding with ProRes standard (apcn) profile
[prores @ 0x28b7b00] encoding with ProRes standard (apcn) profile
[prores @ 0x28ba6a0] encoding with ProRes standard (apcn) profile
[prores @ 0x28bd300] encoding with ProRes standard (apcn) profile
[prores @ 0x28bfea0] encoding with ProRes standard (apcn) profile
[prores @ 0x28c2ae0] encoding with ProRes standard (apcn) profile
[prores @ 0x28c5720] encoding with ProRes standard (apcn) profile
[prores @ 0x28c8360] encoding with ProRes standard (apcn) profile
[prores @ 0x28cafa0] encoding with ProRes standard (apcn) profile
[prores @ 0x28cdbe0] encoding with ProRes standard (apcn) profile
[prores @ 0x28d0820] encoding with ProRes standard (apcn) profile
[prores @ 0x28d3460] encoding with ProRes standard (apcn) profile
[prores @ 0x28d60a0] encoding with ProRes standard (apcn) profile
[prores @ 0x25797c0] encoding with ProRes standard (apcn) profile
Output #0, mov, to '/tmp/output.mov':
   Metadata:
     encoder         : Lavf55.12.100
     Stream #0:0: Video: prores (apcn) (apcn / 0x6E637061), yuv422p10le, 
15696x2048 [SAR 1:1 DAR 981:128], q=2-31, 200 kb/s, 12800 tbn, 25 tbc
Stream mapping:
   Stream #0:0 -> #0:0 (mjpeg -> prores)
Press [q] to stop, [?] for help
Segmentation fault (core dumped)

####

This is the gdb bt output.
####

root at ns61620:/home/klaus/original# gdb /root/progs/ffmpeg/bin/ffmpeg 
core
GNU gdb (Ubuntu/Linaro 7.4-2012.04-0ubuntu2.1) 7.4-2012.04
Copyright (C) 2012 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later 
<http://gnu.org/licenses/gpl.html>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show 
copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu".
For bug reporting instructions, please see:
<http://bugs.launchpad.net/gdb-linaro/>...
Reading symbols from /root/progs/ffmpeg/bin/ffmpeg...done.
[New LWP 19580]
[New LWP 19581]
[New LWP 19566]
[New LWP 19583]
[New LWP 19582]
[New LWP 19585]
[New LWP 19591]
[New LWP 19584]
[New LWP 19589]
[New LWP 19592]
[New LWP 19554]
[New LWP 19602]
[New LWP 19590]
[New LWP 19556]
[New LWP 19572]
[New LWP 19588]
[New LWP 19594]
[New LWP 19569]
[New LWP 19586]
[New LWP 19587]
[New LWP 19576]
[New LWP 19595]
[New LWP 19558]
[New LWP 19577]
[New LWP 19573]
[New LWP 19578]
[New LWP 19575]
[New LWP 19557]
[New LWP 19570]
[New LWP 19603]
[New LWP 19567]
[New LWP 19601]
[New LWP 19599]
[New LWP 19564]
[New LWP 19600]
[New LWP 19598]
[New LWP 19561]
[New LWP 19579]
[New LWP 19597]
[New LWP 19555]
[New LWP 19568]
[New LWP 19593]
[New LWP 19563]
[New LWP 19596]
[New LWP 19571]
[New LWP 19562]
[New LWP 19574]
[New LWP 19560]
[New LWP 19565]
[New LWP 19559]

warning: Can't read pathname for load map: Input/output error.
[Thread debugging using libthread_db enabled]
Using host libthread_db library 
"/lib/x86_64-linux-gnu/libthread_db.so.1".
Core was generated by `ffmpeg -i img%03d.jpg -y -c:v prores -c:a null -r 
25 -an /tmp/output.mov'.
Program terminated with signal 11, Segmentation fault.
#0  0x000000000089e71d in put_bits (value=2, n=3, s=0x7f5dcc286ca0) at 
libavcodec/put_bits.h:169
169	        AV_WB32(s->buf_ptr, bit_buf);

(gdb) bt
#0  0x000000000089e71d in put_bits (value=2, n=3, s=0x7f5dcc286ca0) at 
libavcodec/put_bits.h:169
#1  encode_codeword (pb=0x7f5dcc286ca0, val=<optimized out>, 
codebook=<optimized out>) at libavcodec/proresenc_anatoliy.c:179
#2  0x000000000089f492 in encode_dc_coeffs (qmat=0x2711918, 
blocks_per_slice=32, in=0x7f5dcc285ca0, pb=0x7f5dcc286ca0)
     at libavcodec/proresenc_anatoliy.c:220
#3  encode_slice_plane (mb_count=<optimized out>, src=<optimized out>, 
src_stride=<optimized out>, buf=0x0, buf_size=514359626,
     qmat=0x2711918, chroma=0, avctx=<optimized out>) at 
libavcodec/proresenc_anatoliy.c:309
#4  0x000000000089fabe in encode_slice_data (qp=1, 
v_data_size=<synthetic pointer>, u_data_size=<synthetic pointer>,
     y_data_size=<synthetic pointer>, data_size=514359626, 
buf=0x7f5d33870cea "", mb_count=8, chroma_stride=15728, 
luma_stride=31456,
     dest_v=<optimized out>, dest_u=<optimized out>, dest_y=<optimized 
out>, avctx=0x2562b20) at libavcodec/proresenc_anatoliy.c:324
#5  encode_slice (qp=<synthetic pointer>, unsafe=<optimized out>, 
data_size=514359632, buf=0x7f5d33870ce4 "", mb_count=8, mb_y=0,
     mb_x=<optimized out>, pic=0x28dffc0, avctx=0x2562b20) at 
libavcodec/proresenc_anatoliy.c:408
#6  prores_encode_picture (buf_size=514359640, buf=0x7f5d338690dc "", 
pic=<optimized out>, avctx=0x2562b20)
     at libavcodec/proresenc_anatoliy.c:471
#7  prores_encode_frame (got_packet=0x7f5dcc286eac, pict=<optimized 
out>, pkt=0x7f5dc40008c0, avctx=0x2562b20)
     at libavcodec/proresenc_anatoliy.c:502
#8  prores_encode_frame (avctx=0x2562b20, pkt=0x7f5dc40008c0, 
pict=<optimized out>, got_packet=0x7f5dcc286eac)
     at libavcodec/proresenc_anatoliy.c:489
#9  0x0000000000934413 in avcodec_encode_video2 (avctx=0x2562b20, 
avpkt=0x7f5dc40008c0, frame=0x28dffc0,
     got_packet_ptr=0x7f5dcc286eac) at libavcodec/utils.c:1756
#10 0x0000000000a4ae86 in worker (v=<optimized out>) at 
libavcodec/frame_thread_encoder.c:93
#11 0x00007f5dda244e9a in start_thread () from 
/lib/x86_64-linux-gnu/libpthread.so.0
#12 0x00007f5dd9f71ccd in clone () from /lib/x86_64-linux-gnu/libc.so.6
#13 0x0000000000000000 in ?? ()
(gdb)
###

System:

lsb_release -a
No LSB modules are available.
Distributor ID:	Ubuntu
Description:	Ubuntu 12.04.2 LTS
Release:	12.04
Codename:	precise

dpkg -l yasm
ii  yasm                        1.1.0-1                     modular 
assembler with multiple syntaxes support

Thanks for any help.

Best regards
Aleks

From avetisyan.andranik1992 at mail.ru  Sat Aug 10 14:20:42 2013
From: avetisyan.andranik1992 at mail.ru (=?UTF-8?B?QW5kcmFuaWsgQXZldGlzeWFu?=)
Date: Sat, 10 Aug 2013 16:20:42 +0400
Subject: [FFmpeg-user] =?utf-8?q?Question?=
Message-ID: <1376137242.292511594@f268.i.mail.ru>



Hello

How can i get .avi videos duration from ffmpeg ?in c++
-- 
Andranik Avetisyan

From phil_rhodes at rocketmail.com  Sat Aug 10 14:37:23 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Sat, 10 Aug 2013 05:37:23 -0700 (PDT)
Subject: [FFmpeg-user] =?utf-8?b?77u/UXVlc3Rpb24=?=
In-Reply-To: <1376137242.292511594@f268.i.mail.ru>
References: <1376137242.292511594@f268.i.mail.ru>
Message-ID: <1376138243.11662.YahooMailNeo@web121101.mail.ne1.yahoo.com>

I've no idea, but bear in mind that getting the duration of an AVI file may not be easy or quick in any case. Some AVIs have a frame count in the headers, but many don't, and some that do are inaccurate. If you're always working with files produced by a known application, and you know that application always writes accurate frame counts, there may be a way to do it. If not, the only accurate way is to scan through the entire file frame by frame, which can be very slow on long files.

That's quite apart from how you'd actually get libavformat (presumably the format library, for just a frame count?) to actually do it.

P


________________________________
 From: Andranik Avetisyan <avetisyan.andranik1992 at mail.ru>
To: ffmpeg-user at ffmpeg.org 
Sent: Saturday, 10 August 2013, 13:20
Subject: [FFmpeg-user] ?Question
 



Hello

How can i get .avi videos duration from ffmpeg ?in c++
-- 
Andranik Avetisyan
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From onemda at gmail.com  Sat Aug 10 14:48:08 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 10 Aug 2013 12:48:08 +0000
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <52053303.7070107@gmail.com>
References: <52053303.7070107@gmail.com>
Message-ID: <CAPYw7P5AL1e0rv8am0fU_32YeORL+a4=C0v+dRZomaDqkok=xA@mail.gmail.com>

On 8/9/13, Richard H Lee <ricardohenrylee at gmail.com> wrote:
> I use ffprobe -show_frames to find out the exact frame rate of videos
> that stray away from 24000/1001 fps before conversion to 25 fps so that
> I can get accurate a/v sync when I speed up the audio.
>
> I basically use pkt_pts_time / coded_picture_number of the very last
> frame for the overall frame rate.
>
> The problem is that on large files, it can take quite a while to dump
> all the frame data.

Mainly because you need to decode whole file (at least currently).

>
> Is it possible to just dump the last few frames / seconds of the file?

Yes, by seeking to such location.

> Or would this affect coded_picture_number? (I'm not sure if
> coded_picture_number is counted by keeping track of frames/packets or is
> stored within the packet itself.)

It looks like coded_picture_number is increased by one when decoding
(and only for few codecs).
So its not stored in bitstream, and skiping decoding would change it.

You can try this anyway and report outcome.

From onemda at gmail.com  Sat Aug 10 14:57:49 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 10 Aug 2013 12:57:49 +0000
Subject: [FFmpeg-user] ftp output problem
In-Reply-To: <029801ce9588$ecd3fdd0$c67bf970$@gmail.com>
References: <028a01ce94e7$a9cb9210$fd62b630$@gmail.com>
	<CAPYw7P4MaHOZmb+r4tYBiqDQCVjRFbhdRazitB3d1P=-fZUkhg@mail.gmail.com>
	<029801ce9588$ecd3fdd0$c67bf970$@gmail.com>
Message-ID: <CAPYw7P607Y1_cJCC80hegsnirm0=JSjVw9SqaO64egAGiRh9+Q@mail.gmail.com>

On 8/10/13, jhseo <sjh431 at gmail.com> wrote:
> Thanks for reply.
>
> The output to ftp-server has less than local(hard-disk) output (in case mp4
> extension)
>
> So I compare two files using WinMerge, and I can't find out moov box in ftp
> output file..
>
> Is this case "ftp server that is not supporting seeking for write"?
>
> And, when the "ftp-write-seekable" option is off, "[mp4 @ 0000000002887de0]
> muxer does not support non seekable output" is print out..
>
> Please comment one more.

Unfortunately I have no more comments.
As you found out using ftp-write-seekable does not work correctly for
your ftp server,
e.g. file is badly written.
Note I'm not the one how added ftp protocol support, neither I ever
tried to send files using ffmpeg ftp protocol (seekable or not).

Obviosly disabling ftp-write-seekable will not work at all for mp4 as such
muxer needs seekable output.

Maybe there is bug in ftp protocol implementation in FFmpeg or ftp
server you use
does not support seeking. I'm just guessing from info you gave me.

>
> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org
> [mailto:ffmpeg-user-bounces at ffmpeg.org]
> On Behalf Of Paul B Mahol
> Sent: Friday, August 09, 2013 7:29 PM
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] ftp output problem
>
> On 8/9/13, jhseo <sjh431 at gmail.com> wrote:
>> Hi, all
>>
>> I'm test about ftp output.
>>
>> When I setting to AVI file format(output), file play is well..
>> But, mp4 is not play(using VLC).
>>
>> My option is..
>> ffmpeg -i inputfile -pass 1 -y -strict experimental -vcodec libx264
>> -vprofile main -b:v 3072k -s 320x240 -ftp-write-seekable 1
>> ftp://128.0.0.1/1.mp4
>>
>> somebody help me T.T
>
> What is error?
>
> Perhaps ftp server you are using is not supporing seeking for write?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From dashing.meng at gmail.com  Sat Aug 10 15:15:58 2013
From: dashing.meng at gmail.com (littlebat)
Date: Sat, 10 Aug 2013 21:15:58 +0800
Subject: [FFmpeg-user] ffmpeg and monotonically increasing DTS
In-Reply-To: <CACgtWZP7O25ZTw_5NSKVEq9MVeYyXN4HYK7x97EpxHqk38au_Q@mail.gmail.com>
References: <CACgtWZP7O25ZTw_5NSKVEq9MVeYyXN4HYK7x97EpxHqk38au_Q@mail.gmail.com>
Message-ID: <20130810211558.d944cdd6.dashing.meng@gmail.com>

On Fri, 9 Aug 2013 17:14:51 -0700
Pushkar Pradhan <pradhan.pushkar at gmail.com> wrote:

> I have a mp4 where some of the video frames have the same DTS/PTS and
> ffmpeg refuses to copy it when I try to extract the first few seconds:
> $ ffmpeg -i full_file.mp4  -vcodec copy -acodec copy -t 45
> short_45s.mp4 ffmpeg version git-2012-04-04-41a097a Copyright (c)
> 2000-2012 the FFmpeg developers
>   built on Apr  3 2012 18:08:31 with gcc 4.4.3
>   configuration: --enable-gpl --enable-libfaac --enable-libmp3lame
> --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libtheora
> --enable-libvorbis --enable-libx264 --enable-nonfree --enable-version3
> --enable-x11grab --enable-libvpx
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'full_file.mp4':
> Stream mapping:
>   Stream #0:1 -> #0:0 (copy)
>   Stream #0:0 -> #0:1 (copy)
> [mp4 @ 0x24b98a0] Application provided invalid, non monotonically
> increasing dts to muxer in stream 0: 119974 >= 119974
> av_interleaved_write_frame(): Invalid argument
> 
> However, when I take out the -acodec copy and -vcodec copy, i.e. when
> ffmpeg reencodes it, it does not abort.
> 
> So my question is: Is it invalid according to the MP4 spec to have
> frames with the same DTS/PTS for a video track?
> The error definitely makes sense.

You should try the lastest version of ffmpeg, I have met simlar
problem with old version, but it disappeared when use lastest version. 

From dashing.meng at gmail.com  Sat Aug 10 15:35:13 2013
From: dashing.meng at gmail.com (littlebat)
Date: Sat, 10 Aug 2013 21:35:13 +0800
Subject: [FFmpeg-user] Problems with Audio Codec: almost solved it
In-Reply-To: <1376066926.81169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
References: <1376062385.6588.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAEnqZEV8oMHi1CvetpVsDBaWid4EpwBAaCsu8wSX7CK-SzcxaQ@mail.gmail.com>
	<1376066926.81169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <20130810213513.3bf25f30.dashing.meng@gmail.com>

On Fri, 9 Aug 2013 09:48:46 -0700 (PDT)
James Board <jpboard2 at yahoo.com> wrote:

> Alternatively, if I want to compress the audio, what is a good
> suggestion for an audio codec that is most widely supported?? I want
> to ensure it works with mencoder, mplayer, and whatever tool I might
> use in the future.? What's the best bet?? '-c:a mp2'?? I use Linux
> machines almost exclusively.

You can choose mkv format container if no other considerations, so you
can free to select almost any video and audio codec. It should be no
problem when you use mplayer to play the file encoded by ffmpeg.
Mplayer uses ffmpeg decoder too. But I am not sure if it is the all
cases, because "/usr/bin/ffmpeg -codecs | grep "  EA"" show some audio
codec without decoder, e.g., "  EA    sonicls         Sonic lossless",
I only find "DEA    sonic           Sonic", maybe "sonic" audio decoder
also decode "sonicls"?

From somsaks at gmail.com  Sat Aug 10 16:11:22 2013
From: somsaks at gmail.com (Somsak Sriprayoonsakul)
Date: Sat, 10 Aug 2013 21:11:22 +0700
Subject: [FFmpeg-user] ffmpeg record only 25 secs of audio from RTSP
Message-ID: <CAMKWn_hAp86TBauwdV9gLfwZxSEguaAYmrGTtfy0-3+ucFkQ9g@mail.gmail.com>

Hello,

I want to dump the RTSP stream from my AVTECH AVM328B IP Camera. ffmpeg can
probe the stream but the recorded stream only has first 25 seconds of audio
for some reason.

My ffmpeg version

ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on May 27 2013 02:00:54 with gcc 4.4.7 (GCC) 20120313 (Red Hat
4.4.7-3)
  configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64
--mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect
--enable-gpl --enable-version3 --enable-nonfree --enable-postproc
--enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau
--disable-avisynth --enable-frei0r --enable-libopencv --enable-libdc1394
--enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libnut
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-librtmp --enable-libspeex --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid
--extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
-fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC'
--disable-stripping
  libavutil      52. 18.100 / 52. 18.100
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.104 / 54. 63.104
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 42.103 /  3. 42.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100


The command that I used

ffmpeg -i rtsp://172.20.20.15/live/mpeg4_ulaw/HD720P -acodec copy -vcodec
copy /var/tmp/test_copy.mov


ffprobe output of the stream

ffprobe version 1.2.1 Copyright (c) 2007-2013 the FFmpeg developers
  built on May 27 2013 02:00:54 with gcc 4.4.7 (GCC) 20120313 (Red Hat
4.4.7-3)
  configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64
--mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect
--enable-gpl --enable-version3 --enable-nonfree --enable-postproc
--enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau
--disable-avisynth --enable-frei0r --enable-libopencv --enable-libdc1394
--enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libnut
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-librtmp --enable-libspeex --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid
--extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
-fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC'
--disable-stripping
  libavutil      52. 18.100 / 52. 18.100
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.104 / 54. 63.104
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 42.103 /  3. 42.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[rtsp @ 0x1505480] Estimating duration from bitrate, this may be inaccurate
Input #0, rtsp, from 'rtsp://172.20.20.15/live/mpeg4_ulaw/HD720P':
  Metadata:
    title           : RTSP server
  Duration: N/A, start: 0.000000, bitrate: 64 kb/s
    Stream #0:0: Video: mpeg4, yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 20
tbr, 90k tbn, 20 tbc
    Stream #0:1: Audio: pcm_mulaw, 8000 Hz, 1 channels, s16, 64 kb/s



The result vdo file is playable, just that audio is only available for 25
seconds. Seems like ffmpeg wrongly detect the duration and stop recording
after 25 seconds.

Any suggestion are welcome.

Best Regards,


Somsak

somsaks at gmail.com

From hv.harneetvirk at gmail.com  Sat Aug 10 11:00:03 2013
From: hv.harneetvirk at gmail.com (Harneet Virk)
Date: Sat, 10 Aug 2013 14:30:03 +0530
Subject: [FFmpeg-user] ffmpeg code Problem
Message-ID: <CAMUPNE0FhU3RNf896wh3BOEUMKMW73UAPCxU9deXXYqZh3m30A@mail.gmail.com>

Dear All,

Required specs :
NTSC

MPEG-2 Main Profile.
4:2:0 chroma sampling.
ITU-R BT.601 color space.
15 Mbps.
Long GOP.
640 pixels fixed horizontal dimension.
Variable size vertical dimension depending on aspect ratio of source,
maximum size of 480 pixels;
Native frame rate of original source. 24 progressive frames per second for
film sourced, 23.976 progressive frames for inverse telecine sourced from
film, and 29.97 interlaced frames per second for video sourced.
All content must be provided deinterlaced and tagged as progressive.

ffmpeg Code Using:
ffmpeg -i Mashallah_cropped.mp4 -vcodec mpeg2video -pix_fmt yuv420p -r 24
-b:v 15000k -s 640x274 -ar 48000 -ac 2 -b:a 320k Ek_Tha_Tiger2.mpeg

output result is :
output file - ok
code - ok
audio - ok
 Problem is video bit rate in ffmpeg code we r putting 15000k but in final
output result we r getting only 6565k


Please help me for solve this problem ...


-- 
*Thanks & Regards*

*Harneet Virk*

From jiachielee at live.com  Sat Aug 10 23:08:37 2013
From: jiachielee at live.com (thljcl)
Date: Sat, 10 Aug 2013 14:08:37 -0700 (PDT)
Subject: [FFmpeg-user] How about multi-pass encoding for H264?
In-Reply-To: <tencent_15A69087168E129679010E23@qq.com>
References: <tencent_7A0415656176484179930BCE@qq.com>
	<20130807184012.3e2991f0@lrcd.com>
	<1375962468.80397.YahooMailNeo@web121101.mail.ne1.yahoo.com>
	<tencent_15A69087168E129679010E23@qq.com>
Message-ID: <1376168917622-4660621.post@n4.nabble.com>

<iframe width="560" height="315" src="//www.youtube.com/embed/P1y4osu56Hk"
frameborder="0" allowfullscreen></iframe>

If the encoding speed is not your main concern, I do recommend CRF over
multi-pass encoding.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-about-multi-pass-encoding-for-H264-tp4660564p4660621.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jiachielee at live.com  Sat Aug 10 23:40:25 2013
From: jiachielee at live.com (thljcl)
Date: Sat, 10 Aug 2013 14:40:25 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
Message-ID: <1376170825784-4660622.post@n4.nabble.com>

I do have experiences over changing frame rate from 25 FPS, 29.97 FPS, and 30
FPS to 24 FPS using the method I described. As far as my eyes are concerned,
I do not have the so-called ?unrealistic motion? issues. Some people have
believed that higher frame rate is required for more realistic motion.
Without regards to human eyes or perceptions, there is no such thing as
?absolute motion? as Laws of Physics are equivalent to all inertial frame of
references; depending on which postulate you use to derive the theory of
special relativity, such a statement could be regarded as either
?consequence? or ?postulate?. That said, human visual perception has its own
properties; what we see does not necessarily represents the actual motion
relative to us as an observer. Generally, our eyes take time to capture
visual information; hence our visual perception is actually formed by
information we received from different time. That?s why we observe the
effect of ?motion blurring? from ?fast-moving object?. In a way of speaking,
?motion blurring? effect we observe is similar to the camera, which also
requires time to capture electromagnetic radiation from different time. Due
to our day-to-day experience, it?s hence more realistic to have ?motion
blurring? effect from fast-moving object to our human eyes.
Generally speaking, human eyes are able to differentiate 12-15 images every
second by perceiving them individually. In other words, the frame rate of 24
FPS, 25 FPS, 29.97 FPS, 30 FPS, or 60 FPS all create the illusion of motion
picture, which we cannot actually know the frame rate using our naked eyes.
Theoretically speaking, if a camera is able to capture high-resolution image
at a very high frame rate on high-speed moving object, the ?motion blurring?
effect would be lessened or disappear for individual frames. Hence, dropping
the frames to achieve lower frame rate of 24 FPS would actually result in
?less realistic? motion. It?s possible to add ?special effect? with software
to make the motion ?more realistic? to our eyes. That said, we were talking
about perhaps from 300 FPS to 24 FPS. For the common frame rate used by
different standard such as ?24 FPS?, ?25 FPS?, ?29.97 FPS, etc., conversion
between them would almost certainly would not result any difference to human
naked eyes by duplicating and dropping frames.
As a matter of fact, most LCD displays can only have up to 60 Hz of refresh
rate. That means, even if the video does have very high frame rate, it would
become somewhat unrealistic to human eyes because of the lack of ?motion
blurring? effect and the low refresh rate of the display. Only when both the
display has similarly high refresh rate and at the same time the video does
have very high frame rate, the properties of human visual system would
create ?motion blurring? effect just as we actually see the actual motion of
the object.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660622.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From ricardohenrylee at gmail.com  Sun Aug 11 00:56:17 2013
From: ricardohenrylee at gmail.com (Richard H Lee)
Date: Sat, 10 Aug 2013 23:56:17 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <CAPYw7P5AL1e0rv8am0fU_32YeORL+a4=C0v+dRZomaDqkok=xA@mail.gmail.com>
References: <52053303.7070107@gmail.com>
	<CAPYw7P5AL1e0rv8am0fU_32YeORL+a4=C0v+dRZomaDqkok=xA@mail.gmail.com>
Message-ID: <5206C511.1030307@gmail.com>

On 10/08/2013 13:48, Paul B Mahol wrote:
> It looks like coded_picture_number is increased by one when decoding
> (and only for few codecs).
> So its not stored in bitstream, and skiping decoding would change it.

OK, I see thanks.

> You can try this anyway and report outcome.

Unfortunately, there does not seems to be an option for ffprobe to dump 
frames for a certain point. There is a seek_point option for the movie 
source, but this appears only to be for overlays.

From nicolas.george at normalesup.org  Sun Aug 11 01:21:30 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sun, 11 Aug 2013 01:21:30 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376170825784-4660622.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
Message-ID: <20130810232130.GA11654@phare.normalesup.org>

Le tridi 23 thermidor, an CCXXI, thljcl a ?crit?:
> I do have experiences over changing frame rate from 25 FPS, 29.97 FPS, and 30
> FPS to 24 FPS using the method I described.

That is very different from 24 -> 25 fps, because in one case the extra
frame comes at 5~6 Hz while in the other case it comes at ~1 Hz. An extra
frame at 5~6 Hz is barely noticeable, while an extra frame at 0.1~2 Hz is
very noticeable.

<snip>

How you can have so much knowledge and fail so utterly to actually
understand it is beyond me.

Regards,

-- 
  Nicolas George
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From maxim.levkov at gmail.com  Sun Aug 11 01:57:25 2013
From: maxim.levkov at gmail.com (Maxim Levkov)
Date: Sat, 10 Aug 2013 16:57:25 -0700
Subject: [FFmpeg-user] FFMPEG,
 How do I form an output file name automatically from information of
 an input file?
Message-ID: <CA+oYyUVFD-3L6C0fDbhQtOJst9JiePgXjTswugzKQfNvxiVi+A@mail.gmail.com>

Hello Group,

I'm processing many files and they are sporadically distinct from each
other. I'm trying to format an output filename that carries some
information about the file from the source file; such as, frame size,
frame rate, and other.

I could form the file name programmatically, by invoking series of
format output procedures. Then, record the name and the
characteristics into an output file name, and then form a command line
with ffmpeg for processing. However, I'm curious if there is a more
elegant way to just ask ffmpeg to pass some of these parameters to
format the name of the outputted file.

I found a discussion on the topic:
http://ffmpeg-users.933282.n4.nabble.com/How-to-output-real-frame-number-instead-of-sequentially-added-image-numbers-td4656645.html

However, I'm not sure if any of this was implemented or not.

Regards,
Max

From eclipse7 at gmx.net  Sun Aug 11 03:33:46 2013
From: eclipse7 at gmx.net (Alexander Strasser)
Date: Sun, 11 Aug 2013 03:33:46 +0200
Subject: [FFmpeg-user] some mp3 files can not be decoded by recent FFmpeg
In-Reply-To: <CAJQwZb33FAjiCR5VvJC1a1LFTmqxEyu5oWhqPg0YpyU0g8BhVQ@mail.gmail.com>
References: <CAJQwZb33FAjiCR5VvJC1a1LFTmqxEyu5oWhqPg0YpyU0g8BhVQ@mail.gmail.com>
Message-ID: <20130811013346.GA23399@akuma.Speedport_W_723V_1_28_000>

Hi Kirill, 

  thank you for the report and the sample!

On 2013-08-10 11:31 +0400, Kirill Gavrilov wrote:
> After migration to FFmpeg 2.0 I have observed that several old mp3 files (9
> from 100) can not be played anymore with the same error - no 'data' tag
> found:

  I analyzed the situation and found the culprit. It all breaks down
in the wav demuxer. Your file is an mp3 inside of a WAV container. It
additionally contains ID3 information at the beginning.

  So let me explain what is happening:
  0. wav demuxer was changed to only seek to even positions
     (fixes some "kind of broken" files I assume)
  1. the ID3 chunk in your file is of uneven size
  2. the wav demuxer seeks one byte too short
  3. the 2nd tag is read and results in a wrong tag name and a wrong size
  4. the data tag is never found and the demuxer errors out

  Unfortunately I am not yet sure what would be the best fix here.
Simply reverting the offending commit would result in other files
being not playable that were with it.

  Suggestions welcome.

  Alexander

> ./ffmpeg -i anb3.mp3
> ffmpeg version N-55410-gd4db7c3 Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Aug 10 2013 10:58:01 with gcc 4.2.1 (GCC) (Apple Inc. build
> 5666) (dot 3)
>   configuration: --enable-swscale --enable-shared --disable-static
> --enable-memalign-hack --enable-avfilter --enable-hardcoded-tables
> --enable-pthreads --disable-libopenjpeg --enable-runtime-cpudetect
> --enable-vda --disable-debug --enable-version3
>   libavutil      52. 41.100 / 52. 41.100
>   libavcodec     55. 24.100 / 55. 24.100
>   libavformat    55. 13.102 / 55. 13.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 82.100 /  3. 82.100
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.103 /  0. 17.103
> [wav @ 0x10180e820] Format wav probed with size=2048 and score=99
> [wav @ 0x10180e820] id3v2 ver:2 flags:00 len:85
> [wav @ 0x10180e820] no 'data' tag found
> [AVIOContext @ 0x101111260] Statistics: 32768 bytes read, 0 seeks
> anb3.mp3: Invalid data found when processing input
> 
> There no such issues with older release 0.10.2:
> [wav @ 0x10100e820] max_analyze_duration 5000000 reached at 5015510
> Input #0, wav, from 'anb3.mp3':
>   Duration: 00:03:11.82, bitrate: 12 kb/s
>     Stream #0:0: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, stereo, s16,
> 128 kb/s
> 
> I may suppose that these files were encoded with broken encoder but it is
> weird that only recent FFmpeg can not play them now...
> Could somebody tell me is it regression or new "feature"?
> 
> I have prepared 300 KB cut from one sample:
> http://sview.cifro-city.ru/trash/anb3.mp3
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From jiachielee at live.com  Sun Aug 11 03:20:10 2013
From: jiachielee at live.com (thljcl)
Date: Sat, 10 Aug 2013 18:20:10 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <20130810232130.GA11654@phare.normalesup.org>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
Message-ID: <1376184010720-4660626.post@n4.nabble.com>

Nicolas George-2 wrote
> That is very different from 24 -> 25 fps, because in one case the extra
> frame comes at 5~6 Hz while in the other case it comes at ~1 Hz. An extra
> frame at 5~6 Hz is barely noticeable, while an extra frame at 0.1~2 Hz is
> very noticeable.
> <snip>
> How you can have so much knowledge and fail so utterly to actually
> understand it is beyond me.
> 
> Regards,
> 
> -- 
>   Nicolas George
> 
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> 
> 
> signature.asc (205 bytes)
> &lt;http://ffmpeg-users.933282.n4.nabble.com/attachment/4660624/0/signature.asc&gt;

Hm? Well, let?s put it to actual test, which should be a much better option
than believing in either you or me. Do you currently have a source of 24 FPS
which you think that it has ?high speed motion?? If you seriously think that
using the method I describe would lead to ?unrealistic motion?, can you
share such a source with me and allow me to do the encoding process? After
doing the encoding process, I will upload the output file to SkyDrive and
share it temporarily publicly. We can then all use our eyes to make our
judgment. I do believe that an actual experiment would help me understand
better. Thank you.

Oh, yes? Please make sure that the source you use is not too big or too
lengthy. After all, it?s just meant to prove the hypothesis. I would prefer
your source to have less than 10 minutes in length.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660626.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From phil_rhodes at rocketmail.com  Sun Aug 11 09:53:17 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Sun, 11 Aug 2013 00:53:17 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376184010720-4660626.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>	<1376074113012-4660603.post@n4.nabble.com>	<52057604.30206@gmail.com>	<1376106209176-4660606.post@n4.nabble.com>	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>	<1376170825784-4660622.post@n4.nabble.com>	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
Message-ID: <1376207597.56237.YahooMailNeo@web121101.mail.ne1.yahoo.com>

Just duplicating frames is unlikely to produce results anyone wants to watch.?Duplicating one frame per second to go from 24 (or nearly 24) to 25 was occasionally done in days of yore when people wanted to edit film-originated material on PAL video. It looks very rough.

Properly done frame rate conversions that are intended to be of high quality, regardless of the desired change in rate, are invariably done with motion compensated interpolation. Of course this is not always an exact science either, but it's about the best option and properly done it is generally fairly close to invisible on most subjects I don't know if ffmpeg is capable of doing it.

I don't think there's really any serious argument about any of this, is there?

P

From krueger at lesspain.de  Sun Aug 11 10:10:06 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Sun, 11 Aug 2013 10:10:06 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376207597.56237.YahooMailNeo@web121101.mail.ne1.yahoo.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376207597.56237.YahooMailNeo@web121101.mail.ne1.yahoo.com>
Message-ID: <CAEnqZEXCaVw-U_ey+M-vKFN8iSV1+BuPQBiELPOj82vBvcAzqg@mail.gmail.com>

On Sun, Aug 11, 2013 at 9:53 AM, Phil Rhodes <phil_rhodes at rocketmail.com> wrote:
> Just duplicating frames is unlikely to produce results anyone wants to watch. Duplicating one frame per second to go from 24 (or nearly 24) to 25 was occasionally done in days of yore when people wanted to edit film-originated material on PAL video. It looks very rough.
>
> Properly done frame rate conversions that are intended to be of high quality, regardless of the desired change in rate, are invariably done with motion compensated interpolation. Of course this is not always an exact science either, but it's about the best option and properly done it is generally fairly close to invisible on most subjects I don't know if ffmpeg is capable of doing it.

No, there is currently no way to change frame rate via motion
interpolation (in some programs this is called "optical flow") in
ffmpeg.

From nicolas.george at normalesup.org  Sun Aug 11 10:50:17 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sun, 11 Aug 2013 10:50:17 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376184010720-4660626.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
Message-ID: <20130811085017.GA14544@phare.normalesup.org>

Le tridi 23 thermidor, an CCXXI, thljcl a ?crit?:
> Hm? Well, let?s put it to actual test, which should be a much better option
> than believing in either you or me.

The sliding coloured band of testsrc is already enough to see the problem,
although slightly due to the gradient. It is even more noticeable with a
more contrasted source, such as one produced by drawtext:

ffmpeg -lavfi 'color=black:r=24:d=20,drawtext=fontfile=sans:text=!:fontcolor=white:fontsize=1000:y=-10:x=t*24-200,fps=fps=25' -preset ultrafast -y t.mkv
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From nicolas.george at normalesup.org  Sun Aug 11 10:56:08 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sun, 11 Aug 2013 10:56:08 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <CAEnqZEXCaVw-U_ey+M-vKFN8iSV1+BuPQBiELPOj82vBvcAzqg@mail.gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376207597.56237.YahooMailNeo@web121101.mail.ne1.yahoo.com>
	<CAEnqZEXCaVw-U_ey+M-vKFN8iSV1+BuPQBiELPOj82vBvcAzqg@mail.gmail.com>
Message-ID: <20130811085608.GB14544@phare.normalesup.org>

Le quartidi 24 thermidor, an CCXXI, Robert Kr?ger a ?crit?:
> No, there is currently no way to change frame rate via motion
> interpolation (in some programs this is called "optical flow") in
> ffmpeg.

There could be, and it is a worthy project. But still, even with
interpolation, converting between two frame rated differing by 0.1~2 Hz will
cause visible hiccups.

The same applies to scaling: take an image with sharp edges, it will look
uglier if you scale it by a few pixels than if you scale it a lot.

Regards,

-- 
  Nicolas George
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From phil_rhodes at rocketmail.com  Sun Aug 11 11:07:36 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Sun, 11 Aug 2013 02:07:36 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <20130811085608.GB14544@phare.normalesup.org>
References: <52053303.7070107@gmail.com>	<1376074113012-4660603.post@n4.nabble.com>	<52057604.30206@gmail.com>	<1376106209176-4660606.post@n4.nabble.com>	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>	<1376170825784-4660622.post@n4.nabble.com>	<20130810232130.GA11654@phare.normalesup.org>	<1376184010720-4660626.post@n4.nabble.com>	<1376207597.56237.YahooMailNeo@web121101.mail.ne1.yahoo.com>	<CAEnqZEXCaVw-U_ey+M-vKFN8iSV1+BuPQBiELPOj82vBvcAzqg@mail.gmail.com>
	<20130811085608.GB14544@phare.normalesup.org>
Message-ID: <1376212056.26840.YahooMailNeo@web121105.mail.ne1.yahoo.com>



> But still, even with?interpolation, converting between?
> two frame rated differing by 0.1~2 Hz will?cause visible hiccups.

That's not always the case, although of course it's tricky. There are commercial products and hardware devices which will do it. They end up creating effectively all of the output frames using optical flow, presumably using algorithms tweaked to the specific case of that particular set of frame rates.

But yes, there are special considerations.

P

From soho123.2012 at gmail.com  Sun Aug 11 11:34:12 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Sun, 11 Aug 2013 17:34:12 +0800
Subject: [FFmpeg-user] [HELP][HELP]ffmpeg can not post new video data to
	ffserver
In-Reply-To: <CAFd2cZWMxuu+fMAFnh=A9VJ6n6qTE9BwxWX9yo5qjnX2oqMtEw@mail.gmail.com>
References: <CAFd2cZWMxuu+fMAFnh=A9VJ6n6qTE9BwxWX9yo5qjnX2oqMtEw@mail.gmail.com>
Message-ID: <CAFd2cZXik36M2n1DAHc=bhFu4Xtexyd0=ynPuF6vTkac63Um-Q@mail.gmail.com>

Hi All,

I got the problem about ffmpeg post fail to ffserver.
I use the solution ffmpeg+ ffserver for a stream server.
the stream can work for a few minutes, then ffmpeg can not post video data
via http://localhost:8090/feed1.ffm
the command line I use for ffmpeg:
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format h264
-i /dev/video1 -vcodec
copy http://localhost:8090/feed1.ffm


ffserver can not receive new video data from ffmpeg if the stream is
running for about 30~60 minutes.
I check the errno from ffmpeg av_interleaved_write_frame()
it says "No such file or directory"
errno is 2

Does anyone have ideas?

From soho123.2012 at gmail.com  Sun Aug 11 12:01:51 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Sun, 11 Aug 2013 18:01:51 +0800
Subject: [FFmpeg-user] HELP!HELP! ffmpeg+ffserver stability issue via
	127.0.0.1 http post
Message-ID: <CAFd2cZUKEcdjATurzjJZJJMZOTOToFGuVa0WbaaV1=EA1GjNvQ@mail.gmail.com>

Hi All,

When I try to setup a stream server with ffmpeg+ ffserver.
the problem I got is :
ffmpeg use tcp to connect ffserver. then ffmpeg use http post to send video
data to ffserver via loopback.

the source of video data is from usb webcam,
ffmpeg capture video then post to ffserver via tcp socket through loopback
netdev.
it is not stable via http 127.0.0.1 post.
since sometime ffserver says : there is not new data ready for read (the
result is from poll).
then no any data can be sent to remote vlc client for preview.
the video stream will be disconnected.
Does anyone have idea about http post to 127.0.0.1 from ffmpeg to ffserver?

why it is not stable?

From dashing.meng at gmail.com  Sun Aug 11 12:07:54 2013
From: dashing.meng at gmail.com (littlebat)
Date: Sun, 11 Aug 2013 18:07:54 +0800
Subject: [FFmpeg-user] FFMPEG,
 How do I form an output file name automatically from information of
 an input file?
In-Reply-To: <CA+oYyUVFD-3L6C0fDbhQtOJst9JiePgXjTswugzKQfNvxiVi+A@mail.gmail.com>
References: <CA+oYyUVFD-3L6C0fDbhQtOJst9JiePgXjTswugzKQfNvxiVi+A@mail.gmail.com>
Message-ID: <20130811180754.dbfd20ba.dashing.meng@gmail.com>

On Sat, 10 Aug 2013 16:57:25 -0700
Maxim Levkov <maxim.levkov at gmail.com> wrote:

> Hello Group,
> 
> I'm processing many files and they are sporadically distinct from each
> other. I'm trying to format an output filename that carries some
> information about the file from the source file; such as, frame size,
> frame rate, and other.
> 
> I could form the file name programmatically, by invoking series of
> format output procedures. Then, record the name and the
> characteristics into an output file name, and then form a command line
> with ffmpeg for processing. However, I'm curious if there is a more
> elegant way to just ask ffmpeg to pass some of these parameters to
> format the name of the outputted file.
> 
> I found a discussion on the topic:
> http://ffmpeg-users.933282.n4.nabble.com/How-to-output-real-frame-number-instead-of-sequentially-added-image-numbers-td4656645.html
> 
> However, I'm not sure if any of this was implemented or not.
> 

It seems you must do some programing, ffmpeg seems can't affect the
name of output file.

Shell scripting is a good choice on Linux os as I known.

From adf.lists at gmail.com  Sun Aug 11 12:43:46 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sun, 11 Aug 2013 11:43:46 +0100
Subject: [FFmpeg-user] dts channel check for downmix issue trac 2852
Message-ID: <52076AE2.5070602@gmail.com>

I filed a bug for an issue I have with dca internal downmix -

https://ffmpeg.org/trac/ffmpeg/ticket/2852

I would have liked to test/upload a normal dts channel check from a dvd
but totally failed to find one. Old links I have to mplayer/videolan
samples don't work anymore.

Does any one have or know of such a sample?



From adf.lists at gmail.com  Sun Aug 11 12:47:50 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sun, 11 Aug 2013 11:47:50 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376184010720-4660626.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
Message-ID: <52076BD6.3050607@gmail.com>

thljcl wrote:

> Hm? Well, let?s put it to actual test, which should be a much better
> option than believing in either you or me. Do you currently have a
> source of 24 FPS which you think that it has ?high speed motion?? If
> you seriously think that using the method I describe would lead to
> ?unrealistic motion?, can you share such a source with me and allow
> me to do the encoding process? After doing the encoding process, I
> will upload the output file to SkyDrive and share it temporarily
> publicly. We can then all use our eyes to make our judgment. I do
> believe that an actual experiment would help me understand better.
> Thank you.

Remember also that the 24 -> 25 case under discussion is going to show
better on a 50Hz display so when you say "We can then all use our eyes"
a lot of people will be viewing on a monitor at 60Hz that likely won't
do 50Hz.



From adf.lists at gmail.com  Sun Aug 11 13:24:19 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sun, 11 Aug 2013 12:24:19 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376170825784-4660622.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
Message-ID: <52077463.2020302@gmail.com>

thljcl wrote:
> I do have experiences over changing frame rate from 25 FPS, 29.97
> FPS, and 30 FPS to 24 FPS using the method I described. As far as my
> eyes are concerned, I do not have the so-called ?unrealistic motion?
> issues. Some people have believed that higher frame rate is required
> for more realistic motion. Without regards to human eyes or
> perceptions

Well you totally do not consider the aspect of display technology and
the issue of film judder caused by the need to refresh > framerate to
avoid flicker perception.

Of course film makers know the limitations of there media and do use
blur and tricks, including just not attempting some shots to alleviate
judder perception. This does not mean that this is ideal.

I could link to a Poynton here but I thought you would want something
more recent :-)

http://www.bbc.co.uk/rd/publications/whitepaper209




From soho123.2012 at gmail.com  Sun Aug 11 16:01:58 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Sun, 11 Aug 2013 22:01:58 +0800
Subject: [FFmpeg-user] ffmpeg + ffserver stream server
Message-ID: <CAFd2cZWsZVTV2EOuY5+5YB8yknPhZrztKX42ZTemX8SGrHaPgg@mail.gmail.com>

Hi All,


Does anyone familiar with the communication between ffmpeg and ffserver?
for example :
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format
mjpeg -i /dev/video0 -vcodec copy http://localhost:9080/feed1.ffm

as my understand,
ffmpeg use http post(localhost) to ffserver, then ffserver can use "recv"
system call to get video data.
per the result I tested.
the mechanism about http post to local host is not stable for long time
test.
because ffmpeg do tcp send to ffserver, the send will get stuck when I test
for about 30 minutes.
then ffserver can not get new video for the client preview.

Does anyone have idae about this issue?

From nicolas.george at normalesup.org  Sun Aug 11 16:29:21 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sun, 11 Aug 2013 16:29:21 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <52076BD6.3050607@gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
Message-ID: <20130811142921.GA10827@phare.normalesup.org>

Le quartidi 24 thermidor, an CCXXI, Andy Furniss a ?crit?:
> Remember also that the 24 -> 25 case under discussion is going to show
> better on a 50Hz display so when you say "We can then all use our eyes"
> a lot of people will be viewing on a monitor at 60Hz that likely won't
> do 50Hz.

Good remark. On a 60 Hz display, I re-ran the drawtext scrolling I posted a
few hours ago doing a 29->30 conversion instead 24->25, and it is even more
visible.

Regards,

-- 
  Nicolas George
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From phil_rhodes at rocketmail.com  Sun Aug 11 16:58:07 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Sun, 11 Aug 2013 07:58:07 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <20130811142921.GA10827@phare.normalesup.org>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
Message-ID: <1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>



> a lot of people will be viewing on a monitor at 60Hz that likely won't
> do 50Hz.


Bear in mind that almost no computer video playback software will show video frame-for-frame even if the frame rate of the pictures matches the monitor's refresh rate precisely. They may happen to do so, perhaps even for quite extended periods of time, but eventually it'll always glitch as the clocks on the graphics and sound hardware run in and out of phase.

P

From riventree at gmail.com  Sun Aug 11 14:39:15 2013
From: riventree at gmail.com (ffmpeg@riventree.com)
Date: Sun, 11 Aug 2013 05:39:15 -0700
Subject: [FFmpeg-user] Unsophisticated user has a question
Message-ID: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>

Hello all,

I've encountered something that makes me suspicious that I've lost data. I
have a .MOD file (Grrr Cannon vidcams) that takes up 39M of storage. I
converted it to .mp4 with the simplest possible command:

jde-desktop% ffmpeg -i 002.MOD 002.mp4

And the resultant .mp4 is 9.9M!

A cursory (and uninformed) glance at the output shows a [720x480 / yuv420p
/ 29.97fps] stream in and the same thing out. The video input is
"mpeg2video" and the output is "libx264", the audio is AC3 in and AAC out.

*I need to know if that's just better packing or (this is the problem) I've
lost data in the conversion.*

How can I find out?

Below is the complete output from ffmpeg. Can anyone explain the 4x size
difference with the data there?

-Jeff

ffmpeg version 0.8.6-4:0.8.6-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the
Libav developers
  built on Apr  2 2013 17:02:36 with gcc 4.6.3
*** THIS PROGRAM IS DEPRECATED ***
This program is only provided for compatibility and will be removed in a
future release. Please use avconv instead.
[mpeg @ 0x12699a0] max_analyze_duration reached
Input #0, mpeg, from '002.MOD':
  Duration: 00:00:05.97, start: 0.220756, bitrate: 5272 kb/s
    Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [PAR
32:27 DAR 16:9], 9600 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0.1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s
[buffer @ 0x1271f60] w:720 h:480 pixfmt:yuv420p
[libx264 @ 0x1271220] using SAR=32/27
[libx264 @ 0x1271220] using cpu capabilities: MMX2 SSE2Fast SSSE3 Cache64
[libx264 @ 0x1271220] profile Main, level 3.0
[libx264 @ 0x1271220] 264 - core 120 r2151 a3f4407 - H.264/MPEG-4 AVC codec
- Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1
ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11
fast_pskip=1 chroma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1
interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0
b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=2 keyint=250
keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1
crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.25 aq=1:1.00
Output #0, mp4, to '002.mp4':
  Metadata:
    encoder         : Lavf53.21.1
    Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 32:27 DAR 16:9],
q=-1--1, 30k tbn, 29.97 tbc
    Stream #0.1: Audio: libvo_aacenc, 48000 Hz, stereo, s16, 200 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
  Stream #0.1 -> #0.1
Press ctrl-c to stop encoding
[mpeg2video @ 0x126ba40] warning: first frame is no keyframe
    Last message repeated 1 times    734kB time=4.30 bitrate=1396.2kbits/s

frame=  180 fps= 29 q=29.0 Lsize=     969kB time=5.97 bitrate=1328.6kbits/s

video:816kB audio:147kB global headers:0kB muxing overhead 0.607864%
frame I:1     Avg QP:24.04  size: 59660
[libx264 @ 0x1271220] frame P:66    Avg QP:23.79  size: 10898
[libx264 @ 0x1271220] frame B:113   Avg QP:28.92  size:   494
[libx264 @ 0x1271220] consecutive B-frames:  3.3% 17.8% 63.3% 15.6%
[libx264 @ 0x1271220] mb I  I16..4:  5.9%  0.0% 94.1%
[libx264 @ 0x1271220] mb P  I16..4:  0.0%  0.0%  0.1%  P16..4: 62.1% 11.2%
11.3%  0.0%  0.0%    skip:15.4%
[libx264 @ 0x1271220] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8: 34.0%  0.6%
 0.1%  direct: 0.3%  skip:64.9%  L0:30.8% L1:59.4% BI: 9.8%
[libx264 @ 0x1271220] coded y,uvDC,uvAC intra: 93.2% 91.8% 54.2% inter:
14.9% 16.3% 1.1%
[libx264 @ 0x1271220] i16 v,h,dc,p: 26% 17% 15% 42%
[libx264 @ 0x1271220] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 14% 12%  6%  8%
10%  7% 10%  7%
[libx264 @ 0x1271220] i8c dc,h,v,p: 44% 17% 31%  9%
[libx264 @ 0x1271220] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x1271220] ref P L0: 63.4%  8.1% 16.6% 11.9%
[libx264 @ 0x1271220] ref B L0: 85.8% 14.2%
[libx264 @ 0x1271220] kb/s:1111.83
jde-desktop%

From h.reindl at thelounge.net  Sun Aug 11 18:20:27 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Sun, 11 Aug 2013 18:20:27 +0200
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
Message-ID: <5207B9CB.3080206@thelounge.net>


Am 11.08.2013 14:39, schrieb ffmpeg at riventree.com:
> I've encountered something that makes me suspicious that I've lost data. I
> have a .MOD file (Grrr Cannon vidcams) that takes up 39M of storage. I
> converted it to .mp4 with the simplest possible command:

you *always* lose data with non-lossless codecos at any conversion
any time, always, that's how things are working as you always lose
data when change a jpeg image and save it again

> jde-desktop% ffmpeg -i 002.MOD 002.mp4
> And the resultant .mp4 is 9.9M!

nothign special

> A cursory (and uninformed) glance at the output shows a [720x480 / yuv420p
> / 29.97fps] stream in and the same thing out. The video input is
> "mpeg2video" and the output is "libx264", the audio is AC3 in and AAC out.
> 
> *I need to know if that's just better packing or (this is the problem) I've
> lost data in the conversion.*

http://en.wikipedia.org/wiki/MPEG-2
http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC

> Can anyone explain the 4x size difference with the data there?

it is logical from MPEG3 to H264

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From soho123.2012 at gmail.com  Sun Aug 11 18:38:59 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 12 Aug 2013 00:38:59 +0800
Subject: [FFmpeg-user] communication between ffmpeg and ffserver how to
Message-ID: <CAFd2cZWH4QSebDD-cSUj_7T2xpU+risdHAVnNmZt+syZR8W2rA@mail.gmail.com>

Hi All,


if the example below:
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format
mjpeg -i /dev/video0 -vcodec copy http://localhost:9080/feed1.ffm

ffmpeg will use http post to send ffserver video data , but it should be
involve TCP handshake between ffserver and ffmpeg.
Is it possible to send video data without TCP handshake from ffmpeg?

Is there any communication method suitable for ffmpeg <---->ffserver?

From ricardohenrylee at gmail.com  Sun Aug 11 18:38:33 2013
From: ricardohenrylee at gmail.com (Richard H Lee)
Date: Sun, 11 Aug 2013 17:38:33 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
	<1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
Message-ID: <5207BE09.1020001@gmail.com>

On 11/08/2013 15:58, Phil Rhodes wrote:
> eventually it'll always glitch as the clocks on the graphics and sound hardware run in and out of phase.

but isn't that the whole point of the DTS and PTS. To keep track of the 
packets so that they never go out of sync?

From lou at lrcd.com  Sun Aug 11 18:42:31 2013
From: lou at lrcd.com (Lou)
Date: Sun, 11 Aug 2013 08:42:31 -0800
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
Message-ID: <1376239351.21887.8529491.296FD2ED@webmail.messagingengine.com>

On Sun, Aug 11, 2013, at 04:39 AM, ffmpeg at riventree.com wrote:
>
> ffmpeg version 0.8.6-4:0.8.6-0ubuntu0.12.04.1, Copyright (c) 2000-2013
> the
> Libav developers

This is not a product of the FFmpeg project and is therefore unsupported
here:
<http://stackoverflow.com/questions/9477115/who-can-tell-me-the-difference-and-relation-between-ffmpeg-libav-and-avconv>

You can get recent ffmpeg builds from links on the FFmpeg download page:
https://ffmpeg.org/download.html#LinuxBuilds

Or you can follow a step-by-step guide to compile ffmpeg:
http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide

From phil_rhodes at rocketmail.com  Sun Aug 11 18:53:36 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Sun, 11 Aug 2013 09:53:36 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <5207BE09.1020001@gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
	<1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<5207BE09.1020001@gmail.com>
Message-ID: <1376240016.36891.YahooMailNeo@web121106.mail.ne1.yahoo.com>

This does come up frequently!

The a/v data in the file may never go out of sync, but the actual hardware is not locked together.

Ideally, you might think you'd get (for instance) 60 refreshes per second out of the graphics card, and (for instance) 48,000 samples per second out of the sound card. But small, entirely normal errors in the crystal oscillators that run the hardware mean that eventually either a frame will be dropped or duplicated, or at least dropped or duplicated in terms of one display refresh. The other, better alternative would be to resample the audio and time the whole playback system to the graphics card's refresh rate, but nobody seems to do that, presumably because resampling audio is hard.

Either way, for this reason, there is no guarantee that even if you display a 30fps video file on a 60Hz display, you will get each frame exactly twice. It's impossible, given the fact that the playback software is (usually) locked to the sound card, and the graphics card is not locked to the sound card. As a practical matter, even now it's quite feasible that variabilities in system load and interrupt handling will cause further glitches beyond that intrinsic problem. It can be quite difficult to get modern computers to do things to within a sixtieth of a second.

This means that it is strictly impossible to do critical motion rendering evaluation on a computer monitor with common software, and it is why people like Blackmagic make integrated video playback boards, with both sound and video hardware on one card, clocked from the same crystal and thus in perfect sync.

P

From phil_rhodes at rocketmail.com  Sun Aug 11 18:56:23 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Sun, 11 Aug 2013 09:56:23 -0700 (PDT)
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <5207B9CB.3080206@thelounge.net>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
	<5207B9CB.3080206@thelounge.net>
Message-ID: <1376240183.9929.YahooMailNeo@web121102.mail.ne1.yahoo.com>

>> Can anyone explain the 4x size difference with the data there?


>it is logical from MPEG3 to H264

I suspect you meant MPEG2.

But is this a situation where ffmpeg is intelligently picking a bitrate based on the characteristics of the video, or is it defaulting to a low bitrate, given the user hasn't specified one? In the latter case it may make sense for the the user to specify a higher bitrate.

P

From h.reindl at thelounge.net  Sun Aug 11 19:09:02 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Sun, 11 Aug 2013 19:09:02 +0200
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <1376240183.9929.YahooMailNeo@web121102.mail.ne1.yahoo.com>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
	<5207B9CB.3080206@thelounge.net>
	<1376240183.9929.YahooMailNeo@web121102.mail.ne1.yahoo.com>
Message-ID: <5207C52E.4040901@thelounge.net>



Am 11.08.2013 18:56, schrieb Phil Rhodes:
>>> Can anyone explain the 4x size difference with the data there?
>> it is logical from MPEG3 to H264
> 
> I suspect you meant MPEG2.

yes, typo

> But is this a situation where ffmpeg is intelligently picking a bitrate based on the 
> characteristics of the video, or is it defaulting to a low bitrate, given the user 
> hasn't specified one? In the latter case it may make sense for the the user to specify 
> a higher bitrate

most likely a combination of both

in my expierience it is always a bad idea not to be explicit
but H.222/H.262 will in any case be much larger than H264

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From barsnick at gmx.net  Sun Aug 11 20:22:36 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Sun, 11 Aug 2013 20:22:36 +0200
Subject: [FFmpeg-user] ffmpeg code Problem
In-Reply-To: <CAMUPNE0FhU3RNf896wh3BOEUMKMW73UAPCxU9deXXYqZh3m30A@mail.gmail.com>
References: <CAMUPNE0FhU3RNf896wh3BOEUMKMW73UAPCxU9deXXYqZh3m30A@mail.gmail.com>
Message-ID: <20130811182236.GA10361@sunshine.barsnick.net>

On Sat, Aug 10, 2013 at 14:30:03 +0530, Harneet Virk wrote:
> ffmpeg -i Mashallah_cropped.mp4 -vcodec mpeg2video -pix_fmt yuv420p -r 24
> -b:v 15000k -s 640x274 -ar 48000 -ac 2 -b:a 320k Ek_Tha_Tiger2.mpeg

>  Problem is video bit rate in ffmpeg code we r putting 15000k but in final
> output result we r getting only 6565k

Lucky you, you're saving a lot of bandwidth! ;-)

The input video does not provide enough information to justify 15000k
output. You need to set "-minrate" to the same. I believe (correct me
if I'm wrong) the output will be "padded" to the requested rate without
any gain in quality.

The ffmpeg docs say this:
       ?   To force CBR video output:
                   ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v

Moritz

From werner.robitza at gmail.com  Sun Aug 11 20:31:11 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Sun, 11 Aug 2013 20:31:11 +0200
Subject: [FFmpeg-user] ffmpeg code Problem
In-Reply-To: <20130811182236.GA10361@sunshine.barsnick.net>
References: <CAMUPNE0FhU3RNf896wh3BOEUMKMW73UAPCxU9deXXYqZh3m30A@mail.gmail.com>
	<20130811182236.GA10361@sunshine.barsnick.net>
Message-ID: <CAN0LTaOr2Y05WbwaTXuas1=ZAhhFQPTXoy4pW5KERYaOfVK==A@mail.gmail.com>

> The ffmpeg docs say this:
>        ?   To force CBR video output:
>                    ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate
> 4000k -bufsize 1835k out.m2v
>

Just a note: -b is ambiguous. Use -b:v instead ? the documentation is
outdated here.

From fschwiet at gmail.com  Sun Aug 11 23:51:49 2013
From: fschwiet at gmail.com (Frank Schwieterman)
Date: Sun, 11 Aug 2013 14:51:49 -0700
Subject: [FFmpeg-user] inconsistent ffmpeg results when running same command
	on different machines
Message-ID: <CAPKhsbvY7SkevaGXhv27ohpgK6TvEes-upH9-8pHJe6rnL6DGg@mail.gmail.com>

  I am running the same ffmpeg command against the same input data on PC
and Mac, but am seeing slight differences in the output.  I suspect the
ffmpeg configuration is different...  Is it possible to configure things
such that I'd get the same result from both machines?  Can I reasonably get
a binary match?  I suspect I'll need to keep ffmpeg versions in sync (which
I can do) and keep the configurations in sync (which I'll have to learn).

  Some tracing of the command ran are below, I am clipping 100ms from a
300ms video as a test.  Attached are files with the actual console output.

on my Mac:
ffmpeg returned 0 with result hashed as
db19391b6dc9299c4b85ea6850adfb4146854fdf when invoked with [
    "-nostats",
    "-y",
    "-i",
    "/Users/user/src/videoplayer/media/300ms.mp4",
    "-ss",
    0.1,
    "-to",
    0.2,

"/Users/user/src/videoplayer/temp/media/97c1ff6f-08ec-4c2c-a762-135109bc0996.mp4"
]

on my PC:
ffmpeg returned 0 with result hashed as
34654f2aa267ea9f63d94ae0cd024dcac7240414 when invoked with [
    "-nostats",
    "-y",
    "-i",
    "C:\\src\\videoplayer\\media\\300ms.mp4",
    "-ss",
    0.1,
    "-to",
    0.2,

"C:\\src\\videoplayer\\temp\\media\\7ae5530f-6893-4cec-9c69-1d4a7c90dab8.mp4"
]
-------------- next part --------------
opffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 25 2013 13:33:41 with Apple LLVM version 4.2 (clang-425.0.28) (based on LLVM 3.2svn)
  configuration: --prefix=/opt/boxen/homebrew/Cellar/ffmpeg/1.2.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --enable-avresample --enable-vda --cc=cc --host-cflags= --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libxvid
  libavutil      52. 18.100 / 52. 18.100
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.104 / 54. 63.104
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 42.103 /  3. 42.103
  libswscale      2.  2.100 /  2.  2.100

  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100

Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/user/src/videoplayer/media/300ms.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.63.104
  Duration: 00:00:00.33, start: 0.030667, bitrate: 8488 kb/s

    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 9189 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn, 59.94 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 136 kb/s
    Metadata:
      handler_name    : SoundHandler

[libx264 @ 0x7ff461908e00] using SAR=1/1

[libx264 @ 0x7ff461908e00] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX

[libx264 @ 0x7ff461908e00] profile High, level 4.0

[libx264 @ 0x7ff461908e00] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00

Output #0, mp4, to '/Users/user/src/videoplayer/temp/media/97c1ff6f-08ec-4c2c-a762-135109bc0996.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.63.104

    Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 29.97 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, s16, 128 kb/s
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (aac -> libfaac)
Press [q] to stop, [?] for help

frame=    3 fps=0.0 q=-1.0 Lsize=     132kB time=00:00:00.11 bitrate=9528.7kbits/s dup=6 drop=0    
video:128kB audio:2kB subtitle:0 global headers:0kB muxing overhead 1.026052%

[libx264 @ 0x7ff461908e00] frame I:1     Avg QP:25.48  size: 83580
[libx264 @ 0x7ff461908e00] 
frame P:1     Avg QP:28.10  size: 31075
[libx264 @ 0x7ff461908e00] frame B:1     Avg QP:29.18  size: 16077
[libx264 @ 0x7ff461908e00] consecutive B-frames: 33.3% 66.7%  0.0%  0.0%
[libx264 @ 0x7ff461908e00] mb I  I16..4: 21.5% 53.8% 24.7%

[libx264 @ 0x7ff461908e00] mb P  I16..4:  5.1%  5.6%  2.1%  P16..4: 39.4% 14.2%  5.1%  0.0%  0.0%    skip:28.5%
[libx264 @ 0x7ff461908e00] mb B  I16..4:  0.4%  0.2%  0.2%  B16..8: 40.4%  8.0%  1.6%  direct: 1.8%  skip:47.4%  L0:37.4% L1:54.4% BI: 8.1%
[libx264 @ 0x7ff461908e00] 8x8 transform intra:52.4% inter:67.0%
[libx264 @ 0x7ff461908e00] coded y,uvDC,uvAC intra: 54.2% 69.9% 24.2% inter: 13.9% 18.0% 0.2%
[libx264 @ 0x7ff461908e00] i16 v,h,dc,p: 31% 34%  8% 26%
[libx264 @ 0x7ff461908e00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 25% 13%  6%  5%  6%  7%  7%  8%
[libx264 @ 0x7ff461908e00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 33% 24% 12%  6%  5%  6%  5%  5%  5%
[libx264 @ 0x7ff461908e00] i8c dc,h,v,p: 44% 25% 23%  7%
[libx264 @ 0x7ff461908e00] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x7ff461908e00] kb/s:10448.10
-------------- next part --------------
ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers

  built on Mar 28 2013 00:34:08 with gcc 4.8.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 18.100 / 52. 18.100
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.104 / 54. 63.104
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 42.103 /  3. 42.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100

Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\src\videoplayer\media\300ms.mp4':
  Metadata:
    major_brand     : 
isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41

    encoder         : Lavf54.63.10
4

  Duration: 00:00:00.33, start: 
0.030667, bitrate: 8488 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 9189 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn, 59.94 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 136 kb/s
    Metadata:
      handler_name    : SoundHandler

[libx264 @ 0000000002246da0] using SAR=1/1

[libx264 @ 0000000002246da0] using cpu capabilities: MMX2 SSE2Fast SSEMisalign LZCNT

[libx264 @ 0000000002246da0] profile High, level 4.0

[libx264 @ 0000000002246da0] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00

Output #0, mp4, to 'C:\src\videoplayer\temp\media\7ae5530f-6893-4cec-9c69-1d4a7c90dab8.mp4':
  Metadata:

    major_brand     : isom
    minor_version   : 
512

    compatible_brands: isomiso2avc1mp41

    encoder         : 
Lavf54.63.104


    Stream #0:0
(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1
, 11988 tbn
, 29.97 tbc

    Metadata:
      handler_name    : 
VideoHandler


    Stream #0:1
(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, s16, 128 kb/s

    Metadata:

      handler_name    : 
SoundHandler

Stream mapping:
  Stream #0:0 -> #0:0
 (h264 -> libx264)

  Stream #0:1 -> #0:1
 (aac -> libvo_aacenc)

Press [q] to stop, [?] for help

frame=    3 fps=0.0 q=-1.0 Lsize=     133kB time=00:00:00.12 bitrate=8851.7kbits/s dup=6 drop=0    

video:129kB audio:3kB subtitle:0 global headers:0kB muxing overhead 1.029447%

[libx264 @ 0000000002246da0] frame I:1     Avg QP:25.51  size: 83461
[libx264 @ 0000000002246da0] frame P:1     Avg QP:27.85  size: 30995
[libx264 @ 0000000002246da0] frame B:1     Avg QP:28.70  size: 16517
[libx264 @ 0000000002246da0] consecutive B-frames: 33.3% 66.7%  0.0%  0.0%
[libx264 @ 0000000002246da0] mb I  I16..4: 23.5% 51.7% 24.9%
[libx264 @ 0000000002246da0] mb P  I16..4:  4.8%  5.6%  2.3%  P16..4: 40.2% 14.2%  4.8%  0.0%  0.0%    skip:28.2%
[libx264 @ 0000000002246da0] mb B  I16..4:  0.5%  0.3%  0.3%  B16..8: 40.8%  8.5%  1.5%  direct: 1.7%  skip:46.5%  L0:37.2% L1:54.1% BI: 8.7%
[libx264 @ 0000000002246da0] 8x8 transform intra:50.6% inter:66.4%
[libx264 @ 0000000002246da0] coded y,uvDC,uvAC intra: 54.0% 70.2% 24.0% inter: 14.1% 18.0% 0.3%
[libx264 @ 0000000002246da0] i16 v,h,dc,p: 35% 33%  7% 24%
[libx264 @ 0000000002246da0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 22% 15%  6%  5%  6%  7%  7%  8%
[libx264 @ 0000000002246da0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 32% 24% 13%  5%  5%  6%  6%  5%  5%
[libx264 @ 0000000002246da0] i8c dc,h,v,p: 44% 26% 23%  7%
[libx264 @ 0000000002246da0] Weighted P-Frames: Y:0.0% UV:0.0%

[libx264 @ 0000000002246da0] kb/s:10467.36

From barsnick at gmx.net  Mon Aug 12 00:04:32 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Mon, 12 Aug 2013 00:04:32 +0200
Subject: [FFmpeg-user] inconsistent ffmpeg results when running same
 command on different machines
In-Reply-To: <CAPKhsbvY7SkevaGXhv27ohpgK6TvEes-upH9-8pHJe6rnL6DGg@mail.gmail.com>
References: <CAPKhsbvY7SkevaGXhv27ohpgK6TvEes-upH9-8pHJe6rnL6DGg@mail.gmail.com>
Message-ID: <20130811220432.GA12825@sunshine.barsnick.net>

On Sun, Aug 11, 2013 at 14:51:49 -0700, Frank Schwieterman wrote:
> I suspect the ffmpeg configuration is different...

The ffmpeg output (at the very top) will hint you towards that.

> Is it possible to configure things such that I'd get the same result
> from both machines?

I guess if the identical versions of ffmpeg and all used libraries are
identical (same versions, same configuration, not necessarily same
compilers though), it should work.

Your ffmpeg base code seems in sync (judging from the reported versions
of the libav* libraries).

But since these two differ version-wise:
> [libx264 @ 0x7ff461908e00] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> [libx264 @ 0000000002246da0] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
you can suspect that they won't create bitwise identical output in the
payload.

Moritz

From ricardohenrylee at gmail.com  Mon Aug 12 00:04:12 2013
From: ricardohenrylee at gmail.com (Richard H Lee)
Date: Sun, 11 Aug 2013 23:04:12 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376240016.36891.YahooMailNeo@web121106.mail.ne1.yahoo.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
	<1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<5207BE09.1020001@gmail.com>
	<1376240016.36891.YahooMailNeo@web121106.mail.ne1.yahoo.com>
Message-ID: <52080A5C.2060000@gmail.com>

On 11/08/2013 17:53, Phil Rhodes wrote:
> The other, better alternative would be to resample the audio and time
> the whole playback system to the graphics card's refresh rate

This is actually what I do. Most of the media I acquire is at NTSC
24000/1001 fps, but I play it on a PAL 25 fps monitor. The only way to
get smooth motion is to speed up the frame rate.

If I need to re-encode I pipe the raw video frames through yuvfps to set 
the frame rate or I set the PTS manually with ffmpeg. If I just need to 
change the frame rate without re-encoding, I have use MP4Box, as ffmpeg 
can't use "-vcodec copy" and setpts together. (If anybody knows how to 
do this, let me know.)

> but nobody seems to do that, presumably because resampling audio is
> hard.

And the reason why resampling is hard is because there are no tools out 
there that can resample audio accurately.

I tried every tool, ffmpeg, mencoder, sox, but they all ended up with 
large a/v desync after 10 mins. I think this is due to the rounding off 
of the decimal number I give to the various utilities. The only tool 
that worked was AVISynth, but that is a Windows tool.

(warning, shameless plug)
So I decided to create one myself.
https://github.com/rhlee/speed

First of all, you dump the audio into a wav file, either using mplayer 
or ffmpeg. Then you work out the factor by which you need to speed up 
the audio as a fraction. E.g. when converting from NTSC (24000/1001 fps) 
to PAL (25 fps), the ratio would be 25/(24000/1001) = 1001/960 . So the 
utility would be called as so:
speed 1001/960 input.wav output.wav

If the frame rate is not exact, which occurs with many media, you can 
work out the average frame rate using ffprobe -show_frames and specify 
the factor as a decimal
speed -f 1.02842655493067542736 1/1 in.wav out.wav

Finally, one can pitch shift the audio down using sox, as by speeding up 
the audio, you also raise the pitch.

I have been using this method for the past year or so on 100's of videos 
and it has worked perfectly every time. The only time it failed was when 
the source file was out of sync.

It would be nice if ffmpeg could have this level of accuracy when 
resampling audio, but I guess that would be a feature request.


[Please bear in mind 'speed' (I never got round to naming it properly) 
is the first proper C project I did, so it going to be pretty rough 
round the edges. It handles 2-channel, 8 or 16 bits per sample, PCM or 
float encoding which should cover 99% of audio out there. There is no 
documentation or licensing. The codebase itself is pretty messy. Also it 
could be a lot more efficient and possibly multi-threaded.]

From fschwiet at gmail.com  Mon Aug 12 00:57:49 2013
From: fschwiet at gmail.com (Frank Schwieterman)
Date: Sun, 11 Aug 2013 15:57:49 -0700
Subject: [FFmpeg-user] inconsistent ffmpeg results when running same
 command on different machines
In-Reply-To: <20130811220432.GA12825@sunshine.barsnick.net>
References: <CAPKhsbvY7SkevaGXhv27ohpgK6TvEes-upH9-8pHJe6rnL6DGg@mail.gmail.com>
	<20130811220432.GA12825@sunshine.barsnick.net>
Message-ID: <CAPKhsbv4RfiPa7aAkdsNZayo3uBumu2CURSVT5ORV5_foHggxQ@mail.gmail.com>

  Thanks for the insight Moritz.  I suspect that the libx264 version is
determined by whoever puts together the build, and so picking up windows
and mac builds there isn't a straightforward way to keep those in sync.
 Getting a binary match is not critical, so this is ok.
  One concern though is that the result lengths are different.  I am trying
to extract a clip of a specific length, which is going to be concatenated
for a loop effect.  It is important that the concatenated portion is a
specific time duration because the loop establishes a musical tempo.
  The command I run should extract video from time 0.1 to 0.2, so I'd
expect the output result to be 0.1 seconds long.  From the output, it says
it is 0.11s or 0.12s instead (I see "frame= 3 fps=0.0 q=-1.0 Lsize= 133kB
time=00:00:00.12 bitrate=8851.7kbits/s dup=6 drop=0" in one). I suppose
0.02 error on the duration is reasonable, if the error stays the same as
the clip is extended. Would you know if I could expect a similar fixed-size
error regardless of the duration of the extracted clip?

On Sun, Aug 11, 2013 at 3:04 PM, Moritz Barsnick <barsnick at gmx.net> wrote:

> On Sun, Aug 11, 2013 at 14:51:49 -0700, Frank Schwieterman wrote:
> > I suspect the ffmpeg configuration is different...
>
> The ffmpeg output (at the very top) will hint you towards that.
>
> > Is it possible to configure things such that I'd get the same result
> > from both machines?
>
> I guess if the identical versions of ffmpeg and all used libraries are
> identical (same versions, same configuration, not necessarily same
> compilers though), it should work.
>
> Your ffmpeg base code seems in sync (judging from the reported versions
> of the libav* libraries).
>
> But since these two differ version-wise:
> > [libx264 @ 0x7ff461908e00] 264 - core 125 - H.264/MPEG-4 AVC codec -
> Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1
> ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
> mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11
> fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1
> sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
> constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1
> weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40
> intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0
> qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> > [libx264 @ 0000000002246da0] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4
> AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html -
> options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1
> psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1
> cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6
> lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
> bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1
> b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25
> scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0
> qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> you can suspect that they won't create bitwise identical output in the
> payload.
>
> Moritz
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From krueger at lesspain.de  Mon Aug 12 08:57:18 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Mon, 12 Aug 2013 08:57:18 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <52080A5C.2060000@gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
	<1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<5207BE09.1020001@gmail.com>
	<1376240016.36891.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<52080A5C.2060000@gmail.com>
Message-ID: <CAEnqZEV9HBa4FmOn-hPWJiEma5qDPSFFNkwAFjrY9od=yeZ23w@mail.gmail.com>

On Mon, Aug 12, 2013 at 12:04 AM, Richard H Lee
<ricardohenrylee at gmail.com> wrote:
> On 11/08/2013 17:53, Phil Rhodes wrote:
>>
>> The other, better alternative would be to resample the audio and time
>> the whole playback system to the graphics card's refresh rate
>
>
> This is actually what I do. Most of the media I acquire is at NTSC
> 24000/1001 fps, but I play it on a PAL 25 fps monitor. The only way to
> get smooth motion is to speed up the frame rate.
>
> If I need to re-encode I pipe the raw video frames through yuvfps to set the
> frame rate or I set the PTS manually with ffmpeg. If I just need to change
> the frame rate without re-encoding, I have use MP4Box, as ffmpeg can't use
> "-vcodec copy" and setpts together. (If anybody knows how to do this, let me
> know.)
>
>
>> but nobody seems to do that, presumably because resampling audio is
>> hard.
>
>
> And the reason why resampling is hard is because there are no tools out
> there that can resample audio accurately.
>
> I tried every tool, ffmpeg, mencoder, sox, but they all ended up with large
> a/v desync after 10 mins. I think this is due to the rounding off of the
> decimal number I give to the various utilities. The only tool that worked
> was AVISynth, but that is a Windows tool.

If you have a reproducible test case for ffmpeg, please file a bug
report. I am not aware that this is a known limitation/bug of the
respective ffmpeg features and I have not come across it myself when
changing audio speed to achieve exactly what you describe (then again,
I am not sure I have tested with files > 10 minutes). Anyway, it is
worth taking the time to provide an input file and the exact command
line you used where you could produce an A/V desync.

From anatol2002 at gmail.com  Mon Aug 12 09:45:49 2013
From: anatol2002 at gmail.com (Anatol)
Date: Mon, 12 Aug 2013 10:45:49 +0300
Subject: [FFmpeg-user] inconsistent ffmpeg results when running same
 command on different machines
In-Reply-To: <CAPKhsbv4RfiPa7aAkdsNZayo3uBumu2CURSVT5ORV5_foHggxQ@mail.gmail.com>
References: <CAPKhsbvY7SkevaGXhv27ohpgK6TvEes-upH9-8pHJe6rnL6DGg@mail.gmail.com>
	<20130811220432.GA12825@sunshine.barsnick.net>
	<CAPKhsbv4RfiPa7aAkdsNZayo3uBumu2CURSVT5ORV5_foHggxQ@mail.gmail.com>
Message-ID: <CAPCGgdQZZNGz7ov_Wopy-yEKKMuWfy_DEmjnrwBne-OunybN2w@mail.gmail.com>

The same ffmpeg/Linux binary running on different hardwares or different
Linux version might produce different output files.
We had this with MPEG-TS decoding - there was bug in ffmpeg 0.10, it used
multi-threaded decoding as a default. Some machines handled it properly,
while others had nasty artifacts. It had to do with type of hw acceleration
that each machine had and the drivers that it used.


On Mon, Aug 12, 2013 at 1:57 AM, Frank Schwieterman <fschwiet at gmail.com>wrote:

>   Thanks for the insight Moritz.  I suspect that the libx264 version is
> determined by whoever puts together the build, and so picking up windows
> and mac builds there isn't a straightforward way to keep those in sync.
>  Getting a binary match is not critical, so this is ok.
>   One concern though is that the result lengths are different.  I am trying
> to extract a clip of a specific length, which is going to be concatenated
> for a loop effect.  It is important that the concatenated portion is a
> specific time duration because the loop establishes a musical tempo.
>   The command I run should extract video from time 0.1 to 0.2, so I'd
> expect the output result to be 0.1 seconds long.  From the output, it says
> it is 0.11s or 0.12s instead (I see "frame= 3 fps=0.0 q=-1.0 Lsize= 133kB
> time=00:00:00.12 bitrate=8851.7kbits/s dup=6 drop=0" in one). I suppose
> 0.02 error on the duration is reasonable, if the error stays the same as
> the clip is extended. Would you know if I could expect a similar fixed-size
> error regardless of the duration of the extracted clip?
>
> On Sun, Aug 11, 2013 at 3:04 PM, Moritz Barsnick <barsnick at gmx.net> wrote:
>
> > On Sun, Aug 11, 2013 at 14:51:49 -0700, Frank Schwieterman wrote:
> > > I suspect the ffmpeg configuration is different...
> >
> > The ffmpeg output (at the very top) will hint you towards that.
> >
> > > Is it possible to configure things such that I'd get the same result
> > > from both machines?
> >
> > I guess if the identical versions of ffmpeg and all used libraries are
> > identical (same versions, same configuration, not necessarily same
> > compilers though), it should work.
> >
> > Your ffmpeg base code seems in sync (judging from the reported versions
> > of the libav* libraries).
> >
> > But since these two differ version-wise:
> > > [libx264 @ 0x7ff461908e00] 264 - core 125 - H.264/MPEG-4 AVC codec -
> > Copyleft 2003-2012 - http://www.videolan.org/x264.html - options:
> cabac=1
> > ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1
> psy_rd=1.00:0.00
> > mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0
> deadzone=21,11
> > fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1
> > sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
> > constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1
> > weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40
> > intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60
> qpmin=0
> > qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> > > [libx264 @ 0000000002246da0] 264 - core 130 r2274 c832fe9 -
> H.264/MPEG-4
> > AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html -
> > options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7
> psy=1
> > psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1
> > cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6
> > lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
> > bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1
> > b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25
> > scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0
> > qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> > you can suspect that they won't create bitwise identical output in the
> > payload.
> >
> > Moritz
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From fschwiet at gmail.com  Mon Aug 12 15:10:44 2013
From: fschwiet at gmail.com (Frank Schwieterman)
Date: Mon, 12 Aug 2013 06:10:44 -0700
Subject: [FFmpeg-user] inconsistent ffmpeg results when running same
 command on different machines
In-Reply-To: <CAPCGgdQZZNGz7ov_Wopy-yEKKMuWfy_DEmjnrwBne-OunybN2w@mail.gmail.com>
References: <CAPKhsbvY7SkevaGXhv27ohpgK6TvEes-upH9-8pHJe6rnL6DGg@mail.gmail.com>
	<20130811220432.GA12825@sunshine.barsnick.net>
	<CAPKhsbv4RfiPa7aAkdsNZayo3uBumu2CURSVT5ORV5_foHggxQ@mail.gmail.com>
	<CAPCGgdQZZNGz7ov_Wopy-yEKKMuWfy_DEmjnrwBne-OunybN2w@mail.gmail.com>
Message-ID: <CAPKhsbuZ_ZrS=tDGf65XHG57aJkK6iv9vRDvpORxK_Hqgpb5Dw@mail.gmail.com>

  Ok, thanks for the info.

On Mon, Aug 12, 2013 at 12:45 AM, Anatol <anatol2002 at gmail.com> wrote:

> The same ffmpeg/Linux binary running on different hardwares or different
> Linux version might produce different output files.
> We had this with MPEG-TS decoding - there was bug in ffmpeg 0.10, it used
> multi-threaded decoding as a default. Some machines handled it properly,
> while others had nasty artifacts. It had to do with type of hw acceleration
> that each machine had and the drivers that it used.
>
>
> On Mon, Aug 12, 2013 at 1:57 AM, Frank Schwieterman <fschwiet at gmail.com
> >wrote:
>
> >   Thanks for the insight Moritz.  I suspect that the libx264 version is
> > determined by whoever puts together the build, and so picking up windows
> > and mac builds there isn't a straightforward way to keep those in sync.
> >  Getting a binary match is not critical, so this is ok.
> >   One concern though is that the result lengths are different.  I am
> trying
> > to extract a clip of a specific length, which is going to be concatenated
> > for a loop effect.  It is important that the concatenated portion is a
> > specific time duration because the loop establishes a musical tempo.
> >   The command I run should extract video from time 0.1 to 0.2, so I'd
> > expect the output result to be 0.1 seconds long.  From the output, it
> says
> > it is 0.11s or 0.12s instead (I see "frame= 3 fps=0.0 q=-1.0 Lsize= 133kB
> > time=00:00:00.12 bitrate=8851.7kbits/s dup=6 drop=0" in one). I suppose
> > 0.02 error on the duration is reasonable, if the error stays the same as
> > the clip is extended. Would you know if I could expect a similar
> fixed-size
> > error regardless of the duration of the extracted clip?
> >
> > On Sun, Aug 11, 2013 at 3:04 PM, Moritz Barsnick <barsnick at gmx.net>
> wrote:
> >
> > > On Sun, Aug 11, 2013 at 14:51:49 -0700, Frank Schwieterman wrote:
> > > > I suspect the ffmpeg configuration is different...
> > >
> > > The ffmpeg output (at the very top) will hint you towards that.
> > >
> > > > Is it possible to configure things such that I'd get the same result
> > > > from both machines?
> > >
> > > I guess if the identical versions of ffmpeg and all used libraries are
> > > identical (same versions, same configuration, not necessarily same
> > > compilers though), it should work.
> > >
> > > Your ffmpeg base code seems in sync (judging from the reported versions
> > > of the libav* libraries).
> > >
> > > But since these two differ version-wise:
> > > > [libx264 @ 0x7ff461908e00] 264 - core 125 - H.264/MPEG-4 AVC codec -
> > > Copyleft 2003-2012 - http://www.videolan.org/x264.html - options:
> > cabac=1
> > > ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1
> > psy_rd=1.00:0.00
> > > mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0
> > deadzone=21,11
> > > fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1
> > > sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
> > > constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1
> > > weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40
> > > intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60
> > qpmin=0
> > > qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> > > > [libx264 @ 0000000002246da0] 264 - core 130 r2274 c832fe9 -
> > H.264/MPEG-4
> > > AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html -
> > > options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7
> > psy=1
> > > psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1
> > > cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6
> > > lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
> > > bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1
> > > b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250
> keyint_min=25
> > > scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0
> > > qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> > > you can suspect that they won't create bitwise identical output in the
> > > payload.
> > >
> > > Moritz
> > > _______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From andrey.mochenov at idomoo.com  Mon Aug 12 16:22:25 2013
From: andrey.mochenov at idomoo.com (Andrey Mochenov)
Date: Mon, 12 Aug 2013 17:22:25 +0300
Subject: [FFmpeg-user] HLS example
Message-ID: <CAOJzLorxp1Xb=nAZcqs6mcnAH9c6YkUQ-t+11ZYwLwpM3A+tyw@mail.gmail.com>

Hi gentlemen,

We are trying to write a simple HLS code example based on muxing.c standard
one.
Let be two input streams, video and audio (they can be synthetic, doesn't
matter). Our purpose is to mux them into M3U8 playlist using HLS. Suppose,
duration of every TS segment file be 3 sec.
What are the corresponding codec parameters to be set (may be, segment
duration, etc.)? My be, AVOutputFormat ff_hls_muxer from hlsenc.c file
(from FFmpeg application sources) can be used here? Since there is no
corresponding documentation for HLS FFmpeg functions, I ask you to help us
with it.

Thank you,
Andrey Mochenov.

From jiang.hu at Sunmedia.com.cn  Mon Aug 12 09:42:00 2013
From: jiang.hu at Sunmedia.com.cn (jiang.hu at Sunmedia.com.cn)
Date: Mon, 12 Aug 2013 15:42:00 +0800
Subject: [FFmpeg-user] ERROR: libx264 not found
Message-ID: <OF58868640.FFAC2681-ON48257BC5.002A00B3-48257BC5.002A4A77@sunmedia.com.cn>

Dears?

I have a quest for ffmpeg compile.

[jiang.hu at cdoa06 ~/x264-snapshot-20130808-2245]$ make install
install -d /home/jiang.hu/dist/bin
install x264 /home/jiang.hu/dist/bin
install -d /home/jiang.hu/dist/include
install -d /home/jiang.hu/dist/lib
install -d /home/jiang.hu/dist/lib/pkgconfig
install -m 644 ./x264.h /home/jiang.hu/dist/include
install -m 644 x264_config.h /home/jiang.hu/dist/include
install -m 644 x264.pc /home/jiang.hu/dist/lib/pkgconfig
install -m 644 libx264.a /home/jiang.hu/dist/lib
ranlib /home/jiang.hu/dist/lib/libx264.a
[jiang.hu at cdoa06 ~/x264-snapshot-20130808-2245]$ cd ..
[jiang.hu at cdoa06 ~]$ cd ffmpeg-2.0/
[jiang.hu at cdoa06 ~/ffmpeg-2.0]$ ./configure --disable-avdevice 
--disable-yasm --enable-muxers --enable-demuxers --enable-shared 
--disable-static --enable-avcodec --enable-libx264 --enable-gpl 
--prefix=/home/jiang.hu/dist
ERROR: libx264 not found

If you think configure made a mistake, make sure you are using the latest
version from Git.  If the latest version fails, report the problem to the
ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net.
Include the log file "config.log" produced by configure as this will help
solving the problem.

entironment variable:
setenv PKG_CONFIG_PATH /home/jiang.hu/dist/lib/pkgconfig
setenv LD_LIBRARY_PATH /home/jiang.hu/dist/lib

thx~

??  ?????????
??????????????
SunMedia Technology Co.,Ltd
Tel: 86-28-87848688 EXT 5850
Fax: 86-28-87848686
email:jiang.hu at Sunmedia.com.cn

The preceding e-mail message (including any attachments) may contain confidential information intended for a specific individual and purpose.If you are not an intended recipient of this message, please notify the sender by replying to this message and then delete it from your system.Use,dissemination, distribution, or reproduction of this message by unintended recipients is not authorized and may be unlawful.


From riventree at gmail.com  Mon Aug 12 16:42:40 2013
From: riventree at gmail.com (Jeff Evarts)
Date: Mon, 12 Aug 2013 07:42:40 -0700
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
Message-ID: <CADaSsY35m+07ozr5_bPv3F=vsgkpWk35fL4UQiqm4gxWMEFhSg@mail.gmail.com>

>
> From phil_rhodes at rocketmail.com  Sun Aug 11 18:56:23 2013
> From: phil_rhodes at rocketmail.com (Phil Rhodes)
> Date: Sun, 11 Aug 2013 09:56:23 -0700 (PDT)
> Subject: [FFmpeg-user] Unsophisticated user has a question
> In-Reply-To: <5207B9CB.3080206 at thelounge.net>
> References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw at mail.gmail.com>
> 	<5207B9CB.3080206 at thelounge.net>
> Message-ID: <1376240183.9929.YahooMailNeo at web121102.mail.ne1.yahoo.com>
>
> >> Can anyone explain the 4x size difference with the data there?
>
>
> >it is logical from MPEG3 to H264
>
> I suspect you meant MPEG2.
>
> But is this a situation where ffmpeg is intelligently picking a bitrate based on the characteristics of the video, or is it defaulting to a low bitrate, given the user hasn't specified one? In the latter case it may make sense for the the user to specify a higher bitrate.
>
> P
>
>
It appears I should've been more clear. If I am losing a large amount of
visual data, then this conversion is a bad plan. If I'm not, then the 4x
storage decrease would be a boon. It looks like the original bitrate was
5.2kb/s and the output bitrate was merely 1.3kb/s.

Is there some way for me to understand whether this is "a lot of loss and I
should keep the original" vs "no real loss and save the space".

-Jeff

From manfred.ehgartner at gmx.net  Mon Aug 12 20:48:23 2013
From: manfred.ehgartner at gmx.net (Manfred Ehgartner)
Date: Mon, 12 Aug 2013 20:48:23 +0200
Subject: [FFmpeg-user] convert/replay H263 raw file Form rtp payload
Message-ID: <g2hgw0gbq3vbgjspthrtkb7n.1376333298899@email.android.com>

Hello, I am currently Wirkung in a project, where I have pcaps containing h263 and h264 rtp streams. I dont habe ans signaling, SDP info, just the rtp streams and I need Tod create something playable out oft it. For H264 I succeeded using videosnarf. The output raw file oft videosnarf could be converted Tod a MPEG Or avi file using ffmpeg. Unfortunately videosnarf doesnt support h263, so I tried manually just dump the rtp payload into some file, but this couldnt be read by ffmpeg. Also the output oft rtpbreak couldnt be processed by ffmpeg. So I assume, videosnarf is not simply dumping the rtp payload to a file. How must the h263 Payload be modified that the file can be read using ffmpeg -i ? I am happy about every hint

From pjjuliana216 at gmail.com  Mon Aug 12 22:20:42 2013
From: pjjuliana216 at gmail.com (Pete)
Date: Mon, 12 Aug 2013 20:20:42 +0000 (UTC)
Subject: [FFmpeg-user] FFmpeg CPU usage
Message-ID: <loom.20130812T220952-607@post.gmane.org>

I have a huge number of videos to transcode. The quality of the videos vary 
from low to medium. DVD quality is probably the highest quality. I have ffmpeg 
and libx264 compiled with pthreads and I use -threads 0 in my call to ffmpeg 
but I'm still not seeing 100% usage on all the cores. I need to maximize cpu 
usage since I'm not trying to transcode videos for a month straight. 

Here is the ffmpeg call I'm using:
ffmpeg -y -i "$i" -c:v libx264 -preset fast -b:v "$VID_BIT" -threads 0 -c:a 
libfdk_aac -b:a "$AUD_BIT" -f mp4 /"$OUT_DIR"/"$MD5"

Could the libfdk_aac codec be the issue? I couldn't find anything online that 
explicitly stated whether or not this codec supports threading. I really hope 
it does since I need the audio to be able to stream in a browser and aac works 
very well for that purpose. 

Thanks,
Pete


From manfred.ehgartner at gmx.net  Mon Aug 12 22:48:46 2013
From: manfred.ehgartner at gmx.net (Manfred Ehgartner)
Date: Mon, 12 Aug 2013 22:48:46 +0200
Subject: [FFmpeg-user] convert/replay H263 raw file Form rtp payload
Message-ID: <00bb01ce979d$57236480$056a2d80$@gmx.net>

I am currently working in a project, where I have pcaps containing h263 and
h264 rtp streams. I don't have any signaling, SDP info, just the rtp streams
and I need to create something playable out oft it. For H264 I succeeded
using videosnarf. The output raw file of videosnarf could be converted to a
MPEG or avi file using ffmpeg. Unfortunately, videosnarf doesn't support
h263, so I tried manually just to dump the rtp payload into some file, but
this couldn't be read by ffmpeg correctly. Also the output oft rtpbreak
(similar tool like videosnarf) couldn't be processed by ffmpeg. So I assume,
videosnarf is not simply dumping the rtp payload to a file. How must the
h263 Payload be modified that the file can be read using ffmpeg? I am happy
about every hint

 


From lou at lrcd.com  Mon Aug 12 22:50:05 2013
From: lou at lrcd.com (Lou)
Date: Mon, 12 Aug 2013 12:50:05 -0800
Subject: [FFmpeg-user] FFmpeg CPU usage
In-Reply-To: <loom.20130812T220952-607@post.gmane.org>
References: <loom.20130812T220952-607@post.gmane.org>
Message-ID: <20130812125005.708ef5b7@lrcd.com>

On Mon, 12 Aug 2013 20:20:42 +0000 (UTC)
Pete <pjjuliana216 at gmail.com> wrote:

> I have a huge number of videos to transcode. The quality of the videos vary 
> from low to medium. DVD quality is probably the highest quality. I have ffmpeg 
> and libx264 compiled with pthreads and I use -threads 0 in my call to ffmpeg 
> but I'm still not seeing 100% usage on all the cores. I need to maximize cpu 
> usage since I'm not trying to transcode videos for a month straight. 
> 
> Here is the ffmpeg call I'm using:
> ffmpeg -y -i "$i" -c:v libx264 -preset fast -b:v "$VID_BIT" -threads 0 -c:a 
> libfdk_aac -b:a "$AUD_BIT" -f mp4 /"$OUT_DIR"/"$MD5"

Please provide an actual, unscripted command and the complete ffmpeg
console output.

-threads 0 is default for this encoder.

> I really hope it does since I need the audio to be able to stream in a
> browser and aac works very well for that purpose. 

If your viewers with watch using progressive download then consider
adding "-movflags faststart". This will reloate the moov atom so the
video does not have to be completely downloaded before playback begins.

From pjjuliana216 at gmail.com  Mon Aug 12 23:23:40 2013
From: pjjuliana216 at gmail.com (Pete)
Date: Mon, 12 Aug 2013 21:23:40 +0000 (UTC)
Subject: [FFmpeg-user] FFmpeg CPU usage
References: <loom.20130812T220952-607@post.gmane.org>
	<20130812125005.708ef5b7@lrcd.com>
Message-ID: <loom.20130812T231303-668@post.gmane.org>

> Please provide an actual, unscripted command and the complete ffmpeg
> console output.

By nonscripted I assume you mean the command without the variables. Here it 
is: 

ffmpeg -y -i "inputVid" -c:v libx264 -preset fast -b:v 480k -threads 0 -c:a 
libfdk_aac -b:a 128k -f mp4 "outputVid.mp4"

here is the ffmpeg output from the server I'm using to transcode:

ffmpeg version git-2013-08-06-a68b6ec Copyright (c) 2000-2013 the FFmpeg 
developers
  built on Aug  7 2013 10:24:15 with gcc 4.4.5 (GCC) 20110214 (Red Hat 
4.4.5-6)
  configuration: --prefix=/usr/local/bin/ffmpeg_build/ --extra-cflags=-
I/usr/local/bin/ffmpeg_build/include --extra-ldflags=-
L/usr/local/bin/ffmpeg_build/lib --bindir=/usr/local/bin --extra-libs=-ldl -
-enable-gpl --enable-nonfree --
enable-libfdk_aac --enable-libx264 --enable-pthreads
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 22.100 / 55. 22.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] 
outfile}...
Use -h to get full help or, even better, run 'man ffmpeg'

I compiled it using the instructions here:
https://trac.ffmpeg.org/wiki/CentosCompilationGuide

> If your viewers with watch using progressive download then consider
> adding "-movflags faststart". This will reloate the moov atom so the
> video does not have to be completely downloaded before playback begins. 

I'll look into the "-movflag faststart" option. Thanks for the tip. Let me 
know if you need anything else from me.

-Pete


From seanm at goldeneraproductions.org  Tue Aug 13 00:25:01 2013
From: seanm at goldeneraproductions.org (Sean McShane)
Date: Mon, 12 Aug 2013 15:25:01 -0700
Subject: [FFmpeg-user] VC1 and WMV3 encoding in FFMPEG
Message-ID: <020701ce97aa$c98ae2c0$5ca0a840$@org>

Does anyone know if there is any plan to enable VC1 and/or WMV3 encoding in
FFMPEG?

 

-Sean




From ricardohenrylee at gmail.com  Tue Aug 13 01:27:29 2013
From: ricardohenrylee at gmail.com (Richard H Lee)
Date: Tue, 13 Aug 2013 00:27:29 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <CAEnqZEV9HBa4FmOn-hPWJiEma5qDPSFFNkwAFjrY9od=yeZ23w@mail.gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
	<1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<5207BE09.1020001@gmail.com>
	<1376240016.36891.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<52080A5C.2060000@gmail.com>
	<CAEnqZEV9HBa4FmOn-hPWJiEma5qDPSFFNkwAFjrY9od=yeZ23w@mail.gmail.com>
Message-ID: <52096F61.60007@gmail.com>

On 12/08/2013 07:57, Robert Kr?ger wrote:
> I am not aware that this is a known limitation/bug of the
> respective ffmpeg features and I have not come across it myself when
> changing audio speed to achieve exactly what you describe

Well I decided to double check the problem before filing a bug against 
ffmpeg and I discovered the 'atempo' filter. This works perfectly and 
keeps sync even for a 2 hour long movie.

It seems like atempo filter is a relatively new one, which would explain 
why I didn't come across it during my search for something similar last 
year.

From dashing.meng at gmail.com  Tue Aug 13 02:36:34 2013
From: dashing.meng at gmail.com (littlebat)
Date: Tue, 13 Aug 2013 08:36:34 +0800
Subject: [FFmpeg-user] ERROR: libx264 not found
In-Reply-To: <OF58868640.FFAC2681-ON48257BC5.002A00B3-48257BC5.002A4A77@sunmedia.com.cn>
References: <OF58868640.FFAC2681-ON48257BC5.002A00B3-48257BC5.002A4A77@sunmedia.com.cn>
Message-ID: <20130813083634.88e927c2.dashing.meng@gmail.com>

On Mon, 12 Aug 2013 15:42:00 +0800
jiang.hu at Sunmedia.com.cn wrote:

> Dears?
> 
> I have a quest for ffmpeg compile.
> 
> [jiang.hu at cdoa06 ~/x264-snapshot-20130808-2245]$ make install
> install -d /home/jiang.hu/dist/bin
> install x264 /home/jiang.hu/dist/bin
> install -d /home/jiang.hu/dist/include
> install -d /home/jiang.hu/dist/lib
> install -d /home/jiang.hu/dist/lib/pkgconfig
> install -m 644 ./x264.h /home/jiang.hu/dist/include
> install -m 644 x264_config.h /home/jiang.hu/dist/include
> install -m 644 x264.pc /home/jiang.hu/dist/lib/pkgconfig
> install -m 644 libx264.a /home/jiang.hu/dist/lib
> ranlib /home/jiang.hu/dist/lib/libx264.a
> [jiang.hu at cdoa06 ~/x264-snapshot-20130808-2245]$ cd ..
> [jiang.hu at cdoa06 ~]$ cd ffmpeg-2.0/
> [jiang.hu at cdoa06 ~/ffmpeg-2.0]$ ./configure --disable-avdevice 
> --disable-yasm --enable-muxers --enable-demuxers --enable-shared 
> --disable-static --enable-avcodec --enable-libx264 --enable-gpl 
> --prefix=/home/jiang.hu/dist
> ERROR: libx264 not found
> 
> If you think configure made a mistake, make sure you are using the
> latest version from Git.  If the latest version fails, report the
> problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on
> irc.freenode.net. Include the log file "config.log" produced by
> configure as this will help solving the problem.
> 
> entironment variable:
> setenv PKG_CONFIG_PATH /home/jiang.hu/dist/lib/pkgconfig
> setenv LD_LIBRARY_PATH /home/jiang.hu/dist/lib

You install libx264 into a non-standard place. Try:
[jiang.hu at cdoa06 ~/ffmpeg-2.0]$ ./configure --disable-avdevice 
--disable-yasm --enable-muxers --enable-demuxers --enable-shared 
--disable-static --enable-avcodec --enable-libx264 --enable-gpl 
--prefix=/home/jiang.hu/dist
--extra-cflags="-I/home/jiang.hu/dist/include"
--extra-ldflags="-L/home/jiang.hu/dist/lib"

From rmaximus839 at gmail.com  Tue Aug 13 00:29:19 2013
From: rmaximus839 at gmail.com (Robin Maximus)
Date: Mon, 12 Aug 2013 18:29:19 -0400
Subject: [FFmpeg-user] Cannot get resampling context
Message-ID: <CALdDpUoiDQvxMLREQU_AVfdNRQ8CT47-yOXXX9QRFZKvfHYn-Q@mail.gmail.com>

Hi,

Please can you provide help/hint for this follwing issue:

I did downloaded the free FFmpeg version to make my website to create
automated thumbnails from uploaded clips and it worked for a while. But my
provider (HostGator.com) told me that I have to upgrade to paying FFmpeg to
be able to convert video format to .mp4 (h264).  So they did upgrade but
now, because of this upgrade, my website does not created automated
thumbnails images has it was doing, and I have the  following warning
message:

" Array ( [0] => FFmpeg version SVN-r15261,  2000-2008 Fabrice Bellard, and
al. [1] => configuration: --prefix=/usr --incdir=/usr/include/ffmpeg
--libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man
--arch=x86_64 --extra-cflags=-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
-fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64
-mtune=generic -D_ISOC99_SOURCE -D_POSIX_C_SOURCE=200112 -fasm -std=c99
-fno-math-errno --enable-libdc1394 --enable-libfaac --enable-libfaad
--enable-libgsm --enable-libmp3lame --enable-libtheora --enable-libvorbis
--enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter
--enable-avfilter-lavf --enable-postproc --enable-swscale --enable-pthreads
--disable-static --enable-shared --enable-gpl --disable-debug
--disable-optimizations --disable-stripping [2] => libavutil 49.10. 0 /
49.10. 0 [3] => libavcodec 51.71. 0 / 51.71. 0 [4] => libavformat 52.22. 1
/ 52.38. 0 [5] => libavdevice 52. 1. 0 / 52. 2. 0 [6] => libavfilter 0. 1.
0 / 0. 1. 0 [7] => libswscale 0. 6. 1 / 0. 7. 1 [8] => libpostproc 51. 2. 0
/ 51. 2. 0 [9] => built on Feb 4 2009 19:41:05, gcc: 4.1.2 20071124 (Red
Hat 4.1.2-42) [10] => [wmv3 @ 0x2f98110]Extra data: 8 bits left, value: 0
[11] => [12] => Seems stream 1 codec frame rate differs from container
frame rate: 1000.00 (1000/1) -> 29.97 (30000/1001) [13] => Cannot get
resampling context )     "

Please can you help me as the Host or my programmer are not sure what to
do. Tks a lot, waiting for your reply.

From dashing.meng at gmail.com  Tue Aug 13 02:49:49 2013
From: dashing.meng at gmail.com (littlebat)
Date: Tue, 13 Aug 2013 08:49:49 +0800
Subject: [FFmpeg-user] FFmpeg CPU usage
In-Reply-To: <loom.20130812T220952-607@post.gmane.org>
References: <loom.20130812T220952-607@post.gmane.org>
Message-ID: <20130813084949.d35f6c7e.dashing.meng@gmail.com>

On Mon, 12 Aug 2013 20:20:42 +0000 (UTC)
Pete <pjjuliana216 at gmail.com> wrote:

> I have a huge number of videos to transcode. The quality of the
> videos vary from low to medium. DVD quality is probably the highest
> quality. I have ffmpeg and libx264 compiled with pthreads and I use
> -threads 0 in my call to ffmpeg but I'm still not seeing 100% usage
> on all the cores. I need to maximize cpu usage since I'm not trying
> to transcode videos for a month straight. 
> 
> Here is the ffmpeg call I'm using:
> ffmpeg -y -i "$i" -c:v libx264 -preset fast -b:v "$VID_BIT" -threads
> 0 -c:a libfdk_aac -b:a "$AUD_BIT" -f mp4 /"$OUT_DIR"/"$MD5"
> 
> Could the libfdk_aac codec be the issue? I couldn't find anything
> online that explicitly stated whether or not this codec supports
> threading. I really hope it does since I need the audio to be able to
> stream in a browser and aac works very well for that purpose. 
> 

Support you have 8 cores cpu, just start 8 or more ffmpeg command lines
you will get a 100% cpu usage.

From lou at lrcd.com  Tue Aug 13 03:02:19 2013
From: lou at lrcd.com (Lou)
Date: Mon, 12 Aug 2013 17:02:19 -0800
Subject: [FFmpeg-user] Cannot get resampling context
In-Reply-To: <CALdDpUoiDQvxMLREQU_AVfdNRQ8CT47-yOXXX9QRFZKvfHYn-Q@mail.gmail.com>
References: <CALdDpUoiDQvxMLREQU_AVfdNRQ8CT47-yOXXX9QRFZKvfHYn-Q@mail.gmail.com>
Message-ID: <20130812170219.354fd5e2@lrcd.com>

On Mon, 12 Aug 2013 18:29:19 -0400
Robin Maximus <rmaximus839 at gmail.com> wrote:

> Hi,
> 
> Please can you provide help/hint for this follwing issue:
> 
> I did downloaded the free FFmpeg version to make my website to create
> automated thumbnails from uploaded clips and it worked for a while. But my
> provider (HostGator.com) told me that I have to upgrade to paying FFmpeg to
> be able to convert video format to .mp4 (h264).  So they did upgrade but
> now, because of this upgrade, my website does not created automated
> thumbnails images has it was doing, and I have the  following warning
> message:

All versions of FFmpeg are free.

> " Array ( [0] => FFmpeg version SVN-r15261,  2000-2008 Fabrice Bellard

This is the "upgrade"? This is incredibly outdated and is unsupported
here. There have been at least 40269 revisions since the version you are
attempting to use. You are wasting your time and our time by trying to
use such an ancient build.

I'm not familiar with your hosting situation or really what you are
trying to do but you might be able to use a recent ffmpeg build for
Linux:

http://ffmpeg.org/download.html#LinuxBuilds

Or you could always compile:

https://trac.ffmpeg.org/wiki/CentosCompilationGuide

Also, your actual command is missing.

From anatol2002 at gmail.com  Tue Aug 13 05:51:12 2013
From: anatol2002 at gmail.com (Anatol)
Date: Tue, 13 Aug 2013 06:51:12 +0300
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
Message-ID: <CAPCGgdQ++LtHchHGwN8N6FJEWgk0QiNuTM+3sc+_E4KO0WZOFg@mail.gmail.com>

1. Get newer ffmpeg - 0.8.6 is quite old one. In your case - get newer x264
- it is 2 years old!. There is quite significant quality improvement with
newer ffmpeg/x264.
2. You may try h264 high profile (vs main that u use now) and/or you may
try 2-pass conversion.
3. Make up your mind what is the max bitrate that u can afford. Use - '-b
BITRATE -maxrate BITRATE' to set it.
4. Try those before digging further into crf, qscale, qmin, qmax, ...

x4 bitrate factor between h264 and mpeg2 makes sense, h264 should provide
fairly equal VISUAL quality.

On Sun, Aug 11, 2013 at 3:39 PM, ffmpeg at riventree.com
<riventree at gmail.com>wrote:

> Hello all,
>
> I've encountered something that makes me suspicious that I've lost data. I
> have a .MOD file (Grrr Cannon vidcams) that takes up 39M of storage. I
> converted it to .mp4 with the simplest possible command:
>
> jde-desktop% ffmpeg -i 002.MOD 002.mp4
>
> And the resultant .mp4 is 9.9M!
>
> A cursory (and uninformed) glance at the output shows a [720x480 / yuv420p
> / 29.97fps] stream in and the same thing out. The video input is
> "mpeg2video" and the output is "libx264", the audio is AC3 in and AAC out.
>
> *I need to know if that's just better packing or (this is the problem) I've
> lost data in the conversion.*
>
> How can I find out?
>
> Below is the complete output from ffmpeg. Can anyone explain the 4x size
> difference with the data there?
>
> -Jeff
>
> ffmpeg version 0.8.6-4:0.8.6-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the
> Libav developers
>   built on Apr  2 2013 17:02:36 with gcc 4.6.3
> *** THIS PROGRAM IS DEPRECATED ***
> This program is only provided for compatibility and will be removed in a
> future release. Please use avconv instead.
> [mpeg @ 0x12699a0] max_analyze_duration reached
> Input #0, mpeg, from '002.MOD':
>   Duration: 00:00:05.97, start: 0.220756, bitrate: 5272 kb/s
>     Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [PAR
> 32:27 DAR 16:9], 9600 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
>     Stream #0.1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s
> [buffer @ 0x1271f60] w:720 h:480 pixfmt:yuv420p
> [libx264 @ 0x1271220] using SAR=32/27
> [libx264 @ 0x1271220] using cpu capabilities: MMX2 SSE2Fast SSSE3 Cache64
> [libx264 @ 0x1271220] profile Main, level 3.0
> [libx264 @ 0x1271220] 264 - core 120 r2151 a3f4407 - H.264/MPEG-4 AVC codec
> - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options:
> cabac=1
> ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00
> mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11
> fast_pskip=1 chroma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1
> interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0
> b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=2 keyint=250
> keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1
> crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.25 aq=1:1.00
> Output #0, mp4, to '002.mp4':
>   Metadata:
>     encoder         : Lavf53.21.1
>     Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 32:27 DAR 16:9],
> q=-1--1, 30k tbn, 29.97 tbc
>     Stream #0.1: Audio: libvo_aacenc, 48000 Hz, stereo, s16, 200 kb/s
> Stream mapping:
>   Stream #0.0 -> #0.0
>   Stream #0.1 -> #0.1
> Press ctrl-c to stop encoding
> [mpeg2video @ 0x126ba40] warning: first frame is no keyframe
>     Last message repeated 1 times    734kB time=4.30 bitrate=1396.2kbits/s
>
> frame=  180 fps= 29 q=29.0 Lsize=     969kB time=5.97 bitrate=1328.6kbits/s
>
> video:816kB audio:147kB global headers:0kB muxing overhead 0.607864%
> frame I:1     Avg QP:24.04  size: 59660
> [libx264 @ 0x1271220] frame P:66    Avg QP:23.79  size: 10898
> [libx264 @ 0x1271220] frame B:113   Avg QP:28.92  size:   494
> [libx264 @ 0x1271220] consecutive B-frames:  3.3% 17.8% 63.3% 15.6%
> [libx264 @ 0x1271220] mb I  I16..4:  5.9%  0.0% 94.1%
> [libx264 @ 0x1271220] mb P  I16..4:  0.0%  0.0%  0.1%  P16..4: 62.1% 11.2%
> 11.3%  0.0%  0.0%    skip:15.4%
> [libx264 @ 0x1271220] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8: 34.0%  0.6%
>  0.1%  direct: 0.3%  skip:64.9%  L0:30.8% L1:59.4% BI: 9.8%
> [libx264 @ 0x1271220] coded y,uvDC,uvAC intra: 93.2% 91.8% 54.2% inter:
> 14.9% 16.3% 1.1%
> [libx264 @ 0x1271220] i16 v,h,dc,p: 26% 17% 15% 42%
> [libx264 @ 0x1271220] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 14% 12%  6%  8%
> 10%  7% 10%  7%
> [libx264 @ 0x1271220] i8c dc,h,v,p: 44% 17% 31%  9%
> [libx264 @ 0x1271220] Weighted P-Frames: Y:0.0% UV:0.0%
> [libx264 @ 0x1271220] ref P L0: 63.4%  8.1% 16.6% 11.9%
> [libx264 @ 0x1271220] ref B L0: 85.8% 14.2%
> [libx264 @ 0x1271220] kb/s:1111.83
> jde-desktop%
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From jiachielee at live.com  Tue Aug 13 06:13:15 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 12 Aug 2013 21:13:15 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376184010720-4660626.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
Message-ID: <1376367195683-4660674.post@n4.nabble.com>

thljcl wrote
> Hm? Well, let?s put it to actual test, which should be a much better
> option than believing in either you or me. Do you currently have a source
> of 24 FPS which you think that it has ?high speed motion?? If you
> seriously think that using the method I describe would lead to
> ?unrealistic motion?, can you share such a source with me and allow me to
> do the encoding process? After doing the encoding process, I will upload
> the output file to SkyDrive and share it temporarily publicly. We can then
> all use our eyes to make our judgment. I do believe that an actual
> experiment would help me understand better. Thank you.
> 
> Oh, yes? Please make sure that the source you use is not too big or too
> lengthy. After all, it?s just meant to prove the hypothesis. I would
> prefer your source to have less than 10 minutes in length.

My main desktop PC is an All-In-One HP PC. The refresh rate of that display
is 60 Hz. The television set in my living room seems to be able to support
both refresh rate of 60 Hz and 50 Hz. It is a 50-inch LCD display. It is
connected to the set-top box for the Astro Satellite TV service, which
delivers contents on both 1080i and 720p, depending on the user?s
configuration. The frame rate of the Satellite TV service seems to be 50i or
25p. I have another 11.6-inch tablet/Ultrabook, which has an HDMI output as
well as a HD media player. I occasionally connect the tablet to the
television set. From what I?ve seen so far, I did watch videos of different
frame rates, including but not limited to 23.976 FPS, 24 FPS, 25 FPS, 29.97
FPS, and 30 FPS. Most of the time, I cannot tell the frame rate without
specifically checking the media info. The television programs produced by
different producers throughout the world and theatrical films did use
different frame rates. I supposed that if the software can decode the
content correctly we the users need not bother about the frame rates?

I did specifically choose 24 FPS as my chosen frame rate for the videos I
encoded, regardless of the frame rate of the source. Most of the time, I did
not notice any difference using my naked eyes on my displays. However, to
verify the claim of others that ?unrealistic motion? could happen for 24-25
frame rate conversion by duplicating frame, I performed a test on the source
which has produced for the frame rate of 24 FPS. The source I used is an
open movie created in the year of 2010, called ?Sintel? (Project ?Durian?).
It?s a CGI-created animation using open-source software such as Blender. The
source of ?Sintel? of lossless compression or uncompressed version is at
http://media.xiph.org/sintel/.

>From the sources, I created both 24-FPS version and 25-FPS version, which I
uploaded to http://sdrv.ms/14FXYb9. On my display, I could not tell the
difference with my naked eyes at most parts, other than at the end where the
credits are rolling. The lack of ?motion blurring? causes the obvious
?jarring unrealistic motion? when the credits are rolling for the 25-FPS
version. I only used ffmpeg to duplicate frames. If I edit the frames by
adding some ?motion blurring? or ?transitional effects?, the ?jarring
unrealistic motion? could be somewhat mitigated.

A single experiment is far from conclusive to prove anything. However, from
my experiences as well as this experiment, while duplicating/dropping frames
is not strictly perfect, it?s certainly an acceptable solution for many if
not most cases. The reasoning is mainly the properties of human visual
system. ?Motion picture? is never ?the picture that moves? but our visual
system makes us think that ?it moves?.

Note: I may remove the ?Sintel? from SkyDrive from August 21, 2013 onwards.
Please download them before the deadline if you want to see for yourself.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660674.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From 670292548 at qq.com  Tue Aug 13 06:56:50 2013
From: 670292548 at qq.com (=?ISO-8859-1?B?dW5kZWZpbmVk?=)
Date: Tue, 13 Aug 2013 12:56:50 +0800
Subject: [FFmpeg-user] Live streams
Message-ID: <tencent_2FDAACF0353128D40C289B3C@qq.com>

Hi,
I have solved the problem of "Could not write header for output file" by changing the output format to flv with your last help. Thanks again. But I found an irregular error these days.


If I publish a live stream(in the format of H264 & AAC) to AMS 5, and start a ffmpeg to output another live stream, it works all the time. Here's the cmd(on windows 7):
C:\Users\Administrator>ffmpeg -re -y -i "rtmp://localhost/xolive/FDR3XTT7 live=1" -c:a libvo_aacenc -c:v libx264 -b:v 900k -b:a 96k -s 960*540 -f flv "rtmp://localhost/xolive/FDR3XTT7_540"


However, if I start another ffmpeg to convert the same input stream into another live stream, even if converting another live stream, it fails at times. And the output stream can not be play at times even though the outputting succeed. But, there're no much differences between these cmds.


Here's the uncut log:
C:\Users\Administrator>ffmpeg -re -y -i "rtmp://localhost/xolive/G0NKYAKZ live=1" -c:a libvo_aacenc -c:v libx264 -b:v 90
0k -b:a 96k -s 960*540 -f flv "rtmp://localhost/xolive/G0NKYAKZ_540"
ffmpeg version N-54857-gf9f7f4c Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 21 2013 20:04:40 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfi
g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopenco
re-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-lib
vpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 81.102 /  3. 81.102
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Received FLV packet before play()! Ignoring.
Metadata:
  duration              0.00
  fileSize              0.00
  width                 1280.00
  height                720.00
  videocodecid          avc1
  videodatarate         3600.00
  framerate             30.00
  audiocodecid          mp4a
  audiodatarate         256.00
  audiosamplerate       44100.00
  audiosamplesize       16.00
  audiochannels         2.00
  stereo                TRUE
  encoder               Open Broadcaster Software v0.542b
[flv @ 00000000025cdaa0] Stream discovered after head already parsed
[h264 @ 000000000265b640] no frame!
    Last message repeated 31 times
[h264 @ 000000000265b640] non-existing SPS 32 referenced in buffering period
[h264 @ 000000000265b640] no frame!
    Last message repeated 36 times
rtmp server sent error
[h264 @ 000000000265b640] no frame!
    Last message repeated 171 times
[h264 @ 000000000265b640] slice type too large (24) at 0 0
[h264 @ 000000000265b640] decode_slice_header error
[h264 @ 000000000265b640] no frame!
    Last message repeated 4 times
[flv @ 00000000025cdaa0] decoding for stream 1 failed
[flv @ 00000000025cdaa0] Could not find codec parameters for stream 1 (Video: h264): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, flv, from 'rtmp://localhost/xolive/G0NKYAKZ live=1':
  Metadata:
    fileSize        : 0
    audiochannels   : 2
    encoder         : Open Broadcaster Software v0.542b
  Duration: 00:00:00.00, start: 0.069000, bitrate: N/A
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 262 kb/s
    Stream #0:1: Video: h264, 30 tbr, 1k tbn
Output #0, flv, to 'rtmp://localhost/xolive/G0NKYAKZ_540':
  Metadata:
    fileSize        : 0
    audiochannels   : 2
    encoder         : Lavf55.12.102
    Stream #0:0: Audio: aac (libvo_aacenc) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, s16, 96 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> libvo_aacenc)
Press [q] to stop, [?] for help
[flv @ 0000000004a766e0] Failed to update header with correct duration.
[flv @ 0000000004a766e0] Failed to update header with correct filesize.
size=      37kB time=00:00:03.06 bitrate=  99.4kbits/s
video:0kB audio:35kB subtitle:0 global headers:0kB muxing overhead 6.860684%


The log would still show the converted progress. But it's different from the normal one.
Normal: 
frame=17381 fps= 25 q=-1.0 Lsize=   35321kB time=00:11:36.19 bitrate= 415.6kbits
/s dup=0 drop=162

Error:
size=      63kB time=00:00:05.04 bitrate= 103.0kbits/s





Otherwise, I want to convert the live streams into two or more live streams.






Best Regards,
Lane

From krueger at lesspain.de  Tue Aug 13 08:33:34 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Tue, 13 Aug 2013 08:33:34 +0200
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <52096F61.60007@gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
	<1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<5207BE09.1020001@gmail.com>
	<1376240016.36891.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<52080A5C.2060000@gmail.com>
	<CAEnqZEV9HBa4FmOn-hPWJiEma5qDPSFFNkwAFjrY9od=yeZ23w@mail.gmail.com>
	<52096F61.60007@gmail.com>
Message-ID: <CAEnqZEXHLgRUW5Sr_t4Zs6ap=A2Unga5oQwZ8WEJVwNzJ_c1Jw@mail.gmail.com>

On Tue, Aug 13, 2013 at 1:27 AM, Richard H Lee
<ricardohenrylee at gmail.com> wrote:
> On 12/08/2013 07:57, Robert Kr?ger wrote:
>>
>> I am not aware that this is a known limitation/bug of the
>> respective ffmpeg features and I have not come across it myself when
>> changing audio speed to achieve exactly what you describe
>
>
> Well I decided to double check the problem before filing a bug against
> ffmpeg and I discovered the 'atempo' filter. This works perfectly and keeps
> sync even for a 2 hour long movie.
>
> It seems like atempo filter is a relatively new one, which would explain why
> I didn't come across it during my search for something similar last year.
>

yes, it is relatively new (added a bit more than a year ago) but
still, if your original command line, that tries to achieve the audio
stretching/squeezing using resampling causes a/v desync, it should be
investigated whether your command line is wrong or there is a problem
in the resampling code.

From manfred.ehgartner at gmx.net  Tue Aug 13 11:15:15 2013
From: manfred.ehgartner at gmx.net (Manfred Ehgartner)
Date: Tue, 13 Aug 2013 11:15:15 +0200
Subject: [FFmpeg-user] create avi or mpeg file from rtp dump + sdp file
Message-ID: <5209F923.8000306@gmx.net>

Hello,

I have a file containing rtp packets of a video streaming session (h263,
h264) and additionally the sdp file, my goal is to create a playable
video file out of it (the container doesn't matter) using ffmpeg.

I am not experienced using sdp, but as far as I understood, inside the
sdp file the parameter c = IN IP4 Host defines the network parameters
for the player which is opening the sdp. So the player is using this
parameters to wait for incoming rtp packets fitting to the description
inside the SDP, am I right?

So I would need to replay the rtp stream locally and use ffmpeg to
capture on the packets on the NIC for writing the container file? Or is
it possible somehow directly feed ffmpeg with the sdp + the file
containing the already captured rtp packets and take this two input
parameters for generating the output directly?

Thanks in advance for your help


From werner.robitza at gmail.com  Tue Aug 13 12:09:14 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Tue, 13 Aug 2013 12:09:14 +0200
Subject: [FFmpeg-user] Cannot get resampling context
In-Reply-To: <CALdDpUoiDQvxMLREQU_AVfdNRQ8CT47-yOXXX9QRFZKvfHYn-Q@mail.gmail.com>
References: <CALdDpUoiDQvxMLREQU_AVfdNRQ8CT47-yOXXX9QRFZKvfHYn-Q@mail.gmail.com>
Message-ID: <CAN0LTaMXop8we=25P=uJerMz4-Si6DXT9F4p1_1pQ64M8ieG8Q@mail.gmail.com>

On Tue, Aug 13, 2013 at 12:29 AM, Robin Maximus <rmaximus839 at gmail.com>wrote:

>
> Please can you help me as the Host or my programmer are not sure what to
> do.


Unfortunately this is not an FFmpeg issue, but a hosting issue. Seems
HostGator is notorious for supplying you with outdated versions unless you
have a CentOS 6 VPS from them, which gives you 0.10 or something. If your
host does not allow you to run static builds (like Lou said), or those
cannot run due to an outdated Linux kernel, consider changing your hosting
provider, or at least changing to a CentOS 6 VPS.

(Regardless, it is ridiculous to charge money for five year old free and
open source software.)

I'm afraid there is nothing more we can do on this mailing list.

From belcampo at zonnet.nl  Tue Aug 13 12:40:48 2013
From: belcampo at zonnet.nl (Henk D. Schoneveld)
Date: Tue, 13 Aug 2013 12:40:48 +0200
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <CADaSsY35m+07ozr5_bPv3F=vsgkpWk35fL4UQiqm4gxWMEFhSg@mail.gmail.com>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
	<CADaSsY35m+07ozr5_bPv3F=vsgkpWk35fL4UQiqm4gxWMEFhSg@mail.gmail.com>
Message-ID: <3CD99F6F-E13B-4061-846D-2027C467BF55@zonnet.nl>


On Aug 12, 2013, at 4:42 PM, Jeff Evarts <riventree at gmail.com> wrote:

>> 
>> From phil_rhodes at rocketmail.com  Sun Aug 11 18:56:23 2013
>> From: phil_rhodes at rocketmail.com (Phil Rhodes)
>> Date: Sun, 11 Aug 2013 09:56:23 -0700 (PDT)
>> Subject: [FFmpeg-user] Unsophisticated user has a question
>> In-Reply-To: <5207B9CB.3080206 at thelounge.net>
>> References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw at mail.gmail.com>
>> 	<5207B9CB.3080206 at thelounge.net>
>> Message-ID: <1376240183.9929.YahooMailNeo at web121102.mail.ne1.yahoo.com>
>> 
>>>> Can anyone explain the 4x size difference with the data there?
>> 
>> 
>>> it is logical from MPEG3 to H264
>> 
>> I suspect you meant MPEG2.
>> 
>> But is this a situation where ffmpeg is intelligently picking a bitrate based on the characteristics of the video, or is it defaulting to a low bitrate, given the user hasn't specified one? In the latter case it may make sense for the the user to specify a higher bitrate.
>> 
>> P
>> 
>> 
> It appears I should've been more clear. If I am losing a large amount of
> visual data, then this conversion is a bad plan. If I'm not, then the 4x
> storage decrease would be a boon. It looks like the original bitrate was
> 5.2kb/s and the output bitrate was merely 1.3kb/s.
> 
> Is there some way for me to understand whether this is "a lot of loss and I
> should keep the original" vs "no real loss and save the space".
Real loss means, you can visibly detect it. If you add -crf 22 and inspect both visibly and the size of the file, then if not satisfied with the result, try -crf 20 etc. With -crf 20 file size will be larger and quality will be better. Decide yourself if things are going in the desired direction.
> 
> -Jeff
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From tevans.uk at googlemail.com  Tue Aug 13 15:47:41 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Tue, 13 Aug 2013 14:47:41 +0100
Subject: [FFmpeg-user] Problems with Audio Codec: almost solved it
In-Reply-To: <1376066926.81169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
References: <1376062385.6588.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAEnqZEV8oMHi1CvetpVsDBaWid4EpwBAaCsu8wSX7CK-SzcxaQ@mail.gmail.com>
	<1376066926.81169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1KtYGDhSsf=RJRVfCHL89HnzfZJXvW2Lh=KccTMqS58=Q@mail.gmail.com>

On Fri, Aug 9, 2013 at 5:48 PM, James Board <jpboard2 at yahoo.com> wrote:
>>What exactly does ffmpeg output when you do ffmpeg -i <your-input-file>?
>
>
> For the audio code, it displays: audio pcm_s16le 48000 Hz s1g, 1536 kb/s.
>
> I guess that means I should use '-c:a pcm_s16le'.
>
> Is there an ffmpeg option that tells it to use whatever audio codec
> that the input file used?

Yes, you can do this by saying to copy the audio with "-c:a copy".
Bear in mind that not every format can contain every codec, and so you
cannot for instance put pcm_s16 into an mp4 container, mp4 (according
to wikipedia) can only take MPEG4 Part 3 codecs, limiting you to
things like AAC, MP1, MP2, MP3 and some less well known codecs, and
not ac3 nor pcm.

Other containers are more permissive, eg the nut container will take
almost anything that ffmpeg can produce, but probably won't be played
back by anything other than ffplay and possibly mplayer. mkv is a good
choice.

>
> Alternatively, if I want to compress the audio, what is a good suggestion
> for an audio codec that is most widely supported?  I want to ensure it
> works with mencoder, mplayer, and whatever tool I might use in the
> future.  What's the best bet?  '-c:a mp2'?  I use Linux machines almost
> exclusively.

mp2 is fine, if a little long in the tooth. I would use ac3 if you
want high quality, or aac if you want a small file size.

Cheers

Tom

From jpboard2 at yahoo.com  Tue Aug 13 16:44:45 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 13 Aug 2013 07:44:45 -0700 (PDT)
Subject: [FFmpeg-user] FFmpeg CPU usage
In-Reply-To: <20130813084949.d35f6c7e.dashing.meng@gmail.com>
References: <loom.20130812T220952-607@post.gmane.org>
	<20130813084949.d35f6c7e.dashing.meng@gmail.com>
Message-ID: <1376405085.55602.YahooMailNeo@web164705.mail.gq1.yahoo.com>



>> I have a huge number of videos to transcode. The quality of the
>> videos vary from low to medium. DVD quality is probably the highest
>> quality. I have ffmpeg and libx264 compiled with pthreads and I use
>> -threads 0 in my call to ffmpeg but I'm still not seeing 100% usage
>> on all the cores. I need to maximize cpu usage since I'm not trying
>> to transcode videos for a month straight. 

Do you have enough disk bandwidth to support 8 mpeg encodes at
full speed?? If not, then you won't see 100 percent CPU usage.

From tevans.uk at googlemail.com  Tue Aug 13 16:58:52 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Tue, 13 Aug 2013 15:58:52 +0100
Subject: [FFmpeg-user] FFmpeg CPU usage
In-Reply-To: <loom.20130812T231303-668@post.gmane.org>
References: <loom.20130812T220952-607@post.gmane.org>
	<20130812125005.708ef5b7@lrcd.com>
	<loom.20130812T231303-668@post.gmane.org>
Message-ID: <CAFHbX1Ka5uyyLVMHPrhbUyBjA4=H1T53tKLwnh0DYdVj7wOE7A@mail.gmail.com>

On Mon, Aug 12, 2013 at 10:23 PM, Pete <pjjuliana216 at gmail.com> wrote:
>> Please provide an actual, unscripted command and the complete ffmpeg
>> console output.
>
> By nonscripted I assume you mean the command without the variables. Here it
> is:
>
> ffmpeg -y -i "inputVid" -c:v libx264 -preset fast -b:v 480k -threads 0 -c:a
> libfdk_aac -b:a 128k -f mp4 "outputVid.mp4"
>
> here is the ffmpeg output from the server I'm using to transcode:
>

Close, but no cigar - "complete ffmpeg console output" is the output
of the command you are trying to run. I don't think it matters too
much in this scenario though - you're happy with ffmpeg's output, just
not how long it is taking.

You could try transcoding a few videos with -an (to disable audio) and
see if that dramatically improves fps - otherwise, start checking
whether any component is 100% stressed, eg disk, network, cpu or RAM.
If not, you're likely constrained by CPU.

I know you say that you are not seeing 100% usage across all cores,
bear in mind MT is not 100% efficient. Since you have a veritable
pipeline of videos to transcode, perhaps you should try transcoding N
videos at a time, instead of trying to have N threads transcoding 1
video simultaneously.

Cheers

Tom

From riventree at gmail.com  Tue Aug 13 16:50:09 2013
From: riventree at gmail.com (Jeff Evarts)
Date: Tue, 13 Aug 2013 07:50:09 -0700
Subject: [FFmpeg-user] Unsophisticated user has a question
In-Reply-To: <3CD99F6F-E13B-4061-846D-2027C467BF55@zonnet.nl>
References: <CADaSsY2sQwNO1WPX40hjENxmJ4bqZD28UKUrxvAvnQB8JxrXTw@mail.gmail.com>
	<CADaSsY35m+07ozr5_bPv3F=vsgkpWk35fL4UQiqm4gxWMEFhSg@mail.gmail.com>
	<3CD99F6F-E13B-4061-846D-2027C467BF55@zonnet.nl>
Message-ID: <CADaSsY3PFLyFM_YcS8HdD2=zCd2RptiPYNBhu9iP3j+oXR5ZqQ@mail.gmail.com>

On Tue, Aug 13, 2013 at 3:40 AM, Henk D. Schoneveld <belcampo at zonnet.nl>wrote:

> > Is there some way for me to understand whether this is "a lot of loss
> and I
> > should keep the original" vs "no real loss and save the space".
> Real loss means, you can visibly detect it. If you add -crf 22 and inspect
> both visibly and the size of the file, then if not satisfied with the
> result, try -crf 20 etc. With -crf 20 file size will be larger and quality
> will be better. Decide yourself if things are going in the desired
> direction.
>

Thank you Henk! This is exactly what I needed.

-Jeff

From jpboard2 at yahoo.com  Tue Aug 13 18:36:27 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 13 Aug 2013 09:36:27 -0700 (PDT)
Subject: [FFmpeg-user] FFmpeg CPU usage
In-Reply-To: <1376405085.55602.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <loom.20130812T220952-607@post.gmane.org>	<20130813084949.d35f6c7e.dashing.meng@gmail.com>
	<1376405085.55602.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <1376411787.54052.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> I have a huge number of videos to transcode. The quality of the

>> videos vary from low to medium. DVD quality is probably the highest
>> quality. I have ffmpeg and libx264 compiled with pthreads and I use
>> -threads 0 in my call to ffmpeg but I'm still not seeing 100% usage
>> on all the cores. I need to maximize cpu usage since I'm not trying
>> to transcode videos for a month straight. 

Also, if you have far more videos than CPU cores, you will be far
better off, assuming you have enough disk bandwidth, by running
a single encode of a single video on each core, and encoding multiple
videos concurrently.? I've done this and the limiting factor for me
is disk I/O.

From ricardohenrylee at gmail.com  Tue Aug 13 20:46:15 2013
From: ricardohenrylee at gmail.com (Richard H Lee)
Date: Tue, 13 Aug 2013 19:46:15 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <CAEnqZEXHLgRUW5Sr_t4Zs6ap=A2Unga5oQwZ8WEJVwNzJ_c1Jw@mail.gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<52076BD6.3050607@gmail.com>
	<20130811142921.GA10827@phare.normalesup.org>
	<1376233087.60147.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<5207BE09.1020001@gmail.com>
	<1376240016.36891.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<52080A5C.2060000@gmail.com>
	<CAEnqZEV9HBa4FmOn-hPWJiEma5qDPSFFNkwAFjrY9od=yeZ23w@mail.gmail.com>
	<52096F61.60007@gmail.com>
	<CAEnqZEXHLgRUW5Sr_t4Zs6ap=A2Unga5oQwZ8WEJVwNzJ_c1Jw@mail.gmail.com>
Message-ID: <520A7EF7.1030301@gmail.com>

On 13/08/2013 07:33, Robert Kr?ger wrote:
> yes, it is relatively new (added a bit more than a year ago) but
> still, if your original command line, that tries to achieve the audio
> stretching/squeezing using resampling causes a/v desync, it should be
> investigated whether your command line is wrong or there is a problem
> in the resampling code.

I stand corrected. Before the addition of the atempo audio filter, it 
does not appear there was a way to seed up audio with ffmpeg.

From dzungng89 at gmail.com  Tue Aug 13 18:26:53 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Tue, 13 Aug 2013 11:26:53 -0500
Subject: [FFmpeg-user] Intermediate video format for Concatenation of video
	and images
Message-ID: <CABOWChw0L_stKhoWbw7dwGbESvMSY+nUpYRqqqRHVOp9mYi1yw@mail.gmail.com>

Hi all,

I'm converting image to videos, and then merging them together (because the
length of the image slideshow is random, not fixed).

1) Image --> video:

ffmpeg -loop 1 -f image2 -i image.png -c:v libx264 -tune stillimage -t 30
video1.mp4

2) Concatenate subvideos:

ffmpeg -f concat -i list.txt -c copy concatenated.mp4

Which format should I use when creating subvideos? I found that mpeg lose
the image quality, while using mp4 the output is unskippable.

From leonard at kcfchurch.org  Wed Aug 14 02:47:30 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Tue, 13 Aug 2013 17:47:30 -0700
Subject: [FFmpeg-user] ffmpeg on Ubuntu 12.04 to Wowza server.
Message-ID: <CA+JzsNQMFVt2mC-8PBtxYdJBcA1Wk6UWkHbtraewGVuvK-dLRg@mail.gmail.com>

After finally getting ffmpeg to compile I've been trying for about 2 days
to get ffmpeg to live stream from my DeckLink Mini Recorder card (capture
device) to a WoWza server using rtmp.  Everytime I start ffmpeg I ALWAYS
get this line "Could not write header for output file #0 (incorrect codec
parameters ?): Operation not permitted" and then ffmpeg terminates.

FMLE seems to work fine (even to my local test trial install of WoWza) on
my old and overburdoned Windows XP machine

I'm running a brand new install of Ubuntu 12.04 Server amd64 on a brand
(butt-spanking) new PC.

This is the command I'm issuing (and the response):

OUTSTREAM="-f mp4 rtmp://192.168.7.2:1935/live/kcfstream.mp4"
bmdcapture -m 14 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -i - -copyts \
  -r 30 -s 360x200 -aspect 16:9 \
  -acodec libfdk_aac -ab 64000 -ar 48000 -ac 2 \
  -vcodec libx264 -vprofile baseline -level 3.1 -pix_fmt yuv420p \
  $OUTSTREAM

ffmpeg version git-2013-08-13-8a7295b Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 13 2013 15:08:47 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  configuration: --prefix=/home/suser/ffmpeg_build
--extra-cflags=-I/home/suser/ffmpeg_build/include --e
xtra-ldflags=-L/home/suser/ffmpeg_build/lib --bindir=/home/suser/bin
--extra-libs=-ldl --enable-gpl --en                      able-libass
--enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora
--enable-libvorb                      is --enable-libvpx --enable-libx264
--enable-libfaac --enable-nonfree --enable-encoder=libfaac
  libavutil      52. 42.100 / 52. 42.100
  libavcodec     55. 27.100 / 55. 27.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, nut, from 'pipe:':
  Metadata:
    encoder         : Lavf54.20.3
  Duration: N/A, start: 0.000000, bitrate: 1536 kb/s
    Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1280x720,
59.94 tbr, 59.94 tbn, 59.94 tbc
    Stream #0:1: Audio: pcm_s16le (PSD[16] / 0x10445350), 48000 Hz, stereo,
s16, 1536 kb/s
[libx264 @ 0x21f5f00] using SAR=80/81
[libx264 @ 0x21f5f00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x21f5f00] profile Constrained Baseline, level 3.1
[libx264 @ 0x21f5f00] 264 - core 135 r2 f0c1c53 - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://w
ww.videolan.org/x264.html- options: cabac=0 ref=3 deblock=1:0:0
analyse=0x1:0x111 me=hex subme=7
psy=1                       psy_rd=1.00:0.00 mixed_ref=1 me_range=16
chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11
fast_pskip=                      1 chroma_qp_offset=-2 threads=12
lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
blura                      y_compat=0 constrained_intra=0 bframes=0
weightp=0 keyint=250 keyint_min=25 scenecut=40
intra_refresh=0                       rc_lookahead=40 rc=crf mbtree=1
crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
[mp4 @ 0x21f59a0] muxer does not support non seekable output
Output #0, mp4, to 'rtmp://192.168.7.2:1935/live/kcfstream.mp4':
  Metadata:
    encoder         : Lavf55.13.102
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
360x200 [SAR 80:81 DAR 16:9],                       q=-1--1, 90k tbn, 30 tbc
    Stream #0:1: Audio: aac (libfdk_aac) ([64][0][0][0] / 0x0040), 48000
Hz, stereo, s16, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> libx264)
  Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
Could not write header for output file #0 (incorrect codec parameters ?):
Operation not permitted


Can somebody tell me what I'm doing wrong?

From ajsinclair at gmail.com  Wed Aug 14 03:04:14 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Wed, 14 Aug 2013 11:04:14 +1000
Subject: [FFmpeg-user] VC1 and WMV3 encoding in FFMPEG
In-Reply-To: <020701ce97aa$c98ae2c0$5ca0a840$@org>
References: <020701ce97aa$c98ae2c0$5ca0a840$@org>
Message-ID: <CAOeAR14NUZP6k+ronpjsmX7To+X7ePNgN28HSoapMqP7+-DeQA@mail.gmail.com>

wmv3 should be in there. Use ffmpeg -codecs to list what is in your build.

Last time I checked no one looked like they were going to do vc-1 but
unless you really need vc-1 just use h.264 and you can mux this into an ASF
wrapper in the case you have some special Windows/WMV requirements.


On 13 August 2013 08:25, Sean McShane <seanm at goldeneraproductions.org>wrote:

> Does anyone know if there is any plan to enable VC1 and/or WMV3 encoding in
> FFMPEG?
>
>
>
> -Sean
>
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From ajsinclair at gmail.com  Wed Aug 14 03:07:04 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Wed, 14 Aug 2013 11:07:04 +1000
Subject: [FFmpeg-user] ffmpeg on Ubuntu 12.04 to Wowza server.
In-Reply-To: <CA+JzsNQMFVt2mC-8PBtxYdJBcA1Wk6UWkHbtraewGVuvK-dLRg@mail.gmail.com>
References: <CA+JzsNQMFVt2mC-8PBtxYdJBcA1Wk6UWkHbtraewGVuvK-dLRg@mail.gmail.com>
Message-ID: <CAOeAR15HNdgHoSZs+-eB_6hgqAeJLifS=x0PtOG4-kCa9tUhBA@mail.gmail.com>

Try using -f flv for pushing the file over rtmp and drop the .mp4 off you
wowza entrypoint.




On 14 August 2013 10:47, Leonard Bogard <leonard at kcfchurch.org> wrote:

> After finally getting ffmpeg to compile I've been trying for about 2 days
> to get ffmpeg to live stream from my DeckLink Mini Recorder card (capture
> device) to a WoWza server using rtmp.  Everytime I start ffmpeg I ALWAYS
> get this line "Could not write header for output file #0 (incorrect codec
> parameters ?): Operation not permitted" and then ffmpeg terminates.
>
> FMLE seems to work fine (even to my local test trial install of WoWza) on
> my old and overburdoned Windows XP machine
>
> I'm running a brand new install of Ubuntu 12.04 Server amd64 on a brand
> (butt-spanking) new PC.
>
> This is the command I'm issuing (and the response):
>
> OUTSTREAM="-f mp4 rtmp://192.168.7.2:1935/live/kcfstream.mp4"
> bmdcapture -m 14 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -i - -copyts \
>   -r 30 -s 360x200 -aspect 16:9 \
>   -acodec libfdk_aac -ab 64000 -ar 48000 -ac 2 \
>   -vcodec libx264 -vprofile baseline -level 3.1 -pix_fmt yuv420p \
>   $OUTSTREAM
>
> ffmpeg version git-2013-08-13-8a7295b Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Aug 13 2013 15:08:47 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
>   configuration: --prefix=/home/suser/ffmpeg_build
> --extra-cflags=-I/home/suser/ffmpeg_build/include --e
> xtra-ldflags=-L/home/suser/ffmpeg_build/lib --bindir=/home/suser/bin
> --extra-libs=-ldl --enable-gpl --en                      able-libass
> --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora
> --enable-libvorb                      is --enable-libvpx --enable-libx264
> --enable-libfaac --enable-nonfree --enable-encoder=libfaac
>   libavutil      52. 42.100 / 52. 42.100
>   libavcodec     55. 27.100 / 55. 27.100
>   libavformat    55. 13.102 / 55. 13.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 82.100 /  3. 82.100
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.103 /  0. 17.103
>   libpostproc    52.  3.100 / 52.  3.100
> Guessed Channel Layout for  Input Stream #0.1 : stereo
> Input #0, nut, from 'pipe:':
>   Metadata:
>     encoder         : Lavf54.20.3
>   Duration: N/A, start: 0.000000, bitrate: 1536 kb/s
>     Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1280x720,
> 59.94 tbr, 59.94 tbn, 59.94 tbc
>     Stream #0:1: Audio: pcm_s16le (PSD[16] / 0x10445350), 48000 Hz, stereo,
> s16, 1536 kb/s
> [libx264 @ 0x21f5f00] using SAR=80/81
> [libx264 @ 0x21f5f00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> AVX
> [libx264 @ 0x21f5f00] profile Constrained Baseline, level 3.1
> [libx264 @ 0x21f5f00] 264 - core 135 r2 f0c1c53 - H.264/MPEG-4 AVC codec -
> Copyleft 2003-2013 - http://w
> ww.videolan.org/x264.html- options: cabac=0 ref=3 deblock=1:0:0
> analyse=0x1:0x111 me=hex subme=7
> psy=1                       psy_rd=1.00:0.00 mixed_ref=1 me_range=16
> chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11
> fast_pskip=                      1 chroma_qp_offset=-2 threads=12
> lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
> blura                      y_compat=0 constrained_intra=0 bframes=0
> weightp=0 keyint=250 keyint_min=25 scenecut=40
> intra_refresh=0                       rc_lookahead=40 rc=crf mbtree=1
> crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> [mp4 @ 0x21f59a0] muxer does not support non seekable output
> Output #0, mp4, to 'rtmp://192.168.7.2:1935/live/kcfstream.mp4':
>   Metadata:
>     encoder         : Lavf55.13.102
>     Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
> 360x200 [SAR 80:81 DAR 16:9],                       q=-1--1, 90k tbn, 30
> tbc
>     Stream #0:1: Audio: aac (libfdk_aac) ([64][0][0][0] / 0x0040), 48000
> Hz, stereo, s16, 64 kb/s
> Stream mapping:
>   Stream #0:0 -> #0:0 (rawvideo -> libx264)
>   Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
> Could not write header for output file #0 (incorrect codec parameters ?):
> Operation not permitted
>
>
> Can somebody tell me what I'm doing wrong?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Wed Aug 14 03:22:10 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 14 Aug 2013 01:22:10 +0000
Subject: [FFmpeg-user] VC1 and WMV3 encoding in FFMPEG
In-Reply-To: <CAOeAR14NUZP6k+ronpjsmX7To+X7ePNgN28HSoapMqP7+-DeQA@mail.gmail.com>
References: <020701ce97aa$c98ae2c0$5ca0a840$@org>
	<CAOeAR14NUZP6k+ronpjsmX7To+X7ePNgN28HSoapMqP7+-DeQA@mail.gmail.com>
Message-ID: <CAPYw7P42TCweA2TSXorO0mwC6BL+-zQgvLSh0uPp5ehS-zFmKA@mail.gmail.com>

On 8/14/13, Andrew Sinclair <ajsinclair at gmail.com> wrote:
> wmv3 should be in there. Use ffmpeg -codecs to list what is in your build.
>
> Last time I checked no one looked like they were going to do vc-1 but
> unless you really need vc-1 just use h.264 and you can mux this into an ASF
> wrapper in the case you have some special Windows/WMV requirements.

There is not such thing as wmv3. And user keep asking same questions
over and over again.

Also why are you top posting?

>
>
> On 13 August 2013 08:25, Sean McShane
> <seanm at goldeneraproductions.org>wrote:
>
>> Does anyone know if there is any plan to enable VC1 and/or WMV3 encoding
>> in
>> FFMPEG?
>>
>>
>>
>> -Sean
>>
>>
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From leonard at kcfchurch.org  Wed Aug 14 03:36:38 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Tue, 13 Aug 2013 18:36:38 -0700
Subject: [FFmpeg-user] ffmpeg on Ubuntu 12.04 to Wowza server.
In-Reply-To: <CAOeAR15HNdgHoSZs+-eB_6hgqAeJLifS=x0PtOG4-kCa9tUhBA@mail.gmail.com>
References: <CA+JzsNQMFVt2mC-8PBtxYdJBcA1Wk6UWkHbtraewGVuvK-dLRg@mail.gmail.com>
	<CAOeAR15HNdgHoSZs+-eB_6hgqAeJLifS=x0PtOG4-kCa9tUhBA@mail.gmail.com>
Message-ID: <CA+JzsNRO-hv6gmeSLZjZbh5nNxukR28p3yweQ9n8AK=+UPa9=g@mail.gmail.com>

I wasn't able to drop .mp4 from the entrypoint because the embeded player
is hardcoded to .mp4 but I did try -f flv and it worked.

Doesn't changing it to -f flv degrade the video quality be dint of the
encoder it uses?

And, is it that the rtmp protocol doesn't like .mp4 container?


On Tue, Aug 13, 2013 at 6:07 PM, Andrew Sinclair <ajsinclair at gmail.com>wrote:

> Try using -f flv for pushing the file over rtmp and drop the .mp4 off you
> wowza entrypoint.
>
>
>
>
> On 14 August 2013 10:47, Leonard Bogard <leonard at kcfchurch.org> wrote:
>
> > After finally getting ffmpeg to compile I've been trying for about 2 days
> > to get ffmpeg to live stream from my DeckLink Mini Recorder card (capture
> > device) to a WoWza server using rtmp.  Everytime I start ffmpeg I ALWAYS
> > get this line "Could not write header for output file #0 (incorrect codec
> > parameters ?): Operation not permitted" and then ffmpeg terminates.
> >
> > FMLE seems to work fine (even to my local test trial install of WoWza) on
> > my old and overburdoned Windows XP machine
> >
> > I'm running a brand new install of Ubuntu 12.04 Server amd64 on a brand
> > (butt-spanking) new PC.
> >
> > This is the command I'm issuing (and the response):
> >
> > OUTSTREAM="-f mp4 rtmp://192.168.7.2:1935/live/kcfstream.mp4"
> > bmdcapture -m 14 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -i - -copyts \
> >   -r 30 -s 360x200 -aspect 16:9 \
> >   -acodec libfdk_aac -ab 64000 -ar 48000 -ac 2 \
> >   -vcodec libx264 -vprofile baseline -level 3.1 -pix_fmt yuv420p \
> >   $OUTSTREAM
> >
> > ffmpeg version git-2013-08-13-8a7295b Copyright (c) 2000-2013 the FFmpeg
> > developers
> >   built on Aug 13 2013 15:08:47 with gcc 4.6 (Ubuntu/Linaro
> 4.6.3-1ubuntu5)
> >   configuration: --prefix=/home/suser/ffmpeg_build
> > --extra-cflags=-I/home/suser/ffmpeg_build/include --e
> > xtra-ldflags=-L/home/suser/ffmpeg_build/lib --bindir=/home/suser/bin
> > --extra-libs=-ldl --enable-gpl --en                      able-libass
> > --enable-libfdk-aac --enable-libmp3lame --enable-libopus
> --enable-libtheora
> > --enable-libvorb                      is --enable-libvpx --enable-libx264
> > --enable-libfaac --enable-nonfree --enable-encoder=libfaac
> >   libavutil      52. 42.100 / 52. 42.100
> >   libavcodec     55. 27.100 / 55. 27.100
> >   libavformat    55. 13.102 / 55. 13.102
> >   libavdevice    55.  3.100 / 55.  3.100
> >   libavfilter     3. 82.100 /  3. 82.100
> >   libswscale      2.  4.100 /  2.  4.100
> >   libswresample   0. 17.103 /  0. 17.103
> >   libpostproc    52.  3.100 / 52.  3.100
> > Guessed Channel Layout for  Input Stream #0.1 : stereo
> > Input #0, nut, from 'pipe:':
> >   Metadata:
> >     encoder         : Lavf54.20.3
> >   Duration: N/A, start: 0.000000, bitrate: 1536 kb/s
> >     Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1280x720,
> > 59.94 tbr, 59.94 tbn, 59.94 tbc
> >     Stream #0:1: Audio: pcm_s16le (PSD[16] / 0x10445350), 48000 Hz,
> stereo,
> > s16, 1536 kb/s
> > [libx264 @ 0x21f5f00] using SAR=80/81
> > [libx264 @ 0x21f5f00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> > AVX
> > [libx264 @ 0x21f5f00] profile Constrained Baseline, level 3.1
> > [libx264 @ 0x21f5f00] 264 - core 135 r2 f0c1c53 - H.264/MPEG-4 AVC codec
> -
> > Copyleft 2003-2013 - http://w
> > ww.videolan.org/x264.html- options: cabac=0 ref=3 deblock=1:0:0
> > analyse=0x1:0x111 me=hex subme=7
> > psy=1                       psy_rd=1.00:0.00 mixed_ref=1 me_range=16
> > chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11
> > fast_pskip=                      1 chroma_qp_offset=-2 threads=12
> > lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
> > blura                      y_compat=0 constrained_intra=0 bframes=0
> > weightp=0 keyint=250 keyint_min=25 scenecut=40
> > intra_refresh=0                       rc_lookahead=40 rc=crf mbtree=1
> > crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> > [mp4 @ 0x21f59a0] muxer does not support non seekable output
> > Output #0, mp4, to 'rtmp://192.168.7.2:1935/live/kcfstream.mp4':
> >   Metadata:
> >     encoder         : Lavf55.13.102
> >     Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
> > 360x200 [SAR 80:81 DAR 16:9],                       q=-1--1, 90k tbn, 30
> > tbc
> >     Stream #0:1: Audio: aac (libfdk_aac) ([64][0][0][0] / 0x0040), 48000
> > Hz, stereo, s16, 64 kb/s
> > Stream mapping:
> >   Stream #0:0 -> #0:0 (rawvideo -> libx264)
> >   Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
> > Could not write header for output file #0 (incorrect codec parameters ?):
> > Operation not permitted
> >
> >
> > Can somebody tell me what I'm doing wrong?
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Wed Aug 14 03:41:44 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 14 Aug 2013 01:41:44 +0000
Subject: [FFmpeg-user] Intermediate video format for Concatenation of
 video and images
In-Reply-To: <CABOWChw0L_stKhoWbw7dwGbESvMSY+nUpYRqqqRHVOp9mYi1yw@mail.gmail.com>
References: <CABOWChw0L_stKhoWbw7dwGbESvMSY+nUpYRqqqRHVOp9mYi1yw@mail.gmail.com>
Message-ID: <CAPYw7P6R6_Vf_yhXEnn+=HMcmKXZRtATpLNCKgDJ1j21nR6obQ@mail.gmail.com>

On 8/13/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> Hi all,
>
> I'm converting image to videos, and then merging them together (because the
> length of the image slideshow is random, not fixed).
>
> 1) Image --> video:
>
> ffmpeg -loop 1 -f image2 -i image.png -c:v libx264 -tune stillimage -t 30
> video1.mp4
>
> 2) Concatenate subvideos:
>
> ffmpeg -f concat -i list.txt -c copy concatenated.mp4
>
> Which format should I use when creating subvideos? I found that mpeg lose
> the image quality, while using mp4 the output is unskippable.

With mpeg you can tweak quality. defaults do not target quality.
Also using mp4 should not really matter.

What by "unskippable output" you mean?

From seanm at goldeneraproductions.org  Wed Aug 14 03:50:11 2013
From: seanm at goldeneraproductions.org (Sean)
Date: Tue, 13 Aug 2013 18:50:11 -0700
Subject: [FFmpeg-user] VC1 and WMV3 encoding in FFMPEG
In-Reply-To: <CAOeAR14NUZP6k+ronpjsmX7To+X7ePNgN28HSoapMqP7+-DeQA@mail.gmail.com>
References: <020701ce97aa$c98ae2c0$5ca0a840$@org>
	<CAOeAR14NUZP6k+ronpjsmX7To+X7ePNgN28HSoapMqP7+-DeQA@mail.gmail.com>
Message-ID: <024101ce9890$9cfef430$d6fcdc90$@org>

On 13 August 2013 08:25, Sean McShane <seanm at goldeneraproductions.org>wrote:

> Does anyone know if there is any plan to enable VC1 and/or WMV3 encoding
in
> FFMPEG?
>
>
>
> -Sean
>
>

>>wmv3 should be in there. Use ffmpeg -codecs to list what is in your build.
>>
>>Last time I checked no one looked like they were going to do vc-1 but
>>unless you really need vc-1 just use h.264 and you can mux this into an
ASF
>>wrapper in the case you have some special Windows/WMV requirements.


I am using this Windows build version:
=============================
ffmpeg version N-54960-gf3f4e13 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 26 2013 19:16:37 with gcc 4.7.3 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig 
--enable-frei0r --enable-gnutls --enable-iconv --enable-libass
--enable-libbluray --enable-libcaca --enable-libfreetype 
--enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-libschroedinger --enable-libsoxr --enable-libspeex 
--enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 81.103 /  3. 81.103
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
==============================

When I run the -codecs flag, I get this under the wmv section:
==============================
D.V.L. vc1                  SMPTE VC-1
DEV.L. wmv1                 Windows Media Video 7
DEV.L. wmv2                 Windows Media Video 8
D.V.L. wmv3                 Windows Media Video 9
==============================

Looks like encoding is only enabled for wmv1 (WMV7) and wmv2 (WMV8), but not
for wmv3 (WMV9) and vc1.

Unfortunately, I do have specific Windows/wmv requirements. 

-Sean

_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user






From onemda at gmail.com  Wed Aug 14 04:03:39 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 14 Aug 2013 02:03:39 +0000
Subject: [FFmpeg-user] VC1 and WMV3 encoding in FFMPEG
In-Reply-To: <024101ce9890$9cfef430$d6fcdc90$@org>
References: <020701ce97aa$c98ae2c0$5ca0a840$@org>
	<CAOeAR14NUZP6k+ronpjsmX7To+X7ePNgN28HSoapMqP7+-DeQA@mail.gmail.com>
	<024101ce9890$9cfef430$d6fcdc90$@org>
Message-ID: <CAPYw7P7GjA4xU1B265GSv_R7xzrb_=ri9ZYHThVG6LyG+_iaSQ@mail.gmail.com>

On 8/14/13, Sean <seanm at goldeneraproductions.org> wrote:
> On 13 August 2013 08:25, Sean McShane
> <seanm at goldeneraproductions.org>wrote:
>
>> Does anyone know if there is any plan to enable VC1 and/or WMV3 encoding
> in
>> FFMPEG?
>>
>>
>>
>> -Sean
>>
>>
>
>>>wmv3 should be in there. Use ffmpeg -codecs to list what is in your
>>> build.
>>>
>>>Last time I checked no one looked like they were going to do vc-1 but
>>>unless you really need vc-1 just use h.264 and you can mux this into an
> ASF
>>>wrapper in the case you have some special Windows/WMV requirements.
>
>
> I am using this Windows build version:
> =============================
> ffmpeg version N-54960-gf3f4e13 Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Jul 26 2013 19:16:37 with gcc 4.7.3 (GCC)
> configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig
> --enable-frei0r --enable-gnutls --enable-iconv --enable-libass
> --enable-libbluray --enable-libcaca --enable-libfreetype
> --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
> --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-libopus --enable-librtmp
> --enable-libschroedinger --enable-libsoxr --enable-libspeex
> --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
>   libavutil      52. 40.100 / 52. 40.100
>   libavcodec     55. 19.100 / 55. 19.100
>   libavformat    55. 12.102 / 55. 12.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 81.103 /  3. 81.103
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.103 /  0. 17.103
>   libpostproc    52.  3.100 / 52.  3.100
> ==============================
>
> When I run the -codecs flag, I get this under the wmv section:
> ==============================
> D.V.L. vc1                  SMPTE VC-1
> DEV.L. wmv1                 Windows Media Video 7
> DEV.L. wmv2                 Windows Media Video 8
> D.V.L. wmv3                 Windows Media Video 9
> ==============================
>
> Looks like encoding is only enabled for wmv1 (WMV7) and wmv2 (WMV8), but
> not
> for wmv3 (WMV9) and vc1.
>
> Unfortunately, I do have specific Windows/wmv requirements.

Unfortunatly there are no plans to add such encoders.

From jhbell50 at hotmail.com  Wed Aug 14 01:16:19 2013
From: jhbell50 at hotmail.com (john Bell)
Date: Tue, 13 Aug 2013 19:16:19 -0400
Subject: [FFmpeg-user] cookie Syntax help
Message-ID: <BLU178-W167A142216356CD25A72ECF440@phx.gbl>

I an issuing the following command(with spaces following the ;'s)
 
ffplay -i -cookies "eurosp1sp1keys=t4HxT5V8c4myQle3HO_6YA; eurosp1
sp1keyt=1376432802; path=\; domain=www.hqstream.tv; \n" "http://50.22.223.42/hqs
tream/eurosp1sp1/playlist.m3u8?keys=t4HxT5V8c4myQle3HO_6YA&keyt=1376432802"

and the result is:
ffplay version N-55020-g768e40b Copyright (c) 2003-2013 the FFmpeg developers
 ...
[http @ 03ddac00] Invalid cookie found, no value, path or domain specified
 
I believe the command follows the code requirements as posted below. 
Can anyone tell me where I am going wrong? 
 
The ffmpeg cookie code reads as follows:
 
 438 /** 439  * Create a string containing cookie values for use as a HTTP cookie header 440  * field value for a particular path and domain from the cookie values stored in 441  * the HTTP protocol context. The cookie string is stored in *cookies. 442  * 443  * @return a negative value if an error condition occurred, 0 otherwise 444  */ 445 static int get_cookies(HTTPContext *s, char **cookies, const char *path, 446                        const char *domain) 447 { 448     // cookie strings will look like Set-Cookie header field values.  Multiple 449     // Set-Cookie fields will result in multiple values delimited by a newline 450     int ret = 0; 451     char *next, *cookie, *set_cookies = av_strdup(s->cookies), *cset_cookies = set_cookies; 452  453     if (!set_cookies) return AVERROR(EINVAL); 454  455     *cookies = NULL; 456     while ((cookie = av_strtok(set_cookies, "\n", &next))) { 457         int domain_offset = 0; 458         char *param, *next_param, *cdomain = NULL, *cpath = NULL, *cvalue = NULL; 459         set_cookies = NULL; 460   461         while ((param = av_strtok(cookie, "; ", &next_param))) { 462             cookie = NULL; 463             if        (!av_strncasecmp("path=",   param, 5)) { 464                 av_free(cpath); 465                 cpath = av_strdup(&param[5]); 466             } else if (!av_strncasecmp("domain=", param, 7)) { 467                 av_free(cdomain); 468                 cdomain = av_strdup(&param[7]); 469             } else if (!av_strncasecmp("secure",  param, 6) || 470                        !av_strncasecmp("comment", param, 7) || 471                        !av_strncasecmp("max-age", param, 7) || 472                        !av_strncasecmp("version", param, 7)) { 473                 // ignore Comment, Max-Age, Secure and Version 474             } else { 475                 av_free(cvalue); 476                 cvalue = av_strdup(param); 477             } 478         } 479         if (!cdomain) 480             cdomain = av_strdup(domain); 481  482         // ensure all of the necessary values are valid 483         if (!cdomain || !cpath || !cvalue) { 484             av_log(s, AV_LOG_WARNING, 485                    "Invalid cookie found, no value, path or domain specified\n"); 486             goto done_cookie; 487         } 488  		 	   		  

From leonard at kcfchurch.org  Wed Aug 14 04:24:31 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Tue, 13 Aug 2013 19:24:31 -0700
Subject: [FFmpeg-user] ffmpeg on Ubuntu 12.04 to Wowza server.
In-Reply-To: <CAOeAR15HNdgHoSZs+-eB_6hgqAeJLifS=x0PtOG4-kCa9tUhBA@mail.gmail.com>
References: <CA+JzsNQMFVt2mC-8PBtxYdJBcA1Wk6UWkHbtraewGVuvK-dLRg@mail.gmail.com>
	<CAOeAR15HNdgHoSZs+-eB_6hgqAeJLifS=x0PtOG4-kCa9tUhBA@mail.gmail.com>
Message-ID: <CA+JzsNQaHy2rd=Pub7Z0r41bdAWQGKZJ9vH0DNp+k-=snbTJDw@mail.gmail.com>

After changing to -f flv it started working but then after about 10 minutes
ffmpeg dies with messages:

[flv @ 0x37b6280] Failed to update header with correct duration.
[flv @ 0x37b6280] Failed to update header with correct filesize.

Is it possible I compiled ffmpeg incorrectly?

From cr-ffmpeg at ponzo.net  Wed Aug 14 04:30:19 2013
From: cr-ffmpeg at ponzo.net (Scott Doty)
Date: Tue, 13 Aug 2013 19:30:19 -0700
Subject: [FFmpeg-user] ffmpeg on Ubuntu 12.04 to Wowza server.
In-Reply-To: <CA+JzsNQMFVt2mC-8PBtxYdJBcA1Wk6UWkHbtraewGVuvK-dLRg@mail.gmail.com>
References: <CA+JzsNQMFVt2mC-8PBtxYdJBcA1Wk6UWkHbtraewGVuvK-dLRg@mail.gmail.com>
Message-ID: <520AEBBB.9040806@ponzo.net>

On 08/13/2013 05:47 PM, Leonard Bogard wrote:
> After finally getting ffmpeg to compile I've been trying for about 2 days
> to get ffmpeg to live stream from my DeckLink Mini Recorder card (capture
> device) to a WoWza server using rtmp.  Everytime I start ffmpeg I ALWAYS
> get this line "Could not write header for output file #0 (incorrect codec
> parameters ?): Operation not permitted" and then ffmpeg terminates.
>
> FMLE seems to work fine (even to my local test trial install of WoWza) on
> my old and overburdoned Windows XP machine
>
> I'm running a brand new install of Ubuntu 12.04 Server amd64 on a brand
> (butt-spanking) new PC.
>
> This is the command I'm issuing (and the response):
>
> OUTSTREAM="-f mp4 rtmp://192.168.7.2:1935/live/kcfstream.mp4"
> bmdcapture -m 14 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -i - -copyts \
>   -r 30 -s 360x200 -aspect 16:9 \
>   -acodec libfdk_aac -ab 64000 -ar 48000 -ac 2 \
>   -vcodec libx264 -vprofile baseline -level 3.1 -pix_fmt yuv420p \
>   $OUTSTREAM

> Stream mapping:
>   Stream #0:0 -> #0:0 (rawvideo -> libx264)
>   Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
> Could not write header for output file #0 (incorrect codec parameters ?):
> Operation not permitted
>
>
> Can somebody tell me what I'm doing wrong?
>

I've run into this while trying to stream to Wowza with some* aac audio
libraries (to Twitch.tv or Justin.tv) -- but if I use mp3 for audio,
it's all good.

Maybe try with a different audio codec?

 -Scott

p.s. * "some" being a placeholder for "the ones I tried so long ago that
I no longer remember".


From leonard at kcfchurch.org  Wed Aug 14 07:12:34 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Tue, 13 Aug 2013 22:12:34 -0700
Subject: [FFmpeg-user] "Failed to update header with correct duration."
Message-ID: <CA+JzsNQ1Q=HXmyuOsiG5Nb0-Sz1iM47CJJkqtA-DPRTTzffQAQ@mail.gmail.com>

Whether I stop ffmpeg manually or it stops on its own after some randomly
short period of time fail it displays two error messages:
[flv @ 0x2279ba0] Failed to update header with correct duration
[flv @ 0x2279ba0] Failed to update header with correct filesize.

I'm trying to stream from ffmpeg to a wowza server, here's my start script
and output:

OUTSTREAM="-f flv rtmp://192.168.7.2:1935/live/kcfstream"

OUTAUDIO="-acodec libfdk_aac -af volume=8.0 -ab 64000 -ar 48000 -ac 2"
#OUTAUDIO="-acodec libmp3lame -af volume=8.0 -ab 96000 -ar 44100 -ac 1"

bmdcapture -m 14 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -i - -copyts \
  -r 30 -s 640x360 -aspect 16:9 \
  -threads 2 \
  $OUTAUDIO \
  -vcodec libx264 -vb 500k -vprofile baseline -level 3.1 -pix_fmt yuv420p \
  -strict -2 \
  $OUTSTREAM

~$ ./exp2.sh
ffmpeg version git-2013-08-13-8a7295b Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 13 2013 21:30:35 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  configuration: --prefix=/home/suser/ffmpeg_build
--extra-cflags=-I/home/suser/ffmpeg_build/include
--extra-ldflags=-L/home/suser/ffmpeg_build/lib --bindir=/home/suser/bin
--extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac
--enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libfaac --enable-version3
--enable-nonfree --enable-encoder=libfaac
  libavutil      52. 42.100 / 52. 42.100
  libavcodec     55. 27.100 / 55. 27.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, nut, from 'pipe:':
  Metadata:
    encoder         : Lavf54.20.3
  Duration: N/A, start: 0.000000, bitrate: 1536 kb/s
    Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1280x720,
59.94 tbr, 59.94 tbn, 59.94 tbc
    Stream #0:1: Audio: pcm_s16le (PSD[16] / 0x10445350), 48000 Hz, stereo,
s16, 1536 kb/s
[libx264 @ 0x24b9120] using SAR=1/1
[libx264 @ 0x24b9120] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x24b9120] profile Constrained Baseline, level 3.1
[libx264 @ 0x24b9120] 264 - core 135 r2 f0c1c53 - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0
ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11
fast_pskip=1 chroma_qp_offset=-2 threads=2 lookahead_threads=1
sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25
scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=500
ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, flv, to 'rtmp://192.168.7.2:1935/live/kcfstream':
  Metadata:
    encoder         : Lavf55.13.102
    Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p,
640x360 [SAR 1:1 DAR 16:9], q=-1--1, 500 kb/s, 1k tbn, 30 tbc
    Stream #0:1: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 48000
Hz, stereo, s16, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> libx264)
  Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
^C[flv @ 0x24b8ba0] Failed to update header with correct duration.e=
435.2kbits/s dup=0 drop=209
[flv @ 0x24b8ba0] Failed to update header with correct filesize.
frame=  225 fps= 31 q=-1.0 Lsize=     528kB time=00:00:07.50 bitrate=
576.7kbits/s dup=0 drop=221
video:458kB audio:59kB subtitle:0 global headers:0kB muxing overhead
2.044314%
[libx264 @ 0x24b9120] frame I:11    Avg QP:23.52  size:  3701
[libx264 @ 0x24b9120] frame P:214   Avg QP:29.68  size:  2000
[libx264 @ 0x24b9120] mb I  I16..4: 88.1%  0.0% 11.9%
[libx264 @ 0x24b9120] mb P  I16..4:  4.4%  0.0%  1.2%  P16..4: 28.9%  7.4%
1.8%  0.0%  0.0%    skip:56.2%
[libx264 @ 0x24b9120] final ratefactor: 27.48
[libx264 @ 0x24b9120] coded y,uvDC,uvAC intra: 15.2% 47.5% 7.5% inter: 9.3%
13.1% 1.2%
[libx264 @ 0x24b9120] i16 v,h,dc,p: 41% 23%  6% 31%
[libx264 @ 0x24b9120] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 14% 45% 18%  3%  4%
3%  6%  2%  4%
[libx264 @ 0x24b9120] i8c dc,h,v,p: 62% 20% 10%  8%
[libx264 @ 0x24b9120] ref P L0: 85.8%  8.7%  5.5%
[libx264 @ 0x24b9120] kb/s:499.96
Received signal 2: terminating.

What am I doing wrong?  Thanks in advance for any help.

From j+ffmpeg at lundstocholm.dk  Wed Aug 14 10:09:30 2013
From: j+ffmpeg at lundstocholm.dk (jlundstocholm)
Date: Wed, 14 Aug 2013 01:09:30 -0700 (PDT)
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
Message-ID: <1376467770105-4660699.post@n4.nabble.com>

I am trying to use ffmpeg to remove commercials from films I have recorded on
my DVR. I have used the guide at
http://www.joeldare.com/wiki/video:cut_video_with_ffmpeg .

So I am using it like this:

ffmpeg -ss 213 -i MySourceMovie.m4v testoutput.m4v

Above I would like to cut the first 213 seconds away from the video

The result is good with really good quality.

However - is there any way to prevent ffmpeg to run thru the entire video
when doing this? For a TV-show with 2-3 commercial breaks it would take a
couple of hours to process a 25-minute TV-show.

And - the command aboce results in a testoutput.m4v-file that contains what
I would like to cut out. Is there any way that I can "reverse" this such
that it results in the original video /without/ the cut-out piece instead?
Otherwise I would end up with a number of video files that I would need to
stitch together again.

Thanks,



/Jesper
Copenhagen, Denmark




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From dashing.meng at gmail.com  Wed Aug 14 11:08:43 2013
From: dashing.meng at gmail.com (littlebat)
Date: Wed, 14 Aug 2013 17:08:43 +0800
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <1376467770105-4660699.post@n4.nabble.com>
References: <1376467770105-4660699.post@n4.nabble.com>
Message-ID: <20130814170843.5c08405b.dashing.meng@gmail.com>

On Wed, 14 Aug 2013 01:09:30 -0700 (PDT)
jlundstocholm <j+ffmpeg at lundstocholm.dk> wrote:

> I am trying to use ffmpeg to remove commercials from films I have
> recorded on my DVR. I have used the guide at
> http://www.joeldare.com/wiki/video:cut_video_with_ffmpeg .
> 
> So I am using it like this:
> 
> ffmpeg -ss 213 -i MySourceMovie.m4v testoutput.m4v
> 
> Above I would like to cut the first 213 seconds away from the video
> 
> The result is good with really good quality.
> 
> However - is there any way to prevent ffmpeg to run thru the entire
> video when doing this? For a TV-show with 2-3 commercial breaks it
> would take a couple of hours to process a 25-minute TV-show.

Try -c:v copy -c:a copy , this avoid re-encode the streams.

> 
> And - the command aboce results in a testoutput.m4v-file that
> contains what I would like to cut out. Is there any way that I can
> "reverse" this such that it results in the original video /without/
> the cut-out piece instead? Otherwise I would end up with a number of
> video files that I would need to stitch together again.
> 

It seems no way to avoid re-jointing the videos after you cut some
parts if I have not misunderstanded your meaning. 

From horsley1953 at gmail.com  Wed Aug 14 12:14:50 2013
From: horsley1953 at gmail.com (Tom Horsley)
Date: Wed, 14 Aug 2013 06:14:50 -0400
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <1376467770105-4660699.post@n4.nabble.com>
References: <1376467770105-4660699.post@n4.nabble.com>
Message-ID: <20130814061450.6ec7002a@zooty>

On Wed, 14 Aug 2013 01:09:30 -0700 (PDT)
jlundstocholm wrote:

> I am trying to use ffmpeg to remove commercials from films I have recorded on
> my DVR.

Here's a description of the scripts I've been using:

http://home.comcast.net/~tomhorsley/game/ffmpeg.html

I produce several chunks of the video between the
commercials and use the concat filter to stich them
back together.

From j+ffmpeg at lundstocholm.dk  Wed Aug 14 12:08:10 2013
From: j+ffmpeg at lundstocholm.dk (jlundstocholm)
Date: Wed, 14 Aug 2013 03:08:10 -0700 (PDT)
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <20130814170843.5c08405b.dashing.meng@gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
Message-ID: <1376474890384-4660701.post@n4.nabble.com>

Hi,

> Try -c:v copy -c:a copy , this avoid re-encode the streams.

Ok - should I just add those as parameters to the ffmpeg-call? As in:

ffmpeg -ss 213 -c:v copy -c:a copy -i MySourceMovie.m4v testoutput.m4v

The above makes ffmpeg puke on

"unknown decoder 'copy'

ffmpeg -h tells me that the -c option needs a "codec" paramater?

> It seems no way to avoid re-jointing the videos after you cut some 
> parts if I have not misunderstanded your meaning. 

Ok - this latter part is not a big deal for me since my media center (Plex)
supports streaming of split-up files (as far as I remember, at least)

Thanks for you reply :-)

/Jesper
Copenhagen, Denmark



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From sjh431 at gmail.com  Wed Aug 14 13:41:08 2013
From: sjh431 at gmail.com (jhseo)
Date: Wed, 14 Aug 2013 20:41:08 +0900
Subject: [FFmpeg-user] problem using http input
Message-ID: <014f01ce98e3$2ce8b750$86ba25f0$@gmail.com>

Hi,

 

I used http input at avformat_open_input( ). (mp4 extension)

When ffmpeg get the stream, huge http GET requests occur. ( 31 time per sec,
a/v frame unit(based moov info) )

ffmpeg and the storage use local connection.

And "Content-Length" field did not specified range of end(ex.
Content-Length: 20- )

 

Can I control this case?

Because, the storage is suffer from http requests("swift storage", The
storage is preparing to send file to the end for every request.)

 

Give me a comment please.

 


From onemda at gmail.com  Wed Aug 14 14:18:19 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 14 Aug 2013 12:18:19 +0000
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <1376474890384-4660701.post@n4.nabble.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
Message-ID: <CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>

On 8/14/13, jlundstocholm <j+ffmpeg at lundstocholm.dk> wrote:
> Hi,
>
>> Try -c:v copy -c:a copy , this avoid re-encode the streams.
>
> Ok - should I just add those as parameters to the ffmpeg-call? As in:
>
> ffmpeg -ss 213 -c:v copy -c:a copy -i MySourceMovie.m4v testoutput.m4v
>

It should be:

ffmpeg -ss 213 -i MySourceMovie.m4v -c:v copy -c:a copy testoutput.m4v

> The above makes ffmpeg puke on
>
> "unknown decoder 'copy'
>
> ffmpeg -h tells me that the -c option needs a "codec" paramater?
>
>> It seems no way to avoid re-jointing the videos after you cut some
>> parts if I have not misunderstanded your meaning.
>
> Ok - this latter part is not a big deal for me since my media center (Plex)
> supports streaming of split-up files (as far as I remember, at least)
>
> Thanks for you reply :-)
>
> /Jesper
> Copenhagen, Denmark
>
>
>
> --
> View this message in context:
> http://ffmpeg-users.933282.n4.nabble.com/Using-ffmpeg-to-remove-commercials-from-video-file-tp4660699p4660701.html
> Sent from the FFmpeg-users mailing list archive at Nabble.com.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From francois.visagie at gmail.com  Wed Aug 14 14:36:32 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Wed, 14 Aug 2013 14:36:32 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
In-Reply-To: <20130807180030.GB14079@phare.normalesup.org>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>	<51FEB571.2080701@gmail.com>	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>	<52018A9E.4050604@gmail.com>	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>	<20130807145029.GB25535@phare.normalesup.org>	<005501ce9389$548a5e80$fd9f1b80$@gmail.com>
	<20130807180030.GB14079@phare.normalesup.org>
Message-ID: <000301ce98ea$eb8ae280$c2a0a780$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Nicolas George
> Sent: 07 August 2013 20:01
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] Preserving perceived loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Le decadi 20 thermidor, an CCXXI, Francois Visagie a ?crit :
> > Is it therefore correct to say that:
> > 	* the only input codec-independent way of downmixing to stereo is
> > ?-ac 2?/?-filter:a aformat=channel_layouts=stereo?/?-filter:a
> > aresample=ocl=3? (which now all behave the same?), and
> 
> Yes. Note that it is always safe to specify both that and "-
> request_channels 2": codecs that do not support it will just ignore the option,
> and if the option is supported, the filters will just do nothing.
> 
> > 	* if one wants to preserve perceived input volume one needs to
> adjust
> > gain during encoding?
> 
> Yes, but if you do that, unless your input was never at peak level, you will get
> clipping, and that is probably worse than low volume.
> 
> > Further to that, for a given energy level per input channel, does the
> > current down-mixing mechanism produce differing output energy levels
> > depending on the _number_ of input channels? I.e. is it expected that
> > different input layouts (with the same energy level per channel) would
> > require different gain factors for equally loud outputs, or will one
> > be able to find a suitable gain factor and use that regardless of
> > number of input channels?
> 
> Well, of course, it depends on the number of input channels. If you want to
> mix one channel into one, you do not need to lower the volume. If you want
> to mix forty-two channels into one, you need to divide the amplitude by
> forty-two to avoid clipping. What it does to energy depends on the input. If
> the channels are in phase, the energy is preserved; if they are not, each the
> energy of each channel is divided by forty-two squared, and then the
> energies are summed, the net result is a division by forty-two.
> 

Is it possible to normalise audio levels using ffmpeg? The 'pan' filter documentation mentions:

"If the ?=? in a channel specification is replaced by ?<?, then the gains for that specification will be renormalized so that the total is 1, thus avoiding clipping noise."

I.e., having downmixed to stereo, can one expect correct normalisation from '-filter:a pan=stereo:c0<c0:c1<c1'?

If not, does ffmpeg provide a better mechanism, or is something like that in planning?

Thanks,
Francois

> lswr uses rather tricky coefficients to match the standard loudness and
> geometry of speakers. You can get the exact matrix with -loglevel debug. For
> example, here is the 5.1 -> stereo matrix:
> 
> 0.414214 0.000000 0.292893 0.000000 0.292893 0.000000
> 0.000000 0.414214 0.292893 0.000000 0.000000 0.292893
> 
> That means: out_left = 0.414 front_left + 0.293 center + 0.293 back_left and
> the symmetrical formula for right; note that LFE is discarded.
> 
> Regards,
> 
> --
>   Nicolas George


From nicolas.george at normalesup.org  Wed Aug 14 14:44:02 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 14 Aug 2013 14:44:02 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <000301ce98ea$eb8ae280$c2a0a780$@gmail.com>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>
	<51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>
	<20130807145029.GB25535@phare.normalesup.org>
	<005501ce9389$548a5e80$fd9f1b80$@gmail.com>
	<20130807180030.GB14079@phare.normalesup.org>
	<000301ce98ea$eb8ae280$c2a0a780$@gmail.com>
Message-ID: <20130814124402.GA16262@phare.normalesup.org>

Le septidi 27 thermidor, an CCXXI, Francois Visagie a ?crit?:
> Is it possible to normalise audio levels using ffmpeg? The 'pan' filter
> documentation mentions:

Define "normalize".

> "If the ?=? in a channel specification is replaced by ?<?, then the gains
> for that specification will be renormalized so that the total is 1, thus
> avoiding clipping noise."

Read carefully: "so that the total is 1", "avoiding clipping". That is
exactly what was discussed in this thread.

> I.e., having downmixed to stereo, can one expect correct normalisation
> from '-filter:a pan=stereo:c0<c0:c1<c1'?

This filter does nothing, obviously.

> If not, does ffmpeg provide a better mechanism, or is something like that
> in planning?

There are mechanism, they are mostly good, but you are not specific enough.

Regards,

-- 
  Nicolas George
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From francois.visagie at gmail.com  Wed Aug 14 16:31:46 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Wed, 14 Aug 2013 16:31:46 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
In-Reply-To: <20130814124402.GA16262@phare.normalesup.org>
References: <000901ce9028$6e9d0980$4bd71c80$@gmail.com>	<51FEB571.2080701@gmail.com>	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>	<52018A9E.4050604@gmail.com>	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>	<20130807145029.GB25535@phare.normalesup.org>	<005501ce9389$548a5e80$fd9f1b80$@gmail.com>	<20130807180030.GB14079@phare.normalesup.org>	<000301ce98ea$eb8ae280$c2a0a780$@gmail.com>
	<20130814124402.GA16262@phare.normalesup.org>
Message-ID: <004001ce98fb$06644190$132cc4b0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Nicolas George
> Sent: 14 August 2013 14:44
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] Preserving perceived loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Le septidi 27 thermidor, an CCXXI, Francois Visagie a ?crit :
> > Is it possible to normalise audio levels using ffmpeg? The 'pan'
> > filter documentation mentions:
> 
> Define "normalize".
> 
> > "If the ?=? in a channel specification is replaced by ?<?, then the
> > gains for that specification will be renormalized so that the total is
> > 1, thus avoiding clipping noise."
> 
> Read carefully: "so that the total is 1", "avoiding clipping". That is exactly what
> was discussed in this thread.
> 
> > I.e., having downmixed to stereo, can one expect correct normalisation
> > from '-filter:a pan=stereo:c0<c0:c1<c1'?
> 
> This filter does nothing, obviously.
> 
> > If not, does ffmpeg provide a better mechanism, or is something like
> > that in planning?
> 
> There are mechanism, they are mostly good, but you are not specific
> enough.
> 

Apologies, I do not know enough about this field to express myself clearly. I'll try to improve: I'm looking for an ffmpeg mechanism that automatically determines and uniformly applies to the audio stream a gain factor that ensures maximum output levels without clipping. An added benefit would be the ability to specify the maximum permitted output level as a percentage of maximum possible output level. In other words, something similar to DGIndex's normalisation:
"Immediately, a pass will be made over the input files and a gain factor (pre-scale ratio) will be determined and stored for later use. This gain factor is such that when the audio is amplified by this factor, the highest sound peak will be set to the percentage of the maximum possible audio level set by the normalization selected. So, if your normalization is set to 50%, the loudest sound will be set to 50% of maximum."

How can this be accomplished in ffmpeg?

Thanks,
Francois


From nicolas.george at normalesup.org  Wed Aug 14 16:44:59 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 14 Aug 2013 16:44:59 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
 audio from 5.1 AC3 to stereo AAC
In-Reply-To: <004001ce98fb$06644190$132cc4b0$@gmail.com>
References: <51FEB571.2080701@gmail.com>
	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>
	<52018A9E.4050604@gmail.com>
	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>
	<20130807145029.GB25535@phare.normalesup.org>
	<005501ce9389$548a5e80$fd9f1b80$@gmail.com>
	<20130807180030.GB14079@phare.normalesup.org>
	<000301ce98ea$eb8ae280$c2a0a780$@gmail.com>
	<20130814124402.GA16262@phare.normalesup.org>
	<004001ce98fb$06644190$132cc4b0$@gmail.com>
Message-ID: <20130814144459.GA31873@phare.normalesup.org>

Le septidi 27 thermidor, an CCXXI, Francois Visagie a ?crit?:
> Apologies, I do not know enough about this field to express myself
> clearly. I'll try to improve: I'm looking for an ffmpeg mechanism that
> automatically determines and uniformly applies to the audio stream a gain
> factor that ensures maximum output levels without clipping.

What you are looking for is the output of volumedetect. See the
documentation for explanations on what every field means.

>							      An added
> benefit would be the ability to specify the maximum permitted output level
> as a percentage of maximum possible output level.

volumedetect provides the mean volume of the file, you can do the arithmetic
from there. ebur128 provides a more realistic measure of the mean volume,
but it is slower.

Any such method requires two passes to work correctly, since volume peaks
can be present at the very end of the stream.

Regards,

-- 
  Nicolas George
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From diljit77 at gmail.com  Wed Aug 14 18:53:06 2013
From: diljit77 at gmail.com (Diljit Singh)
Date: Wed, 14 Aug 2013 22:23:06 +0530
Subject: [FFmpeg-user] Controlling moov atom file size
Message-ID: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>

Dear All,

Please suggest the way to control moov atom file size while transcoding a
media file with the help of ffmpeg.

Regards,
Diljit Singh

Sent from my iPad

From werner.robitza at gmail.com  Wed Aug 14 19:56:17 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Wed, 14 Aug 2013 19:56:17 +0200
Subject: [FFmpeg-user] Controlling moov atom file size
In-Reply-To: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>
References: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>
Message-ID: <CAN0LTaNwzmga5NnwwKbSgX=Z50q6aQnUMZVGLQ9a2RUtcRBkzw@mail.gmail.com>

> moov atom file size
>

This is a little ambiguous. The moov atom is not a file.
Do you want to control the size of the atom?

See the -moov_size option ? should achieve what you need.

From diljit77 at gmail.com  Wed Aug 14 20:14:02 2013
From: diljit77 at gmail.com (Diljit Singh)
Date: Wed, 14 Aug 2013 23:44:02 +0530
Subject: [FFmpeg-user] Controlling moov atom file size
In-Reply-To: <CAN0LTaNwzmga5NnwwKbSgX=Z50q6aQnUMZVGLQ9a2RUtcRBkzw@mail.gmail.com>
References: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>
	<CAN0LTaNwzmga5NnwwKbSgX=Z50q6aQnUMZVGLQ9a2RUtcRBkzw@mail.gmail.com>
Message-ID: <CAKH46aRCzc4AMQRxp+71td4iVoc0gP=tKTf-VXAZTEau9EC_oA@mail.gmail.com>

Thanks,

Actually the problem which we were facing was high latency while playing
Video through CDN. This was happening due to the fact that content was not
hot in the cache of edge servers. It was suggest by the tech team that moov
file is large and it should reduced below a certain threshold.

Somewhere in mails on this user group i also read that there is another
flag "-movflags faststart" which can be used to achieve similar purpose.

please suggest which could be better approach.

regards,
Diljit Singh


On Wed, Aug 14, 2013 at 11:26 PM, Werner Robitza
<werner.robitza at gmail.com>wrote:

> > moov atom file size
> >
>
> This is a little ambiguous. The moov atom is not a file.
> Do you want to control the size of the atom?
>
> See the -moov_size option ? should achieve what you need.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From h.reindl at thelounge.net  Wed Aug 14 20:19:08 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Wed, 14 Aug 2013 20:19:08 +0200
Subject: [FFmpeg-user] Controlling moov atom file size
In-Reply-To: <CAKH46aRCzc4AMQRxp+71td4iVoc0gP=tKTf-VXAZTEau9EC_oA@mail.gmail.com>
References: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>
	<CAN0LTaNwzmga5NnwwKbSgX=Z50q6aQnUMZVGLQ9a2RUtcRBkzw@mail.gmail.com>
	<CAKH46aRCzc4AMQRxp+71td4iVoc0gP=tKTf-VXAZTEau9EC_oA@mail.gmail.com>
Message-ID: <520BCA1C.3070101@thelounge.net>



Am 14.08.2013 20:14, schrieb Diljit Singh:
> Actually the problem which we were facing was high latency while playing
> Video through CDN. This was happening due to the fact that content was not
> hot in the cache of edge servers. It was suggest by the tech team that moov
> file is large and it should reduced below a certain threshold.
> 
> Somewhere in mails on this user group i also read that there is another
> flag "-movflags faststart" which can be used to achieve similar purpose.
> 
> please suggest which could be better approach

"-movflags +faststart" is nothing else than
http://www.stoimen.com/blog/2010/11/12/how-to-make-mp4-progressive-with-qt-faststart/
http://stackoverflow.com/questions/8061798/post-processing-in-ffmpeg-to-move-moov-atom-in-mp4-files-qt-faststart

"file size" is plain stupid in this context since "file size" have a specific
meaning and nothing to do with faststart

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From werner.robitza at gmail.com  Wed Aug 14 20:38:56 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Wed, 14 Aug 2013 20:38:56 +0200
Subject: [FFmpeg-user] Controlling moov atom file size
In-Reply-To: <CAKH46aRCzc4AMQRxp+71td4iVoc0gP=tKTf-VXAZTEau9EC_oA@mail.gmail.com>
References: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>
	<CAN0LTaNwzmga5NnwwKbSgX=Z50q6aQnUMZVGLQ9a2RUtcRBkzw@mail.gmail.com>
	<CAKH46aRCzc4AMQRxp+71td4iVoc0gP=tKTf-VXAZTEau9EC_oA@mail.gmail.com>
Message-ID: <CAN0LTaMEjmVmyaE574CuEA56+X-F10vKALdfCtrfaCPtHOeBdg@mail.gmail.com>

> It was suggest by the tech team that moov file is large and it should
> reduced below a certain threshold.
>

Unless they can't tell you what exact size the maximum is, I doubt this
will significantly decrease the latency.


> Somewhere in mails on this user group i also read that there is another
> flag "-movflags faststart" which can be used to achieve similar purpose.
>

This moves the moov atom to the beginning of the file, which is essential
for streaming. Otherwise you'd have to download the entire file before the
player could start showing you video.
You should always do that when preparing content for streaming.

Please do not top-post on this mailing list. (
http://en.wikipedia.org/wiki/Posting_style#Top-posting)

From diljit77 at gmail.com  Wed Aug 14 20:43:46 2013
From: diljit77 at gmail.com (Diljit Singh)
Date: Thu, 15 Aug 2013 00:13:46 +0530
Subject: [FFmpeg-user] Controlling moov atom file size
In-Reply-To: <CAN0LTaMEjmVmyaE574CuEA56+X-F10vKALdfCtrfaCPtHOeBdg@mail.gmail.com>
References: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>
	<CAN0LTaNwzmga5NnwwKbSgX=Z50q6aQnUMZVGLQ9a2RUtcRBkzw@mail.gmail.com>
	<CAKH46aRCzc4AMQRxp+71td4iVoc0gP=tKTf-VXAZTEau9EC_oA@mail.gmail.com>
	<CAN0LTaMEjmVmyaE574CuEA56+X-F10vKALdfCtrfaCPtHOeBdg@mail.gmail.com>
Message-ID: <CAKH46aRamBcxYdfnERibu_dinveFYWJo_+cq=HDa2fu_wd4jFw@mail.gmail.com>

as per them current size of MOOV is 5.5 MB, but they fetch content from
origin server in sections of 2 MB. So, as per them it should be less than 2
MB or close to it.

Regards,
Diljit Singh


On Thu, Aug 15, 2013 at 12:08 AM, Werner Robitza
<werner.robitza at gmail.com>wrote:

> > It was suggest by the tech team that moov file is large and it should
> > reduced below a certain threshold.
> >
>
> Unless they can't tell you what exact size the maximum is, I doubt this
> will significantly decrease the latency.
>
>
> > Somewhere in mails on this user group i also read that there is another
> > flag "-movflags faststart" which can be used to achieve similar purpose.
> >
>
> This moves the moov atom to the beginning of the file, which is essential
> for streaming. Otherwise you'd have to download the entire file before the
> player could start showing you video.
> You should always do that when preparing content for streaming.
>
> Please do not top-post on this mailing list. (
> http://en.wikipedia.org/wiki/Posting_style#Top-posting)
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From h.reindl at thelounge.net  Wed Aug 14 20:57:27 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Wed, 14 Aug 2013 20:57:27 +0200
Subject: [FFmpeg-user] Controlling moov atom file size
In-Reply-To: <CAKH46aRamBcxYdfnERibu_dinveFYWJo_+cq=HDa2fu_wd4jFw@mail.gmail.com>
References: <CAKH46aSNe1WpLSwM_9QhD0CNZLR0rFRrwiqwJOE1Gx+kSL8g9g@mail.gmail.com>
	<CAN0LTaNwzmga5NnwwKbSgX=Z50q6aQnUMZVGLQ9a2RUtcRBkzw@mail.gmail.com>
	<CAKH46aRCzc4AMQRxp+71td4iVoc0gP=tKTf-VXAZTEau9EC_oA@mail.gmail.com>
	<CAN0LTaMEjmVmyaE574CuEA56+X-F10vKALdfCtrfaCPtHOeBdg@mail.gmail.com>
	<CAKH46aRamBcxYdfnERibu_dinveFYWJo_+cq=HDa2fu_wd4jFw@mail.gmail.com>
Message-ID: <520BD317.6090005@thelounge.net>

Am 14.08.2013 20:43, schrieb Diljit Singh:
> as per them current size of MOOV is 5.5 MB, but they fetch content from
> origin server in sections of 2 MB. So, as per them it should be less than 2
> MB or close to it.

this depends on the video length

i doubt that my 5.4 small H264 videos generated with ffmpeg
have a MOOV with 5 MB

hence that is why you should RTFM what MOOV means and people
which suggest it should be a specific size too.............

http://www.adobe.com/devnet/video/articles/mp4_movie_atom.html

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From leonard at kcfchurch.org  Wed Aug 14 22:13:16 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Wed, 14 Aug 2013 13:13:16 -0700
Subject: [FFmpeg-user] ffmpeg only works for one user.
Message-ID: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>

I followed the instructions at
http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to install ffmpeg
but the only user it works on is the user I logged into to install it
with.  All users can have access to ffmpeg itself but when they try to do
anything it fails with "Unknown encoder 'libx264'".  I don't get that error
with the user it was installed with.

How do I fix this?

From h.reindl at thelounge.net  Wed Aug 14 22:34:47 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Wed, 14 Aug 2013 22:34:47 +0200
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
Message-ID: <520BE9E7.70207@thelounge.net>



Am 14.08.2013 22:13, schrieb Leonard Bogard:
> I followed the instructions at
> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to install ffmpeg
> but the only user it works on is the user I logged into to install it
> with.  All users can have access to ffmpeg itself but when they try to do
> anything it fails with "Unknown encoder 'libx264'".  I don't get that error
> with the user it was installed with.
> 
> How do I fix this?

our crystal ball is broken

so how and *where* did you compile/install x264?
how are the permissions of any folder down to the x264 library?


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From leonard at kcfchurch.org  Wed Aug 14 23:15:37 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Wed, 14 Aug 2013 14:15:37 -0700
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <520BE9E7.70207@thelounge.net>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
Message-ID: <CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>

On Wed, Aug 14, 2013 at 1:34 PM, Reindl Harald <h.reindl at thelounge.net>wrote:

>
>
> Am 14.08.2013 22:13, schrieb Leonard Bogard:
> > I followed the instructions at
> > http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to install
> ffmpeg
> > but the only user it works on is the user I logged into to install it
> > with.  All users can have access to ffmpeg itself but when they try to do
> > anything it fails with "Unknown encoder 'libx264'".  I don't get that
> error
> > with the user it was installed with.
> >
> > How do I fix this?
>
> our crystal ball is broken
>
> so how and *where* did you compile/install x264?
> how are the permissions of any folder down to the x264 library?
>
>
>
I compiled it to the instructions laid out in the fore mentioned link
provided, therefore any permissions set on any folders/files are what you
would expect following the directions.

Where would I find these folders of which you speak?  Are you talking about
the folders that the libraries/binaries are compiled in or the installation
location?  If you're referring to the places where they were downloaded and
compiled, then they all have the permissions of the user they were compiled
with. If you're talking about the install location, I have no idea, I don't
know where the general locations for installed binaries or libraries are.

From h.reindl at thelounge.net  Wed Aug 14 23:24:41 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Wed, 14 Aug 2013 23:24:41 +0200
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
Message-ID: <520BF599.4090505@thelounge.net>



Am 14.08.2013 23:15, schrieb Leonard Bogard:
> On Wed, Aug 14, 2013 at 1:34 PM, Reindl Harald <h.reindl at thelounge.net>wrote:
> 
>> Am 14.08.2013 22:13, schrieb Leonard Bogard:
>>> I followed the instructions at
>>> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to install
>> ffmpeg
>>> but the only user it works on is the user I logged into to install it
>>> with.  All users can have access to ffmpeg itself but when they try to do
>>> anything it fails with "Unknown encoder 'libx264'".  I don't get that
>> error
>>> with the user it was installed with.
>>>
>>> How do I fix this?
>>
>> our crystal ball is broken
>>
>> so how and *where* did you compile/install x264?
>> how are the permissions of any folder down to the x264 library?
>>
> I compiled it to the instructions laid out in the fore mentioned link
> provided, therefore any permissions set on any folders/files are what you
> would expect following the directions.

i do not expect anything from Ubuntu instructions as Redhat user

> Where would I find these folders of which you speak?  Are you talking about
> the folders that the libraries/binaries are compiled in or the installation
> location?  If you're referring to the places where they were downloaded and
> compiled, then they all have the permissions of the user they were compiled
> with. If you're talking about the install location, I have no idea, I don't
> know where the general locations for installed binaries or libraries are

if you have no idea where the binaries installed it will be hard to help
how can anybody install self compiled things without knowing?
does the referred instruction does not state this?

if not why are you using a random instruction on the web without knowing
what happens at your system?

however, "ldd /path/to/the/ffmpeg/binary/ffmpeg" with the user
where it works will list you linked libraries like below

[harry at srv-rhsoft:~]$ ldd /usr/local/bin/ffmpeg
        linux-vdso.so.1 =>  (0x00007fff493fe000)
        libdl.so.2 => /lib64/libdl.so.2 (0x00007f86b1e5f000)
        libpthread.so.0 => /lib64/libpthread.so.0 (0x00007f86b1c43000)
        libvpx.so.1 => /lib64/libvpx.so.1 (0x00007f86b1989000)
        libtheoraenc.so.1 => /lib64/libtheoraenc.so.1 (0x00007f86b175c000)
        libtheoradec.so.1 => /lib64/libtheoradec.so.1 (0x00007f86b154c000)
        libmp3lame.so.0 => /lib64/libmp3lame.so.0 (0x00007f86b1288000)
        libm.so.6 => /lib64/libm.so.6 (0x00007f86b0f87000)
        libbz2.so.1 => /lib64/libbz2.so.1 (0x00007f86b0d77000)
        libz.so.1 => /lib64/libz.so.1 (0x00007f86b0b54000)
        libc.so.6 => /lib64/libc.so.6 (0x00007f86b079c000)
        /lib64/ld-linux-x86-64.so.2 (0x00007f86b207d000)
        libogg.so.0 => /lib64/libogg.so.0 (0x00007f86b0595000)
        libgomp.so.1 => /lib64/libgomp.so.1 (0x00007f86b0385000)
        librt.so.1 => /lib64/librt.so.1 (0x00007f86b017d000)

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From adf.lists at gmail.com  Thu Aug 15 00:01:59 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Wed, 14 Aug 2013 23:01:59 +0100
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376367195683-4660674.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376367195683-4660674.post@n4.nabble.com>
Message-ID: <520BFE57.7000307@gmail.com>

thljcl wrote:

> From the sources, I created both 24-FPS version and 25-FPS version,
> which I uploaded to http://sdrv.ms/14FXYb9. On my display, I could
> not tell the difference with my naked eyes at most parts, other than
> at the end where the credits are rolling. The lack of ?motion
> blurring? causes the obvious ?jarring unrealistic motion? when the
> credits are rolling for the 25-FPS version. I only used ffmpeg to
> duplicate frames. If I edit the frames by adding some ?motion
> blurring? or ?transitional effects?, the ?jarring unrealistic motion?
> could be somewhat mitigated.

I did have a look and the result was as I expected = very noticeable
"stop" every second viewing on both 50Hz and 60Hz.

Of course not every scene was affected but a lot more that just the
credits. The opening fly in looked terrible.

I accept you may not notice this as much, peoples perceptions are
different when it comes to many different things related to video. I've
read that some don't even notice tearing without vsync, I certainly do.
Some people are more sensitive to flicker that others. Film judder is
another case - some people just don't seem to see that  deinterlacing
50i to 25p looks really juddery on some shots, I am also very sensitive
to that one.

I don't think that motion blur could help the 24 -> 25 dup, I guess it
could in the case of 50i -> 25p.

> A single experiment is far from conclusive to prove anything.
> However, from my experiences as well as this experiment, while
> duplicating/dropping frames is not strictly perfect, it?s certainly
> an acceptable solution for many if not most cases. The reasoning is
> mainly the properties of human visual system. ?Motion picture? is
> never ?the picture that moves? but our visual system makes us think
> that ?it moves?.

I would never dup 24 -> 25, if you need to make something for pal, speed
it up - that's the way it's always been done.


From jiachielee at live.com  Thu Aug 15 00:27:15 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 14 Aug 2013 15:27:15 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <520BFE57.7000307@gmail.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376367195683-4660674.post@n4.nabble.com>
	<520BFE57.7000307@gmail.com>
Message-ID: <1376519235178-4660721.post@n4.nabble.com>

With the access of the sources, I can inspect the every single frame that
makes up the video. Not every scene is noticeable. And I don?t claim that I
exactly have the best eyesight to make conclusive judgment on behalf of the
others. The less noticeable scenes, are either static (or ?motionless?) or
with adequate ?motion blurring?.
There are reasons why I?m not exactly in favor of speeding up the video; the
playback speed is altered although I admit that not every person can
actually ?feel? the ?speeding up? at first look. Speeding the audio would
affect the ?pitch? and other effects. I guess the best solution is to
reproduce the video frame-by-frame for 25 FPS.
To me personally, I would support the standardization of frame rate at 24
FPS for all videos/films created. For historical reasons, NTSC standard uses
29.97 FPS (interlaced) while PAL standard uses 25 FPS; the theatrical films
are created at 24 FPS. On a PC, it does not matter which frame rate the
video has used. Due to the existence of different standards, there exist PAL
DVD and NTSC DVD, both only supports interlaced contents natively. On LCD
display, undoubtedly progressive-scan video can better preserve image
quality for every single frame. The real reasons for the old standards are
still in use is more about traditions or what people already have or are
used to. Even though H.264 offers better compression compared with H.262,
Blu-ray still has to support H.262 for backward compatibility reasons.
Speaking of which, I?ve never thought of converting to 25 FPS before because
I?m in favor of standardizing on 24 FPS. I used to convert from 25 FPS and
30 FPS to 24 FPS and never quite experience ?unrealistic motion? before. To
be honest, I don?t even know the reason for converting to 25 FPS from 24 FPS
other than for traditional analog broadcast. I usually keep the original
frame rate; when I do change the frame rate to 24 FPS, it?s when I actually
edit the videos (up-scaling, cropping, etc.)
Historically, it?s necessary to change the frame rate when we transfer the
film materials for TV viewing to either 29.97 FPS or 25 FPS. Is it really
necessary to change frame rate when I view film materials or digitally
created 24-FPS video on LCD display? I guess not. I conduct the experiment
mainly out of curiosity. Anyway, it?s a great learning experience.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660721.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From leonard at kcfchurch.org  Thu Aug 15 00:50:54 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Wed, 14 Aug 2013 15:50:54 -0700
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <520BF599.4090505@thelounge.net>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
Message-ID: <CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>

On Wed, Aug 14, 2013 at 2:24 PM, Reindl Harald <h.reindl at thelounge.net>wrote:

>
>
> Am 14.08.2013 23:15, schrieb Leonard Bogard:
> > On Wed, Aug 14, 2013 at 1:34 PM, Reindl Harald <h.reindl at thelounge.net
> >wrote:
> >
> >> Am 14.08.2013 22:13, schrieb Leonard Bogard:
> >>> I followed the instructions at
> >>> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to install
> >> ffmpeg
> >>> but the only user it works on is the user I logged into to install it
> >>> with.  All users can have access to ffmpeg itself but when they try to
> do
> >>> anything it fails with "Unknown encoder 'libx264'".  I don't get that
> >> error
> >>> with the user it was installed with.
> >>>
> >>> How do I fix this?
> >>
> >> our crystal ball is broken
> >>
> >> so how and *where* did you compile/install x264?
> >> how are the permissions of any folder down to the x264 library?
> >>
> > I compiled it to the instructions laid out in the fore mentioned link
> > provided, therefore any permissions set on any folders/files are what you
> > would expect following the directions.
>
> i do not expect anything from Ubuntu instructions as Redhat user
>
> > Where would I find these folders of which you speak?  Are you talking
> about
> > the folders that the libraries/binaries are compiled in or the
> installation
> > location?  If you're referring to the places where they were downloaded
> and
> > compiled, then they all have the permissions of the user they were
> compiled
> > with. If you're talking about the install location, I have no idea, I
> don't
> > know where the general locations for installed binaries or libraries are
>
> if you have no idea where the binaries installed it will be hard to help
> how can anybody install self compiled things without knowing?
> does the referred instruction does not state this?
>
> if not why are you using a random instruction on the web without knowing
> what happens at your system?
>
> however, "ldd /path/to/the/ffmpeg/binary/ffmpeg" with the user
> where it works will list you linked libraries like below
>
> [harry at srv-rhsoft:~]$ ldd /usr/local/bin/ffmpeg
>         linux-vdso.so.1 =>  (0x00007fff493fe000)
>         libdl.so.2 => /lib64/libdl.so.2 (0x00007f86b1e5f000)
>         libpthread.so.0 => /lib64/libpthread.so.0 (0x00007f86b1c43000)
>         libvpx.so.1 => /lib64/libvpx.so.1 (0x00007f86b1989000)
>         libtheoraenc.so.1 => /lib64/libtheoraenc.so.1 (0x00007f86b175c000)
>         libtheoradec.so.1 => /lib64/libtheoradec.so.1 (0x00007f86b154c000)
>         libmp3lame.so.0 => /lib64/libmp3lame.so.0 (0x00007f86b1288000)
>         libm.so.6 => /lib64/libm.so.6 (0x00007f86b0f87000)
>         libbz2.so.1 => /lib64/libbz2.so.1 (0x00007f86b0d77000)
>         libz.so.1 => /lib64/libz.so.1 (0x00007f86b0b54000)
>         libc.so.6 => /lib64/libc.so.6 (0x00007f86b079c000)
>         /lib64/ld-linux-x86-64.so.2 (0x00007f86b207d000)
>         libogg.so.0 => /lib64/libogg.so.0 (0x00007f86b0595000)
>         libgomp.so.1 => /lib64/libgomp.so.1 (0x00007f86b0385000)
>         librt.so.1 => /lib64/librt.so.1 (0x00007f86b017d000)
>
>
It's all root:root:

~$ whereis ffmpeg
ffmpeg: /usr/local/bin/ffmpeg
~$ ldd /usr/local/bin/ffmpeg
        linux-vdso.so.1 =>  (0x00007fff185ff000)
        libva.so.1 => /usr/lib/x86_64-linux-gnu/libva.so.1
(0x00007f73ed2da000)
        libasound.so.2 => /usr/lib/x86_64-linux-gnu/libasound.so.2
(0x00007f73ecfed000)
        libSDL-1.2.so.0 => /usr/lib/x86_64-linux-gnu/libSDL-1.2.so.0
(0x00007f73ecd53000)
        libm.so.6 => /lib/x86_64-linux-gnu/libm.so.6 (0x00007f73eca57000)
        libz.so.1 => /lib/x86_64-linux-gnu/libz.so.1 (0x00007f73ec840000)
        librt.so.1 => /lib/x86_64-linux-gnu/librt.so.1 (0x00007f73ec637000)
        libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0
(0x00007f73ec41a000)
        libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f73ec05b000)
        libdl.so.2 => /lib/x86_64-linux-gnu/libdl.so.2 (0x00007f73ebe56000)
        libpulse-simple.so.0 =>
/usr/lib/x86_64-linux-gnu/libpulse-simple.so.0 (0x00007f73ebc52000)
        libpulse.so.0 => /usr/lib/x86_64-linux-gnu/libpulse.so.0
(0x00007f73eba0a000)
        libX11.so.6 => /usr/lib/x86_64-linux-gnu/libX11.so.6
(0x00007f73eb6d4000)
        libXext.so.6 => /usr/lib/x86_64-linux-gnu/libXext.so.6
(0x00007f73eb4c3000)
        libcaca.so.0 => /usr/lib/x86_64-linux-gnu/libcaca.so.0
(0x00007f73eb1f8000)
        /lib64/ld-linux-x86-64.so.2 (0x00007f73ed4fc000)
        libpulsecommon-1.1.so => /usr/lib/x86_64-linux-gnu/
libpulsecommon-1.1.so (0x00007f73eaf99000)
        libjson.so.0 => /usr/lib/x86_64-linux-gnu/libjson.so.0
(0x00007f73ead91000)
        libdbus-1.so.3 => /lib/x86_64-linux-gnu/libdbus-1.so.3
(0x00007f73eab4d000)
        libxcb.so.1 => /usr/lib/x86_64-linux-gnu/libxcb.so.1
(0x00007f73ea92e000)
        libslang.so.2 => /lib/x86_64-linux-gnu/libslang.so.2
(0x00007f73ea5bd000)
        libncursesw.so.5 => /lib/x86_64-linux-gnu/libncursesw.so.5
(0x00007f73ea390000)
        libtinfo.so.5 => /lib/x86_64-linux-gnu/libtinfo.so.5
(0x00007f73ea168000)
        libwrap.so.0 => /lib/x86_64-linux-gnu/libwrap.so.0
(0x00007f73e9f5f000)
        libsndfile.so.1 => /usr/lib/x86_64-linux-gnu/libsndfile.so.1
(0x00007f73e9cf8000)
        libasyncns.so.0 => /usr/lib/x86_64-linux-gnu/libasyncns.so.0
(0x00007f73e9af1000)
        libXau.so.6 => /usr/lib/x86_64-linux-gnu/libXau.so.6
(0x00007f73e98ee000)
        libXdmcp.so.6 => /usr/lib/x86_64-linux-gnu/libXdmcp.so.6
(0x00007f73e96e8000)
        libnsl.so.1 => /lib/x86_64-linux-gnu/libnsl.so.1
(0x00007f73e94cd000)
        libFLAC.so.8 => /usr/lib/x86_64-linux-gnu/libFLAC.so.8
(0x00007f73e9283000)
        libvorbisenc.so.2 => /usr/lib/x86_64-linux-gnu/libvorbisenc.so.2
(0x00007f73e8db4000)
        libvorbis.so.0 => /usr/lib/x86_64-linux-gnu/libvorbis.so.0
(0x00007f73e8b87000)
        libogg.so.0 => /usr/lib/x86_64-linux-gnu/libogg.so.0
(0x00007f73e8980000)
        libresolv.so.2 => /lib/x86_64-linux-gnu/libresolv.so.2
(0x00007f73e8764000)
lrwxrwxrwx 1 root root 17 Mar 29  2012 /usr/lib/x86_64-linux-gnu/libva.so.1
-> libva.so.1.3200.0
lrwxrwxrwx 1 root root 18 Jan 24  2013
/usr/lib/x86_64-linux-gnu/libasound.so.2 -> libasound.so.2.0.0
lrwxrwxrwx 1 root root 20 Jan 27  2012
/usr/lib/x86_64-linux-gnu/libSDL-1.2.so.0 -> libSDL-1.2.so.0.11.3
lrwxrwxrwx 1 root root 12 Jan 28  2013 /lib/x86_64-linux-gnu/libm.so.6 ->
libm-2.15.so
lrwxrwxrwx 1 root root 15 Nov 10  2011 /lib/x86_64-linux-gnu/libz.so.1 ->
libz.so.1.2.3.4
lrwxrwxrwx 1 root root 13 Jan 28  2013 /lib/x86_64-linux-gnu/librt.so.1 ->
librt-2.15.so
lrwxrwxrwx 1 root root 18 Jan 28  2013
/lib/x86_64-linux-gnu/libpthread.so.0 -> libpthread-2.15.so
lrwxrwxrwx 1 root root 12 Jan 28  2013 /lib/x86_64-linux-gnu/libc.so.6 ->
libc-2.15.so
lrwxrwxrwx 1 root root 13 Jan 28  2013 /lib/x86_64-linux-gnu/libdl.so.2 ->
libdl-2.15.so
lrwxrwxrwx 1 root root 24 May 28 12:24
/usr/lib/x86_64-linux-gnu/libpulse-simple.so.0 -> libpulse-simple.so.0.0.3
lrwxrwxrwx 1 root root 18 May 28 12:24
/usr/lib/x86_64-linux-gnu/libpulse.so.0 -> libpulse.so.0.13.5
lrwxrwxrwx 1 root root 15 May 29 14:31
/usr/lib/x86_64-linux-gnu/libX11.so.6 -> libX11.so.6.3.0
lrwxrwxrwx 1 root root 16 May 28 06:38
/usr/lib/x86_64-linux-gnu/libXext.so.6 -> libXext.so.6.4.0
lrwxrwxrwx 1 root root 18 Mar 10  2012
/usr/lib/x86_64-linux-gnu/libcaca.so.0 -> libcaca.so.0.99.17
-rw-r--r-- 1 root root 385240 May 28 12:24 /usr/lib/x86_64-linux-gnu/
libpulsecommon-1.1.so
lrwxrwxrwx 1 root root 16 Aug 12  2011
/usr/lib/x86_64-linux-gnu/libjson.so.0 -> libjson.so.0.0.1
lrwxrwxrwx 1 root root 18 Jun 13 07:55 /lib/x86_64-linux-gnu/libdbus-1.so.3
-> libdbus-1.so.3.5.8
lrwxrwxrwx 1 root root 15 May 27 14:13
/usr/lib/x86_64-linux-gnu/libxcb.so.1 -> libxcb.so.1.1.0
lrwxrwxrwx 1 root root 17 Oct 18  2011 /lib/x86_64-linux-gnu/libslang.so.2
-> libslang.so.2.2.4
lrwxrwxrwx 1 root root 18 Nov 18  2011
/lib/x86_64-linux-gnu/libncursesw.so.5 -> libncursesw.so.5.9
lrwxrwxrwx 1 root root 15 Nov 18  2011 /lib/x86_64-linux-gnu/libtinfo.so.5
-> libtinfo.so.5.9
lrwxrwxrwx 1 root root 16 Aug 12  2011 /lib/x86_64-linux-gnu/libwrap.so.0
-> libwrap.so.0.7.6
lrwxrwxrwx 1 root root 20 Jan 25  2012
/usr/lib/x86_64-linux-gnu/libsndfile.so.1 -> libsndfile.so.1.0.25
lrwxrwxrwx 1 root root 19 Aug 12  2011
/usr/lib/x86_64-linux-gnu/libasyncns.so.0 -> libasyncns.so.0.3.1
lrwxrwxrwx 1 root root 15 Nov  9  2011
/usr/lib/x86_64-linux-gnu/libXau.so.6 -> libXau.so.6.0.0
lrwxrwxrwx 1 root root 17 Nov  9  2011
/usr/lib/x86_64-linux-gnu/libXdmcp.so.6 -> libXdmcp.so.6.0.0
lrwxrwxrwx 1 root root 14 Jan 28  2013 /lib/x86_64-linux-gnu/libnsl.so.1 ->
libnsl-2.15.so
lrwxrwxrwx 1 root root 16 Oct 17  2011
/usr/lib/x86_64-linux-gnu/libFLAC.so.8 -> libFLAC.so.8.2.0
lrwxrwxrwx 1 root root 21 Feb 20  2012
/usr/lib/x86_64-linux-gnu/libvorbisenc.so.2 -> libvorbisenc.so.2.0.8
lrwxrwxrwx 1 root root 18 Feb 20  2012
/usr/lib/x86_64-linux-gnu/libvorbis.so.0 -> libvorbis.so.0.4.5
lrwxrwxrwx 1 root root 15 Aug 12  2011
/usr/lib/x86_64-linux-gnu/libogg.so.0 -> libogg.so.0.7.1
lrwxrwxrwx 1 root root 17 Jan 28  2013 /lib/x86_64-linux-gnu/libresolv.so.2
-> libresolv-2.15.so
lrwxrwxrwx 1 root root 32 Jan 28  2013 /lib64/ld-linux-x86-64.so.2 ->
/lib/x86_64-linux-gnu/ld-2.15.so
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x 24 root root   28672 Aug 13 22:48 x86_64-linux-gnu
drwxr-xr-x  3 root root  12288 Aug 13 21:05 x86_64-linux-gnu

From lou at lrcd.com  Thu Aug 15 01:03:13 2013
From: lou at lrcd.com (Lou)
Date: Wed, 14 Aug 2013 15:03:13 -0800
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
Message-ID: <1376521393.5894.9915143.5805C2B6@webmail.messagingengine.com>

On Wed, Aug 14, 2013, at 12:13 PM, Leonard Bogard wrote:
> I followed the instructions at
> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to install
> ffmpeg but the only user it works on is the user I logged into to install it

The guide locally installs the ffmpeg binary to the ~/bin directory of
the user following the guide. A system installation is not used by the
guide. This prevents conflicts with the old, fake "ffmpeg" package in
the repository, or more importantly, usually prevents interfering with
packages that depend on the old, fake "ffmpeg" package in the
repository.

> All users can have access to ffmpeg itself but when they try to do
> anything it fails with "Unknown encoder 'libx264'".  I don't get that
> error with the user it was installed with.

This means they are using the so-called "ffmpeg" from the repository.

> How do I fix this?

You have several options. Here are some ideas:

* Perform a "system installation" with checkinstall (the following is
somewhat outdated but will give you an idea):
  <http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide?version=59>

* Install the compiled ffmpeg (or simply a ffmpeg build):
  <https://ffmpeg.org/download.html#LinuxBuilds>
  ...in a custom location and use a persistent environmental variable to
  allow all users to use it:
  <https://help.ubuntu.com/community/EnvironmentVariables#Persistent_environment_variables>

From pkoshevoy at gmail.com  Thu Aug 15 01:50:47 2013
From: pkoshevoy at gmail.com (Pavel Koshevoy)
Date: Wed, 14 Aug 2013 17:50:47 -0600
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376519235178-4660721.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>
	<1376074113012-4660603.post@n4.nabble.com>
	<52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376367195683-4660674.post@n4.nabble.com>
	<520BFE57.7000307@gmail.com>
	<1376519235178-4660721.post@n4.nabble.com>
Message-ID: <520C17D7.6070003@gmail.com>

On 8/14/2013 4:27 PM, thljcl wrote:
> With the access of the sources, I can inspect the every single frame that
> makes up the video. Not every scene is noticeable. And I don?t claim that I
> exactly have the best eyesight to make conclusive judgment on behalf of the
> others. The less noticeable scenes, are either static (or ?motionless?) or
> with adequate ?motion blurring?.
> There are reasons why I?m not exactly in favor of speeding up the video; the
> playback speed is altered although I admit that not every person can
> actually ?feel? the ?speeding up? at first look. Speeding the audio would
> affect the ?pitch? and other effects.

atempo preserves the pitch, and for small tempo changes you won't notice 
any difference.

     Pavel.

From jiachielee at live.com  Thu Aug 15 02:37:19 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 14 Aug 2013 17:37:19 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <520C17D7.6070003@gmail.com>
References: <52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376367195683-4660674.post@n4.nabble.com>
	<520BFE57.7000307@gmail.com>
	<1376519235178-4660721.post@n4.nabble.com>
	<520C17D7.6070003@gmail.com>
Message-ID: <1376527039826-4660725.post@n4.nabble.com>

It?s technically impossible to change the playback speed of audio without
affecting pitch, length, or other issues due to round-off
errors/quantization. As you said, though, it?s counting on less noticeable
perceptional difference. The waveforms of the audio would change
accordingly, albeit possibly very minor if done correctly, when you speed up
the audio. The original sound (analog) information is being encoded so that
it could be represented by combinations of digits. With enough sampling
rate, floating points, etc., the accuracy could be achieved that our
speakers could reproduce the audio signal that we still perceive pretty
close to the original signal. I do believe that digital audio is the right
solution compared with analog storage medium for audio; but the debate for
analog vs. digital does still persist in these days, especially among
audiophiles. I am personally in favor of digital audio because our
perception does have limitation; we cannot listen to sound wave of all
frequencies; we cannot see photons of every frequency either even though
technically the frequency of photon is frame-dependent.
As a matter of fact, when the audio is still in production stage, it?s very
common to have twice the sampling rate of the eventual sampling rate for
distribution. Some of the high-cost or high-value production of audios
producers may still keep ?master copy? of such sampling rate. At the
production stage, they may re-mix the audios, speed up, changing speed,
pitch, etc. The famous song ?Billie Jeans? has the mix of Michael Jackson?s
voice of various pitches. Of course, in the early days, the audio is being
produced without the assistance of computer; but Michael Jackson and many
other producers later do use computers to assist in audio production. Of
course, the audios for distribution would have its sampling rate to be
down-sampled. In fact, recently the rise of MP3 online store makes the audio
quality to be even worse for commercially-released songs due to lossy
compression. I bet the teenagers couldn?t care less with their headset.
Reproduction of frame-by-frame is not a perfect solution either, so to
speak. Only when the producers work hard, smart, and careful, then only we
can achieve high-quality reproduction. It needs substantial testing, of
course. But I do admit such reproduction is much laborious compared with
changing the playback speed of audio.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660725.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From pkoshevoy at gmail.com  Thu Aug 15 04:01:09 2013
From: pkoshevoy at gmail.com (Pavel Koshevoy)
Date: Wed, 14 Aug 2013 20:01:09 -0600
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376527039826-4660725.post@n4.nabble.com>
References: <52057604.30206@gmail.com>
	<1376106209176-4660606.post@n4.nabble.com>
	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376367195683-4660674.post@n4.nabble.com>
	<520BFE57.7000307@gmail.com>
	<1376519235178-4660721.post@n4.nabble.com>
	<520C17D7.6070003@gmail.com>
	<1376527039826-4660725.post@n4.nabble.com>
Message-ID: <520C3665.1020104@gmail.com>

On 8/14/13 6:37 PM, thljcl wrote:
> It?s technically impossible to change the playback speed of audio without
> affecting pitch, length, or other issues due to round-off
> errors/quantization.

atempo implements WSOLA algorithm.  It's not resampling -- it's 
cut-shift-align-overlap-blend.  Its effect is to change the duration of 
the track by duplicating or de-duplicating some repeating patterns in 
the audio waveform.  For voice-only audio its artifacts are quiet 
unnoticeable even at relatively large tempo changes.  I've tested it by 
playing music at 80% tempo and found the sound acceptable.  Higher tempo 
artifacts are even harder to notice. Going from 24 to 25 fps is about 
4.17% tempo change, well within the range where atempo was tested.

Pavel.


From dashing.meng at gmail.com  Thu Aug 15 04:42:03 2013
From: dashing.meng at gmail.com (littlebat)
Date: Thu, 15 Aug 2013 10:42:03 +0800
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
	<CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
Message-ID: <20130815104203.e255b849.dashing.meng@gmail.com>

On Wed, 14 Aug 2013 15:50:54 -0700
Leonard Bogard <leonard at kcfchurch.org> wrote:

> On Wed, Aug 14, 2013 at 2:24 PM, Reindl Harald
> <h.reindl at thelounge.net>wrote:
> 
> >
> >
> > Am 14.08.2013 23:15, schrieb Leonard Bogard:
> > > On Wed, Aug 14, 2013 at 1:34 PM, Reindl Harald
> > > <h.reindl at thelounge.net
> > >wrote:
> > >
> > >> Am 14.08.2013 22:13, schrieb Leonard Bogard:
> > >>> I followed the instructions at
> > >>> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to
> > >>> install
> > >> ffmpeg
> > >>> but the only user it works on is the user I logged into to
> > >>> install it with.  All users can have access to ffmpeg itself
> > >>> but when they try to
> > do
> > >>> anything it fails with "Unknown encoder 'libx264'".  I don't
> > >>> get that
> > >> error
> > >>> with the user it was installed with.
> > >>>
> > >>> How do I fix this?
> > >>
> > >> our crystal ball is broken
> > >>
> > >> so how and *where* did you compile/install x264?
> > >> how are the permissions of any folder down to the x264 library?
> > >>
> > > I compiled it to the instructions laid out in the fore mentioned
> > > link provided, therefore any permissions set on any folders/files
> > > are what you would expect following the directions.
> >
> > i do not expect anything from Ubuntu instructions as Redhat user
> >
> > > Where would I find these folders of which you speak?  Are you
> > > talking
> > about
> > > the folders that the libraries/binaries are compiled in or the
> > installation
> > > location?  If you're referring to the places where they were
> > > downloaded
> > and
> > > compiled, then they all have the permissions of the user they were
> > compiled
> > > with. If you're talking about the install location, I have no
> > > idea, I
> > don't
> > > know where the general locations for installed binaries or
> > > libraries are
> >
> > if you have no idea where the binaries installed it will be hard to
> > help how can anybody install self compiled things without knowing?
> > does the referred instruction does not state this?
> >
> > if not why are you using a random instruction on the web without
> > knowing what happens at your system?
> >
> > however, "ldd /path/to/the/ffmpeg/binary/ffmpeg" with the user
> > where it works will list you linked libraries like below
> >
> > [harry at srv-rhsoft:~]$ ldd /usr/local/bin/ffmpeg
> >         linux-vdso.so.1 =>  (0x00007fff493fe000)
> >         libdl.so.2 => /lib64/libdl.so.2 (0x00007f86b1e5f000)
> >         libpthread.so.0 => /lib64/libpthread.so.0
> > (0x00007f86b1c43000) libvpx.so.1 => /lib64/libvpx.so.1
> > (0x00007f86b1989000) libtheoraenc.so.1 => /lib64/libtheoraenc.so.1
> > (0x00007f86b175c000) libtheoradec.so.1 => /lib64/libtheoradec.so.1
> > (0x00007f86b154c000) libmp3lame.so.0 => /lib64/libmp3lame.so.0
> > (0x00007f86b1288000) libm.so.6 => /lib64/libm.so.6
> > (0x00007f86b0f87000) libbz2.so.1 => /lib64/libbz2.so.1
> > (0x00007f86b0d77000) libz.so.1 => /lib64/libz.so.1
> > (0x00007f86b0b54000) libc.so.6 => /lib64/libc.so.6
> > (0x00007f86b079c000) /lib64/ld-linux-x86-64.so.2
> > (0x00007f86b207d000) libogg.so.0 => /lib64/libogg.so.0
> > (0x00007f86b0595000) libgomp.so.1 => /lib64/libgomp.so.1
> > (0x00007f86b0385000) librt.so.1 => /lib64/librt.so.1
> > (0x00007f86b017d000)
> >
> >
> It's all root:root:
> 
> ~$ whereis ffmpeg
> ffmpeg: /usr/local/bin/ffmpeg

No, it isn't the ffmpeg you installed follow the instruction at
http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide . I have a
glance at this instruction, the correct ffmpeg here should
be: /home/INSTALL_USER/bin/ffmpeg .

So, try "/home/INSTALL_USER/bin/ffmpeg" when login with
another user. 

From jiachielee at live.com  Thu Aug 15 04:35:58 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 14 Aug 2013 19:35:58 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <520C3665.1020104@gmail.com>
References: <CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>
	<1376170825784-4660622.post@n4.nabble.com>
	<20130810232130.GA11654@phare.normalesup.org>
	<1376184010720-4660626.post@n4.nabble.com>
	<1376367195683-4660674.post@n4.nabble.com>
	<520BFE57.7000307@gmail.com>
	<1376519235178-4660721.post@n4.nabble.com>
	<520C17D7.6070003@gmail.com>
	<1376527039826-4660725.post@n4.nabble.com>
	<520C3665.1020104@gmail.com>
Message-ID: <1376534158577-4660727.post@n4.nabble.com>

As I said, it?s all about the perceptional difference and the actual
difference. As long as you or any other users can accept the result, it?s
fine with me. The duplicating of the ?wave patterns??
Strictly speaking, a periodic function (wave) can be represented by Fourier
series, which is an infinite series. Infinite series is important in many
aspects of mathematics and Physics; as many of the common functions or
mathematical constants are defined by different infinite series. When it
comes to actual computation, do you think we can actually compute ?infinite?
number of terms? We merely calculate the approximated answers, up to the
necessary accuracy; of course, we can only determine necessary accuracy from
empirical results. Mathematical periodic functions do exist; but when we
actually do the calculation, we always use approximation.
In the case of converting of analog signal (sound wave) to digital signals
(0 and 1), the only possible empirical results we can get are subjective
perceptional tests, which differ from one person to another. Technically
speaking, it?s impossible for any calculator or computer to compute the
actual ?wave patterns?, so to speak. We only use good approximation. LPCM
audio data does not have the actual ?wave patterns? from the analog signal,
which is the main reasoning behind the analog vs. digital debate. Through
resampling, we get digital data, which we later use to re-create the
approximated ?wave patterns?, i.e. audio signal.
One way to increase accuracy of the approximated results is to avoid
rounding-off during the calculation. For computer, however, using floating
point or decimal number is inevitable but we can increase the number of
floating points to reduce errors; just like the producers use twice the
sampling rate during the production stage.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffprobe-show-frames-and-coded-picture-number-tp4660602p4660727.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From highgod0401 at gmail.com  Thu Aug 15 04:58:51 2013
From: highgod0401 at gmail.com (Wei Gao)
Date: Thu, 15 Aug 2013 10:58:51 +0800
Subject: [FFmpeg-user] [FFmpeg-devel]  question: Hi,
 I want to ask a question about capture audio from audio card
In-Reply-To: <CAL1QdWcZu0ptbwXZvfvLhfurCGTVRmRHjkDCfC2t+SqahfmbtQ@mail.gmail.com>
References: <CAAGF25gj93ogvjUh5dh3+LSu-j4m_HJZrGHJCb6d=A6GHX09+Q@mail.gmail.com>
	<CAL1QdWdSZJJdthc8pA0OyCWyNCx634LJDEDqd_EdcPbDTeEHPA@mail.gmail.com>
	<CAAGF25jZKXiYiBufCG1r5dZN=Y7J-pGoeeetf6-h4HtjmdLpsA@mail.gmail.com>
	<CAL1QdWfXseON2nCSpXhWNgB3h8N94APV5AABcx8C9DGUGKJgyA@mail.gmail.com>
	<CAAGF25gkomsr=Pw5Wg6Xbt6Wc=yosXwYmXT2dkFtMDK96kFR8g@mail.gmail.com>
	<CAL1QdWcZu0ptbwXZvfvLhfurCGTVRmRHjkDCfC2t+SqahfmbtQ@mail.gmail.com>
Message-ID: <CAAGF25jNVEMum2Zg6VeW_tDCV+fmVcF04Fn1+=xoqPMwWurwNw@mail.gmail.com>

2013/8/3 Roger Pack <rogerdpack2 at gmail.com>

> I guess it is a bug in windows 8...will file it though, thank you.
>
> Hi, Roger Pack, sorry for the late reply because busy these days. Do you
mean it is a bug in ffmpeg? Because we run perfect on win7 using dshow.

do you want any information that I can submit to you such as any debug
informantion?

Thanks very much

Best regards

> >
> >> >> > _______________________________________________
> >> >> > ffmpeg-devel mailing list
> >> >> > ffmpeg-devel at ffmpeg.org
> >> >> > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> >> >> >
> >> >>
> >> >
> >>
> > _______________________________________________
> > ffmpeg-devel mailing list
> > ffmpeg-devel at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> >
>

From leonard at kcfchurch.org  Thu Aug 15 05:05:31 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Wed, 14 Aug 2013 20:05:31 -0700
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <20130815104203.e255b849.dashing.meng@gmail.com>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
	<CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
	<20130815104203.e255b849.dashing.meng@gmail.com>
Message-ID: <CA+JzsNQc1vZr+j54MA_dPVW+yiHbaAuEokMOPESS5vKZsR=4Aw@mail.gmail.com>

On Wed, Aug 14, 2013 at 7:42 PM, littlebat <dashing.meng at gmail.com> wrote:

> On Wed, 14 Aug 2013 15:50:54 -0700
> Leonard Bogard <leonard at kcfchurch.org> wrote:
>
> > On Wed, Aug 14, 2013 at 2:24 PM, Reindl Harald
> > <h.reindl at thelounge.net>wrote:
> >
> > >
> > >
> > > Am 14.08.2013 23:15, schrieb Leonard Bogard:
> > > > On Wed, Aug 14, 2013 at 1:34 PM, Reindl Harald
> > > > <h.reindl at thelounge.net
> > > >wrote:
> > > >
> > > >> Am 14.08.2013 22:13, schrieb Leonard Bogard:
> > > >>> I followed the instructions at
> > > >>> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide to
> > > >>> install
> > > >> ffmpeg
> > > >>> but the only user it works on is the user I logged into to
> > > >>> install it with.  All users can have access to ffmpeg itself
> > > >>> but when they try to
> > > do
> > > >>> anything it fails with "Unknown encoder 'libx264'".  I don't
> > > >>> get that
> > > >> error
> > > >>> with the user it was installed with.
> > > >>>
> > > >>> How do I fix this?
> > > >>
> > > >> our crystal ball is broken
> > > >>
> > > >> so how and *where* did you compile/install x264?
> > > >> how are the permissions of any folder down to the x264 library?
> > > >>
> > > > I compiled it to the instructions laid out in the fore mentioned
> > > > link provided, therefore any permissions set on any folders/files
> > > > are what you would expect following the directions.
> > >
> > > i do not expect anything from Ubuntu instructions as Redhat user
> > >
> > > > Where would I find these folders of which you speak?  Are you
> > > > talking
> > > about
> > > > the folders that the libraries/binaries are compiled in or the
> > > installation
> > > > location?  If you're referring to the places where they were
> > > > downloaded
> > > and
> > > > compiled, then they all have the permissions of the user they were
> > > compiled
> > > > with. If you're talking about the install location, I have no
> > > > idea, I
> > > don't
> > > > know where the general locations for installed binaries or
> > > > libraries are
> > >
> > > if you have no idea where the binaries installed it will be hard to
> > > help how can anybody install self compiled things without knowing?
> > > does the referred instruction does not state this?
> > >
> > > if not why are you using a random instruction on the web without
> > > knowing what happens at your system?
> > >
> > > however, "ldd /path/to/the/ffmpeg/binary/ffmpeg" with the user
> > > where it works will list you linked libraries like below
> > >
> > > [harry at srv-rhsoft:~]$ ldd /usr/local/bin/ffmpeg
> > >         linux-vdso.so.1 =>  (0x00007fff493fe000)
> > >         libdl.so.2 => /lib64/libdl.so.2 (0x00007f86b1e5f000)
> > >         libpthread.so.0 => /lib64/libpthread.so.0
> > > (0x00007f86b1c43000) libvpx.so.1 => /lib64/libvpx.so.1
> > > (0x00007f86b1989000) libtheoraenc.so.1 => /lib64/libtheoraenc.so.1
> > > (0x00007f86b175c000) libtheoradec.so.1 => /lib64/libtheoradec.so.1
> > > (0x00007f86b154c000) libmp3lame.so.0 => /lib64/libmp3lame.so.0
> > > (0x00007f86b1288000) libm.so.6 => /lib64/libm.so.6
> > > (0x00007f86b0f87000) libbz2.so.1 => /lib64/libbz2.so.1
> > > (0x00007f86b0d77000) libz.so.1 => /lib64/libz.so.1
> > > (0x00007f86b0b54000) libc.so.6 => /lib64/libc.so.6
> > > (0x00007f86b079c000) /lib64/ld-linux-x86-64.so.2
> > > (0x00007f86b207d000) libogg.so.0 => /lib64/libogg.so.0
> > > (0x00007f86b0595000) libgomp.so.1 => /lib64/libgomp.so.1
> > > (0x00007f86b0385000) librt.so.1 => /lib64/librt.so.1
> > > (0x00007f86b017d000)
> > >
> > >
> > It's all root:root:
> >
> > ~$ whereis ffmpeg
> > ffmpeg: /usr/local/bin/ffmpeg
>
> No, it isn't the ffmpeg you installed follow the instruction at
> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide . I have a
> glance at this instruction, the correct ffmpeg here should
> be: /home/INSTALL_USER/bin/ffmpeg .
>
> So, try "/home/INSTALL_USER/bin/ffmpeg" when login with
> another user.
>

Simple solution. GAH!  Thanks, it worked.

From h.reindl at thelounge.net  Thu Aug 15 05:13:34 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Thu, 15 Aug 2013 05:13:34 +0200
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <CA+JzsNQc1vZr+j54MA_dPVW+yiHbaAuEokMOPESS5vKZsR=4Aw@mail.gmail.com>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
	<CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
	<20130815104203.e255b849.dashing.meng@gmail.com>
	<CA+JzsNQc1vZr+j54MA_dPVW+yiHbaAuEokMOPESS5vKZsR=4Aw@mail.gmail.com>
Message-ID: <520C475E.8090300@thelounge.net>



Am 15.08.2013 05:05, schrieb Leonard Bogard:
>>> ~$ whereis ffmpeg
>>> ffmpeg: /usr/local/bin/ffmpeg
>>
>> No, it isn't the ffmpeg you installed follow the instruction at
>> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide . I have a
>> glance at this instruction, the correct ffmpeg here should
>> be: /home/INSTALL_USER/bin/ffmpeg .
>>
>> So, try "/home/INSTALL_USER/bin/ffmpeg" when login with
>> another user.
> 
> Simple solution. GAH!  Thanks, it worked

now ask yourself two things careful:

* why in the world has one user permission to the userhome of another one
  on your machine

* why do you use instructions installing in your userhome if you
  want to share the binary with other users?

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From dashing.meng at gmail.com  Thu Aug 15 05:35:09 2013
From: dashing.meng at gmail.com (littlebat)
Date: Thu, 15 Aug 2013 11:35:09 +0800
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <520C475E.8090300@thelounge.net>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
	<CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
	<20130815104203.e255b849.dashing.meng@gmail.com>
	<CA+JzsNQc1vZr+j54MA_dPVW+yiHbaAuEokMOPESS5vKZsR=4Aw@mail.gmail.com>
	<520C475E.8090300@thelounge.net>
Message-ID: <20130815113509.248654ae.dashing.meng@gmail.com>

On Thu, 15 Aug 2013 05:13:34 +0200
Reindl Harald <h.reindl at thelounge.net> wrote:

> 
> 
> Am 15.08.2013 05:05, schrieb Leonard Bogard:
> >>> ~$ whereis ffmpeg
> >>> ffmpeg: /usr/local/bin/ffmpeg
> >>
> >> No, it isn't the ffmpeg you installed follow the instruction at
> >> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide . I have
> >> a glance at this instruction, the correct ffmpeg here should
> >> be: /home/INSTALL_USER/bin/ffmpeg .
> >>
> >> So, try "/home/INSTALL_USER/bin/ffmpeg" when login with
> >> another user.
> > 
> > Simple solution. GAH!  Thanks, it worked
> 
> now ask yourself two things careful:
> 
> * why in the world has one user permission to the userhome of another
> one on your machine
> 
> * why do you use instructions installing in your userhome if you
>   want to share the binary with other users?
> 
Yes, if your ffmpeg will be shared with many users on one machine,
e.g., a media server. It is best to install it in a system-wide place,
and avoid to conflict with the Ubuntu repository version ffmpeg,
e.g., /opt/custom/. But this need some modification again the
instruction for Ubuntu above, and also need you add the ffmpeg path
ahead of system path, e.g., append line: export PATH="/opt/custom/bin:
$PATH" into your "/etc/profile" file.

From leonard at kcfchurch.org  Thu Aug 15 06:33:05 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Wed, 14 Aug 2013 21:33:05 -0700
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <520C475E.8090300@thelounge.net>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
	<CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
	<20130815104203.e255b849.dashing.meng@gmail.com>
	<CA+JzsNQc1vZr+j54MA_dPVW+yiHbaAuEokMOPESS5vKZsR=4Aw@mail.gmail.com>
	<520C475E.8090300@thelounge.net>
Message-ID: <CA+JzsNSiw1ezmCarp3TxLguGzq7az0L=Sv4uCcwxUb+omrfvcA@mail.gmail.com>

On Wed, Aug 14, 2013 at 8:13 PM, Reindl Harald <h.reindl at thelounge.net>wrote:

>
>
> Am 15.08.2013 05:05, schrieb Leonard Bogard:
> >>> ~$ whereis ffmpeg
> >>> ffmpeg: /usr/local/bin/ffmpeg
> >>
> >> No, it isn't the ffmpeg you installed follow the instruction at
> >> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide . I have a
> >> glance at this instruction, the correct ffmpeg here should
> >> be: /home/INSTALL_USER/bin/ffmpeg .
> >>
> >> So, try "/home/INSTALL_USER/bin/ffmpeg" when login with
> >> another user.
> >
> > Simple solution. GAH!  Thanks, it worked
>
> now ask yourself two things careful:
>
> * why in the world has one user permission to the userhome of another one
>   on your machine
>

To be honest, I have no idea, and at this point I really don't care since
there isn't going to be a large userspace anyways.  Infact, only www-data
(the account the webserver is running from) needs access.  I absolutely
agree that its not ideal and infact if there was a step-by-step WORKING
instructions (that I could find) for installing compiled ffmpeg with the
libraries needed globally I would have done it.  Infact, I would have been
just fine with the fake ffmpeg if it would have worked.


> * why do you use instructions installing in your userhome if you
>   want to share the binary with other users?
>

Because it was the only one I could get working the way I needed it.  Had
their been a complete install instructions that worked, I would have used
it.

Point of note: the third party PPA that someone suggested didn't work
either.  I forget what all I had to change and strip out of my original
ffmpeg startup script but in the end it just would not work.  I'm not sure
but I think that version of ffmpeg wasn't able to discern the audio stream
from my Blackmagic Decklink Mini Recorder (through bmdcapture).  I don't
know why (if my assumption is correct) since it does seem to be compiled
from the latest version of ffmpeg.

I am still very amenable to and can still try to install a proper globally
accessible compiled ffmpeg (with libfdk-aac, libfaac, libx264, libmp3lame,
libopus, libvpx.)  However, I need to go live with this new machine Sunday
and it needs to be working from the web page for easy and fast access.

From h.reindl at thelounge.net  Thu Aug 15 09:42:59 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Thu, 15 Aug 2013 09:42:59 +0200
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <CA+JzsNSiw1ezmCarp3TxLguGzq7az0L=Sv4uCcwxUb+omrfvcA@mail.gmail.com>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
	<CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
	<20130815104203.e255b849.dashing.meng@gmail.com>
	<CA+JzsNQc1vZr+j54MA_dPVW+yiHbaAuEokMOPESS5vKZsR=4Aw@mail.gmail.com>
	<520C475E.8090300@thelounge.net>
	<CA+JzsNSiw1ezmCarp3TxLguGzq7az0L=Sv4uCcwxUb+omrfvcA@mail.gmail.com>
Message-ID: <520C8683.9010103@thelounge.net>



Am 15.08.2013 06:33, schrieb Leonard Bogard:
> On Wed, Aug 14, 2013 at 8:13 PM, Reindl Harald <h.reindl at thelounge.net>wrote:
>> Am 15.08.2013 05:05, schrieb Leonard Bogard:
>>>>> ~$ whereis ffmpeg
>>>>> ffmpeg: /usr/local/bin/ffmpeg
>>>>
>>>> No, it isn't the ffmpeg you installed follow the instruction at
>>>> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide . I have a
>>>> glance at this instruction, the correct ffmpeg here should
>>>> be: /home/INSTALL_USER/bin/ffmpeg .
>>>>
>>>> So, try "/home/INSTALL_USER/bin/ffmpeg" when login with
>>>> another user.
>>>
>>> Simple solution. GAH!  Thanks, it worked
>>
>> now ask yourself two things careful:
>>
>> * why in the world has one user permission to the userhome of another one
>>   on your machine
>>
> 
> To be honest, I have no idea, and at this point I really don't care since
> there isn't going to be a large userspace anyways.  Infact, only www-data
> (the account the webserver is running from) needs access

*wow* this should become a production webserver and you have no idea
why everybody and his brother can access userhomes?

why do you not hire someone with a clue?

>> * why do you use instructions installing in your userhome if you
>> want to share the binary with other users?
> 
> Because it was the only one I could get working the way I needed it. Had
> their been a complete install instructions that worked, I would have used

what about asking for help with specific querstions and problem
descriptions if you are about?

this way below you get ffmpeg with a *static linked* recent x264
well you need to switch the path because obviously the fool built
you installed package is still using /usr/local/ and enable some
things i do not need

but the point is i can install the this way built package parallel
to the Fedora ones (even both as RPM because i never ever would build
as root or install without making a distribution package) and i replaced
the rpm-macros with direct commands and paths for you........

./configure \
 --host=x86_64-redhat-linux \
 --prefix=/usr/local \
 --exec-prefix=/usr/local \
 --bindir=/usr/local/bin \
 --libdir=/usr/local/lib64 \
 --includedir=/usr/local/include \
 --extra-ldflags="-Wl,-z,now -Wl,-z,relro,-z,noexecstack" \
 --enable-static \
 --disable-shared \
 --disable-avs \
 --disable-cli \
 --disable-ffms \
 --disable-gpac \
 --disable-lavf \
 --disable-opencl \
 --disable-swscale
make
make install

../configure \
 --prefix=/usr/local \
 --bindir=/usr/local/bin \
 --datadir=/usr/local/share/ffmpeg \
 --incdir=/usr/local/include/ffmpeg \
 --libdir=/usr/local/lib64 \
 --mandir=/usr/local/man \
 --arch=x86_64 \
 --extra-cflags="-I/usr/local/x264" \
 --extra-ldflags="-I/usr/local/x264 -L/usr/local/x264 -ldl -Wl,-z,now -Wl,-z,relro,-z,noexecstack" \
 --enable-nonfree \
 --enable-gpl \
 --enable-version3 \
 --enable-libmp3lame \
 --enable-libtheora \
 --enable-libx264 \
 --enable-libvpx \
 --disable-libopus \
 --enable-static \
 --enable-libvorbis \
 --enable-libfaac \
 --enable-runtime-cpudetect \
 --disable-librtmp \
 --disable-shared \
 --disable-libxvid \
 --disable-libspeex \
 --disable-libopencore-amrnb \
 --disable-libopencore-amrwb \
 --disable-libopenjpeg \
 --disable-libopencv \
 --disable-libschroedinger \
 --disable-libgsm \
 --disable-libfreetype \
 --disable-hwaccels \
 --disable-libdc1394 \
 --disable-devices \
 --disable-amd3dnow \
 --disable-protocol=gopher \
 --disable-ffplay \
 --disable-ffprobe \
 --disable-avdevice \
 --disable-debug \
 --disable-htmlpages \
 --disable-podpages \
 --disable-txtpages
make
make install

> Point of note: the third party PPA that someone suggested didn't work either

noboddy can help with "didn't help either"
*what exactly was the commandline and the uncutted output*

> I forget what all I had to change and strip out of my original
> ffmpeg startup script but in the end it just would not work.

what is a ffmpe startup script?

> I am still very amenable to and can still try to install a proper globally
> accessible compiled ffmpeg (with libfdk-aac, libfaac, libx264, libmp3lame,
> libopus, libvpx.)  

i do not believe you need libfdk-aac *and* libfaac
they are doing *the same*

> However, I need to go live with this new machine Sunday
> and it needs to be working from the web page for easy and fast access.

do yourself and the internet a favor and do *not* connect servers
in such a braindead timeline to the internet while you have no
clue how your operating system works - why does everybody and his
brother think he needs to play admin?

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From francois.visagie at gmail.com  Thu Aug 15 10:00:27 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Thu, 15 Aug 2013 10:00:27 +0200
Subject: [FFmpeg-user] Preserving perceived loudness when downmixing
	audio from 5.1 AC3 to stereo AAC
In-Reply-To: <20130814144459.GA31873@phare.normalesup.org>
References: <51FEB571.2080701@gmail.com>	<006101ce91ca$aeb9da40$0c2d8ec0$@gmail.com>	<52018A9E.4050604@gmail.com>	<002101ce934c$f6c07a50$e4416ef0$@gmail.com>	<20130807145029.GB25535@phare.normalesup.org>	<005501ce9389$548a5e80$fd9f1b80$@gmail.com>	<20130807180030.GB14079@phare.normalesup.org>	<000301ce98ea$eb8ae280$c2a0a780$@gmail.com>	<20130814124402.GA16262@phare.normalesup.org>	<004001ce98fb$06644190$132cc4b0$@gmail.com>
	<20130814144459.GA31873@phare.normalesup.org>
Message-ID: <003e01ce998d$847c7330$8d755990$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Nicolas George
> Sent: 14 August 2013 16:45
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] Preserving perceived loudness when
> downmixing audio from 5.1 AC3 to stereo AAC
> 
> Le septidi 27 thermidor, an CCXXI, Francois Visagie a ?crit?:
> > Apologies, I do not know enough about this field to express myself
> > clearly. I'll try to improve: I'm looking for an ffmpeg mechanism that
> > automatically determines and uniformly applies to the audio stream a
> > gain factor that ensures maximum output levels without clipping.
> 
> What you are looking for is the output of volumedetect. See the
> documentation for explanations on what every field means.
> 
> >							      An added
> > benefit would be the ability to specify the maximum permitted output
> >level  as a percentage of maximum possible output level.
> 
> volumedetect provides the mean volume of the file, you can do the
> arithmetic from there. ebur128 provides a more realistic measure of the
> mean volume, but it is slower.
> 
> Any such method requires two passes to work correctly, since volume peaks
> can be present at the very end of the stream.

Thank you.

Kind regards,
Francois


From phil_rhodes at rocketmail.com  Thu Aug 15 11:11:38 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Thu, 15 Aug 2013 02:11:38 -0700 (PDT)
Subject: [FFmpeg-user] ffprobe -show_frames and coded_picture_number
In-Reply-To: <1376519235178-4660721.post@n4.nabble.com>
References: <52053303.7070107@gmail.com>	<1376074113012-4660603.post@n4.nabble.com>	<52057604.30206@gmail.com>	<1376106209176-4660606.post@n4.nabble.com>	<CAEnqZEVB27Rbwi+atqxS+YmVfg7wohjkNiAKxOqPhNOu=x1iKg@mail.gmail.com>	<1376170825784-4660622.post@n4.nabble.com>	<20130810232130.GA11654@phare.normalesup.org>	<1376184010720-4660626.post@n4.nabble.com>	<1376367195683-4660674.post@n4.nabble.com>	<520BFE57.7000307@gmail.com>
	<1376519235178-4660721.post@n4.nabble.com>
Message-ID: <1376557898.65035.YahooMailNeo@web121101.mail.ne1.yahoo.com>

> Speeding the audio would?affect the ?pitch? and other effects.?

This is invariably fixed using clever digital algorithms in modern conversions.

> To me personally, I would support the standardization of frame rate at 24

>?FPS for all videos/films created.?

Ah, wouldn't we all.

Actually, we wouldn't, as things like broadcast sports would suffer horribly from stroboscopic motion.

> The real reasons for the old standards are?still in use is more about?
> traditions or what people already have or are?used to.?

Well, hang on a minute. The reason the old standards are still in use is because the world is still full of standard-definition TV sets and related equipment. Why d'you think HD broacast formats still include the fractional frame rates? So that they can be downconverted to standard-def in realtime without having to deal with complicated frame rate conformation issues.

Nobody likes the issues that arise from this; they're just less objectionable issues than the alternative.

> Is it really?necessary to change frame rate when I view film materials or digitally
>?created 24-FPS video on LCD display??

The "frame rate" (that is, the duration for which each frame is displayed) certainly is being changed, regardless of anyone's opinion on the matter.?

P

From andrey.krieger.utkin at gmail.com  Thu Aug 15 12:09:40 2013
From: andrey.krieger.utkin at gmail.com (Andrey Utkin)
Date: Thu, 15 Aug 2013 13:09:40 +0300
Subject: [FFmpeg-user] problem using http input
In-Reply-To: <014f01ce98e3$2ce8b750$86ba25f0$@gmail.com>
References: <014f01ce98e3$2ce8b750$86ba25f0$@gmail.com>
Message-ID: <CANZNk80vZfQDLs4GQTO5LNR6QYukgp3GRDBymt8kVZ5ea=S3Mw@mail.gmail.com>

Please give a full ffmpeg command and its output, and dump of HTTP
traffic sniff.

-- 
Andrey Utkin

From soho123.2012 at gmail.com  Thu Aug 15 16:01:00 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Thu, 15 Aug 2013 22:01:00 +0800
Subject: [FFmpeg-user] how to mux mjpeg video into udp format
Message-ID: <CAFd2cZUwRZ04iRZzz2Fc199itzTTJKfQecOw019mQmtciyQ-Hg@mail.gmail.com>

Hi all,

the ffserver.conf I use for mjpeg stream is "http".
I can use the url "http://192.168.1.254:8090/test.mjpeg" to get mjpeg stream

the stream config is :
<Stream test.mjpeg>
Feed feed1.ffm
Format mjpeg
VideoSize 1280x720
VideoFrameRate 30
VideoCodec mjpeg
NoAudio
</Stream>
can I mux mjpeg into "udp"?

From mehmetalilk at hotmail.com  Thu Aug 15 14:04:58 2013
From: mehmetalilk at hotmail.com (Mehmet Ali)
Date: Thu, 15 Aug 2013 15:04:58 +0300
Subject: [FFmpeg-user] ffmpeg recording problem while
Message-ID: <DUB107-W3A0BA4DE132B931C28D26B0460@phx.gbl>

Hi there,
I demux from an ip cam h264 video stream and draw it on screen. It is working well but when i record the same stream ( just demux and record not decode) ffplay.exe cant play the file..Why this code doesnt work? If i have to write somethings like header before frames to file please clarify me.. Thank you for your help.. 
My Code like this:
        //create file	pVideo=fopen("VideoTest.264","ab+");		//start a loop and using demux of stream capture video and decode by decoder of stream			if(!stream.mDecoder.AllocateFrame(frame))	{		TRACE("frame Allocate error\n");		return 0;	}		if(!stream.mDecoder.AllocateFrame(scaledFrame))	{		TRACE("scaledFrame Allocate Error\n");		return 0;	}
	if(!stream.mDecoder.InitializeFrame(scaledFrame,stream.mDemuxer.mStreamInfo.mWidth,stream.mDemuxer.mStreamInfo.mHeight,mPxl))	{		TRACE("scaledFrame ?nitialize Error\n");		return 0;	}	DWORD start,end,onesecst;	int framerate=0;	bool startmes=true;	CString frm;	bool isRun=true;	int ret=-1;
		static int frameno=0;	while(isRun) 	{		if(startmes)		{			onesecst=GetTickCount();			startmes=false;		}		start=GetTickCount();				ret=WaitForSingleObject(stream.mHandletoStop,1); //HACK timeout ile ??kmal? reset event ?a??r?l?nca		if(ret==WAIT_OBJECT_0)		{			TRACE("Exit event fired!.\n");			isRun=false;		}
		memset(videoBuffer,0,256*1024);
		if((encodedFrameSize=stream.mDemuxer.GetFrame(videoBuffer))<0)		{			TRACE("GetFrame Error\n");			break;		}		if(!stream.mDecoder.Decode(videoBuffer,encodedFrameSize,frame))		{			TRACE("Decode error\n");			continue;		}
		if(pthis->WillSend)		SetEvent(pthis->SendToCMS);
		int totalWritten=0;		int rest=encodedFrameSize;		DWORD written=0;        while(rest>0)        {            written=fwrite(videoBuffer+totalWritten,1,rest,pVideo);            if(written<0)                break;
            totalWritten+=written;            rest-=written;        }		if(!stream.mDecoder.Scale(frame,scaledFrame,stream.mDemuxer.mStreamInfo.mWidth,stream.mDemuxer.mStreamInfo.mHeight,mPxl))		{			TRACE("scale error\n");			break;		}				stream.mDrawer->Display(scaledFrame->data[0],stream.mRect);
		end=GetTickCount();		int abc=end - onesecst;				framerate++;		totalEncodedSize+=encodedFrameSize;		if(end - onesecst>=1000)		{

			frm.Format("Frame Rate:%d\n",framerate);			pthis->sttcFrameRate.SetWindowText(frm);				frm.Format("Bit Rate:%lu kBps\n",8*totalEncodedSize/(1024));			pthis->sttcBitRate.SetWindowText(frm);			startmes=true;			framerate=0;			totalEncodedSize=0;		}

			}
I used to play file with ffplay this string
E:\ffmpeg\20130703\bin>ffplay.exe -f h264 -analyzeduration 258522508 -probesize958852258 VideoTest.264
ffplay.exe return some warnings
[h264 @ 032741a0] non-existing PPS referenced[h264 @ 032741a0] non-existing PPS 0 referenced[h264 @ 032741a0] decode_slice_header error[h264 @ 032741a0] no frame![h264 @ 032741a0] non-existing PPS referenced[h264 @ 032741a0] non-existing PPS 0 referenced[h264 @ 032741a0] decode_slice_header error[h264 @ 032741a0] no frame![h264 @ 032741a0] non-existing PPS referenced[h264 @ 032741a0] non-existing PPS 0 referenced[h264 @ 032741a0] decode_slice_header error[h264 @ 032741a0] no frame![h264 @ 032741a0] non-existing PPS referenced[h264 @ 032741a0] non-existing PPS 0 referenced[h264 @ 032741a0] decode_slice_header error[h264 @ 032741a0] no frame![h264 @ 032741a0] non-existing PPS referenced[h264 @ 032741a0] non-existing PPS 0 referenced[h264 @ 032741a0] decode_slice_header error[h264 @ 032741a0] no frame![h264 @ 032741a0] non-existing PPS referenced[h264 @ 0326bc40] decoding for stream 0 failed[h264 @ 0326bc40] Could not find codec parameters for stream 0 (Video: h264): unspecified sizeConsider increasing the value for the 'analyzeduration' and 'probesize' optionsVideoTest.264: could not find codec parameters

My working environment 
FFmpeg version: 2013-07-22 git-419a3d8VS 2012 Mfc App.Windows 8
Mehmet Ali 		 	   		  

From dzungng89 at gmail.com  Thu Aug 15 20:55:23 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Thu, 15 Aug 2013 13:55:23 -0500
Subject: [FFmpeg-user] Convert image to MPEG transport stream
Message-ID: <CABOWChz4m8cZq0jUDbRrgQj9ZvNFp251y2-H9b-fWaWt5vOhjA@mail.gmail.com>

How to convert an image to video under TS format?

This command is used to convert to H264 format:

ffmpeg -loop 1 -f image2 -i image.png -c:v libx264 -t 30 output.mp4

From lou at lrcd.com  Thu Aug 15 21:11:50 2013
From: lou at lrcd.com (Lou)
Date: Thu, 15 Aug 2013 11:11:50 -0800
Subject: [FFmpeg-user] ffmpeg only works for one user.
In-Reply-To: <520C8683.9010103@thelounge.net>
References: <CA+JzsNQaY-uU+i_x8tWz+Htr0=Z864U5J5UQOOi+UM0KyHAbbw@mail.gmail.com>
	<520BE9E7.70207@thelounge.net>
	<CA+JzsNRdmkrYEQ3tdXZVtm++DHZqQnWowiw5VqzFzmBf8Z0xsQ@mail.gmail.com>
	<520BF599.4090505@thelounge.net>
	<CA+JzsNT0Y6knmvWgevi6JUouqa5YyjWTJ29DeVo68tjTXqqNAQ@mail.gmail.com>
	<20130815104203.e255b849.dashing.meng@gmail.com>
	<CA+JzsNQc1vZr+j54MA_dPVW+yiHbaAuEokMOPESS5vKZsR=4Aw@mail.gmail.com>
	<520C475E.8090300@thelounge.net>
	<CA+JzsNSiw1ezmCarp3TxLguGzq7az0L=Sv4uCcwxUb+omrfvcA@mail.gmail.com>
	<520C8683.9010103@thelounge.net>
Message-ID: <20130815111150.713e60c8@lrcd.com>

On Thu, 15 Aug 2013 09:42:59 +0200
Reindl Harald <h.reindl at thelounge.net> wrote:

> *wow* this should become a production webserver and you have no idea
> why everybody and his brother can access userhomes?
> 
> why do you not hire someone with a clue?

This type of attitude is not necessary for this mailing list.

From werner.robitza at gmail.com  Thu Aug 15 21:55:13 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Thu, 15 Aug 2013 21:55:13 +0200
Subject: [FFmpeg-user] Convert image to MPEG transport stream
In-Reply-To: <CABOWChz4m8cZq0jUDbRrgQj9ZvNFp251y2-H9b-fWaWt5vOhjA@mail.gmail.com>
References: <CABOWChz4m8cZq0jUDbRrgQj9ZvNFp251y2-H9b-fWaWt5vOhjA@mail.gmail.com>
Message-ID: <CAN0LTaP74T8vpuF2C9+1E0kM99eYmYcOU4bqCgR29zo=4yp1cQ@mail.gmail.com>

On Thu, Aug 15, 2013 at 8:55 PM, Dzung Nguyen <dzungng89 at gmail.com> wrote:

> How to convert an image to video under TS format?
> This command is used to convert to H264 format:
>

Simply change the extension to .ts instead.

Please note that H.264 is not a "format" ? it's a video codec. Your command
encodes an image with the x264 encoder (a H.264 encoder) to the MP4
container/format (MPEG-4 Part 14 to be precise).

From sjh431 at gmail.com  Fri Aug 16 04:14:01 2013
From: sjh431 at gmail.com (jhseo)
Date: Fri, 16 Aug 2013 11:14:01 +0900
Subject: [FFmpeg-user] problem using http input
In-Reply-To: <CANZNk80vZfQDLs4GQTO5LNR6QYukgp3GRDBymt8kVZ5ea=S3Mw@mail.gmail.com>
References: <014f01ce98e3$2ce8b750$86ba25f0$@gmail.com>
	<CANZNk80vZfQDLs4GQTO5LNR6QYukgp3GRDBymt8kVZ5ea=S3Mw@mail.gmail.com>
Message-ID: <023801ce9a26$49751e10$dc5f5a30$@gmail.com>

Thank for an interest.

I use ffmpeg dll files, but under ffmpeg command requests same way(HTTP
GET).

-. ffmpeg command
 ffmpeg -i http://xxx.xxx.xxx.xxx/test.MOV -pass 1 -y -vcodec libx264
-vprofile baseline -b:v 512k -s 320x240 out.mp4


-. output
ffmpeg version N-55393-g3b2e99f Copyright (c) 2000-2013 the FFmpeg
developers  built on Aug  8 2013 21:37:55 with gcc 4.7.3 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib
--enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv
--enable-libass --enable-libbluray
--enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc
--enable-libmodplug --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp
--enable-libschroedinger --enable-libsoxr --enable-libspeex
--enable-libtheora --enable-libtwolame
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libx264
--enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 24.100 / 55. 24.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'http://xxx.xxx.xxx.xxx/test.MOV':
  Metadata:
    major_brand     : qt
    minor_version   : 0
    compatible_brands: qt
    model           : iPhone 5
    model-kor       : iPhone 5
    encoder         : 6.1.4
    encoder-kor     : 6.1.4
    make            : Apple
    make-kor        : Apple
  Duration: 00:01:37.65, start: 0.000000, bitrate: 17292 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1920x1080, 17220 kb/s, 29.97 fps, 29.97 tbr, 600 tbn, 1200 tbc
    Metadata:
      handler_name    : Core Media Data Handler
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp,
62 kb/s
    Metadata:
      handler_name    : Core Media Data Handler
[libx264 @ 00000000002d7cc0] using cpu capabilities: MMX2 SSE2Fast SSSE3
SSE4.2 AVX
[libx264 @ 00000000002d7cc0] profile Constrained Baseline, level 2.0
[libx264 @ 00000000002d7cc0] 264 - core 135 r2345 f0c1c53 - H.264/MPEG-4 AVC
codec - Copyleft 2003-2
013 - http://www.videolan.org/x264.html - options: cabac=0 ref=1
deblock=1:0:0 analyse=0x1:0 me=dia
subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0
8x8dct=0 cqm=0 deadzone
=21,11 fast_pskip=1 chroma_qp_offset=0 threads=18 lookahead_threads=1
sliced_threads=0 nr=0 decimate
=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0
keyint=250 keyint_min=25 sce
necut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=512
ratetol=1.0 qcomp=0.60 qpmin=0
qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'out.mp4':
  Metadata:
    major_brand     : qt
    minor_version   : 0
    compatible_brands: qt
    make-kor        : Apple
    model           : iPhone 5
    model-kor       : iPhone 5
    make            : Apple
    encoder-kor     : 6.1.4
    encoder         : Lavf55.13.102
    Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021),
yuv420p, 320x240, q=-1--1, pass 1, 512 kb/s, 30k tbn, 29.97 tbc
    Metadata:
      handler_name    : Core Media Data Handler
    Stream #0:1(und): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
44100 Hz, mono, s16, 128 kb/s
    Metadata:
      handler_name    : Core Media Data Handler
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (aac -> libvo_aacenc)
Press [q] to stop, [?] for help
frame= 2927 fps= 34 q=-1.0 Lsize=    7669kB time=00:01:37.69 bitrate=
643.1kbits/s
video:6060kB audio:1527kB subtitle:0 global headers:0kB muxing overhead
1.081423%
[libx264 @ 00000000002d7cc0] frame I:15    Avg QP:21.27  size:  7174
[libx264 @ 00000000002d7cc0] frame P:2912  Avg QP:24.67  size:  2094
[libx264 @ 00000000002d7cc0] mb I  I16..4: 35.5%  0.0% 64.5%
[libx264 @ 00000000002d7cc0] mb P  I16..4: 10.0%  0.0%  0.0%  P16..4: 67.2%
0.0%  0.0%  0.0%  0.0%   skip:22.8%
[libx264 @ 00000000002d7cc0] final ratefactor: 21.90
[libx264 @ 00000000002d7cc0] coded y,uvDC,uvAC intra: 38.4% 36.8% 11.5%
inter: 33.4% 18.0% 1.3%
[libx264 @ 00000000002d7cc0] i16 v,h,dc,p: 50% 21% 21%  9%
[libx264 @ 00000000002d7cc0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 22% 14%  6%
6%  8%  7%  7%  7%
[libx264 @ 00000000002d7cc0] i8c dc,h,v,p: 64% 16% 16%  4%
[libx264 @ 00000000002d7cc0] kb/s:508.26


-. http dump:
  v. the problem is "Range" field does not specify end of range.. my storage
is ready to send every request until end of file.
  v. follow request occur 20~30 time per second.
  v. start values of "Range" field are..
     1.mdat position 2.moov position 3.start position of a/v packet
      (this position info get from mp4 container)

(time: 8.465443 sec)
GET /test.MOV HTTP/1.1
User-Agent: Lavf54.1.100
Accept: */*
Range: bytes=0-
Connection: close
Host: xxx.xxx.xxx.xxx

(time: 8.468504 sec)
GET /test.MOV HTTP/1.1
User-Agent: Lavf54.1.100
Accept: */*
Range: bytes=211028448-
Connection: close
Host: xxx.xxx.xxx.xxx

(time: 8.478031 sec)
GET /test.MOV HTTP/1.1
User-Agent: Lavf54.1.100
Accept: */*
Range: bytes=36-
Connection: close
Host: xxx.xxx.xxx.xxx

(time: 8.486259 sec)
GET /test.MOV HTTP/1.1
User-Agent: Lavf54.1.100
Accept: */*
Range: bytes=17479-
Connection: close
Host: xxx.xxx.xxx.xxx

(time: 8.548716 sec)
GET /test.MOV HTTP/1.1
User-Agent: Lavf54.1.100
Accept: */*
Range: bytes=7390-
Connection: close
Host: xxx.xxx.xxx.xxx

(time: 8.564293 sec)
GET /test.MOV HTTP/1.1
User-Agent: Lavf54.1.100
Accept: */*
Range: bytes=275875-
Connection: close
Host: xxx.xxx.xxx.xxx

(time: 8.595498 sec)
GET /test.MOV HTTP/1.1
User-Agent: Lavf54.1.100
Accept: */*
Range: bytes=7573-
Connection: close
Host: xxx.xxx.xxx.xxx

.
.
.
.
.

-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org]
On Behalf Of Andrey Utkin
Sent: Thursday, August 15, 2013 7:10 PM
To: FFmpeg user questions
Subject: Re: [FFmpeg-user] problem using http input

Please give a full ffmpeg command and its output, and dump of HTTP traffic
sniff.

--
Andrey Utkin
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From sjh431 at gmail.com  Fri Aug 16 08:03:02 2013
From: sjh431 at gmail.com (jhseo)
Date: Fri, 16 Aug 2013 15:03:02 +0900
Subject: [FFmpeg-user] about "multiple_requests" option
Message-ID: <024601ce9a46$468e6b50$d3ab41f0$@gmail.com>

Hi, all.

 

I'm testing for "multiple_requests" option.

 

1. I set "multiple_requests" on "1" in case of using http input file.

2. when it get the bit-stream data, "Connection" field of http request has
value "keep-alive".(it was identified by wireshark packet capture.)

3. source ports are changed per http request connection.(increased by one)

 

Is it right?

 

It is beyond my expectation..

( If "multiple_request" set "1", ffmpeg use one port(at least, less than
when the option set "0") ).

 

Please advise my ignorance.


From doenerbudenchef at yahoo.de  Fri Aug 16 17:07:34 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Fri, 16 Aug 2013 08:07:34 -0700 (PDT)
Subject: [FFmpeg-user] Convert progressive MXF to interlaced
Message-ID: <1376665654187-4660746.post@n4.nabble.com>

Hi all,

I just want to convert a progressive MXF-File (XDCAM MPEG HD422, 1080p,
25fps) to interlaced 50fps, tff. It is important to leave the MXF container
with all meta data as it is. What is the best way to do that?

Thanks in advance.






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From mathieu.multicam+ffmpeg at gmail.com  Fri Aug 16 11:36:00 2013
From: mathieu.multicam+ffmpeg at gmail.com (MatCam)
Date: Fri, 16 Aug 2013 02:36:00 -0700 (PDT)
Subject: [FFmpeg-user] Merging H263 videos
Message-ID: <1376645760323-4660745.post@n4.nabble.com>

Hi I'm currently trying to merge 2 h263 videos.
When i try to merge them i firstly had only the first video available in the
merged version.

So following FAQ of FFMPEG, I tried to use intermediate format to avoid
merging problem due to file headers.

After many test here my command  (mpg is my intermediate format, is ts
better or faster ?)
ffmpeg.exe -threads 8 -i "1.mpg|2.mpg" -f mpeg -bufsize 1000k -maxrate 5000k
-b:v 500k  -vcodec h263 -s 704x576  output.mp4

As you can see I do weird stuff with bitrate and other rates to avoid errors
but even when i have no errors my video is empty (good duration but without
my video ...)

Could you show me an example of command line to simply merge Two ts (or mpg)
files in a h263 one ?

Thanks.

Sincerely
mathieu



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Merging-H263-videos-tp4660745.html
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From jhbell50 at hotmail.com  Fri Aug 16 15:44:15 2013
From: jhbell50 at hotmail.com (john Bell)
Date: Fri, 16 Aug 2013 09:44:15 -0400
Subject: [FFmpeg-user] Command error
Message-ID: <BLU178-W29F262C38E515E6E1BF91FCF470@phx.gbl>

ffmpeg.exe -cookies "gandbolsteauaaakeys=ZCV4upgVjq5_RjK9yPYNUQ; gandbolsteauaaakeyt=1376674191; path=\; domain=37.221.172.85; \n" -i "http://37.221.172.85/hqstream/gandbolsteauaaa/playlist.m3u8?keys=ZCV4upgVjq5_RjK9yPYNUQ&keyt=1376674191" 
 
fails with
 
[http @ 03ddac00] Invalid cookie found, no value, path or domain specified
 
but I belive the command conforms to the code for verifying the value, path and domain.
 
Any suggestions?


 		 	   		  

From bradleycbotkin at gmail.com  Fri Aug 16 19:45:30 2013
From: bradleycbotkin at gmail.com (Botkin Brad)
Date: Fri, 16 Aug 2013 13:45:30 -0400
Subject: [FFmpeg-user] multiple programs (ServiceXX) in a single TS
Message-ID: <D4265A9D-7F83-4AE1-BE8A-E8EFEC11C1C5@gmail.com>

Is there any way to coerce ffmpeg to copy existing PMTs from input streams into the output stream.  I have two ATSC TS streams produced thusly:

CONVERT FLVs ---> ATSC TS, using Program #4:
./ffmpeg -i foo1.flv \
		-c:a aac -strict -2 -b:a 128k \
		-c:v mpeg2video -r 30000/1001 \
		-mpegts_service_id 3 -muxrate 19392636 -f mpegts -y \
		foo1.ts



./ffmpeg -i foo2.flv \
		-c:a aac -strict -2 -b:a 128k \
		-c:v mpeg2video -r 30000/1001 \
		-mpegts_service_id 4 -muxrate 19392636 -f mpegts -y \
		foo2.ts

Admittedly, I needn't have created the full 19.38 Mb/s framerate for each of them, but that was part of an earlier test just to make sure each TS would play fine through a system. And they do.

What I'd like it so merge them into a single TS that preserves both PMTs from the input files.  I don't seem to be able to reference the ServiceXX stream like I can the video and audio streams in the input files.

Is this possible with ffmpeg?
Thanks.
--Brad

From lou at lrcd.com  Fri Aug 16 20:19:04 2013
From: lou at lrcd.com (Lou)
Date: Fri, 16 Aug 2013 10:19:04 -0800
Subject: [FFmpeg-user] Command error
In-Reply-To: <BLU178-W29F262C38E515E6E1BF91FCF470@phx.gbl>
References: <BLU178-W29F262C38E515E6E1BF91FCF470@phx.gbl>
Message-ID: <20130816101904.2b4d8fe6@lrcd.com>

On Fri, 16 Aug 2013 09:44:15 -0400
john Bell <jhbell50 at hotmail.com> wrote:

> ffmpeg.exe -cookies "gandbolsteauaaakeys=ZCV4upgVjq5_RjK9yPYNUQ; gandbolsteauaaakeyt=1376674191; path=\; domain=37.221.172.85; \n" -i "http://37.221.172.85/hqstream/gandbolsteauaaa/playlist.m3u8?keys=ZCV4upgVjq5_RjK9yPYNUQ&keyt=1376674191" 
>  
> fails with
>  
> [http @ 03ddac00] Invalid cookie found, no value, path or domain specified
>  
> but I belive the command conforms to the code for verifying the value, path and domain.
>  
> Any suggestions?

You need to include the complete ffmpeg console output and not just a
segment.

From vvubin at gmail.com  Fri Aug 16 21:31:06 2013
From: vvubin at gmail.com (Wu-bin Zhen)
Date: Fri, 16 Aug 2013 15:31:06 -0400
Subject: [FFmpeg-user] Buffer Size for Stream Copy
Message-ID: <CAGK10ZNKQPOjRjC=RP2h0isuudyUVQe1w2Jb50mSoy1dU2yCFg@mail.gmail.com>

Hello ffmpeg,

I am using ffmpeg to grab h264 video stream and aac audio stream from an
internet based hardware encoder with the codec copy option, and it works
pretty well. Now, I start to wonder if I can use the '-bufsize' option for
the stream copy, just in case when the network is temporally slow down, or
the encoder is suddenly chocking for some reason. I googled a little bit
about the 'bufsize' option, and found that most people use the option for
encoding.

So my question is: Do you think using 'bufsize' option will be helpful for
the stream copy? If so, what would be the recommended buffer size? My
internet encoder produces video stream with 4000kbits/sec, and audio stream
with 128kbits/sec.

Thanks, and have a great weekend.
-w

From j+ffmpeg at lundstocholm.dk  Fri Aug 16 21:37:40 2013
From: j+ffmpeg at lundstocholm.dk (jlundstocholm)
Date: Fri, 16 Aug 2013 12:37:40 -0700 (PDT)
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
Message-ID: <1376681860279-4660751.post@n4.nabble.com>

Hi Paul,

> ffmpeg -ss 213 -i MySourceMovie.m4v -c:v copy -c:a copy testoutput.m4v

This works great and drastically reduces the splitting-time - thank you :-)

For the record - my final script is this - if someone else should need to
try the same thing. It took a bit of trial'n error to figure out the correct
syntax and usage of the -to switch.

FFMPEG takes less than two minutes to remove commercials from a 2GB movie
splitting in 5 files.



At some point I would like to use a different switch than -to since it takes
a /duration/ and not a specific time in the movie - hence forcing me to
calculate this duration. But that's for (another) rainy day.





--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Using-ffmpeg-to-remove-commercials-from-video-file-tp4660699p4660751.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From alirazeen at alirazeen.com  Fri Aug 16 22:32:28 2013
From: alirazeen at alirazeen.com (ali)
Date: Fri, 16 Aug 2013 13:32:28 -0700 (PDT)
Subject: [FFmpeg-user] x264 stream decoding question
Message-ID: <1376685148025-4660752.post@n4.nabble.com>

Hi folks,

I am using ffmpeg to decode a live x264 stream. This stream consists of only
I and P frames and does not contain any B frames. I'm noticing a weird
requirement in the decoding. 

It seems like the ffmpeg decoder only completes decoding a current frame X
if it has also received X+1. The very first frame in the stream is an
I-frame. However, that I-frame is not fully decoded until the next frame,
the first P frame in the GOP, is received. Is this behaviour expected? Isn't
an I-frame independent of other frames? Why should the next P frame be
required before the current I-frame is shown?

Has anyone else observed anything similar? Is there also a way to change the
behaviour of the decoder?

I am aware that I am not providing a lot of information on this post
regarding my current setup but that is because I am not aware what
information is required to answer this post. Please let me know if I should
provide any additional details.

Thanks!

Best,
Ali



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/x264-stream-decoding-question-tp4660752.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From werner.robitza at gmail.com  Sat Aug 17 09:52:38 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Sat, 17 Aug 2013 09:52:38 +0200
Subject: [FFmpeg-user] Merging H263 videos
In-Reply-To: <1376645760323-4660745.post@n4.nabble.com>
References: <1376645760323-4660745.post@n4.nabble.com>
Message-ID: <CAN0LTaN+N3we5AwEki4uTd3mLThmV5OPgF0hvJoHg-JnDrjLtw@mail.gmail.com>

On Fri, Aug 16, 2013 at 11:36 AM, MatCam
<mathieu.multicam+ffmpeg at gmail.com>wrote:

> Hi I'm currently trying to merge 2 h263 videos.
> When i try to merge them i firstly had only the first video available in
> the
> merged version.
>
> So following FAQ of FFMPEG, I tried to use intermediate format to avoid
> merging problem due to file headers.
>
> After many test here my command  (mpg is my intermediate format, is ts
> better or faster ?)
> ffmpeg.exe -threads 8 -i "1.mpg|2.mpg" -f mpeg -bufsize 1000k -maxrate
> 5000k

-b:v 500k  -vcodec h263 -s 704x576  output.mp4
>

We don't know how you converted your files to .mpg, but "-f mpeg" is wrong
here, because it specifies an MPEG-1 program stream, while the MP4
extension is for MPEG-4 Part 14 containers.
Don't forget that encoding your input videos to MPEG-1 or MPEG-2 will
result in quality degradation. Concatenating them again, and re-encoding
with a bitrate of 500 kBit/s to H.263 will degrade your quality even
further. Even if you got your concatenated video to play, it might look
really bad.


> As you can see I do weird stuff with bitrate and other rates to avoid
> errors
> but even when i have no errors my video is empty (good duration but without
> my video ...)
>

It'd be good if you told us what errors you get, instead of trying to add
various options in order to avoid them. Most ffmpeg conversions should work
"out of the box" without additional options. Maybe you can include the
full, uncut output of the above command. Then we know what material we're
working with here.

Thank you

From werner.robitza at gmail.com  Sat Aug 17 09:58:50 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Sat, 17 Aug 2013 09:58:50 +0200
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <1376681860279-4660751.post@n4.nabble.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
Message-ID: <CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>

> For the record - my final script is this - if someone else should need to
> try the same thing. It took a bit of trial'n error to figure out the
> correct
> syntax and usage of the -to switch.
>

Where is your script?


> At some point I would like to use a different switch than -to since it
> takes
> a /duration/ and not a specific time in the movie - hence forcing me to
> calculate this duration. But that's for (another) rainy day.
>

-to does not take a duration ? it takes the timestamp of where you would
like
encoding to end. This is exactly the option you need.

-t takes the encoding duration. Depending on your programming/scripting
language of choice it's rather easy to calculate the difference between two
HH:MM:SS.mmmm timestamps.

From francois.visagie at gmail.com  Sat Aug 17 11:39:19 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Sat, 17 Aug 2013 11:39:19 +0200
Subject: [FFmpeg-user] Merging H263 videos
In-Reply-To: <1376645760323-4660745.post@n4.nabble.com>
References: <1376645760323-4660745.post@n4.nabble.com>
Message-ID: <000001ce9b2d$a9862ee0$fc928ca0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of MatCam
> Sent: 16 August 2013 11:36
> To: ffmpeg-user at ffmpeg.org
> Subject: [FFmpeg-user] Merging H263 videos
> 
> Hi I'm currently trying to merge 2 h263 videos.
> When i try to merge them i firstly had only the first video available in
the
> merged version.
> 
> So following FAQ of FFMPEG, I tried to use intermediate format to avoid
> merging problem due to file headers.
> 
> After many test here my command  (mpg is my intermediate format, is ts
> better or faster ?) ffmpeg.exe -threads 8 -i "1.mpg|2.mpg" -f mpeg
-bufsize
> 1000k -maxrate 5000k -b:v 500k  -vcodec h263 -s 704x576  output.mp4

The syntax you're using here seems similar to that of the physical concat
protocol (http://www.ffmpeg.org/ffmpeg-all.html#concat-3) but not quite
correct (more precisely, I've never tried using it without the preceding
'concat:'. The physical concatenation protocol is IMO not the correct
mechanism to use here in any event, since as far as I can work out it does a
byte-level concatenation, which works for things like DVD VOB files.

In this case of separately muxed input files, headers will likely create
problems with physical concatenation as you point out. A more suitable
mechanism to try would be the virtual concatenation script demuxer
(http://www.ffmpeg.org/ffmpeg-all.html#concat-1). This demuxer seems to
operate at the packet level, hence one would expect header information to be
removed from the concatenation. The concat filter
(http://www.ffmpeg.org/ffmpeg-all.html#concat-2) may also be worth a look
although it seems more involved to set up and maintaining sync might become
an issue.

> 
> As you can see I do weird stuff with bitrate and other rates to avoid
errors
> but even when i have no errors my video is empty (good duration but
> without my video ...)

As pointed out elsewhere it's wrong to use the '-f mpeg' option with MP4
output. Since ffmpeg will correctly guess the intended output format from
the .mp4 file extension, you can just omit the '-f' option altogether. If
other problems remain afterwards, post your command line and full console
output.

> 
> Could you show me an example of command line to simply merge Two ts (or
> mpg) files in a h263 one ?
> 
> Thanks.
> 
> Sincerely
> mathieu
> 
> 
> 
> --
> View this message in context: http://ffmpeg-
> users.933282.n4.nabble.com/Merging-H263-videos-tp4660745.html
> Sent from the FFmpeg-users mailing list archive at Nabble.com.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From jpboard2 at yahoo.com  Sat Aug 17 15:51:58 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 17 Aug 2013 06:51:58 -0700 (PDT)
Subject: [FFmpeg-user] Problems with Select Filter
Message-ID: <1376747518.34172.YahooMailNeo@web164702.mail.gq1.yahoo.com>

I captured a video file in raw format ('mplayer -identify' says the format is:
[rawuyvy] vfm: raw (RAW UYVY)).? Then I did some cut edits with ffmpeg using
the 'Select' video filter, and at the same time set the output codec to
ffvhuff.? The problem is that those cut edits were not at the exact frames I
specified with the 'Select' video filter.? When I play the video, the cuts
happen at the wrong places (mostly towards the end of the file).

I'm trying to figure out why this occurs.? When I try to play the edited file
with mplayer, it displays these complaints:
??? AVI: ODML: Building ODML index (2 superindexchunks).
??? AVI: ODML: Broken (incomplete?) file detected. Will use traditional index.
??? AVI: Generated index table for 10983 chunks!
Does that help identify the problem?

However, if I first compress the raw video file with ffvhuff codec, save that
to an intermediate file, and then apply the same exact edits with the Select
filter to the ffvhuff-compressed file, then the file is edited correctly, and
I don't see the above ODML error message.

I want to figure this problem out so I don't inadvertantly reproduce it,
but not notice the incorrect edits until long after the source material
is deleted.? What is the problem?

From jpboard2 at yahoo.com  Sat Aug 17 15:56:38 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 17 Aug 2013 06:56:38 -0700 (PDT)
Subject: [FFmpeg-user] Detecting Scene Changes In Source Video?
Message-ID: <1376747798.41445.YahooMailNeo@web164702.mail.gq1.yahoo.com>

Is there a way to use ffmpeg to detect scene changes in a video file?? I know
some of the codecs must do this already, but I don't know if that output is
readily available?? Maybe in the info file produced when 2-pass encoding?

From onemda at gmail.com  Sat Aug 17 16:01:26 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 17 Aug 2013 14:01:26 +0000
Subject: [FFmpeg-user] Problems with Select Filter
In-Reply-To: <1376747518.34172.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1376747518.34172.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P4vGU8i9P1vSZC4nmtw8yfojNpB52+JDyeQ5_pHc_KKsw@mail.gmail.com>

On 8/17/13, James Board <jpboard2 at yahoo.com> wrote:
> I captured a video file in raw format ('mplayer -identify' says the format
> is:
> [rawuyvy] vfm: raw (RAW UYVY)).  Then I did some cut edits with ffmpeg
> using
> the 'Select' video filter, and at the same time set the output codec to
> ffvhuff.  The problem is that those cut edits were not at the exact frames
> I
> specified with the 'Select' video filter.  When I play the video, the cuts
> happen at the wrong places (mostly towards the end of the file).
>
> I'm trying to figure out why this occurs.  When I try to play the edited
> file
> with mplayer, it displays these complaints:
>     AVI: ODML: Building ODML index (2 superindexchunks).
>     AVI: ODML: Broken (incomplete?) file detected. Will use traditional
> index.
>     AVI: Generated index table for 10983 chunks!
> Does that help identify the problem?

Do not use avi.

>
> However, if I first compress the raw video file with ffvhuff codec, save
> that
> to an intermediate file, and then apply the same exact edits with the
> Select
> filter to the ffvhuff-compressed file, then the file is edited correctly,
> and
> I don't see the above ODML error message.
>
> I want to figure this problem out so I don't inadvertantly reproduce it,
> but not notice the incorrect edits until long after the source material
> is deleted.  What is the problem?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From jpboard2 at yahoo.com  Sat Aug 17 16:17:43 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 17 Aug 2013 07:17:43 -0700 (PDT)
Subject: [FFmpeg-user] Problems with Select Filter
In-Reply-To: <CAPYw7P4vGU8i9P1vSZC4nmtw8yfojNpB52+JDyeQ5_pHc_KKsw@mail.gmail.com>
References: <1376747518.34172.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<CAPYw7P4vGU8i9P1vSZC4nmtw8yfojNpB52+JDyeQ5_pHc_KKsw@mail.gmail.com>
Message-ID: <1376749063.79033.YahooMailNeo@web164701.mail.gq1.yahoo.com>



>> Does that help identify the problem?
>
>Do not use avi.

Okay.? What should I use?

From jpboard2 at yahoo.com  Sat Aug 17 16:32:35 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 17 Aug 2013 07:32:35 -0700 (PDT)
Subject: [FFmpeg-user] Problems with Select Filter: File Formats
In-Reply-To: <CAPYw7P4vGU8i9P1vSZC4nmtw8yfojNpB52+JDyeQ5_pHc_KKsw@mail.gmail.com>
References: <1376747518.34172.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<CAPYw7P4vGU8i9P1vSZC4nmtw8yfojNpB52+JDyeQ5_pHc_KKsw@mail.gmail.com>
Message-ID: <1376749955.62363.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> I'm trying to figure out why this occurs.? When I try to play the edited

>> file
>> with mplayer, it displays these complaints:
>>? ?  AVI: ODML: Building ODML index (2 superindexchunks).
>>? ?  AVI: ODML: Broken (incomplete?) file detected. Will use traditional
>> index.
>>? ?  AVI: Generated index table for 10983 chunks!
>> Does that help identify the problem?
>
>Do not use avi.

Here are the steps I typically do, each of which creates a new intermediate
file.? 

	1. Capture from Component video or HDMI.
	2. Inverse telecine or de-inetrlace, and simultaenously compress to ffvhuff.
	3. Edit out commercials and other crap using ffmpeg and the 'Select' video filter.
	4. Encode with an mpeg 4 codec (like libx264).
Currently all files are AVI, but I don't really care what the intermediate
file formats are.? What would be a better choice for the intermediate
file formats?? I want the edit step to be frame-exact, and I also want the
format that runs the fastest as well.? Are some formats faster than others?

Also, what's a good choice for the final output format?? This should
be fairly portable across machines, and future-proof.? However, I do
most of my work on Linux machine.

From simon.mares at gmail.com  Sun Aug 18 04:04:32 2013
From: simon.mares at gmail.com (simon mares)
Date: Sun, 18 Aug 2013 04:04:32 +0200
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
Message-ID: <52102BB0.3000506@gmail.com>

Hello, is it possible to generate a pre-defined moov atom for fixed 
lenght of fe. 8 hours with use of 
http://ffmpeg.org/ffmpeg-all.html#MOV_002fMP4_002fISMV arguments? Video 
would never be seeked beyond the real duration, as I would not allow.

Reason for doing this is that I'm experimenting with something and I 
would like to playback the file that I'm still recording, so it's kind 
of a live-streaming, but it's only used locally. As I'm only transcoding 
from .ts to .mp4 file no CPU is used, this is also my goal.

Codec is h264, 720p, 25pfs. Video is generated with blackmagic pro-recorder.

Any help or tip is highly appreciable!

Simon

From basteon at gmail.com  Sun Aug 18 10:14:34 2013
From: basteon at gmail.com (basteon)
Date: Sun, 18 Aug 2013 08:14:34 +0000
Subject: [FFmpeg-user] "Failed to update header with correct duration."
In-Reply-To: <CA+JzsNQ1Q=HXmyuOsiG5Nb0-Sz1iM47CJJkqtA-DPRTTzffQAQ@mail.gmail.com>
References: <CA+JzsNQ1Q=HXmyuOsiG5Nb0-Sz1iM47CJJkqtA-DPRTTzffQAQ@mail.gmail.com>
Message-ID: <CAEJHKkVsgq74rqyMCV2iC+QyLzZp1FRownKKqsy-M+=ucYcnCQ@mail.gmail.com>

Hi,
If you still not solve this. That when bmdcapture loosing one frame rtmp
could be confused about timebase of that frame. If you using -copytb.

That will happening for video, audio or both?
for flv you can try nellymoser.


On 14 August 2013 05:12, Leonard Bogard <leonard at kcfchurch.org> wrote:

> Whether I stop ffmpeg manually or it stops on its own after some randomly
> short period of time fail it displays two error messages:
> [flv @ 0x2279ba0] Failed to update header with correct duration
> [flv @ 0x2279ba0] Failed to update header with correct filesize.
>
> I'm trying to stream from ffmpeg to a wowza server, here's my start script
> and output:
>
> OUTSTREAM="-f flv rtmp://192.168.7.2:1935/live/kcfstream"
>
> OUTAUDIO="-acodec libfdk_aac -af volume=8.0 -ab 64000 -ar 48000 -ac 2"
> #OUTAUDIO="-acodec libmp3lame -af volume=8.0 -ab 96000 -ar 44100 -ac 1"
>
> bmdcapture -m 14 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -i - -copyts \
>   -r 30 -s 640x360 -aspect 16:9 \
>   -threads 2 \
>   $OUTAUDIO \
>   -vcodec libx264 -vb 500k -vprofile baseline -level 3.1 -pix_fmt yuv420p \
>   -strict -2 \
>   $OUTSTREAM
>
> ~$ ./exp2.sh
> ffmpeg version git-2013-08-13-8a7295b Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Aug 13 2013 21:30:35 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
>   configuration: --prefix=/home/suser/ffmpeg_build
> --extra-cflags=-I/home/suser/ffmpeg_build/include
> --extra-ldflags=-L/home/suser/ffmpeg_build/lib --bindir=/home/suser/bin
> --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac
> --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis
> --enable-libvpx --enable-libx264 --enable-libfaac --enable-version3
> --enable-nonfree --enable-encoder=libfaac
>   libavutil      52. 42.100 / 52. 42.100
>   libavcodec     55. 27.100 / 55. 27.100
>   libavformat    55. 13.102 / 55. 13.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 82.100 /  3. 82.100
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.103 /  0. 17.103
>   libpostproc    52.  3.100 / 52.  3.100
> Guessed Channel Layout for  Input Stream #0.1 : stereo
> Input #0, nut, from 'pipe:':
>   Metadata:
>     encoder         : Lavf54.20.3
>   Duration: N/A, start: 0.000000, bitrate: 1536 kb/s
>     Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 1280x720,
> 59.94 tbr, 59.94 tbn, 59.94 tbc
>     Stream #0:1: Audio: pcm_s16le (PSD[16] / 0x10445350), 48000 Hz, stereo,
> s16, 1536 kb/s
> [libx264 @ 0x24b9120] using SAR=1/1
> [libx264 @ 0x24b9120] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> AVX
> [libx264 @ 0x24b9120] profile Constrained Baseline, level 3.1
> [libx264 @ 0x24b9120] 264 - core 135 r2 f0c1c53 - H.264/MPEG-4 AVC codec -
> Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0
> ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00
> mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11
> fast_pskip=1 chroma_qp_offset=-2 threads=2 lookahead_threads=1
> sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
> constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25
> scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=500
> ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> Output #0, flv, to 'rtmp://192.168.7.2:1935/live/kcfstream':
>   Metadata:
>     encoder         : Lavf55.13.102
>     Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p,
> 640x360 [SAR 1:1 DAR 16:9], q=-1--1, 500 kb/s, 1k tbn, 30 tbc
>     Stream #0:1: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 48000
> Hz, stereo, s16, 64 kb/s
> Stream mapping:
>   Stream #0:0 -> #0:0 (rawvideo -> libx264)
>   Stream #0:1 -> #0:1 (pcm_s16le -> libfdk_aac)
> ^C[flv @ 0x24b8ba0] Failed to update header with correct duration.e=
> 435.2kbits/s dup=0 drop=209
> [flv @ 0x24b8ba0] Failed to update header with correct filesize.
> frame=  225 fps= 31 q=-1.0 Lsize=     528kB time=00:00:07.50 bitrate=
> 576.7kbits/s dup=0 drop=221
> video:458kB audio:59kB subtitle:0 global headers:0kB muxing overhead
> 2.044314%
> [libx264 @ 0x24b9120] frame I:11    Avg QP:23.52  size:  3701
> [libx264 @ 0x24b9120] frame P:214   Avg QP:29.68  size:  2000
> [libx264 @ 0x24b9120] mb I  I16..4: 88.1%  0.0% 11.9%
> [libx264 @ 0x24b9120] mb P  I16..4:  4.4%  0.0%  1.2%  P16..4: 28.9%  7.4%
> 1.8%  0.0%  0.0%    skip:56.2%
> [libx264 @ 0x24b9120] final ratefactor: 27.48
> [libx264 @ 0x24b9120] coded y,uvDC,uvAC intra: 15.2% 47.5% 7.5% inter: 9.3%
> 13.1% 1.2%
> [libx264 @ 0x24b9120] i16 v,h,dc,p: 41% 23%  6% 31%
> [libx264 @ 0x24b9120] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 14% 45% 18%  3%  4%
> 3%  6%  2%  4%
> [libx264 @ 0x24b9120] i8c dc,h,v,p: 62% 20% 10%  8%
> [libx264 @ 0x24b9120] ref P L0: 85.8%  8.7%  5.5%
> [libx264 @ 0x24b9120] kb/s:499.96
> Received signal 2: terminating.
>
> What am I doing wrong?  Thanks in advance for any help.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From basteon at gmail.com  Sun Aug 18 10:28:46 2013
From: basteon at gmail.com (basteon)
Date: Sun, 18 Aug 2013 08:28:46 +0000
Subject: [FFmpeg-user] convert/replay H263 raw file Form rtp payload
In-Reply-To: <00bb01ce979d$57236480$056a2d80$@gmx.net>
References: <00bb01ce979d$57236480$056a2d80$@gmx.net>
Message-ID: <CAEJHKkUqES66p9uSG3gdD8owZ-s-QrSH0Vmwas6tZ7xVviP+nw@mail.gmail.com>

hi,
No, ffmpeg unable to read pcap files. But if you want you can make or found
somewhere pcap player for playing captured scream as rtp stream for ffmpeg.


On 12 August 2013 20:48, Manfred Ehgartner <manfred.ehgartner at gmx.net>wrote:

> I am currently working in a project, where I have pcaps containing h263 and
> h264 rtp streams. I don't have any signaling, SDP info, just the rtp
> streams
> and I need to create something playable out oft it. For H264 I succeeded
> using videosnarf. The output raw file of videosnarf could be converted to a
> MPEG or avi file using ffmpeg. Unfortunately, videosnarf doesn't support
> h263, so I tried manually just to dump the rtp payload into some file, but
> this couldn't be read by ffmpeg correctly. Also the output oft rtpbreak
> (similar tool like videosnarf) couldn't be processed by ffmpeg. So I
> assume,
> videosnarf is not simply dumping the rtp payload to a file. How must the
> h263 Payload be modified that the file can be read using ffmpeg? I am happy
> about every hint
>
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From basteon at gmail.com  Sun Aug 18 10:45:20 2013
From: basteon at gmail.com (basteon)
Date: Sun, 18 Aug 2013 08:45:20 +0000
Subject: [FFmpeg-user] Activating RTCP feedback with FFMpeg for VP8 rtp
In-Reply-To: <CAB_tPj1+q3FEHfCnAqPissfgZ1MuH07aKHiOihbEr9DosEwVEg@mail.gmail.com>
References: <CAB_tPj1+q3FEHfCnAqPissfgZ1MuH07aKHiOihbEr9DosEwVEg@mail.gmail.com>
Message-ID: <CAEJHKkWCF2mAVjNv95GS0i68Dp-ssHeq+=haW_wngGGZ66JDMA@mail.gmail.com>

hi,
Looks like patch wasn't accepted or checked.
i.e. that patch not available in
http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=rtcp_feedback

and second link you gave about rtsp feedback, not rtcp, that's different.


On 9 August 2013 16:02, Pierrick Grasland <pierrick.grasland at gmail.com>wrote:

> Hello,
>
> I'm trying to receive a vp8 / G711? rtp stream with FFMpeg.
>
> My SDP is the following :
>
> v=0
> o=- 1376063087593 1 IN IP4 127.0.0.1
> s=-
> t=0 0
> m=audio 50008 RTP/AVP 0
> c=IN IP4 192.168.2.196
> a=rtcp:50009 IN IP4 192.168.2.196
> a=rtpmap:0 PCMU/8000
> a=sendrecv
> m=video 50010 RTP/AVP 120
> c=IN IP4 192.168.2.196
> a=rtcp:50011 IN IP4 192.168.2.196
> a=rtpmap:120 VP8/90000
> a=sendrecv
> a=rtcp-fb:* nack
> a=rtcp-fb:* ccm fir
>
>
> FFMpeg will correctly start, using the following command : ffmpeg
> -reorder_queue_size 64 -analyzeduration 15000000 -probesize 50000000 -i
> 3.sdp -f webm {1}.webm -ss 00:00:01.000 -f image2 -vframes 1 {1}.jpg ({1}
> is replaced by the correct output filename).
>
> ffmpeg version N-51976-g8f3c440 Copyright (c) 2000-2013 the FFmpeg
> developers
> built on Apr 25 2013 19:24:31 with gcc 4.4.5 (Debian 4.4.5-8)
> configuration: --enable-gpl --enable-libtheora --enable-libvorbis
> --enable-libvpx --enable-nonfree --enable-version3 --enable-libopus
> --enable-libx264
>   libavutil      52. 26.100 / 52. 26.100
>   libavcodec     55.  2.100 / 55.  2.100
>   libavformat    55.  2.100 / 55.  2.100
>   libavdevice    55.  0.100 / 55.  0.100
>   libavfilter     3. 53.101 /  3. 53.101
>   libswscale      2.  2.100 /  2.  2.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  3.100 / 52.  3.100
>
>
> [sdp @ 0x218b800] Estimating duration from bitrate, this may be inaccurate
> Guessed Channel Layout for  Input Stream #0.0 : mono
> Input #0, sdp, from
> '/home/pierrick/wsrtc-distribution-0.8.8-ZeWall-r8899/sdp/3.sdp':
>   Metadata:
>     title           : -
>   Duration: 00:00:00.04, start: 0.000000, bitrate: 63 kb/s
>     Stream #0:0: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
>     Stream #0:1: Video: vp8, yuv420p, 480x640, 90k tbr, 90k tbn, 90k tbc
> [auto-inserted scaler 0 @ 0x21f0d00] The <w>:<h>:flags=<flags> option
> syntax is deprecated. Use either <w>:<h>:<flags> or
> w=<w>:h=<h>:flags=<flags>.
> [image2 @ 0x218f880] Frame rate very high for a muxer not efficiently
> supporting it.
> Please consider specifying a lower framerate, a different muxer or -vsync 2
> [libvpx @ 0x21efd00] v1.0.0
> Output #0, webm, to 'call_fx_1_3_2013-08-09_17-44-47.webm':
>   Metadata:
>     title           : -
>     encoder         : Lavf55.2.100
>     Stream #0:0: Video: vp8, yuv420p, 480x640, q=-1--1, 200 kb/s, 1k tbn,
> 90k tbc
>     Stream #0:1: Audio: vorbis, 8000 Hz, mono, fltp
> Output #1, image2, to 'call_fx_1_3_2013-08-09_17-44-47.jpg':
>   Metadata:
>     title           : -
>     encoder         : Lavf55.2.100
>     Stream #1:0: Video: mjpeg, yuvj420p, 480x640, q=2-31, 200 kb/s, 90k
> tbn, 90k tbc
> Stream mapping:
>   Stream #0:1 -> #0:0 (vp8 -> libvpx)
>   Stream #0:0 -> #0:1 (pcm_mulaw -> libvorbis)
>   Stream #0:1 -> #1:0 (vp8 -> mjpeg)
> Press [q] to stop, [?] for help
>
> For some minutes (between 30 s and 3 minutes), this works fine, generating
> this kind of output :
>
> frame= 3868 fps= 27 q=0.0 q=0.0 size=    4044kB time=00:02:21.86 bitrate=
> 233.5kbits/s dup=0 drop=3837
> frame= 3885 fps= 27 q=0.0 q=0.0 size=    4057kB time=00:02:22.44 bitrate=
> 233.3kbits/s dup=0 drop=3854
> frame= 3900 fps= 27 q=0.0 q=0.0 size=    4067kB time=00:02:22.98 bitrate=
> 233.0kbits/s dup=0 drop=3869
> frame= 3906 fps= 27 q=0.0 q=0.0 size=    4071kB time=00:02:23.53 bitrate=
> 232.4kbits/s dup=0 drop=3875
>
>
> But, after a delay, I'm encountering a packet loss (due to using WiFi) and
> FFMpeg generate one of the following message :
> - Received no start marker; dropping frame
> - Missed a picture, sequence broken.
>
> The output will now be this :
>
> frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:24.03 bitrate=
> 231.6kbits/s dup=0 drop=3876
> frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:24.54 bitrate=
> 230.8kbits/s dup=0 drop=3876
> frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:25.05 bitrate=
> 230.0kbits/s dup=0 drop=3876
> frame= 3907 fps= 27 q=0.0 q=0.0 size=    4072kB time=00:02:25.60 bitrate=
> 229.1kbits/s dup=0 drop=3876
>
> And my webm file show a freeze at this time, and doesn't appear to restart
> correctly after (freeze can be up to several minutes).
>
>
> Analyzing my network with wireshark will show that my loss are limited to
> one or two packets only, but the wrong one.
>
> I found with google that I can use RTCP feedback for VP8, which seems to be
> possible with FFMpeg :
>
> http://ffmpeg.org/pipermail/ffmpeg-cvslog/2013-January/059123.html
>
> http://ffmpeg.org/doxygen/trunk/structRTSPStream.html#a9f61d90f147f37ed608f1d151fdcdde8
>
> How can I activate RTCP feedback ? Does this support NACK / PLI / FIR
> request ?
>
> Regards,
> --
> Pierrick Grasland
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From soho123.2012 at gmail.com  Sun Aug 18 15:06:43 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Sun, 18 Aug 2013 21:06:43 +0800
Subject: [FFmpeg-user] HELP! ffserver polling get no any fd ready for read,
	even ffmpeg post connection
Message-ID: <CAFd2cZUUB8F0==SP0EmpH1+4__0xeRpdVVsa25=778mGiTmOTQ@mail.gmail.com>

HI all,

the problem I got for a long time.

when ffserver usd poll to get the fd that ready for read/write, then each
connection can get service.
but sometimes,
ffserver get a condition:
c->poll_entry->revents does not include any events about POLLIN, even the
post connection from ffmpeg.
ffmpeg keep posting video data to ffserver, then the feed file can get new
video data.

I check the error code from ffmpeg.
It says , send timeout that the receive side does not read the data on time.

ffserver says there is no fd ready for read.
ffmpeg says that the receive side does not read data on time.

Does any one have idea?



.

From mokacao at gmail.com  Sun Aug 18 11:07:24 2013
From: mokacao at gmail.com (mokacao)
Date: Sun, 18 Aug 2013 02:07:24 -0700 (PDT)
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
Message-ID: <1376816843722-4660765.post@n4.nabble.com>

I'm doing a convert picture to MP4 video, the original image is PIX_FMT_BGRA
format, now converted from RGB to YUV420P, I found many black spot around
the picture, the follow is part of the core code, please help me to how to
clear these black dots, or can tell me what information is missing, thank
you.

libffmpeg::SwsContext* swsContext = libffmpeg::sws_getContext( srcW, srcH,
libffmpeg::PIX_FMT_BGRA,codecContext->width, codecContext->height,
libffmpeg::PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);

libffmpeg::sws_scale( swsContext , srcData, srcLinesize, 0, frame->Height,
data->VideoFrame->data, data->VideoFrame->linesize );

original png file
<http://ffmpeg-users.933282.n4.nabble.com/file/n4660765/P0085.png> 

yuv420p image file(scale and clip)
<http://ffmpeg-users.933282.n4.nabble.com/file/n4660765/yuv420p.png> 





--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-can-I-convert-rbg-to-yuv420p-loseless-tp4660765.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From adf.lists at gmail.com  Mon Aug 19 00:21:47 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sun, 18 Aug 2013 23:21:47 +0100
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
In-Reply-To: <1376816843722-4660765.post@n4.nabble.com>
References: <1376816843722-4660765.post@n4.nabble.com>
Message-ID: <521148FB.3020805@gmail.com>

mokacao wrote:
> I'm doing a convert picture to MP4 video, the original image is PIX_FMT_BGRA
> format, now converted from RGB to YUV420P, I found many black spot around
> the picture, the follow is part of the core code, please help me to how to
> clear these black dots, or can tell me what information is missing, thank
> you.
>
> libffmpeg::SwsContext* swsContext = libffmpeg::sws_getContext( srcW, srcH,
> libffmpeg::PIX_FMT_BGRA,codecContext->width, codecContext->height,
> libffmpeg::PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);
>
> libffmpeg::sws_scale( swsContext , srcData, srcLinesize, 0, frame->Height,
> data->VideoFrame->data, data->VideoFrame->linesize );
>
> original png file
> <http://ffmpeg-users.933282.n4.nabble.com/file/n4660765/P0085.png>
>
> yuv420p image file(scale and clip)
> <http://ffmpeg-users.933282.n4.nabble.com/file/n4660765/yuv420p.png>

Well I don't know if you can do better with ffmpeg or not as I haven't 
tried and others may be able to suggest tweaks.

Referring  to the title of this post RGB -> 420 can't be lossless.

Chroma subsampling roughly works on "normal" video images, but if you 
push it with saturated graphics it can artifact.
Broadcasters will take this sort of thing into account when designing 
text overlays etc.

Below is just as some sort of explanation - I don't know if it's why in 
your particular case/commands used.

http://sas-origin.onstreammedia.com/origin/smpte0109/New%20Folder%201/Chan.pdf


From ddhung at vasc.com.vn  Mon Aug 19 11:34:52 2013
From: ddhung at vasc.com.vn (=?utf-8?B?xJDhurduZyDEkOG7qWMgSMawbmc=?=)
Date: Mon, 19 Aug 2013 16:34:52 +0700
Subject: [FFmpeg-user] Error SEI truncated in H264
Message-ID: <001f01ce9cbf$5ee44d70$1cace850$@vasc.com.vn>

Hi All.

How can I fix this error ?SEI truncated? in ffmpeg? The below is all text code which I am using to trascode.

Thank a lot!

 

[root at localhost Running_files]# ffmpeg -isync -i udp://233.19.204.85:5501 -filter:v yadif=0:-1:1,fps=25 -vcodec libx264 -vprofile main -level 30 -preset slower -pass 1 -b:v 1000k -maxrate 1000k -bufsize 2000k -s:v 720x576 -r:v 25 -g 64 -crf 24 -threads 0 -ac 2 -ar 48000 -b:a 128k -filter:a volume=2 -f mpegts -muxrate 1400k udp://233.19.200.74:5501?pkt_size=1316ffmpeg version N-55519-g7ec7d62 Copyright (c) 2000-2013 the FFmpeg developers

  built on Aug 14 2013 16:00:20 with gcc 4.1.2 (GCC) 20080704 (Red Hat 4.1.2-54)

  configuration: --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvpx --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --disable-ffplay --enable-shared --enable-gpl --enable-postproc --enable-nonfree --enable-avfilter --enable-pthreads --extra-cflags=-fPIC

  libavutil      52. 42.100 / 52. 42.100

  libavcodec     55. 27.100 / 55. 27.100

  libavformat    55. 13.102 / 55. 13.102

  libavdevice    55.  3.100 / 55.  3.100

  libavfilter     3. 82.100 /  3. 82.100

  libswscale      2.  4.100 /  2.  4.100

  libswresample   0. 17.103 /  0. 17.103

  libpostproc    52.  3.100 / 52.  3.100

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] non-existing PPS referenced

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] non-existing PPS 0 referenced

[h264 @ 0x3694720] decode_slice_header error

[h264 @ 0x3694720] no frame!

[h264 @ 0x3694720] SEI truncated

[h264 @ 0x3694720] mmco: unref short failure

    Last message repeated 1 times

[h264 @ 0x3694720] SEI truncated

    Last message repeated 1 times

[h264 @ 0x3694720] mmco: unref short failure

    Last message repeated 1 times

[h264 @ 0x3694720] SEI truncated

    Last message repeated 5 times

[h264 @ 0x3694720] mmco: unref short failure

    Last message repeated 1 times

[h264 @ 0x3694720] SEI truncated

   Last message repeated 14 times

Input #0, mpegts, from 'udp://233.19.204.85:5501':

  Duration: N/A, start: 5284.301678, bitrate: 224 kb/s

  Program 1 

    Stream #0:0[0x21]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 12:11 DAR 15:11], 25 fps, 50 tbr, 90k tbn, 50 tbc

    Stream #0:1[0x22](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 224 kb/s

[libx264 @ 0x3890060] using SAR=12/11

[libx264 @ 0x3890060] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2

[libx264 @ 0x3890060] profile Main, level 3.0

Output #0, mpegts, to 'udp://233.19.200.74:5501?pkt_size=1316':

  Metadata:

    encoder         : Lavf55.13.102

    Stream #0:0: Video: h264 (libx264), yuv420p, 720x576 [SAR 12:11 DAR 15:11], q=-1--1, pass 1, 1000 kb/s, 90k tbn, 25 tbc

    Stream #0:1(eng): Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s

Stream mapping:

  Stream #0:0 -> #0:0 (h264 -> libx264)

  Stream #0:1 -> #0:1 (mp2 -> mp2)

Press [q] to stop, [?] for help

[h264 @ 0x62509c0] SEI truncated

[h264 @ 0x62509c0] Missing reference picture, default is 0

[h264 @ 0x62509c0] decode_slice_header error

[h264 @ 0x6295640] SEI truncated

[h264 @ 0x6295640] Missing reference picture, default is 65492

    Last message repeated 1 times

[h264 @ 0x6303fe0] SEI truncated

[h264 @ 0x63729a0] SEI truncated

[h264 @ 0x63e1360] SEI truncated

[h264 @ 0x644fd20] SEI truncated

[h264 @ 0x64be6e0] SEI truncated

[h264 @ 0x64be6e0] reference picture missing during reorder

    Last message repeated 1 times

[h264 @ 0x64be6e0] Missing reference picture, default is 65492

    Last message repeated 1 times

[h264 @ 0x652d0a0] mmco: unref short failure

    Last message repeated 1 times

[h264 @ 0x652d0a0] SEI truncated

[h264 @ 0x652d0a0] reference picture missing during reorder

[h264 @ 0x652d0a0] Missing reference picture, default is 65493

[h264 @ 0x659ba60] SEI truncated

[h264 @ 0x660a420] SEI truncated

[h264 @ 0x6678de0] SEI truncated

[h264 @ 0x66e77a0] SEI truncated

[h264 @ 0x6756160] SEI truncated

[h264 @ 0x67c4b20] SEI truncated

[h264 @ 0x68334e0] SEI truncated

[h264 @ 0x68a1ea0] mmco: unref short failure

    Last message repeated 1 times

[h264 @ 0x68a1ea0] SEI truncated

[h264 @ 0x62509c0] SEI truncated

[h264 @ 0x6295640] SEI truncated

[h264 @ 0x6303fe0] SEI truncated

[h264 @ 0x63729a0] SEI truncated

[h264 @ 0x63e1360] SEI truncated

[h264 @ 0x644fd20] SEI truncated

[h264 @ 0x64be6e0] SEI truncated

[h264 @ 0x652d0a0] SEI truncated

[h264 @ 0x659ba60] SEI truncated

[h264 @ 0x660a420] SEI truncated

[h264 @ 0x6678de0] SEI truncated

[h264 @ 0x66e77a0] SEI truncated

[h264 @ 0x6756160] SEI truncated

[h264 @ 0x67c4b20] SEI truncated

[h264 @ 0x68334e0] SEI truncated

[h264 @ 0x68a1ea0] SEI truncated

[h264 @ 0x62509c0] SEI truncated

[h264 @ 0x6295640] SEI truncated

[h264 @ 0x6303fe0] SEI truncated

[h264 @ 0x63729a0] SEI truncated

[h264 @ 0x63e1360] SEI truncated

[h264 @ 0x644fd20] SEI truncated

[h264 @ 0x64be6e0] SEI truncated

[h264 @ 0x652d0a0] SEI truncated

[h264 @ 0x659ba60] SEI truncated

[h264 @ 0x660a420] SEI truncated

[h264 @ 0x6678de0] SEI truncated

[h264 @ 0x66e77a0] SEI truncated

[h264 @ 0x6756160] SEI truncated

[h264 @ 0x67c4b20] SEI truncated

[h264 @ 0x68334e0] SEI truncated

[h264 @ 0x68a1ea0] SEI truncated

[h264 @ 0x62509c0] SEI truncated

[h264 @ 0x6295640] SEI truncated

[h264 @ 0x6303fe0] SEI truncated

[h264 @ 0x63729a0] SEI truncated

[h264 @ 0x63e1360] SEI truncated

[h264 @ 0x644fd20] SEI truncated

[h264 @ 0x64be6e0] SEI truncated

[h264 @ 0x652d0a0] SEI truncated

[h264 @ 0x659ba60] SEI truncated

[h264 @ 0x660a420] SEI truncated

[h264 @ 0x6678de0] SEI truncated

[h264 @ 0x66e77a0] SEI truncated

[h264 @ 0x6756160] SEI truncated

[h264 @ 0x67c4b20] SEI truncated

[h264 @ 0x68334e0] SEI truncated

[h264 @ 0x68a1ea0] SEI truncated

[h264 @ 0x62509c0] SEI truncated

[h264 @ 0x6295640] SEI truncated

[h264 @ 0x6303fe0] SEI truncated

[h264 @ 0x63729a0] SEI truncated

[h264 @ 0x63e1360] SEI truncated

[h264 @ 0x644fd20] SEI truncated

[h264 @ 0x64be6e0] SEI truncated

[h264 @ 0x652d0a0] SEI truncated

[h264 @ 0x659ba60] SEI truncated

[h264 @ 0x660a420] SEI truncated

[h264 @ 0x6678de0] SEI truncated

[h264 @ 0x66e77a0] SEI truncated

[h264 @ 0x6756160] SEI truncated

[h264 @ 0x67c4b20] SEI truncated      0kB time=00:00:01.42 bitrate=   0.0kbits/s dup=49 drop=0    

[h264 @ 0x68334e0] SEI truncated

[h264 @ 0x68a1ea0] SEI truncated

[h264 @ 0x62509c0] SEI truncated

[h264 @ 0x6295640] SEI truncated

[h264 @ 0x6303fe0] SEI truncated

[h264 @ 0x63729a0] SEI truncated

[h264 @ 0x63e1360] SEI truncated

[h264 @ 0x644fd20] SEI truncated

[h264 @ 0x64be6e0] SEI truncated

[h264 @ 0x652d0a0] SEI truncated

[h264 @ 0x659ba60] SEI truncated

[h264 @ 0x660a420] SEI truncated

[h264 @ 0x6678de0] SEI truncated

[h264 @ 0x66e77a0] SEI truncated

[h264 @ 0x6756160] SEI truncated

[h264 @ 0x67c4b20] SEI truncated

[h264 @ 0x68334e0] SEI truncated

[h264 @ 0x68a1ea0] SEI truncated

[h264 @ 0x62509c0] SEI truncated

[h264 @ 0x6295640] SEI truncated

[h264 @ 0x6303fe0] SEI truncated

[h264 @ 0x63729a0] SEI truncated

[h264 @ 0x63e1360] SEI truncated

frame=   66 fps=0.0 q=-1.0 Lsize=     467kB time=00:00:02.56 bitrate=1494.6kbits/s dup=49 drop=0    

video:91kB audio:30kB subtitle:0 global headers:0kB muxing overhead 286.776218%

[libx264 @ 0x3890060] frame I:2     Avg QP:19.65  size: 38948

[libx264 @ 0x3890060] frame P:18    Avg QP:19.85  size:   632

[libx264 @ 0x3890060] frame B:46    Avg QP:26.93  size:    80

[libx264 @ 0x3890060] consecutive B-frames:  6.1%  3.0%  0.0% 90.9%

[libx264 @ 0x3890060] mb I  I16..4: 29.8%  0.0% 70.2%

[libx264 @ 0x3890060] mb P  I16..4:  0.2%  0.0%  0.0%  P16..4:  6.6%  0.0%  0.0%  0.0%  0.0%    skip:93.2%

[libx264 @ 0x3890060] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  0.3%  0.0%  0.0%  direct: 0.6%  skip:99.1%  L0:26.4% L1:52.3% BI:21.3%

[libx264 @ 0x3890060] direct mvs  spatial:43.5% temporal:56.5%

[libx264 @ 0x3890060] coded y,uvDC,uvAC intra: 88.5% 90.3% 60.2% inter: 0.4% 1.6% 0.0%

[libx264 @ 0x3890060] i16 v,h,dc,p: 49% 20% 19% 11%

[libx264 @ 0x3890060] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 38% 21% 12%  5%  5%  6%  3%  6%  4%

[libx264 @ 0x3890060] i8c dc,h,v,p: 35% 27% 29%  8%

[libx264 @ 0x3890060] Weighted P-Frames: Y:0.0% UV:0.0%

[libx264 @ 0x3890060] kb/s:281.62

 

 

 

 

Best Regards

??ng ??c H?ng

| Broadcast Technical Department | IPTV Center | VASC Software and Media Company |

| 97 Nguy?n Ch? Thanh | ??ng ?a District | Hanoi | Vietnam|

| Tel (+84) 37722728 ext 681/684 | Mobile (+84) 915 070 921| 

|  <mailto:ddhung at vasc.com.vn> ddhung at vasc.com.vn |  <http://www.mytv.com.vn/> www.mytv.com.vn |

 


From stefasab at gmail.com  Mon Aug 19 12:22:40 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Mon, 19 Aug 2013 12:22:40 +0200
Subject: [FFmpeg-user] Detecting Scene Changes In Source Video?
In-Reply-To: <1376747798.41445.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1376747798.41445.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <20130819102240.GB6843@barisone>

On date Saturday 2013-08-17 06:56:38 -0700, James Board wrote:
> Is there a way to use ffmpeg to detect scene changes in a video file?? I know
> some of the codecs must do this already, but I don't know if that output is
> readily available?? Maybe in the info file produced when 2-pass encoding?

Check the select filter and the scene paramenter.
-- 
FFmpeg = Fast Friendly Muttering Puristic Eretic Goblin

From manfred.ehgartner at gmx.net  Mon Aug 19 14:48:15 2013
From: manfred.ehgartner at gmx.net (Manfred Ehgartner)
Date: Mon, 19 Aug 2013 14:48:15 +0200
Subject: [FFmpeg-user] convert/replay H263 raw file Form rtp payload
In-Reply-To: <CAEJHKkUqES66p9uSG3gdD8owZ-s-QrSH0Vmwas6tZ7xVviP+nw@mail.gmail.com>
References: <00bb01ce979d$57236480$056a2d80$@gmx.net>
	<CAEJHKkUqES66p9uSG3gdD8owZ-s-QrSH0Vmwas6tZ7xVviP+nw@mail.gmail.com>
Message-ID: <5212140F.4040700@gmx.net>

Hello,
thanks for the clarification.

Best regards
Manfred

On 18.08.2013 10:28, basteon wrote:
> hi,
> No, ffmpeg unable to read pcap files. But if you want you can make or found
> somewhere pcap player for playing captured scream as rtp stream for ffmpeg.
>
>
> On 12 August 2013 20:48, Manfred Ehgartner <manfred.ehgartner at gmx.net>wrote:
>
>> I am currently working in a project, where I have pcaps containing h263 and
>> h264 rtp streams. I don't have any signaling, SDP info, just the rtp
>> streams
>> and I need to create something playable out oft it. For H264 I succeeded
>> using videosnarf. The output raw file of videosnarf could be converted to a
>> MPEG or avi file using ffmpeg. Unfortunately, videosnarf doesn't support
>> h263, so I tried manually just to dump the rtp payload into some file, but
>> this couldn't be read by ffmpeg correctly. Also the output oft rtpbreak
>> (similar tool like videosnarf) couldn't be processed by ffmpeg. So I
>> assume,
>> videosnarf is not simply dumping the rtp payload to a file. How must the
>> h263 Payload be modified that the file can be read using ffmpeg? I am happy
>> about every hint
>>
>>
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From jpboard2 at yahoo.com  Mon Aug 19 16:36:29 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 19 Aug 2013 07:36:29 -0700 (PDT)
Subject: [FFmpeg-user] Detecting Scene Changes In Source Video?
In-Reply-To: <20130819102240.GB6843@barisone>
References: <1376747798.41445.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<20130819102240.GB6843@barisone>
Message-ID: <1376922989.71196.YahooMailNeo@web164706.mail.gq1.yahoo.com>

>> Is there a way to use ffmpeg to detect scene changes in a video file?? I know

>> some of the codecs must do this already, but I don't know if that output is
>> readily available?? Maybe in the info file produced when 2-pass encoding?
>
>Check the select filter and the scene paramenter.

That might work.? The example they have will produce a video with
only the first frame of every scene.? Can I somehow instead output the
frame numbers of the first frame in every scene?

Maybe someone that knows the filter syntax, or other syntax better can
help?

From tevans.uk at googlemail.com  Mon Aug 19 16:39:14 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Mon, 19 Aug 2013 15:39:14 +0100
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
In-Reply-To: <52102BB0.3000506@gmail.com>
References: <52102BB0.3000506@gmail.com>
Message-ID: <CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>

On Sun, Aug 18, 2013 at 3:04 AM, simon mares <simon.mares at gmail.com> wrote:
> Hello, is it possible to generate a pre-defined moov atom for fixed lenght
> of fe. 8 hours with use of
> http://ffmpeg.org/ffmpeg-all.html#MOV_002fMP4_002fISMV arguments? Video
> would never be seeked beyond the real duration, as I would not allow.
>
> Reason for doing this is that I'm experimenting with something and I would
> like to playback the file that I'm still recording, so it's kind of a
> live-streaming, but it's only used locally. As I'm only transcoding from .ts
> to .mp4 file no CPU is used, this is also my goal.
>
> Codec is h264, 720p, 25pfs. Video is generated with blackmagic pro-recorder.
>

Just use "-movflags faststart".

Cheers

Tom

From gmstima at gmail.com  Mon Aug 19 17:16:24 2013
From: gmstima at gmail.com (=?KOI8-R?B?8s/Nwc4g88HX3sXOy88=?=)
Date: Mon, 19 Aug 2013 18:16:24 +0300
Subject: [FFmpeg-user] msvc build ffmpeg with x264 linked error: unresolved
	_x264_build_depth
Message-ID: <CAGpdKU+LKL7mggcWJQOeEnAxGNCv0-aZxSvwUbLLn18E6DXXiw@mail.gmail.com>

Hi all,

I try to compile ffmpeg with x264 support on windows. My steps:

1. Download and compile yasm. From msys:
    ./configure --prefix=../build && make && make install
2. Download and compile x264. From msys:
    ./configure --prefix=../build --enable-shared --disable-cli
--enable-win32thread --Wl,--output-def=libx264.def
    make && make install
3. From msvc command line lib /machine:i386 /def:libx264.def
/out:libx264.lib
4. Download c99-to-c99 wrapper(was added to PATH) and inttypes (was puted
to build/include).
5. Run msys form msvc command line. Switch linker (mv /usr/bin/link.exe
/usr/bin/mingw_link.exe).
6. Run:
   ./configure --prefix=../build --toolchain=msvc --extra-cflags="-I
../build/include" --extra-ldflags="/LIBPATH:../build/lib" --enable-gpl
--enable-libx264 --enable-memalign-hack --enable-shared
    (configure is ok)
   make
   unresolved external: libx264.o: __x264_bit_depth_


Can anyone help with this?

Best Regards,
R. Savchenko.

From krueger at lesspain.de  Mon Aug 19 18:11:08 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Mon, 19 Aug 2013 18:11:08 +0200
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
In-Reply-To: <52102BB0.3000506@gmail.com>
References: <52102BB0.3000506@gmail.com>
Message-ID: <CAEnqZEU51HMx7THxjEGra8d+8PB4vSZDaBL4upS33yEaOabV8A@mail.gmail.com>

On Sun, Aug 18, 2013 at 4:04 AM, simon mares <simon.mares at gmail.com> wrote:
> Hello, is it possible to generate a pre-defined moov atom for fixed lenght
> of fe. 8 hours with use of
> http://ffmpeg.org/ffmpeg-all.html#MOV_002fMP4_002fISMV arguments? Video
> would never be seeked beyond the real duration, as I would not allow.

If you mean, generate a moov atom that allows random access for the
first 8 hours will not work if you do not know your sample (i.e.
frame) sizes in advance (which may be doable in theory with a raw
video stream). As far as my understanding of the MP4 file format, what
you ask is not possible (provided I did not misunderstand your
requirement).

From krueger at lesspain.de  Mon Aug 19 18:13:56 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Mon, 19 Aug 2013 18:13:56 +0200
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
In-Reply-To: <CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>
References: <52102BB0.3000506@gmail.com>
	<CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>
Message-ID: <CAEnqZEW=2xGcS26+-EpoN-_r6RXp=+rLG5Zm-0JFoT6iNSLrZw@mail.gmail.com>

On Mon, Aug 19, 2013 at 4:39 PM, Tom Evans <tevans.uk at googlemail.com> wrote:
> On Sun, Aug 18, 2013 at 3:04 AM, simon mares <simon.mares at gmail.com> wrote:
>> Hello, is it possible to generate a pre-defined moov atom for fixed lenght
>> of fe. 8 hours with use of
>> http://ffmpeg.org/ffmpeg-all.html#MOV_002fMP4_002fISMV arguments? Video
>> would never be seeked beyond the real duration, as I would not allow.
>>
>> Reason for doing this is that I'm experimenting with something and I would
>> like to playback the file that I'm still recording, so it's kind of a
>> live-streaming, but it's only used locally. As I'm only transcoding from .ts
>> to .mp4 file no CPU is used, this is also my goal.
>>
>> Codec is h264, 720p, 25pfs. Video is generated with blackmagic pro-recorder.
>>
>
> Just use "-movflags faststart".
Are you sure this will result in a file he can play back while it is
being written? If the docs are correct, I doubt it will.

From onemda at gmail.com  Mon Aug 19 18:24:40 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 19 Aug 2013 16:24:40 +0000
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
In-Reply-To: <CAEnqZEW=2xGcS26+-EpoN-_r6RXp=+rLG5Zm-0JFoT6iNSLrZw@mail.gmail.com>
References: <52102BB0.3000506@gmail.com>
	<CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>
	<CAEnqZEW=2xGcS26+-EpoN-_r6RXp=+rLG5Zm-0JFoT6iNSLrZw@mail.gmail.com>
Message-ID: <CAPYw7P6cDvXB9oWX9MKCf7G+W0ScJd=hPNo6qeCBzxLB4QH-mg@mail.gmail.com>

On 8/19/13, Robert Krueger <krueger at lesspain.de> wrote:
> On Mon, Aug 19, 2013 at 4:39 PM, Tom Evans <tevans.uk at googlemail.com>
> wrote:
>> On Sun, Aug 18, 2013 at 3:04 AM, simon mares <simon.mares at gmail.com>
>> wrote:
>>> Hello, is it possible to generate a pre-defined moov atom for fixed
>>> lenght
>>> of fe. 8 hours with use of
>>> http://ffmpeg.org/ffmpeg-all.html#MOV_002fMP4_002fISMV arguments? Video
>>> would never be seeked beyond the real duration, as I would not allow.
>>>
>>> Reason for doing this is that I'm experimenting with something and I
>>> would
>>> like to playback the file that I'm still recording, so it's kind of a
>>> live-streaming, but it's only used locally. As I'm only transcoding from
>>> .ts
>>> to .mp4 file no CPU is used, this is also my goal.
>>>
>>> Codec is h264, 720p, 25pfs. Video is generated with blackmagic
>>> pro-recorder.
>>>
>>
>> Just use "-movflags faststart".
> Are you sure this will result in a file he can play back while it is
> being written? If the docs are correct, I doubt it will.
>

Of course it will not play back while it is still recording.
That flag just move moov atom from end of file to start of file at end
of muxing.

From simon.mares at gmail.com  Mon Aug 19 18:53:57 2013
From: simon.mares at gmail.com (simon mares)
Date: Mon, 19 Aug 2013 18:53:57 +0200
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
In-Reply-To: <CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>
References: <52102BB0.3000506@gmail.com>
	<CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>
Message-ID: <52124DA5.3060306@gmail.com>

On 08/19/2013 04:39 PM, Tom Evans wrote:
> On Sun, Aug 18, 2013 at 3:04 AM, simon mares <simon.mares at gmail.com> wrote:
>> Hello, is it possible to generate a pre-defined moov atom for fixed lenght
>> of fe. 8 hours with use of
>> http://ffmpeg.org/ffmpeg-all.html#MOV_002fMP4_002fISMV arguments? Video
>> would never be seeked beyond the real duration, as I would not allow.
>>
>> Reason for doing this is that I'm experimenting with something and I would
>> like to playback the file that I'm still recording, so it's kind of a
>> live-streaming, but it's only used locally. As I'm only transcoding from .ts
>> to .mp4 file no CPU is used, this is also my goal.
>>
>> Codec is h264, 720p, 25pfs. Video is generated with blackmagic pro-recorder.
>>
> Just use "-movflags faststart".
>
> Cheers
>
> Tom
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Hey
Thank you for answer.
I tried this, indeed ffmpeg did move moov atom in front of movie data 
(mdat) but when I open this video after 18 seconds of recoding and video 
*is still being recorded* then duration is 18 seconds in player, but 
video is still recording. What I would need that I would predefine this 
headers for some hours ahead.

Please note that I'm encoding from named pipe, so mp4 file is generated 
on-the-fly.

One more explanation

*Current state*:
1) Recording started (recording_01.mp4) from named pipe from video encoder
2) Video is being recorded for 18 seconds now and it continues to record
3) I open recording_01.mp4 in video player, duration is 18 seconds and 
I'm watching from start, 5 seconds
4) ffmpeg is still encoding to recording_01.mp4 ... duration 18, 19, 20, 
21 s, ...
5) when I come to 19 seconds video stops being played (point 3)

after 5 seconds of playing

0s   5s   10s   15s   20s
|-------------------------------| 23 sec rec_01.mp4
|----->----------------| 5 sec video player, duration 18 sec

*What I would like is following*:
1) Recording started (recording_01.mp4) from named pipe from video encoder
2) Video is being recorded for 18 seconds now and it continues to record
3) I open recording_01.mp4 in video player, duration is *3600* seconds 
and I'm watching from start
4) ffmpeg is still encoding to recording_01.mp4 ... duration 18, 19, 20, 
21 s, ...
5) when I come to 19 seconds video *doesnt stop* being played

0s   5s   10s   15s   20s
|-------------------------------| 23 sec rec_01.mp4
|----->---------------------------------- .... -----| 5 sec video 
player, duration 1 hour

Check points 3 and 5, where I put "*".

I hope it's more understandable now.

Cheers,

Simon








From h.reindl at thelounge.net  Mon Aug 19 18:59:49 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Mon, 19 Aug 2013 18:59:49 +0200
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
In-Reply-To: <52124DA5.3060306@gmail.com>
References: <52102BB0.3000506@gmail.com>
	<CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>
	<52124DA5.3060306@gmail.com>
Message-ID: <52124F05.7010707@thelounge.net>



Am 19.08.2013 18:53, schrieb simon mares:
>> Just use "-movflags faststart".
>>
> I tried this, indeed ffmpeg did move moov atom in front of movie data (mdat) but when I open this video after 18
> seconds of recoding and video *is still being recorded* then duration is 18 seconds in player, but video is still
> recording. What I would need that I would predefine this headers for some hours ahead.
> 
> Please note that I'm encoding from named pipe, so mp4 file is generated on-the-fly

you can't pre-define headers of a unknown output file
on which base do you want to guess the data?

maybe some luck with this because it does the same as "-movflags faststart"
with files which have the moov at the end - but not really sure if
it generates one on-the-fly to make the client happy while the
output file is still generated
http://h264.code-shop.com/trac/wiki/Mod-H264-Streaming-Apache-Version2




-------------- next part --------------
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From dzungng89 at gmail.com  Mon Aug 19 19:55:50 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Mon, 19 Aug 2013 12:55:50 -0500
Subject: [FFmpeg-user] Concatenation buffer underflow problem
Message-ID: <CABOWChw67g29xukSkkno-QDESBbnrkJBSgqexDUA8kBfdtCWjQ@mail.gmail.com>

I created subvideos from image, using the following command:

ffmpeg -loop 1 -f image2 -i image.png -r 29.97 -t 10 video1.mpg

Then concatenate them using:

ffmpeg -f concat -i list.txt -c copy output.mpg

where list.txt has the list of subvideo files.

The problem is that the 2nd command has buffer underflow problem: packet
too large, ignoring buffer limit. How to fix it?

From jpboard2 at yahoo.com  Mon Aug 19 20:41:09 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 19 Aug 2013 11:41:09 -0700 (PDT)
Subject: [FFmpeg-user] Detecting Scene Changes In Source Video?
In-Reply-To: <20130819102240.GB6843@barisone>
References: <1376747798.41445.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<20130819102240.GB6843@barisone>
Message-ID: <1376937669.75064.YahooMailNeo@web164704.mail.gq1.yahoo.com>

>> Is there a way to use ffmpeg to detect scene changes in a video file?? I know

>> some of the codecs must do this already, but I don't know if that output is
>> readily available?? Maybe in the info file produced when 2-pass encoding?
>
>Check the select filter and the scene paramenter.

How about something like
??? ffmpeg -i IN.avi -vf select='gt(scene\,0.4',print(n)
How can I make the above command work so that it prints the
frame numbers of scene changes?

From j+ffmpeg at lundstocholm.dk  Mon Aug 19 20:26:32 2013
From: j+ffmpeg at lundstocholm.dk (jlundstocholm)
Date: Mon, 19 Aug 2013 11:26:32 -0700 (PDT)
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
Message-ID: <1376936792242-4660782.post@n4.nabble.com>

Hi Werner,

Thanks for your reply.

> Where is your script? 

My bad - I meant "configuration of ffmpeg" - for now I'm doing it by hand.

> -to does not take a duration ? it takes the timestamp of where you would 
> like encoding to end. This is exactly the option you need.

Yes, the manual tells me:

/-to time_stop       record or transcode stop time/

But can you explain to me why the to-switches I applied to ffmpeg behaves
like a duration? ffmpeg produces 5 files to me - with the lengths

00:15:18
00:18:58
00:16:12
00:18:44
00:18:08

It sure behaves like a duration, but of course I might be using ffmpeg
wrongly ... it is certainly an option.

Thanks,

/Jesper
Copenhagen, Denmark



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From adf.lists at gmail.com  Mon Aug 19 21:16:35 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Mon, 19 Aug 2013 20:16:35 +0100
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
In-Reply-To: <1376816843722-4660765.post@n4.nabble.com>
References: <1376816843722-4660765.post@n4.nabble.com>
Message-ID: <52126F13.8010204@gmail.com>

mokacao wrote:
> I'm doing a convert picture to MP4 video, the original image is
> PIX_FMT_BGRA format, now converted from RGB to YUV420P, I found many
> black spot around the picture, the follow is part of the core code,
> please help me to how to clear these black dots, or can tell me what
> information is missing, thank you.
>
> libffmpeg::SwsContext* swsContext = libffmpeg::sws_getContext( srcW,
> srcH, libffmpeg::PIX_FMT_BGRA,codecContext->width,
> codecContext->height, libffmpeg::PIX_FMT_YUV420P, SWS_BICUBIC, NULL,
> NULL, NULL);
>
> libffmpeg::sws_scale( swsContext , srcData, srcLinesize, 0,
> frame->Height, data->VideoFrame->data, data->VideoFrame->linesize );
>
> original png file
> <http://ffmpeg-users.933282.n4.nabble.com/file/n4660765/P0085.png>
>
> yuv420p image file(scale and clip)
> <http://ffmpeg-users.933282.n4.nabble.com/file/n4660765/yuv420p.png>

It does seem that there is an issue here.

Out of curiosity I made a 420 from your png, played it with ffplay, took
an xwd and displayed it with display from image magick so I could zoom.

It looks OK and doesn't have any of the blackness of your example. It
does show some darkening of the boundary red pixels.

I decided to try some other 420 -> rgb convertions with mplayer,
vdpau,xv and gl all looked the same as ffplay.

-vo x11, though, looked like your example so it seems there is an issue
somewhere. Of course mplayer is not ffmpeg and I don't know what code is
being run when it says -

[swscaler @ 0x7fba4aac65b0] using unscaled yuv420p -> bgra special converter

but it would seem you have found something similar with ffmpeg -
assuming you are actually using ffmpeg to display/convert back to rgb
the image?

From werner.robitza at gmail.com  Mon Aug 19 21:35:17 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Mon, 19 Aug 2013 21:35:17 +0200
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <1376936792242-4660782.post@n4.nabble.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
Message-ID: <CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>

On Mon, Aug 19, 2013 at 8:26 PM, jlundstocholm <j+ffmpeg at lundstocholm.dk> wrote:
>
> But can you explain to me why the to-switches I applied to ffmpeg behaves
> like a duration? ffmpeg produces 5 files to me - with the lengths
>
> 00:15:18
> 00:18:58
> 00:16:12
> 00:18:44
> 00:18:08

Maybe I'm missing context here, but what exact ffmpeg calls are you
using that produce said files?

As far as I know the -to option works exactly as intended.

From j+ffmpeg at lundstocholm.dk  Mon Aug 19 22:15:09 2013
From: j+ffmpeg at lundstocholm.dk (jlundstocholm)
Date: Mon, 19 Aug 2013 13:15:09 -0700 (PDT)
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
	<CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
Message-ID: <1376943309388-4660786.post@n4.nabble.com>

Hi Werner,

> Maybe I'm missing context here, but what exact ffmpeg calls are you 
> using that produce said files?





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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Using-ffmpeg-to-remove-commercials-from-video-file-tp4660699p4660786.html
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From b_bruggeman at ziggo.nl  Mon Aug 19 12:46:47 2013
From: b_bruggeman at ziggo.nl (bert1974)
Date: Mon, 19 Aug 2013 03:46:47 -0700 (PDT)
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
In-Reply-To: <1376816843722-4660765.post@n4.nabble.com>
References: <1376816843722-4660765.post@n4.nabble.com>
Message-ID: <1376909207070-4660771.post@n4.nabble.com>

PIX_FMT_BGRA contains alpha(=transparency) information.. to get the
transparent edges right, you will have to blend/draw on a background color
when converting from bgra to rgb. don't know how to do with SwsContext,
maybe convert/blend rgba->rgb manual, than convert rgb to yuv using
SwsContext

(it looks like the image in your mail is blended on black instead of white)



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From chaitanyag at gopalsystems.com  Sun Aug 18 19:33:20 2013
From: chaitanyag at gopalsystems.com (Chaitanya Gopathi)
Date: Sun, 18 Aug 2013 23:03:20 +0530
Subject: [FFmpeg-user] MPEG1/2 does not support 3/1 fps error
Message-ID: <CABRdPaKjcVRDV8Hwd=PYQk_Pi+E4xurP6FMjV5BK0aomXiDu-A@mail.gmail.com>

Hi,

getting below error while generating output video. Could you please help me
to resolve this issue. Attached output file and configuration file for more
details.

[mpeg1video @ 0x1999afe0]MPEG1/2 does not support 3/1 fps


-- 
Thanks & Regards,
Chaitanya
-------------- next part --------------
root at server [~]# ffmpeg -i /home/xestatt1/www/app/webroot/mobile/uploads/Aug172013181921.3gp http://localhost:8091/feed1.ffm
FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  built on Jan 29 2012 23:55:02 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
  configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  libavutil     50.15. 1 / 50.15. 1
  libavcodec    52.72. 2 / 52.72. 2
  libavformat   52.64. 2 / 52.64. 2
  libavdevice   52. 2. 0 / 52. 2. 0
  libavfilter    1.19. 0 /  1.19. 0
  libswscale     0.11. 0 /  0.11. 0
  libpostproc   51. 2. 0 / 51. 2. 0

Seems stream 0 codec frame rate differs from container frame rate: 1200.00 (1200/1) -> 20.00 (20/1)
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/xestatt1/www/app/webroot/mobile/uploads/Aug172013181921.3gp':
  Metadata:
    major_brand     : qt
    minor_version   : 0
    compatible_brands: qt
  Duration: 00:00:05.01, start: 0.000000, bitrate: 171 kb/s
    Stream #0.0(und): Video: h264, yuv420p, 192x144, 141 kb/s, 20.01 fps, 20 tbr, 600 tbn, 1200 tbc
    Stream #0.1(und): Audio: aac, 22050 Hz, mono, s16, 23 kb/s
[mpeg1video @ 0xae0ffe0]MPEG1/2 does not support 3/1 fps
Output #0, ffm, to 'http://localhost:8091/feed1.ffm':
    Stream #0.0(und): Audio: mp2, 44100 Hz, 1 channels, s16, 32 kb/s
    Stream #0.1(und): Video: mpeg1video, yuv420p, 160x128, q=2-31, 64 kb/s, 1000k tbn, 3 tbc
    Stream #0.2(und): Audio: libmp3lame, 22050 Hz, 1 channels, s16, 64 kb/s
    Stream #0.3(und): Video: msmpeg4, yuv420p, 352x240, q=2-31, 256 kb/s, 1000k tbn, 30 tbc
Stream mapping:
  Stream #0.1 -> #0.0
  Stream #0.0 -> #0.1
  Stream #0.1 -> #0.2
  Stream #0.0 -> #0.3
Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height
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From mathieu.multicam+ffmpeg at gmail.com  Mon Aug 19 08:18:45 2013
From: mathieu.multicam+ffmpeg at gmail.com (MatCam)
Date: Sun, 18 Aug 2013 23:18:45 -0700 (PDT)
Subject: [FFmpeg-user] Merging H263 videos
In-Reply-To: <000001ce9b2d$a9862ee0$fc928ca0$@gmail.com>
References: <1376645760323-4660745.post@n4.nabble.com>
	<000001ce9b2d$a9862ee0$fc928ca0$@gmail.com>
Message-ID: <1376893125719-4660768.post@n4.nabble.com>

Thanks for both responses !

I'm agree, trying to solve problem by adding more and more properties to my
request to ffmpeg is not the best way to successfully make my video...

So if I understand, using TS will avoid loss of quality ?
so my first step is to follow instruction to concat 2 h264 video in one
using TS:

*D:\ffmpeg-20130813-git-bbbd959-win64-static\bin>ffmpeg.exe -i
D:\Records\2013-08-16_084044\Source\2013-08-16_084044.mp4 -c copy -bsf:v
h264_mp4toannexb -f mpegts intermediate1.ts*

ffmpeg version N-55515-gbbbd959 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 13 2013 18:06:32 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libblu
ray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc
--enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenjpeg --enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-libtwolame
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-lib
vpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 42.100 / 52. 42.100
  libavcodec     55. 27.100 / 55. 27.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'D:\Records\2013-08-16_084044\Source\2013-08-16_084044.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf54.31.100
  Duration: 00:00:05.07, start: 0.000000, bitrate: 5476 kb/s
    Stream #0:0(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D),
yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 5390 kb/s, 50 fps, 50 tbr, 12800 tbn,
50 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo,
fltp, 128 kb/s
    Metadata:
      handler_name    : SoundHandler
Output #0, mpegts, to 'intermediate1.ts':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf55.13.102
    Stream #0:0(und): Video: mpeg4 (mp4v / 0x7634706D), yuv420p, 1280x720
[SAR 1:1 DAR 16:9], q=2-31, 5390 kb/s, 50 fps, 90k tbn, 50 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 128
kb/s
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[NULL @ 0000000004e20620] Warning: SPS NALU missing or invalid. The
resulting stream may not play.
[NULL @ 0000000004e20620] Warning: PPS NALU missing or invalid. The
resulting stream may not play.
Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec
copy: Invalid argument
[adts @ 000000000251c360] Encoder did not produce proper pts, making some
up.
frame=  251 fps=0.0 q=-1.0 Lsize=    3632kB time=00:00:05.07
bitrate=5859.8kbits/s
video:3303kB audio:79kB subtitle:0 global headers:0kB muxing overhead
7.365148%

here is my first problem.

Secondly, what are the parameter i have to use to make the same in h263 ?
is there a filter h264_mp4toannexb for h263 for making the same to concat
files ?

Thanks



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From adf.lists at gmail.com  Tue Aug 20 01:16:27 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Tue, 20 Aug 2013 00:16:27 +0100
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
In-Reply-To: <1376909207070-4660771.post@n4.nabble.com>
References: <1376816843722-4660765.post@n4.nabble.com>
	<1376909207070-4660771.post@n4.nabble.com>
Message-ID: <5212A74B.1050300@gmail.com>

bert1974 wrote:
> PIX_FMT_BGRA contains alpha(=transparency) information.. to get the
> transparent edges right, you will have to blend/draw on a background color
> when converting from bgra to rgb. don't know how to do with SwsContext,
> maybe convert/blend rgba->rgb manual, than convert rgb to yuv using
> SwsContext
>
> (it looks like the image in your mail is blended on black instead of white)

I don't think this is it as the alpha seems to be 255 everywhere on the 
png and the artifacts seem to arise on conversion back to rgb.

Testing on 420 produced from the png with no sws options.

Converting back to rgb with ffmpeg -

ffmpeg -s 890x768 -pix_fmt yuv420p -i 420.yuv -sws_flags 
+accurate_rnd+full_chroma_int -pix_fmt rgb24 420-ar-fc.rgb

is the best looking.

-sws_flags +accurate_rnd still has noticable "black" (really grey) 
pixels but is better than no sws options or
-sws_flags +full_chroma_int which produces exactly the same output 
(md5sum) as no sws options.



From mokacao at gmail.com  Tue Aug 20 05:11:47 2013
From: mokacao at gmail.com (mokacao)
Date: Mon, 19 Aug 2013 20:11:47 -0700 (PDT)
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
In-Reply-To: <52126F13.8010204@gmail.com>
References: <1376816843722-4660765.post@n4.nabble.com>
	<52126F13.8010204@gmail.com>
Message-ID: <1376968307925-4660789.post@n4.nabble.com>

I generated a test that contains a variety of colors pictures, red/green have
black dot.

Here is part of the core code:

void writeframe( Bitmap^ frame)
{
        libffmpeg::AVOutputFormat* outputFormat =
libffmpeg::av_guess_format( NULL, "test.mp4", NULL );

        if ( !outputFormat )
        {
                outputFormat = libffmpeg::av_guess_format( "mp4", NULL, NULL
);
        }

        libffmpeg::AVFrame*         videoFrame = NULL;
        libffmpeg::AVFormatContext* formatContext =
libffmpeg::avformat_alloc_context( );
        formatContext->oformat = outputFormat;

        libffmpeg::AVCodec *codec =
libffmpeg::avcodec_find_encoder(libffmpeg::CODEC_ID_H264);
        libffmpeg::AVCodecContext* codecContex;
        libffmpeg::AVStream* videoStream = libffmpeg::avformat_new_stream(
formatContext, codec );

        codecContex = videoStream->codec;
        codecContex->codec_id   = libffmpeg::CODEC_ID_H264;
        codecContex->codec_type = libffmpeg::AVMEDIA_TYPE_VIDEO;
        codecContex->bit_rate = 200;
        codecContex->width    = 1024;
        codecContex->height   = 768;
        codecContex->time_base.num = 1;
        codecContex->time_base.den = 25;

        libffmpeg::AVCodecContext* codecContext = videoStream->codec;

        libffmpeg::avcodec_open2( codecContext, codec, NULL );

        libffmpeg::AVFrame* picture;
        void* picture_buf;
        int size;

        picture = libffmpeg::avcodec_alloc_frame( );

        size = libffmpeg::avpicture_get_size( codecContext->pix_fmt,
codecContext->width, codecContext->height );
        picture_buf = libffmpeg::av_malloc( size );

        libffmpeg::avpicture_fill( (libffmpeg::AVPicture *) picture,
(libffmpeg::uint8_t *) picture_buf, codecContext->pix_fmt,
codecContext->width, codecContext->height );

        videoFrame = picture;

        ibffmpeg::SwsContext* swsContext = libffmpeg::sws_getContext( srcW,
srcH, libffmpeg::PIX_FMT_BGRA,codecContext->width, codecContext->height,
libffmpeg::PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);

        BitmapData^ bitmapData = frame->LockBits(
System::Drawing::Rectangle( 0, 0, frame->Width, frame->Height ),
                ImageLockMode::ReadOnly,
                System::Drawing::Imaging::PixelFormat::Format32bppArgb );

        libffmpeg::uint8_t* ptr = reinterpret_cast<libffmpeg::uint8_t*>(
static_cast<void*>( bitmapData->Scan0 ) );

        libffmpeg::uint8_t* srcData[4] = { ptr, NULL, NULL, NULL };
        int srcLinesize[4] = { bitmapData->Stride, 0, 0, 0 };

        libffmpeg::sws_scale( swsContext, srcData, srcLinesize, 0,
frame->Height, videoFrame->data, videoFrame->linesize );

        frame->UnlockBits( bitmapData );

        yuv420p_save(videoFrame, videoStream->codec);

}

void yuv420p_save(libffmpeg::AVFrame *pFrame, libffmpeg::AVCodecContext
*pCodecCtx)
{
        FILE *pfout = NULL;
        char ffrvout[128] = { 0 };
        sprintf(ffrvout, "%s", "c:\\test.yuv"); 

        pfout = fopen(ffrvout, "wb+"); 

        int i = 0;

        int width = pCodecCtx->width, height = pCodecCtx->height;
        int height_half = height / 2, width_half = width / 2;
        int y_wrap = pFrame->linesize[0];
        int u_wrap = pFrame->linesize[1];
        int v_wrap = pFrame->linesize[2];

        unsigned char *y_buf = pFrame->data[0];
        unsigned char *u_buf = pFrame->data[1];
        unsigned char *v_buf = pFrame->data[2];

        //save y
        for (i = 0; i < height; i++)
                fwrite(y_buf + i * y_wrap, 1, width, pfout);
        fprintf(stderr, "===>save Y success\n");
        //save u
        for (i = 0; i < height_half; i++)
                fwrite(u_buf + i * u_wrap, 1, width_half, pfout);
        fprintf(stderr, "===>save U success\n");
        //save v
        for (i = 0; i < height_half; i++)
                fwrite(v_buf + i * v_wrap, 1, width_half, pfout);
        fprintf(stderr, "===>save V success\n");

        fflush(pfout);
        fclose(pfout); 
}

original png file
<http://ffmpeg-users.933282.n4.nabble.com/file/n4660789/P0001.png> 
yuv420p zoom in 4x, use the software yuv
viewer(http://sourceforge.net/projects/yuviewer/), and screenshot
<http://ffmpeg-users.933282.n4.nabble.com/file/n4660789/P0001_yuv_zoom_in_4x.png> 
original png file to yuv420p use above code
P0001.yuv <http://ffmpeg-users.933282.n4.nabble.com/file/n4660789/P0001.yuv>  



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-can-I-convert-rbg-to-yuv420p-loseless-tp4660765p4660789.html
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From werner.robitza at gmail.com  Tue Aug 20 09:10:28 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Tue, 20 Aug 2013 09:10:28 +0200
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <1376943309388-4660786.post@n4.nabble.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
	<CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
	<1376943309388-4660786.post@n4.nabble.com>
Message-ID: <CAN0LTaNwyh1TUMxZ5LR6tpDQK66CKPt=i4zaQa6PfbWUPjwLyQ@mail.gmail.com>

On Mon, Aug 19, 2013 at 10:15 PM, jlundstocholm
<j+ffmpeg at lundstocholm.dk> wrote:
>
>> Maybe I'm missing context here, but what exact ffmpeg calls are you
>> using that produce said files?
>
>
>
>
>
> --
> View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Using-ffmpeg-to-remove-commercials-from-video-file-tp4660699p4660786.html

Now I know what's going on. Your message turns up empty here. This
would explain why I didn't see your script earlier ? it was swallowed
somewhere in the mail chain.
Please disable rich text formatting in your mail client and send as
plain-text only.

Ideally, can you show us the full, uncut console output for a command
where you think the -to option is not working properly?

Thanks!

From j+ffmpeg at lundstocholm.dk  Tue Aug 20 09:20:20 2013
From: j+ffmpeg at lundstocholm.dk (jlundstocholm)
Date: Tue, 20 Aug 2013 00:20:20 -0700 (PDT)
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <CAN0LTaNwyh1TUMxZ5LR6tpDQK66CKPt=i4zaQa6PfbWUPjwLyQ@mail.gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
	<CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
	<1376943309388-4660786.post@n4.nabble.com>
	<CAN0LTaNwyh1TUMxZ5LR6tpDQK66CKPt=i4zaQa6PfbWUPjwLyQ@mail.gmail.com>
Message-ID: <1376983220384-4660791.post@n4.nabble.com>

Hi Werner,

> Now I know what's going on. Your message turns up empty here. This 
> would explain why I didn't see your script earlier ? it was swallowed 
> somewhere in the mail chain. 
> Please disable rich text formatting in your mail client and send as 
> plain-text only. 

Well, I was actually using the web-ui on nabble.com - I assumed it would "do
the right thing" in terms of creating both a "rich" version of the post and
a plain-text version as well. I'll switch to regular mail instead when I
have produced the output you are asking for.

:o)

/Jesper
Copenhagen, Denmark



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Sent from the FFmpeg-users mailing list archive at Nabble.com.

From 4a4553504552 at gmail.com  Tue Aug 20 09:54:44 2013
From: 4a4553504552 at gmail.com (Jesper Lund Stocholm)
Date: Tue, 20 Aug 2013 09:54:44 +0200
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <1376983220384-4660791.post@n4.nabble.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
	<CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
	<1376943309388-4660786.post@n4.nabble.com>
	<CAN0LTaNwyh1TUMxZ5LR6tpDQK66CKPt=i4zaQa6PfbWUPjwLyQ@mail.gmail.com>
	<1376983220384-4660791.post@n4.nabble.com>
Message-ID: <CAKDDL-LUEf3X_YvZoZTx9sa9Nph7URkKaibBkxBeL5Fzc4upOw@mail.gmail.com>

Hi Werner,



> > Now I know what's going on. Your message turns up empty here. This
> > would explain why I didn't see your script earlier ? it was swallowed
> > somewhere in the mail chain.
> > Please disable rich text formatting in your mail client and send as
> > plain-text only.
>
>
So the commands I use are these:

ffmpeg -ss "00:00:00.000" -i "movie.m4v" -to "00:15:18" -c:v copy -c:a copy
"result.pt1.m4v"
ffmpeg -ss "00:22:29.500" -i "movie.m4v" -to "00:18:58" -c:v copy -c:a copy
"result.pt2.m4v"
ffmpeg -ss "00:50:24.500" -i "movie.m4v" -to "00:16:12" -c:v copy -c:a copy
"result.pt3.m4v"
ffmpeg -ss "01:14:48.500" -i "movie.m4v" -to "00:18:44" -c:v copy -c:a copy
"result.pt4.m4v"
ffmpeg -ss "01:41:35.000" -i "movie.m4v" -to "00:18:08" -c:v copy -c:a copy
"result.pt5.m4v"


Complete console output for the second command (file two) is this:

c:\users\jst\desktop\ffmpeg -ss "00:22:29.500" -i "movie.m4v" -to
"00:18:58" -c:v copy -c:a copy "c:\users\jst\desktop\output.pt2.m4v"
ffmpeg version N-55393-g3b2e99f Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug  8 2013 21:37:55 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
--enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
--ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 24.100 / 55. 24.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000002ea860] stream 0, timescale not set
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'movie.m4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    creation_time   : 2012-11-18 22:23:57
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog)
Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i
virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske
igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog)
Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i
virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske
igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog)
Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i
virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske
igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    date            : 2009-01-01
  Duration: 02:01:39.36, start: 0.000000, bitrate: 5089 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1280x720,
 4953 kb/s, 25 fps, 25 tbr, 2500 tbn, 5k tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo,
fltp, 12
5 kb/s
    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
    Stream #0:2: Video: mjpeg, yuvj420p, 512x748 [SAR 1:1 DAR 128:187], 90k
tbr,
 90k tbn, 90k tbc
Output #0, ipod, to 'c:\users\jst\desktop\movie.pt2.m
4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    date            : 2009-01-01
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog)
Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i
virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske
igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog)
Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i
virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske
igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog)
Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i
virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske
igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    encoder         : Lavf55.13.102
    Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 1280x720,
q=2-31
, 4953 kb/s, 25 fps, 10k tbn, 2500 tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 125
kb/s

    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 5403 fps=0.0 q=-1.0 size=  137903kB time=00:03:36.70
bitrate=5213.0kbits/
frame=11494 fps=11493 q=-1.0 size=  285293kB time=00:07:40.42
bitrate=5076.1kbit
frame=17750 fps=11833 q=-1.0 size=  415500kB time=00:11:50.78
bitrate=4788.7kbit
frame=23625 fps=11529 q=-1.0 size=  551621kB time=00:15:45.79
bitrate=4777.9kbit
frame=28452 fps=11201 q=-1.0 Lsize=  702461kB time=00:18:58.00
bitrate=5056.7kbi
ts/s
video:684096kB audio:17515kB subtitle:0 global headers:0kB muxing overhead
0.121
056%


 (also attached as unicode text file)



Thank you so much for your input on this

-- 
Jesper Lund Stocholm
www.idippedut.dk
SC34/WG4 http://www.itscj.ipsj.or.jp/sc34/wg4/
-------------- next part --------------
?c:\users\jst\desktop\ffmpeg -ss "00:22:29.500" -i "movie.m4v" -to "00:18:58" -c:v copy -c:a copy "c:\users\jst\desktop\output.pt2.m4v"
ffmpeg version N-55393-g3b2e99f Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  8 2013 21:37:55 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 24.100 / 55. 24.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000002ea860] stream 0, timescale not set
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'The Princess and the Frog (2009).m4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    creation_time   : 2012-11-18 22:23:57
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    date            : 2009-01-01
  Duration: 02:01:39.36, start: 0.000000, bitrate: 5089 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720,
 4953 kb/s, 25 fps, 25 tbr, 2500 tbn, 5k tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 12
5 kb/s
    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
    Stream #0:2: Video: mjpeg, yuvj420p, 512x748 [SAR 1:1 DAR 128:187], 90k tbr,
 90k tbn, 90k tbc
Output #0, ipod, to 'c:\users\jst\desktop\movie.pt2.m
4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    date            : 2009-01-01
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    encoder         : Lavf55.13.102
    Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 1280x720, q=2-31
, 4953 kb/s, 25 fps, 10k tbn, 2500 tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 125 kb/s

    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 5403 fps=0.0 q=-1.0 size=  137903kB time=00:03:36.70 bitrate=5213.0kbits/
frame=11494 fps=11493 q=-1.0 size=  285293kB time=00:07:40.42 bitrate=5076.1kbit
frame=17750 fps=11833 q=-1.0 size=  415500kB time=00:11:50.78 bitrate=4788.7kbit
frame=23625 fps=11529 q=-1.0 size=  551621kB time=00:15:45.79 bitrate=4777.9kbit
frame=28452 fps=11201 q=-1.0 Lsize=  702461kB time=00:18:58.00 bitrate=5056.7kbi
ts/s
video:684096kB audio:17515kB subtitle:0 global headers:0kB muxing overhead 0.121
056%

From werner.robitza at gmail.com  Tue Aug 20 10:47:48 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Tue, 20 Aug 2013 10:47:48 +0200
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <CAKDDL-LUEf3X_YvZoZTx9sa9Nph7URkKaibBkxBeL5Fzc4upOw@mail.gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
	<CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
	<1376943309388-4660786.post@n4.nabble.com>
	<CAN0LTaNwyh1TUMxZ5LR6tpDQK66CKPt=i4zaQa6PfbWUPjwLyQ@mail.gmail.com>
	<1376983220384-4660791.post@n4.nabble.com>
	<CAKDDL-LUEf3X_YvZoZTx9sa9Nph7URkKaibBkxBeL5Fzc4upOw@mail.gmail.com>
Message-ID: <CAN0LTaNX+VFJRgdt53yoHK0jWYkh-DwfkbKAL25tXbSxjnPChg@mail.gmail.com>

On Tue, Aug 20, 2013 at 9:54 AM, Jesper Lund Stocholm
<4a4553504552 at gmail.com> wrote:
> So the commands I use are these:
>
> ffmpeg -ss "00:00:00.000" -i "movie.m4v" -to "00:15:18" -c:v copy -c:a copy
> "result.pt1.m4v"
> ffmpeg -ss "00:22:29.500" -i "movie.m4v" -to "00:18:58" -c:v copy -c:a copy
> "result.pt2.m4v"
> ffmpeg -ss "00:50:24.500" -i "movie.m4v" -to "00:16:12" -c:v copy -c:a copy
> "result.pt3.m4v"
> ffmpeg -ss "01:14:48.500" -i "movie.m4v" -to "00:18:44" -c:v copy -c:a copy
> "result.pt4.m4v"
> ffmpeg -ss "01:41:35.000" -i "movie.m4v" -to "00:18:08" -c:v copy -c:a copy
> "result.pt5.m4v"

I see now.

Note that -ss specifies the start time and -to the end time.
You cannot start at 22:29 and end at 18:58. Unless you expect ffmpeg
to travel back in time :)

If you want it to start at 22:29 and encode for 18 minutes and 58
seconds, you need -t, not -to.

From krueger at lesspain.de  Tue Aug 20 11:31:12 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Tue, 20 Aug 2013 11:31:12 +0200
Subject: [FFmpeg-user] mp4 generate moov atom at beginning of recording
In-Reply-To: <52124DA5.3060306@gmail.com>
References: <52102BB0.3000506@gmail.com>
	<CAFHbX1+9z1fKiQSDNJM5EQa2VTn9UFZH4K_mgTcfHevyyExRpA@mail.gmail.com>
	<52124DA5.3060306@gmail.com>
Message-ID: <CAEnqZEU-b28cM9Grnbg_g0ovHgA6B744yQxU-HtiTy-JfJF7Sw@mail.gmail.com>

On Mon, Aug 19, 2013 at 6:53 PM, simon mares <simon.mares at gmail.com> wrote:
> On 08/19/2013 04:39 PM, Tom Evans wrote:
>>
>> On Sun, Aug 18, 2013 at 3:04 AM, simon mares <simon.mares at gmail.com>
>> wrote:
>>>
>>> Hello, is it possible to generate a pre-defined moov atom for fixed
>>> lenght
>>> of fe. 8 hours with use of
>>> http://ffmpeg.org/ffmpeg-all.html#MOV_002fMP4_002fISMV arguments? Video
>>> would never be seeked beyond the real duration, as I would not allow.
>>>
>>> Reason for doing this is that I'm experimenting with something and I
>>> would
>>> like to playback the file that I'm still recording, so it's kind of a
>>> live-streaming, but it's only used locally. As I'm only transcoding from
>>> .ts
>>> to .mp4 file no CPU is used, this is also my goal.
>>>
>>> Codec is h264, 720p, 25pfs. Video is generated with blackmagic
>>> pro-recorder.
>>>
>> Just use "-movflags faststart".
>>
>> Cheers
>>
>> Tom
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> Hey
> Thank you for answer.
> I tried this, indeed ffmpeg did move moov atom in front of movie data (mdat)
> but when I open this video after 18 seconds of recoding and video *is still
> being recorded* then duration is 18 seconds in player, but video is still
> recording. What I would need that I would predefine this headers for some
> hours ahead.
>
> Please note that I'm encoding from named pipe, so mp4 file is generated
> on-the-fly.
>
> One more explanation
>
> *Current state*:
> 1) Recording started (recording_01.mp4) from named pipe from video encoder
> 2) Video is being recorded for 18 seconds now and it continues to record
> 3) I open recording_01.mp4 in video player, duration is 18 seconds and I'm
> watching from start, 5 seconds
> 4) ffmpeg is still encoding to recording_01.mp4 ... duration 18, 19, 20, 21
> s, ...
> 5) when I come to 19 seconds video stops being played (point 3)
>
> after 5 seconds of playing
>
> 0s   5s   10s   15s   20s
> |-------------------------------| 23 sec rec_01.mp4
> |----->----------------| 5 sec video player, duration 18 sec
>
> *What I would like is following*:
> 1) Recording started (recording_01.mp4) from named pipe from video encoder
> 2) Video is being recorded for 18 seconds now and it continues to record
> 3) I open recording_01.mp4 in video player, duration is *3600* seconds and
> I'm watching from start
> 4) ffmpeg is still encoding to recording_01.mp4 ... duration 18, 19, 20, 21
> s, ...
> 5) when I come to 19 seconds video *doesnt stop* being played
>
> 0s   5s   10s   15s   20s
> |-------------------------------| 23 sec rec_01.mp4
> |----->---------------------------------- .... -----| 5 sec video player,
> duration 1 hour
>
> Check points 3 and 5, where I put "*".
>
> I hope it's more understandable now.
>

In theory this is a perfect use case for fragmented mp4 but its
usefulness in your case depends on what you are going to use the
generated mp4 files for. I would guess only a fraction of programs
supporting mp4 as a container properly deal with fragmented mp4 files.
You could simply test. Check the part of the documentation that you
already found on the correct command line parameters for fragmented
mp4 (essentially it generates a header every X seconds or bytes and a
player that supports it can jump through those headers like through a
linked list and should be able to do random access in the fragments
that have a complete header).

From 4a4553504552 at gmail.com  Tue Aug 20 12:13:41 2013
From: 4a4553504552 at gmail.com (Jesper Lund Stocholm)
Date: Tue, 20 Aug 2013 12:13:41 +0200
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <CAN0LTaNX+VFJRgdt53yoHK0jWYkh-DwfkbKAL25tXbSxjnPChg@mail.gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
	<CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
	<1376943309388-4660786.post@n4.nabble.com>
	<CAN0LTaNwyh1TUMxZ5LR6tpDQK66CKPt=i4zaQa6PfbWUPjwLyQ@mail.gmail.com>
	<1376983220384-4660791.post@n4.nabble.com>
	<CAKDDL-LUEf3X_YvZoZTx9sa9Nph7URkKaibBkxBeL5Fzc4upOw@mail.gmail.com>
	<CAN0LTaNX+VFJRgdt53yoHK0jWYkh-DwfkbKAL25tXbSxjnPChg@mail.gmail.com>
Message-ID: <CAKDDL-J07Xus=0D6KKsjnu7dvpNiVVj3Z45m7DppGm6-nODhdw@mail.gmail.com>

Hi Werner,


2013/8/20 Werner Robitza <werner.robitza at gmail.com>
>
>
> Note that -ss specifies the start time and -to the end time.
> You cannot start at 22:29 and end at 18:58. Unless you expect ffmpeg
> to travel back in time :)
>
> If you want it to start at 22:29 and encode for 18 minutes and 58
> seconds, you need -t, not -to.
>
>
Yes, I would think that as well. The thing is just that if I use the
timestamp instead of the duration, e.g.


ffmpeg -ss "00:22:29.500" -i "movie.m4v" -to "00:41:27" -c:v copy -c:a copy
"output.pt2_to_4127.m4v"
ffmpeg -ss "00:22:29.500" -i "movie.m4v" -t "00:41:27" -c:v copy -c:a copy
"output.pt2_t_4127.m4v"

ffmpeg produces exactly identical files and console output (apart from the
actual copying of stream that vary a little bit (I assume that is because
of my machine doing other stuff as well)).

Both files are 41:27 minutes in length.


-- 
Jesper Lund Stocholm
www.idippedut.dk
SC34/WG4 http://www.itscj.ipsj.or.jp/sc34/wg4/
-------------- next part --------------
?
C:\Users\jst\Downloads\ffmpeg-20130809-git-3b2e99f-win64-static\ffmpeg-20130809-
git-3b2e99f-win64-static\bin>"ffmpeg commercial removal.bat"

C:\Users\jst\Downloads\ffmpeg-20130809-git-3b2e99f-win64-static\ffmpeg-20130809-
git-3b2e99f-win64-static\bin>ffmpeg -ss "00:22:29.500" -i "The Princess and the
Frog (2009).m4v" -t "00:41:27" -c:v copy -c:a copy "c:\users\jst\desktop\The Pri
ncess and the Frog (2009).pt2_t_4127.m4v"
ffmpeg version N-55393-g3b2e99f Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  8 2013 21:37:55 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 24.100 / 55. 24.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000028fa880] stream 0, timescale not set
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'The Princess and the Frog (2009).m4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    creation_time   : 2012-11-18 22:23:57
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    date            : 2009-01-01
  Duration: 02:01:39.36, start: 0.000000, bitrate: 5089 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720,
 4953 kb/s, 25 fps, 25 tbr, 2500 tbn, 5k tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 12
5 kb/s
    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
    Stream #0:2: Video: mjpeg, yuvj420p, 512x748 [SAR 1:1 DAR 128:187], 90k tbr,
 90k tbn, 90k tbc
Output #0, ipod, to 'c:\users\jst\desktop\The Princess and the Frog (2009).pt2_t
_4127.m4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    date            : 2009-01-01
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    encoder         : Lavf55.13.102
    Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 1280x720, q=2-31
, 4953 kb/s, 25 fps, 10k tbn, 2500 tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 125 kb/s

    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 6411 fps=0.0 q=-1.0 size=  162923kB time=00:04:16.64 bitrate=5200.5kbits/
frame=13091 fps=13090 q=-1.0 size=  325306kB time=00:08:44.42 bitrate=5081.6kbit
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frame=41898 fps=5779 q=-1.0 size= 1050695kB time=00:27:56.44 bitrate=5134.3kbits
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frame=43646 fps=5248 q=-1.0 size= 1091937kB time=00:29:06.05 bitrate=5123.1kbits
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frame=47973 fps=5081 q=-1.0 size= 1196073kB time=00:31:59.12 bitrate=5105.6kbits
frame=48427 fps=4828 q=-1.0 size= 1207055kB time=00:32:17.54 bitrate=5103.5kbits
frame=49511 fps=4677 q=-1.0 size= 1234372kB time=00:33:00.54 bitrate=5105.6kbits
frame=50433 fps=4544 q=-1.0 size= 1256989kB time=00:33:37.92 bitrate=5102.9kbits
frame=52197 fps=4475 q=-1.0 size= 1297157kB time=00:34:48.06 bitrate=5089.1kbits
frame=54015 fps=4440 q=-1.0 size= 1336514kB time=00:36:01.28 bitrate=5065.8kbits
frame=55272 fps=4342 q=-1.0 size= 1363888kB time=00:36:51.46 bitrate=5052.3kbits
frame=56111 fps=4242 q=-1.0 size= 1382921kB time=00:37:24.78 bitrate=5046.8kbits
frame=59485 fps=4333 q=-1.0 size= 1471297kB time=00:39:39.90 bitrate=5064.4kbits
frame=60391 fps=4239 q=-1.0 size= 1492583kB time=00:40:16.26 bitrate=5060.4kbits
frame=60433 fps=4069 q=-1.0 size= 1494021kB time=00:40:17.79 bitrate=5062.1kbits
frame=60574 fps=3945 q=-1.0 size= 1496604kB time=00:40:23.89 bitrate=5058.0kbits
frame=60647 fps=3813 q=-1.0 size= 1497990kB time=00:40:26.50 bitrate=5057.3kbits
frame=60732 fps=3682 q=-1.0 size= 1499395kB time=00:40:29.57 bitrate=5055.6kbits
frame=60768 fps=3567 q=-1.0 size= 1500439kB time=00:40:31.10 bitrate=5056.0kbits
frame=60829 fps=3449 q=-1.0 size= 1502157kB time=00:40:33.66 bitrate=5056.4kbits
frame=61840 fps=3404 q=-1.0 size= 1527772kB time=00:41:14.11 bitrate=5058.6kbits
frame=62177 fps=3396 q=-1.0 Lsize= 1535678kB time=00:41:27.00 bitrate=5058.4kbit
s/s
video:1495633kB audio:38193kB subtitle:0 global headers:0kB muxing overhead 0.12
0815%
-------------- next part --------------
?
C:\Users\jst\Downloads\ffmpeg-20130809-git-3b2e99f-win64-static\ffmpeg-20130809-
git-3b2e99f-win64-static\bin>"ffmpeg commercial removal.bat"

C:\Users\jst\Downloads\ffmpeg-20130809-git-3b2e99f-win64-static\ffmpeg-20130809-
git-3b2e99f-win64-static\bin>ffmpeg -ss "00:22:29.500" -i "The Princess and the
Frog (2009).m4v" -to "00:41:27" -c:v copy -c:a copy "c:\users\jst\desktop\The Pr
incess and the Frog (2009).pt2_to_4127.m4v"
ffmpeg version N-55393-g3b2e99f Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug  8 2013 21:37:55 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 24.100 / 55. 24.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000036a880] stream 0, timescale not set
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'The Princess and the Frog (2009).m4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    creation_time   : 2012-11-18 22:23:57
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    date            : 2009-01-01
  Duration: 02:01:39.36, start: 0.000000, bitrate: 5089 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720,
 4953 kb/s, 25 fps, 25 tbr, 2500 tbn, 5k tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 12
5 kb/s
    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
    Stream #0:2: Video: mjpeg, yuvj420p, 512x748 [SAR 1:1 DAR 128:187], 90k tbr,
 90k tbn, 90k tbc
Output #0, ipod, to 'c:\users\jst\desktop\The Princess and the Frog (2009).pt2_t
o_4127.m4v':
  Metadata:
    major_brand     : M4VH
    minor_version   : 1
    compatible_brands: M4VHM4A mp42isom
    date            : 2009-01-01
    title           : Prinsessen og Fr??en
    description     : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    synopsis        : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    comment         : 2009, Animation - (The Princess and the Frog) Amerikansk a
nimationsfilm fra 2009. Den smukke prinsesse Tiana m??der en fr??, som i virkeli
gheden er en prins, der er desperat for at blive forvandlet til et menneske igen
. Et sk??bnesvangert kys tryller ogs?
    media_type      : 0
    network         : Kanal 5 DK
    genre           : TV Shows
    encoder         : Lavf55.13.102
    Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuv420p, 1280x720, q=2-31
, 4953 kb/s, 25 fps, 10k tbn, 2500 tbc
    Metadata:
      creation_time   : 2012-11-18 22:23:58
      handler_name    : Apple Video Media Handler
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 125 kb/s

    Metadata:
      creation_time   : 2012-11-18 22:24:00
      handler_name    : Apple Sound Media Handler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 6261 fps=0.0 q=-1.0 size=  159049kB time=00:04:10.50 bitrate=5201.3kbits/
frame=12977 fps=12976 q=-1.0 size=  322465kB time=00:08:39.81 bitrate=5081.9kbit
frame=19635 fps=13089 q=-1.0 size=  449628kB time=00:13:05.54 bitrate=4688.9kbit
frame=24978 fps=12488 q=-1.0 size=  601562kB time=00:16:39.55 bitrate=4930.2kbit
frame=30527 fps=12210 q=-1.0 size=  761213kB time=00:20:21.76 bitrate=5104.0kbit
frame=36336 fps=12099 q=-1.0 size=  908624kB time=00:24:13.70 bitrate=5120.3kbit
frame=36941 fps=10545 q=-1.0 size=  924700kB time=00:24:38.27 bitrate=5124.3kbit
frame=37496 fps=9362 q=-1.0 size=  940516kB time=00:25:00.29 bitrate=5135.5kbits
frame=38513 fps=8548 q=-1.0 size=  958363kB time=00:25:41.25 bitrate=5093.8kbits
frame=39195 fps=7778 q=-1.0 size=  975467kB time=00:26:08.38 bitrate=5095.1kbits
frame=39957 fps=7176 q=-1.0 size=  997110kB time=00:26:38.59 bitrate=5109.7kbits
frame=40548 fps=6664 q=-1.0 size= 1013557kB time=00:27:02.14 bitrate=5118.6kbits
frame=41361 fps=6282 q=-1.0 size= 1039396kB time=00:27:34.91 bitrate=5145.1kbits
frame=42169 fps=5931 q=-1.0 size= 1057557kB time=00:28:07.17 bitrate=5134.9kbits
frame=43198 fps=5676 q=-1.0 size= 1081768kB time=00:28:48.17 bitrate=5127.9kbits
frame=43867 fps=5406 q=-1.0 size= 1097500kB time=00:29:15.26 bitrate=5122.1kbits
frame=44644 fps=5169 q=-1.0 size= 1115915kB time=00:29:46.50 bitrate=5117.0kbits
frame=45598 fps=4972 q=-1.0 size= 1138442kB time=00:30:24.38 bitrate=5111.9kbits
frame=46618 fps=4814 q=-1.0 size= 1161309kB time=00:31:05.34 bitrate=5100.1kbits
frame=47728 fps=4639 q=-1.0 size= 1189348kB time=00:31:49.89 bitrate=5101.4kbits
frame=48888 fps=4531 q=-1.0 size= 1218503kB time=00:32:36.46 bitrate=5102.1kbits
frame=49611 fps=4393 q=-1.0 size= 1236475kB time=00:33:04.64 bitrate=5103.8kbits
frame=50476 fps=4273 q=-1.0 size= 1257933kB time=00:33:39.97 bitrate=5101.6kbits
frame=51680 fps=4169 q=-1.0 size= 1284564kB time=00:34:27.58 bitrate=5089.6kbits
frame=52331 fps=4058 q=-1.0 size= 1300988kB time=00:34:53.70 bitrate=5090.4kbits
frame=53084 fps=3963 q=-1.0 size= 1317921kB time=00:35:23.90 bitrate=5083.3kbits
frame=53911 fps=3860 q=-1.0 size= 1333700kB time=00:35:56.67 bitrate=5066.0kbits
frame=54732 fps=3780 q=-1.0 size= 1352161kB time=00:36:29.95 bitrate=5058.0kbits
frame=56054 fps=3739 q=-1.0 size= 1381878kB time=00:37:22.69 bitrate=5047.7kbits
frame=57303 fps=3699 q=-1.0 size= 1404381kB time=00:38:12.86 bitrate=5017.6kbits
frame=57906 fps=3609 q=-1.0 size= 1422798kB time=00:38:36.93 bitrate=5030.6kbits
frame=58298 fps=3510 q=-1.0 size= 1435109kB time=00:38:52.31 bitrate=5040.7kbits
frame=58728 fps=3432 q=-1.0 size= 1448444kB time=00:39:09.70 bitrate=5049.9kbits
frame=59184 fps=3359 q=-1.0 size= 1461727kB time=00:39:27.62 bitrate=5057.6kbits
frame=60297 fps=3308 q=-1.0 size= 1489368kB time=00:40:12.16 bitrate=5058.1kbits
frame=61460 fps=3281 q=-1.0 size= 1518076kB time=00:40:58.75 bitrate=5057.9kbits
frame=62177 fps=3252 q=-1.0 Lsize= 1535678kB time=00:41:27.00 bitrate=5058.4kbit
s/s
video:1495633kB audio:38193kB subtitle:0 global headers:0kB muxing overhead 0.12
0815%

From nicolas.george at normalesup.org  Tue Aug 20 12:31:48 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Tue, 20 Aug 2013 12:31:48 +0200
Subject: [FFmpeg-user] Concatenation buffer underflow problem
In-Reply-To: <CABOWChw67g29xukSkkno-QDESBbnrkJBSgqexDUA8kBfdtCWjQ@mail.gmail.com>
References: <CABOWChw67g29xukSkkno-QDESBbnrkJBSgqexDUA8kBfdtCWjQ@mail.gmail.com>
Message-ID: <20130820103148.GC25866@phare.normalesup.org>

Le duodi 2 fructidor, an CCXXI, Dzung Nguyen a ?crit?:
> The problem is that the 2nd command has buffer underflow problem: packet
> too large, ignoring buffer limit. How to fix it?

Your message is mostly useless, as you post nowhere near enough information
to reproduce your problem. Your complete command lines and full console
output are a bare minimum.

As is, I can only reply two generic warnings: first, do not rely on default
values, specify codecs and encoding parameters; second, do not use MPEG-PS,
especially as an intermediary format.

Regards,

-- 
  Nicolas George
-------------- next part --------------
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From werner.robitza at gmail.com  Tue Aug 20 12:38:57 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Tue, 20 Aug 2013 12:38:57 +0200
Subject: [FFmpeg-user] Using ffmpeg to remove commercials from video file
In-Reply-To: <CAKDDL-J07Xus=0D6KKsjnu7dvpNiVVj3Z45m7DppGm6-nODhdw@mail.gmail.com>
References: <1376467770105-4660699.post@n4.nabble.com>
	<20130814170843.5c08405b.dashing.meng@gmail.com>
	<1376474890384-4660701.post@n4.nabble.com>
	<CAPYw7P6hEqCAs9JAydjK2Jr6z8QfGvF-w6Au45L31JKZoikT6A@mail.gmail.com>
	<1376681860279-4660751.post@n4.nabble.com>
	<CAN0LTaMgpJ8sy3L5eLN-hWoWxeLvqkh+nurT-XkLiwdf3en86w@mail.gmail.com>
	<1376936792242-4660782.post@n4.nabble.com>
	<CAN0LTaN-LY6VfWWEySK6Y_=69RL1XQ3NbrK-qotDwybPR3xj7A@mail.gmail.com>
	<1376943309388-4660786.post@n4.nabble.com>
	<CAN0LTaNwyh1TUMxZ5LR6tpDQK66CKPt=i4zaQa6PfbWUPjwLyQ@mail.gmail.com>
	<1376983220384-4660791.post@n4.nabble.com>
	<CAKDDL-LUEf3X_YvZoZTx9sa9Nph7URkKaibBkxBeL5Fzc4upOw@mail.gmail.com>
	<CAN0LTaNX+VFJRgdt53yoHK0jWYkh-DwfkbKAL25tXbSxjnPChg@mail.gmail.com>
	<CAKDDL-J07Xus=0D6KKsjnu7dvpNiVVj3Z45m7DppGm6-nODhdw@mail.gmail.com>
Message-ID: <CAN0LTaPqXvMJML+KDbgLYyuo0_FYmOEZ6NJcJKVcGYSMK-tT5A@mail.gmail.com>

On Tue, Aug 20, 2013 at 12:13 PM, Jesper Lund Stocholm
<4a4553504552 at gmail.com> wrote:
> ffmpeg produces exactly identical files and console output (apart from the
> actual copying of stream that vary a little bit (I assume that is because
> of my machine doing other stuff as well)).
>
> Both files are 41:27 minutes in length.

I reproduced this now.

Here's the problem: When you use -ss before the input file, it skips
to the specified point, and then assumes this as being position
00:00:00 (i.e., the start of the file).
So, these are the same, because for ffmpeg the file starts at 0:

ffmpeg -ss 00:01:00 -i input -t 00:02:00 output
ffmpeg -ss 00:01:00 -i input -to 00:02:00 output

This however works and creates a file with 2 min length and 1 min
length, respectively:

ffmpeg -i input -ss 00:01:00 -t 00:02:00 output
ffmpeg -i input -ss 00:01:00 -to 00:02:00 output

Hope you understand what I mean.

From codecomplete at free.fr  Tue Aug 20 13:35:30 2013
From: codecomplete at free.fr (Gilles)
Date: Tue, 20 Aug 2013 13:35:30 +0200
Subject: [FFmpeg-user] Splitting at key frames?
Message-ID: <slk619t0dt67290v5rt0e38s55qjk0fc6v@4ax.com>

Hello

	I need to split a 67mn FLV video (vp6f + mp3) into two parts (40mn
and the remainder starting 10s before), so used the following
commands:

(2400s = 40mn)
ffmpeg.exe -i input.flv -ss 00:00:00 -t 2400 -c:v copy -c:a copy
output1.1.of.2.flv
ffmpeg.exe -i input.flv -ss 00:39:50 -c:v copy -c:a copy
output1.2.of.2.flv

The reason for starting the second video 10s before is to make the
transition easier to follow by adding some overlap.

It mostly works but I noticed that the second video freezes for a few
seconds at the beginning: Could it be that ffmpeg doesn't split at key
frames?

FWIW...
1. I used the latest Windows ffmpeg available at this time:

ffmpeg version N-55644-g68b63a3 Copyright (c) 2000-2013 the FFmpeg
developers
built on Aug 19 2013 20:27:12 with gcc 4.7.3 (GCC)

2. The whole input file is temporarily available here:
http://dl.free.fr/j57cduLA7

Thank you.


From anithasusan at tataelxsi.co.in  Tue Aug 20 16:04:42 2013
From: anithasusan at tataelxsi.co.in (Anitha Susan Varghese)
Date: Tue, 20 Aug 2013 14:04:42 +0000
Subject: [FFmpeg-user] Problem with playing audio
Message-ID: <25563E537418E342A6DDC83C793113FC100F5404@SIXPRD0410MB359.apcprd04.prod.outlook.com>

Hi

Myself trying to play an audio file using libavcodec in qt5 .While trying to play i am unable to play the the file.
while decoding the file using av_read_frame()
audioStream->index=0
and packet.stream_index is some higher nos .so avcodec_decode_audio4() is not getting executed.Why is it so?

This is my program. My output is also given below.Please help me in solving this.
-------------------------------------------------------------------------------------------------------------------------------------
#include <QApplication>
#include <QtDebug>
#include <ao/ao.h>

extern "C" {
#include "libavutil/samplefmt.h"
#include "libavcodec/avcodec.h"
#include "libavutil/mathematics.h"
#include "libavutil/avutil.h"
#include "libavformat/avformat.h"
#include "libswscale/swscale.h"
#include "libavutil/dict.h"
#include <ao/ao.h>
#include "libavformat/version.h"
}
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
#define CODEC_TYPE_VIDEO AVMEDIA_TYPE_VIDEO



void die( const char* msg)
{
qDebug() <<  msg ;
}

int main(int argc, char *argv[])
{
    QApplication app(argc, argv);
    const char* input_filename=argv[1];

    static int driver;
    static ao_device *audio_device;
    int i;
    int stream_id=-1;
    av_register_all();
    avcodec_register_all();
    AVFormatContext* container=NULL;
    AVStream* audioStream=NULL;
       //container=avformat_alloc_context();

     if(avformat_open_input(&container,input_filename,NULL,NULL)<0){
                die("Could not open file");
        }
       else
       die("File Opend");


    if(avformat_find_stream_info(container,NULL)<0){
                die("Could not find file info");
        }


    av_dump_format(container,0,input_filename,false);
    for(i=0; i < container->nb_streams; i++){

        if(container->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO){
                stream_id=i;
                audioStream=container->streams[i];

                qWarning() <<"audioStream"<<audioStream;
                qWarning() <<"Stream id:"<<stream_id;
                 break;
                 }
        }

    if(stream_id==-1){
        die("Could not find Audio Stream");
    }


    AVCodecContext *ctx=container->streams[stream_id]->codec;
    qWarning() <<"Codec ptr:"<< ctx;
    AVCodec *codec=avcodec_find_decoder(ctx->codec_id);
 if(codec==NULL){
        die("cannot find codec!");
   }

  if(avcodec_open2(ctx,codec,NULL)<0){
        die("Codec cannot be opended!");
   }
   else
        qWarning()<<"Codec Opened";


    ao_initialize();
    driver = ao_default_driver_id();
    qWarning()<<"DRIVER ID" <<driver;
   ao_sample_format sformat;
   AVPacket dummy_packet;
   av_read_frame(container,&dummy_packet);
   AVSampleFormat sfmt=ctx->sample_fmt;
//assign device sample rate depend on the input stream

   qWarning()<<"Sample format"<<sfmt;

   if(sfmt==AV_SAMPLE_FMT_U8){
        sformat.bits=8;
        qWarning()<<"Sample format"<<sformat.bits;
   }
   else if(sfmt==AV_SAMPLE_FMT_S16){
        sformat.bits=16;
        qWarning()<<"Sample format"<<sformat.bits;
  }
  else if(sfmt==AV_SAMPLE_FMT_S32){
        sformat.bits=32;
        qWarning()<<"Sample format:AV_SAMPLE_FMT_S32"<<sformat.bits;
  }
else if(sfmt==AV_SAMPLE_FMT_FLT){
        sformat.bits=32;
        qWarning()<<"Sample format:AV_SAMPLE_FMT_FLT"<<sformat.bits;
 else if(sfmt==AV_SAMPLE_FMT_DBL){
        sformat.bits=64;
        qWarning()<<"Sample format"<<sformat.bits;
  }
  else if(sfmt==AV_SAMPLE_FMT_U8P){
        sformat.bits=8;
        qWarning()<<"Sample format"<<sformat.bits;
  }
  else if(sfmt==AV_SAMPLE_FMT_S16P){
        sformat.bits=16;
        qWarning()<<"Sample format"<<sformat.bits;
  }

   else if(sfmt==AV_SAMPLE_FMT_S32P){
        sformat.bits=32;
        qWarning()<<"Sample format:AV_SAMPLE_FMT_S32P"<<sformat.bits;
  }
  else if(sfmt==AV_SAMPLE_FMT_FLTP){
        sformat.bits=32;
        qWarning()<<"Sample format:AV_SAMPLE_FMT_FLTP"<<sformat.bits;
  }
 else if(sfmt==AV_SAMPLE_FMT_DBLP){
        sformat.bits=64;
        qWarning()<<"Sample format"<<sformat.bits;
  }


   sformat.channels=ctx->channels;

  //sformat.channels=2;
  qWarning()<<"Sample format channels"<<sformat.channels;

  sformat.rate=ctx->sample_rate;
  qWarning()<<"Sample format rate"<<sformat.rate;

  sformat.byte_format=AO_FMT_NATIVE;
  qWarning()<<"Sample format "<<sformat.byte_format;
 sformat.matrix=0;

 //seek back to the beginning of stream
  av_seek_frame(container,stream_id,0,AVSEEK_FLAG_ANY);

  audio_device=ao_open_live(driver,&sformat,NULL);
//end of init AO LIB

//data packet read from the stream
  AVPacket packet;
  av_init_packet(&packet);

  int buffer_size=AVCODEC_MAX_AUDIO_FRAME_SIZE+ FF_INPUT_BUFFER_PADDING_SIZE;;

  qWarning()<<"buffersize "<<buffer_size;
  uint8_t buffer[buffer_size];
  packet.data=buffer;
  packet.size =buffer_size;

//frame ,where the decoded data will be written
  AVFrame *frame=avcodec_alloc_frame();
 int len;
  int frameFinished=0;

  qWarning()<<"CODEC"<<audioStream->codec;
  while(av_read_frame(container,&packet)>=0)
  {
//        qWarning()<<audioStream->index;
        if(packet.stream_index==audioStream->index){
        len=avcodec_decode_audio4(ctx,frame,&frameFinished,&packet);
        qWarning()<<"Enterd if loop";
      //  qWarning() <<len;
        if(frameFinished){
//play the decoded byteis
                qWarning()<<"PLAYING";
                ao_play(audio_device, (char*)frame->extended_data[0],frame->linesize[0] );
  }else{
}
}
 avformat_close_input(&container);
  ao_shutdown();
  return app.exec();
}
----------------------------------------------------------------------------------------------------------------------------------
My output
--------------------------------------------------------------------------------------------------------------------                     File Opend
Input #0, ogg, from '/home/tel/Downloads/desktop-login.ogg':
  Duration: 00:00:07.72, start: 0.000000, bitrate: 108 kb/s
    Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp, 112 kb/s
audioStream 0x88b3f20
Stream id: 0
Codec ptr: 0x88b4100
Codec Opened
DRIVER ID 1
Sample format 3
Sample format:AV_SAMPLE_FMT_FLT 32
Sample format channels 0
Sample format rate 0
Sample format  4
buffersize  192008
CODEC 0x88b4100


--------------------------------------------------------------------------------------------------------------
Regards,
Anitha Susan Varghese


From b_bruggeman at ziggo.nl  Tue Aug 20 16:44:48 2013
From: b_bruggeman at ziggo.nl (Bert Bruggeman)
Date: Tue, 20 Aug 2013 16:44:48 +0200
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
In-Reply-To: <5212A74B.1050300@gmail.com>
References: <1376816843722-4660765.post@n4.nabble.com>
	<1376909207070-4660771.post@n4.nabble.com>
	<5212A74B.1050300@gmail.com>
Message-ID: <D816E5E9F3B447F6A26ACEAD0736340A@BertTHUIS>

if its not the alpha, it?s the subsampling used for yuv420
may be you could try yuv444p? that is 24 bits/pixel just as rgb-> kind of 
losless.

-----Original Message----- 
From: Andy Furniss
Sent: Tuesday, August 20, 2013 1:16 AM
To: FFmpeg user questions
Cc: bert1974
Subject: Re: [FFmpeg-user] How can I convert rbg to yuv420p loseless?

bert1974 wrote:
> PIX_FMT_BGRA contains alpha(=transparency) information.. to get the
> transparent edges right, you will have to blend/draw on a background color
> when converting from bgra to rgb. don't know how to do with SwsContext,
> maybe convert/blend rgba->rgb manual, than convert rgb to yuv using
> SwsContext
>
> (it looks like the image in your mail is blended on black instead of 
> white)

I don't think this is it as the alpha seems to be 255 everywhere on the
png and the artifacts seem to arise on conversion back to rgb.

Testing on 420 produced from the png with no sws options.

Converting back to rgb with ffmpeg -

ffmpeg -s 890x768 -pix_fmt yuv420p -i 420.yuv -sws_flags
+accurate_rnd+full_chroma_int -pix_fmt rgb24 420-ar-fc.rgb

is the best looking.

-sws_flags +accurate_rnd still has noticable "black" (really grey)
pixels but is better than no sws options or
-sws_flags +full_chroma_int which produces exactly the same output
(md5sum) as no sws options.


From jhbell50 at hotmail.com  Tue Aug 20 17:28:33 2013
From: jhbell50 at hotmail.com (john Bell)
Date: Tue, 20 Aug 2013 11:28:33 -0400
Subject: [FFmpeg-user] Command error
Message-ID: <BLU178-W190042E3B46D1337398772CF430@phx.gbl>

On Fri, 16 Aug 2013 09:44:15 -0400
john Bell <jhbell50 at hotmail.com> wrote:

"> ffmpeg.exe -cookies "gandbolsteauaaakeys=ZCV4upgVjq5_RjK9yPYNUQ; gandbolsteauaaakeyt=1376674191; path=\; domain=37.221.172.85; \n" -i "http://37.221.172.85/hqstream/gandbolsteauaaa/playlist.m3u8?keys=ZCV4upgVjq5_RjK9yPYNUQ&keyt=1376674191" 
>  
> fails with
>  
> [http @ 03ddac00] Invalid cookie found, no value, path or domain specified
>  
> but I belive the command conforms to the code for verifying the value, path and domain.
>  
> Any suggestions?


You need to include the complete ffmpeg console output and not just a segment."
 
========================================================
Here is the complete output from a current command:
 
C:\Users\John>"C:\Program Files\Serviio\lib\ffmpeg.exe" -cookies "shakhtyor23523
keys=lcHUmfJk15tjkDszLgWO_w; shakhtyor23523keyt=1377019659; path=\; domain=50.22
.223.47; \n" -i "http://50.22.223.47/hqstream/shakhtyor23523/playlist.m3u8?keys=
lcHUmfJk15tjkDszLgWO_w&keyt=1377019659"
ffmpeg version N-54096-ge41bf19 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jun 18 2013 10:49:28 with gcc 4.7.2 (GCC)
  configuration: --enable-static --disable-shared --disable-ffplay --disable-ffs
erver --enable-memalign-hack --enable-libmp3lame --enable-libass --enable-librtm
p --enable-fontconfig --enable-libfreetype --enable-zlib --enable-libx264 --extr
a-libs='-lrtmp -lpolarssl -lws2_32 -lwinmm -lexpat -lfreetype -lfribidi -lz' --a
rch=x86 --enable-runtime-cpudetect --enable-pthreads --target-os=mingw32 --cross
-prefix=i686-w64-mingw32- --enable-gpl --pkg-config=pkg-config
  libavutil      52. 37.101 / 52. 37.101
  libavcodec     55. 16.100 / 55. 16.100
  libavformat    55.  9.100 / 55.  9.100
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 77.101 /  3. 77.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[http @ 018827C0] Invalid cookie found, no value, path or domain specified
    Last message repeated 1 times
http://50.22.223.47/hqstream/shakhtyor23523/playlist.m3u8?keys=lcHUmfJk15tjkDszL
gWO_w&keyt=1377019659: Invalid data found when processing input


 		 	   		  

From mr_ax at hotmail.com  Tue Aug 20 09:03:01 2013
From: mr_ax at hotmail.com (Mr.Hackman 191)
Date: Tue, 20 Aug 2013 14:03:01 +0700
Subject: [FFmpeg-user] i try about 2-3 day and about 1 week seach in google
	not working
Message-ID: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>

make distclean LT_INIT: command not found

when install 

libfdk_aac

i try about 2-3 day and about 1 week seach in google not working 



in cent os6.4


how to fix?


 		 	   		  
From mr_ax at hotmail.com  Tue Aug 20 10:40:35 2013
From: mr_ax at hotmail.com (Mr.Hackman 191)
Date: Tue, 20 Aug 2013 15:40:35 +0700
Subject: [FFmpeg-user] ffmpeg 2 yum install
In-Reply-To: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>
References: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>
Message-ID: <SNT145-W57A6F73DE438274C2564FA96430@phx.gbl>

ffmpeg 2 yum install

how?
 		 	   		  
From vvubin at gmail.com  Tue Aug 20 21:07:38 2013
From: vvubin at gmail.com (Wu-bin Zhen)
Date: Tue, 20 Aug 2013 15:07:38 -0400
Subject: [FFmpeg-user] Buffer Size for Stream Copy
In-Reply-To: <CAGK10ZNKQPOjRjC=RP2h0isuudyUVQe1w2Jb50mSoy1dU2yCFg@mail.gmail.com>
References: <CAGK10ZNKQPOjRjC=RP2h0isuudyUVQe1w2Jb50mSoy1dU2yCFg@mail.gmail.com>
Message-ID: <CAGK10ZOMJ7KuvAx1A+p99VaJ=aqqzw982BuEJ-EdSPPXusLS3g@mail.gmail.com>

On Fri, Aug 16, 2013 at 3:31 PM, Wu-bin Zhen <vvubin at gmail.com> wrote:

> Hello ffmpeg,
>
> I am using ffmpeg to grab h264 video stream and aac audio stream from an
> internet based hardware encoder with the codec copy option, and it works
> pretty well. Now, I start to wonder if I can use the '-bufsize' option for
> the stream copy, just in case when the network is temporally slow down, or
> the encoder is suddenly chocking for some reason. I googled a little bit
> about the 'bufsize' option, and found that most people use the option for
> encoding.
>
> So my question is: Do you think using 'bufsize' option will be helpful for
> the stream copy? If so, what would be the recommended buffer size? My
> internet encoder produces video stream with 4000kbits/sec, and audio stream
> with 128kbits/sec.
>
> Thanks, and have a great weekend.
> -w
>

Sorry if I didn't explain clearly. So ffmpeg works fine when I specify the
option '-bufsize', but later I realized that it actually doesn't care the
value of the bufsize option. The ffmpeg command below works fine without
complaining anything about the bufsize value:

> ffmpeg -i http://192.168.1.224:1881/stream.ts -bufsize helloworld -vcodec
copy -acodec copy buffer_test.ts

So I start to wonder if ffmpeg doesn't do anything about '-bufsize' for the
stream copy.
Please advise. Thanks
-w

From alacki93 at gmail.com  Tue Aug 20 21:19:01 2013
From: alacki93 at gmail.com (=?UTF-8?B?QXJ0dXIgxYHEhWNraQ==?=)
Date: Tue, 20 Aug 2013 21:19:01 +0200
Subject: [FFmpeg-user] Delayed sound in screencast
Message-ID: <5213C125.6060706@gmail.com>

Hi

I'm trying do screencast with ffmpeg but I've problem with delayed sound
(audio is 1-2 seconds after video). I've tried different tricks with
-async, -vsync, -threads etc, but it didn't helped.

Here are log from console and short (~2MB) sample video with problem.
https://dl.dropboxusercontent.com/u/30413262/ffmpeg-async-problem.7z

Thx for help.


From adf.lists at gmail.com  Wed Aug 21 00:04:49 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Tue, 20 Aug 2013 23:04:49 +0100
Subject: [FFmpeg-user] How can I convert rbg to yuv420p loseless?
In-Reply-To: <1376968307925-4660789.post@n4.nabble.com>
References: <1376816843722-4660765.post@n4.nabble.com>
	<52126F13.8010204@gmail.com>
	<1376968307925-4660789.post@n4.nabble.com>
Message-ID: <5213E801.4020104@gmail.com>

mokacao wrote:
> I generated a test that contains a variety of colors pictures,
> red/green have black dot.
>
> Here is part of the core code:

It's the same using ffmpeg directly, but there's probably nothing wrong
with the rgb -> yuv420p and using the flags below for this conversion
does not affect the outcome.

A more interesting test would be to find something else that can make
yuv420p from rgb.

> original png file
> <http://ffmpeg-users.933282.n4.nabble.com/file/n4660789/P0001.png>
> yuv420p zoom in 4x, use the software yuv
> viewer(http://sourceforge.net/projects/yuviewer/), and screenshot

So yuviewer does the same yuv -> rgb scale as ffmpeg in default mode.

> <http://ffmpeg-users.933282.n4.nabble.com/file/n4660789/P0001_yuv_zoom_in_4x.png>
>
>
 > original png file to yuv420p use above code
> P0001.yuv
> <http://ffmpeg-users.933282.n4.nabble.com/file/n4660789/P0001.yuv>

Playing your .yuv looks fine with ffplay and mplayer + three different
ways of using my gpu to scale/csc.

It's likely that anyone playing your yuv with a normal player that will
choose h/w to scale/csc won't see the issue.

Due to the height being >=720 some code may assume it's 709 rather than
601 colourspace, but that's nothing to do with this issue.

ffmpeg -s 1024x768 -pix_fmt yuv420p -i P0001.yuv P0001-noopt.png

Will show the artifacts you see

ffmpeg -s 1024x768 -pix_fmt yuv420p -i P0001.yuv -sws_flags
+accurate_rnd+full_chroma_int P0001-arfc.png

Doesn't produce them, so it looks like the yuv its self is sound, but
the accuracy of yuviewer/ffmpeg default is the issue when converting
back to rgb.


From onemda at gmail.com  Wed Aug 21 01:31:42 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 20 Aug 2013 23:31:42 +0000
Subject: [FFmpeg-user] Buffer Size for Stream Copy
In-Reply-To: <CAGK10ZOMJ7KuvAx1A+p99VaJ=aqqzw982BuEJ-EdSPPXusLS3g@mail.gmail.com>
References: <CAGK10ZNKQPOjRjC=RP2h0isuudyUVQe1w2Jb50mSoy1dU2yCFg@mail.gmail.com>
	<CAGK10ZOMJ7KuvAx1A+p99VaJ=aqqzw982BuEJ-EdSPPXusLS3g@mail.gmail.com>
Message-ID: <CAPYw7P6Ns4XqagdVnbn3ezX-YjiL8t_hCDm=90ybfW+B1bxQsw@mail.gmail.com>

On 8/20/13, Wu-bin Zhen <vvubin at gmail.com> wrote:
> On Fri, Aug 16, 2013 at 3:31 PM, Wu-bin Zhen <vvubin at gmail.com> wrote:
>
>> Hello ffmpeg,
>>
>> I am using ffmpeg to grab h264 video stream and aac audio stream from an
>> internet based hardware encoder with the codec copy option, and it works
>> pretty well. Now, I start to wonder if I can use the '-bufsize' option
>> for
>> the stream copy, just in case when the network is temporally slow down,
>> or
>> the encoder is suddenly chocking for some reason. I googled a little bit
>> about the 'bufsize' option, and found that most people use the option for
>> encoding.
>>
>> So my question is: Do you think using 'bufsize' option will be helpful
>> for
>> the stream copy? If so, what would be the recommended buffer size? My
>> internet encoder produces video stream with 4000kbits/sec, and audio
>> stream
>> with 128kbits/sec.
>>
>> Thanks, and have a great weekend.
>> -w
>>
>
> Sorry if I didn't explain clearly. So ffmpeg works fine when I specify the
> option '-bufsize', but later I realized that it actually doesn't care the
> value of the bufsize option. The ffmpeg command below works fine without
> complaining anything about the bufsize value:
>
>> ffmpeg -i http://192.168.1.224:1881/stream.ts -bufsize helloworld -vcodec
> copy -acodec copy buffer_test.ts
>
> So I start to wonder if ffmpeg doesn't do anything about '-bufsize' for the
> stream copy.
> Please advise. Thanks

As explainded in documentation and by ffmpeg itself, 'bufsize' is used
for audio/video encoding only.

-bufsize           <int>        E..VA. set ratecontrol buffer size (in
bits) (from INT_MIN to INT_MAX) (default 0

From dashing.meng at gmail.com  Wed Aug 21 02:39:59 2013
From: dashing.meng at gmail.com (littlebat)
Date: Wed, 21 Aug 2013 08:39:59 +0800
Subject: [FFmpeg-user] i try about 2-3 day and about 1 week seach in
 google not working
In-Reply-To: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>
References: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>
Message-ID: <20130821083959.39586578.dashing.meng@gmail.com>

On Tue, 20 Aug 2013 14:03:01 +0700
Mr.Hackman 191 <mr_ax at hotmail.com> wrote:

> make distclean LT_INIT: command not found
> 
> when install 
> 
> libfdk_aac
> 
> i try about 2-3 day and about 1 week seach in google not working 
> 
> 
> 
> in cent os6.4
> 
> 
> how to fix?
> 

You need to include the uncutted complete ffmpeg console output.

From dashing.meng at gmail.com  Wed Aug 21 02:43:40 2013
From: dashing.meng at gmail.com (littlebat)
Date: Wed, 21 Aug 2013 08:43:40 +0800
Subject: [FFmpeg-user] i try about 2-3 day and about 1 week seach in
 google not working
In-Reply-To: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>
References: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>
Message-ID: <20130821084340.2a9817d2.dashing.meng@gmail.com>

On Tue, 20 Aug 2013 14:03:01 +0700
Mr.Hackman 191 <mr_ax at hotmail.com> wrote:

> make distclean LT_INIT: command not found
> 
> when install 
> 
> libfdk_aac
> 
> i try about 2-3 day and about 1 week seach in google not working 
> 
> 
> 
> in cent os6.4
> 
> 
> how to fix?
> 

You need to include the uncutted complete command console output.

From cehoyos at ag.or.at  Wed Aug 21 12:10:53 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 21 Aug 2013 10:10:53 +0000 (UTC)
Subject: [FFmpeg-user] Delayed sound in screencast
References: <5213C125.6060706@gmail.com>
Message-ID: <loom.20130821T120936-514@post.gmane.org>

Artur ??cki <alacki93 <at> gmail.com> writes:

> I'm trying do screencast with ffmpeg but I've problem 
> with delayed sound (audio is 1-2 seconds after video). 
> I've tried different tricks with -async, -vsync, 
> -threads etc, but it didn't helped.

Please post your command line together with the complete, 
uncut console output here on the mailing list, external 
resources tend to disappear but the information may still 
be relevant after years.

Carl Eugen


From cehoyos at ag.or.at  Wed Aug 21 12:12:15 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 21 Aug 2013 10:12:15 +0000 (UTC)
Subject: [FFmpeg-user] MPEG1/2 does not support 3/1 fps error
References: <CABRdPaKjcVRDV8Hwd=PYQk_Pi+E4xurP6FMjV5BK0aomXiDu-A@mail.gmail.com>
Message-ID: <loom.20130821T121125-127@post.gmane.org>

Chaitanya Gopathi <chaitanyag <at> gopalsystems.com> writes:

> [mpeg1video  <at>  0x1999afe0]MPEG1/2 does not support 3/1 fps

> FFmpeg version 0.6.5

This is severely outdated (and was always buggy), please 
test current git head.

Carl Eugen


From alacki93 at gmail.com  Wed Aug 21 13:17:10 2013
From: alacki93 at gmail.com (=?UTF-8?B?QXJ0dXIgxYHEhWNraQ==?=)
Date: Wed, 21 Aug 2013 13:17:10 +0200
Subject: [FFmpeg-user] Delayed sound in screencast
In-Reply-To: <loom.20130821T120936-514@post.gmane.org>
References: <5213C125.6060706@gmail.com>
	<loom.20130821T120936-514@post.gmane.org>
Message-ID: <5214A1B6.5090807@gmail.com>

W dniu 21.08.2013 12:10, Carl Eugen Hoyos pisze:
> Artur ??cki <alacki93 <at> gmail.com> writes:
>
>> I'm trying do screencast with ffmpeg but I've problem 
>> with delayed sound (audio is 1-2 seconds after video). 
>> I've tried different tricks with -async, -vsync, 
>> -threads etc, but it didn't helped.
> Please post your command line together with the complete, 
> uncut console output here on the mailing list, external 
> resources tend to disappear but the information may still 
> be relevant after years.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
OK. Here is full log from console.

artur at laptop:~/Wideo> ffmpeg -f pulse -ac 2 -i default -f x11grab -r 30
-s 1366x768 -i :0.0 -vcodec libx264 -acodec libvorbis -y test.mkv
ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug 17 2013 07:01:31 with gcc 4.7 (SUSE Linux)
  configuration: --shlibdir=/usr/lib64 --prefix=/usr
--mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared
--disable-static --enable-debug --disable-stripping
--extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2
-fstack-protector -funwind-tables -fasynchronous-unwind-tables -fPIC
-I/usr/include/gsm' --enable-gpl --enable-x11grab --enable-version3
--enable-pthreads --enable-avfilter --enable-libpulse --enable-libvpx
--enable-libopus --enable-libass --enable-libmp3lame --enable-libvorbis
--enable-libtheora --enable-libspeex --enable-libxvid --enable-libx264
--enable-libschroedinger --enable-libgsm --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-postproc --enable-libdc1394
--enable-librtmp --enable-libfreetype --enable-avresample
--enable-libtwolame --enable-libvo-aacenc
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 79.101 /  3. 79.101
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, pulse, from 'default':
  Duration: N/A, start: 0.019300, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
[x11grab @ 0x1a69760] device: :0.0 -> display: :0.0 x: 0 y: 0 width:
1366 height: 768
[x11grab @ 0x1a69760] shared memory extension found
Input #1, x11grab, from ':0.0':
  Duration: N/A, start: 1377024472.739504, bitrate: 1007124 kb/s
    Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768,
1007124 kb/s, 30 tbr, 1000k tbn, 30 tbc
No pixel format specified, yuv444p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x1a87fc0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x1a87fc0] profile High 4:4:4 Predictive, level 3.2, 4:4:4 8-bit
[libx264 @ 0x1a87fc0] 264 - core 135 - H.264/MPEG-4 AVC codec - Copyleft
2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3
deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0
deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=6
lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1
b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250
keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf
mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40
aq=1:1.00
Output #0, matroska, to 'test.mkv':
  Metadata:
    encoder         : Lavf55.12.100
    Stream #0:0: Video: h264 (libx264) (H264 / 0x34363248), yuv444p,
1366x768, q=-1--1, 1k tbn, 30 tbc
    Stream #0:1: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 48000
Hz, stereo, fltp
Stream mapping:
  Stream #1:0 -> #0:0 (rawvideo -> libx264)
  Stream #0:0 -> #0:1 (pcm_s16le -> libvorbis)
Press [q] to stop, [?] for help
[swscaler @ 0x1a4a880] Warning: data is not aligned! This can lead to a
speedloss
frame= 1505 fps= 27 q=-1.0 Lsize=    2288kB time=00:00:50.10 bitrate=
374.2kbits/s   
video:2116kB audio:141kB subtitle:0 global headers:4kB muxing overhead
1.207635%
[libx264 @ 0x1a87fc0] frame I:7     Avg QP:18.82  size: 88676
[libx264 @ 0x1a87fc0] frame P:423   Avg QP:21.29  size:  3028
[libx264 @ 0x1a87fc0] frame B:1075  Avg QP:32.81  size:   246
[libx264 @ 0x1a87fc0] consecutive B-frames:  4.3%  0.9%  1.2% 93.6%
[libx264 @ 0x1a87fc0] mb I  I16..4: 29.8% 39.1% 31.1%
[libx264 @ 0x1a87fc0] mb P  I16..4:  1.2%  1.2%  0.7%  P16..4:  1.7% 
0.6%  0.3%  0.0%  0.0%    skip:94.2%
[libx264 @ 0x1a87fc0] mb B  I16..4:  0.0%  0.1%  0.0%  B16..8:  1.7% 
0.1%  0.0%  direct: 0.1%  skip:98.0%  L0:46.6% L1:49.8% BI: 3.6%
[libx264 @ 0x1a87fc0] 8x8 transform intra:41.0% inter:60.7%
[libx264 @ 0x1a87fc0] coded y,u,v intra: 23.0% 6.6% 6.3% inter: 0.3%
0.1% 0.1%
[libx264 @ 0x1a87fc0] i16 v,h,dc,p: 36% 60%  1%  3%
[libx264 @ 0x1a87fc0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 38% 11% 43%  1% 
1%  1%  1%  2%  2%
[libx264 @ 0x1a87fc0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 32% 32% 14%  3% 
3%  4%  5%  3%  4%
[libx264 @ 0x1a87fc0] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x1a87fc0] ref P L0: 65.3%  8.8% 18.4%  7.6%
[libx264 @ 0x1a87fc0] ref B L0: 77.0% 20.4%  2.6%
[libx264 @ 0x1a87fc0] ref B L1: 95.6%  4.4%
[libx264 @ 0x1a87fc0] kb/s:345.41


From bradleycbotkin at gmail.com  Wed Aug 21 13:49:35 2013
From: bradleycbotkin at gmail.com (Botkin Brad)
Date: Wed, 21 Aug 2013 07:49:35 -0400
Subject: [FFmpeg-user] multiple programs (ServiceXX) in a single TS?
Message-ID: <29718736-5621-4BA0-8316-B84F3C7E0F46@gmail.com>

Is there any way to coerce ffmpeg to copy existing PMTs from input streams into the output stream? I know I can move specific metadata from inputs to output, but I need to have two complete Program Map Tables.  I have two ATSC TS streams produced thusly:

CONVERT FLVs ---> ATSC TS, using Program #4:
./ffmpeg -i foo1.flv \
		-c:a aac -strict -2 -b:a 128k \
		-c:v mpeg2video -r 30000/1001 \
		-mpegts_service_id 3 -muxrate 19392636 -f mpegts -y \
		foo1.ts



./ffmpeg -i foo2.flv \
		-c:a aac -strict -2 -b:a 128k \
		-c:v mpeg2video -r 30000/1001 \
		-mpegts_service_id 4 -muxrate 19392636 -f mpegts -y \
		foo2.ts

Admittedly, I needn't have created the full 19.38 Mb/s framerate for each of them, but that was part of an earlier test just to make sure each TS would play fine through a system. And they do.

What I'd like it so merge them into a single TS that preserves both PMTs from the input files.  I don't seem to be able to reference the ServiceXX stream like I can the video and audio streams in the input files.

Is this possible with ffmpeg?
Thanks.
--Brad

From cehoyos at ag.or.at  Wed Aug 21 13:53:40 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 21 Aug 2013 11:53:40 +0000 (UTC)
Subject: [FFmpeg-user] Delayed sound in screencast
References: <5213C125.6060706@gmail.com>
	<loom.20130821T120936-514@post.gmane.org>
	<5214A1B6.5090807@gmail.com>
Message-ID: <loom.20130821T135044-175@post.gmane.org>

Artur ??cki <alacki93 <at> gmail.com> writes:

> ffmpeg -f pulse -ac 2 -i default -f x11grab -r 30
> -s 1366x768 -i :0.0 -vcodec libx264 -acodec libvorbis 
> -y test.mkv

> ffmpeg version 2.0.1

Unrelated: If you are a user, please test git head 
before posting here.

> Input #0, pulse, from 'default':
>   Duration: N/A, start: 0.019300, bitrate: 1536 kb/s
                          ^^^^^^^^

> Input #1, x11grab, from ':0.0':
>   Duration: N/A, start: 1377024472.739504, bitrate: 1007124 kb/s
                          ^^^^^^^^^^^^^^^^^

This is the reason for desync.
Can you try alsa?
You can also test -use_wallclock_as_timestamps, I don't know 
if it helps.

Carl Eugen


From alacki93 at gmail.com  Wed Aug 21 14:59:19 2013
From: alacki93 at gmail.com (=?UTF-8?B?QXJ0dXIgxYHEhWNraQ==?=)
Date: Wed, 21 Aug 2013 14:59:19 +0200
Subject: [FFmpeg-user] Delayed sound in screencast
In-Reply-To: <loom.20130821T135044-175@post.gmane.org>
References: <5213C125.6060706@gmail.com>
	<loom.20130821T120936-514@post.gmane.org>
	<5214A1B6.5090807@gmail.com>
	<loom.20130821T135044-175@post.gmane.org>
Message-ID: <5214B9A7.8010102@gmail.com>

W dniu 21.08.2013 13:53, Carl Eugen Hoyos pisze:
> Artur ??cki <alacki93 <at> gmail.com> writes:
>
>> ffmpeg -f pulse -ac 2 -i default -f x11grab -r 30
>> -s 1366x768 -i :0.0 -vcodec libx264 -acodec libvorbis 
>> -y test.mkv
>> ffmpeg version 2.0.1
> Unrelated: If you are a user, please test git head 
> before posting here.
>
>> Input #0, pulse, from 'default':
>>   Duration: N/A, start: 0.019300, bitrate: 1536 kb/s
>                           ^^^^^^^^
>
>> Input #1, x11grab, from ':0.0':
>>   Duration: N/A, start: 1377024472.739504, bitrate: 1007124 kb/s
>                           ^^^^^^^^^^^^^^^^^
>
> This is the reason for desync.
> Can you try alsa?
> You can also test -use_wallclock_as_timestamps, I don't know 
> if it helps.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Now delay is smaller but sounds is "slow".

artur at laptop:~/Wideo> ffmpeg -f alsa -ac 2 -i default -f x11grab -r 30
-s 1366x768 -i :0.0 -vcodec libx264 -acodec libvorbis -y
-use_wallclock_as_timestamps 1 test.mkv
ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug 17 2013 07:01:31 with gcc 4.7 (SUSE Linux)
  configuration: --shlibdir=/usr/lib64 --prefix=/usr
--mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared
--disable-static --enable-debug --disable-stripping
--extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2
-fstack-protector -funwind-tables -fasynchronous-unwind-tables -fPIC
-I/usr/include/gsm' --enable-gpl --enable-x11grab --enable-version3
--enable-pthreads --enable-avfilter --enable-libpulse --enable-libvpx
--enable-libopus --enable-libass --enable-libmp3lame --enable-libvorbis
--enable-libtheora --enable-libspeex --enable-libxvid --enable-libx264
--enable-libschroedinger --enable-libgsm --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-postproc --enable-libdc1394
--enable-librtmp --enable-libfreetype --enable-avresample
--enable-libtwolame --enable-libvo-aacenc
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 79.101 /  3. 79.101
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, alsa, from 'default':
  Duration: N/A, start: 1377089711.958044, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
[x11grab @ 0x104e2a0] device: :0.0 -> display: :0.0 x: 0 y: 0 width:
1366 height: 768
[x11grab @ 0x104e2a0] shared memory extension found
Input #1, x11grab, from ':0.0':
  Duration: N/A, start: 1377089712.038361, bitrate: 1007124 kb/s
    Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768,
1007124 kb/s, 30 tbr, 1000k tbn, 30 tbc
No pixel format specified, yuv444p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x106b700] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x106b700] profile High 4:4:4 Predictive, level 3.2, 4:4:4 8-bit
[libx264 @ 0x106b700] 264 - core 135 - H.264/MPEG-4 AVC codec - Copyleft
2003-2013 - http://www.videolan.org/x264.html - options: cabac=1 ref=3
deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0
deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=6
lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0
bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1
b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250
keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf
mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40
aq=1:1.00
Output #0, matroska, to 'test.mkv':
  Metadata:
    encoder         : Lavf55.12.100
    Stream #0:0: Video: h264 (libx264) (H264 / 0x34363248), yuv444p,
1366x768, q=-1--1, 1k tbn, 30 tbc
    Stream #0:1: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 48000
Hz, stereo, fltp
Stream mapping:
  Stream #1:0 -> #0:0 (rawvideo -> libx264)
  Stream #0:0 -> #0:1 (pcm_s16le -> libvorbis)
Press [q] to stop, [?] for help
[swscaler @ 0x102cac0] Warning: data is not aligned! This can lead to a
speedloss
Queue input is backward in time=     269kB time=00:00:01.60
bitrate=1377.7kbits/s   
[libvorbis @ 0x106f800] Queue input is backward in
time                                                                 
    Last message repeated 24 times
Queue input is backward in time=    1548kB time=00:00:38.54 bitrate=
329.0kbits/s   
[libvorbis @ 0x106f800] Queue input is backward in
time                                                                 
    Last message repeated 55 times
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38612; changing to 38626. This may result in
incorrect timestamps in the output
file.                                                                  
[libvorbis @ 0x106f800] Queue input is backward in
time                                                                 
    Last message repeated 15 times
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38582; changing to 38626. This may result in
incorrect timestamps in the output file.
[libvorbis @ 0x106f800] Queue input is backward in time
    Last message repeated 10 times
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38561; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38547; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38539; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38539; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38544; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38553; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38567; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38584; changing to 38626. This may result in
incorrect timestamps in the output file.
[matroska @ 0x106bd80] Non-monotonous DTS in output stream 0:1;
previous: 38626, current: 38603; changing to 38626. This may result in
incorrect timestamps in the output file.
frame= 1629 fps= 29 q=-1.0 Lsize=    2082kB time=00:00:54.69 bitrate=
311.8kbits/s   
video:1602kB audio:443kB subtitle:0 global headers:4kB muxing overhead
1.607300%
[libx264 @ 0x106b700] frame I:9     Avg QP:17.30  size:125414
[libx264 @ 0x106b700] frame P:427   Avg QP:20.45  size:   940
[libx264 @ 0x106b700] frame B:1193  Avg QP:32.43  size:    92
[libx264 @ 0x106b700] consecutive B-frames:  2.2%  0.4%  0.2% 97.2%
[libx264 @ 0x106b700] mb I  I16..4: 26.9% 35.6% 37.5%
[libx264 @ 0x106b700] mb P  I16..4:  0.5%  0.4%  0.2%  P16..4:  0.7% 
0.1%  0.1%  0.0%  0.0%    skip:97.9%
[libx264 @ 0x106b700] mb B  I16..4:  0.0%  0.1%  0.0%  B16..8:  0.7% 
0.0%  0.0%  direct: 0.0%  skip:99.2%  L0:52.6% L1:46.8% BI: 0.6%
[libx264 @ 0x106b700] 8x8 transform intra:37.9% inter:45.6%
[libx264 @ 0x106b700] coded y,u,v intra: 22.7% 4.6% 5.0% inter: 0.1%
0.0% 0.0%
[libx264 @ 0x106b700] i16 v,h,dc,p: 48% 51%  1%  1%
[libx264 @ 0x106b700] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 34% 27% 37%  0% 
0%  0%  0%  0%  0%
[libx264 @ 0x106b700] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 37% 29% 13%  3% 
3%  3%  5%  3%  4%
[libx264 @ 0x106b700] Weighted P-Frames: Y:0.5% UV:0.5%
[libx264 @ 0x106b700] ref P L0: 62.7% 12.7% 17.8%  6.7%  0.1%
[libx264 @ 0x106b700] ref B L0: 53.9% 44.1%  2.0%
[libx264 @ 0x106b700] ref B L1: 94.2%  5.8%
[libx264 @ 0x106b700] kb/s:241.66
artur at laptop:~/Wideo>


-- 

===========
Pozdrawiam
Artur ??cki
===========


From dzungng89 at gmail.com  Wed Aug 21 17:32:43 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 10:32:43 -0500
Subject: [FFmpeg-user] Concatenation of 50-100 files
Message-ID: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>

I'm using ffmpeg concat protocol to join many subvideo files. FFMPEG keeps
getting dropped packets and can't finish the process.

ffmpeg -f concat -i list.txt -q:v 1 concated.mpg

Should I simply cat subvideo files, or concat in group of 10 and then
concat again? In the first case, which encoding can be catted as files?

From dzungng89 at gmail.com  Wed Aug 21 17:39:04 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 10:39:04 -0500
Subject: [FFmpeg-user] Convert from image to video very slow
Message-ID: <CABOWChyvj7+2VMmXUrdLYF6kcVmkTYyc2a3nEubRPnFkN2wFuQ@mail.gmail.com>

When I convert from image to video, it's very slow, especially if the time
is several minutes long. This command takes almost 30'

ffmpeg -loop 1 -f image 2 -i image -q:v 1 -r 29.97 -t 438 output.mpg

Should I change the frame rate of the input image? How to speed up the
process?

From onemda at gmail.com  Wed Aug 21 17:46:45 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 21 Aug 2013 15:46:45 +0000
Subject: [FFmpeg-user] Concatenation of 50-100 files
In-Reply-To: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>
References: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>
Message-ID: <CAPYw7P58MGYqMhpMOxoomUD0VH6bjcNQQsadEQxsJmGCq2JqqA@mail.gmail.com>

On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> I'm using ffmpeg concat protocol to join many subvideo files. FFMPEG keeps
> getting dropped packets and can't finish the process.
>
> ffmpeg -f concat -i list.txt -q:v 1 concated.mpg

You are not using concat protocol but concat demuxer.

Did you read documentation?

>
> Should I simply cat subvideo files, or concat in group of 10 and then
> concat again? In the first case, which encoding can be catted as files?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From jen142 at promessage.com  Wed Aug 21 17:54:06 2013
From: jen142 at promessage.com (jen142 at promessage.com)
Date: Wed, 21 Aug 2013 08:54:06 -0700
Subject: [FFmpeg-user] Converted DVDs in .mkv format lose Pause, FastForward,
 Skip, etc functions when I restream as mpeg to DLNA
Message-ID: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>

Hi,

My video collection was converted from DVDs onto my hard drive with
Handbrake, which uses ffmpeg.

It was a real "adventure" figuring it out but finally it works great!

Now on the drive all my videos are in "Matroska" .mkv format.

When I play them in a player like VLC or XBMC on a PC they play
perfectly, both audio and video.  All the functions like "Pause", "Fast
Forward", "Rewind", "Chapter Skip" all work OK.

In my house I have two TVs connected to BluRay player, one Sony and one
Panasonic, that have "DLNA renderers" built in.

I setup a "Mediatomb" DLNA server to get the videos onto my TVs.  At
first I could see folders but not the videos.  After a while I learned
that "mkv" is not supported by DLNA, and that I had to "transcode" the
videos.  Or it could be "remuxing".  I'm just confused about how to tell
really.

In the Mediatomb application I can do the transcode by piping it through
ffmpeg again, in real time.

The command is in a script and it looks like this:

exec ffmpeg -i $1 -vcodec mpeg2video -b 4096k -r 25 -acodec copy -f
mpegts - > $2

After doing that I can now get PERFECT! audio and video to display on my
TVs through the DLNA renderer builtin the Bluray players.

The only problem is that even though on the TV I see the popup for the
"Pause", "Fast Forward", "Rewind", "Chapter Skip", and when I press the
buttons on my remote I see those controls highlight on the screen, the
functions don't work anymore.  I can only start & completely stop the
video - none of the other functions.

I found lots of threads in various locations that talk about the missing
"Fast Forward" etc functions, for example these

https://trac.ffmpeg.org/ticket/2330
http://www.videoredo.net/msgBoard/archive/index.php/t-31105.html
http://forum.videohelp.com/threads/352097-After-remux-with-MKVMerge-to-add-subs-scanning-is-slow?p=2212090&viewfull=1#post2212090

But I don't understand from them what to really do to get these
functions back.  There's something about "seek" and "clusters" but I'm
already in over my head :-(

Can anyone help figure out any more or different commands I need to use
to not lose the functions when I stream the videos to the TVs like this?

Thanks a lot :-)

Jen


From dzungng89 at gmail.com  Wed Aug 21 18:42:49 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 11:42:49 -0500
Subject: [FFmpeg-user] Concatenation of 50-100 files
In-Reply-To: <CAPYw7P58MGYqMhpMOxoomUD0VH6bjcNQQsadEQxsJmGCq2JqqA@mail.gmail.com>
References: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>
	<CAPYw7P58MGYqMhpMOxoomUD0VH6bjcNQQsadEQxsJmGCq2JqqA@mail.gmail.com>
Message-ID: <CABOWChznbguxmVw5eCmzWS1sqP9=cOvZ_=Pp1MgFh5Geqk2eBg@mail.gmail.com>

You're right, it's the concat demuxer. So I guess concat protocol ( or
simply cat of files) would be more robust? I read that mpg files is
cattable, but after cat files the final video has no time information
(unseekable, can't play).

This is how I create the subvideo:

ffmpeg -loop 1 -f image2 -i image.png -q:v 0 out1.mpg




On Wed, Aug 21, 2013 at 10:46 AM, Paul B Mahol <onemda at gmail.com> wrote:

> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> > I'm using ffmpeg concat protocol to join many subvideo files. FFMPEG
> keeps
> > getting dropped packets and can't finish the process.
> >
> > ffmpeg -f concat -i list.txt -q:v 1 concated.mpg
>
> You are not using concat protocol but concat demuxer.
>
> Did you read documentation?
>
> >
> > Should I simply cat subvideo files, or concat in group of 10 and then
> > concat again? In the first case, which encoding can be catted as files?
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From dzungng89 at gmail.com  Wed Aug 21 20:58:08 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 13:58:08 -0500
Subject: [FFmpeg-user] Concatenation of mpg files
Message-ID: <CABOWChyDij86ccscNOX+bTeiWoURxaWgHnquuZTNHz+iGryCvA@mail.gmail.com>

I concat video files using cat command:

cat *.mpg >output.mpg

And the final video is not playable. How should be the encoding of *.mpg
be? This is how I create them:

ffmpeg -f image2 -loop 1 -i input.png -q:v 0 out1.mpg

From jpboard2 at yahoo.com  Wed Aug 21 21:02:08 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 21 Aug 2013 12:02:08 -0700 (PDT)
Subject: [FFmpeg-user] Video Capture: how to get started?
Message-ID: <1377111728.9683.YahooMailNeo@web164703.mail.gq1.yahoo.com>

I have a BlackMagic Intensity Pro Video Capture Card.? It
has it's own video capture software, but I'd like to use ffmpeg,
instead.

How would I get started?? I'm on a Linux machine.? What
would an example capture command look like (assuming 1920x1080
video, and 29.97 fps, etc)?? Also, assuming that works, how do I tell
ffmpeg to stop capturing???

Thanks

From lou at lrcd.com  Wed Aug 21 20:57:04 2013
From: lou at lrcd.com (Lou)
Date: Wed, 21 Aug 2013 10:57:04 -0800
Subject: [FFmpeg-user] Concatenation of mpg files
In-Reply-To: <CABOWChyDij86ccscNOX+bTeiWoURxaWgHnquuZTNHz+iGryCvA@mail.gmail.com>
References: <CABOWChyDij86ccscNOX+bTeiWoURxaWgHnquuZTNHz+iGryCvA@mail.gmail.com>
Message-ID: <20130821105704.61ff9f3c@lrcd.com>

On Wed, 21 Aug 2013 13:58:08 -0500
Dzung Nguyen <dzungng89 at gmail.com> wrote:

> I concat video files using cat command:
> 
> cat *.mpg >output.mpg
> 
> And the final video is not playable. How should be the encoding of *.mpg
> be? This is how I create them:

Not playable by what player(s)?

> ffmpeg -f image2 -loop 1 -i input.png -q:v 0 out1.mpg

I believe this would loop indefinitely. "-f image2" is superfluous. The
complete ffmpeg console output for your command is missing.

From lou at lrcd.com  Wed Aug 21 21:04:02 2013
From: lou at lrcd.com (Lou)
Date: Wed, 21 Aug 2013 11:04:02 -0800
Subject: [FFmpeg-user] Convert from image to video very slow
In-Reply-To: <CABOWChyvj7+2VMmXUrdLYF6kcVmkTYyc2a3nEubRPnFkN2wFuQ@mail.gmail.com>
References: <CABOWChyvj7+2VMmXUrdLYF6kcVmkTYyc2a3nEubRPnFkN2wFuQ@mail.gmail.com>
Message-ID: <20130821110402.00a6d40e@lrcd.com>

On Wed, 21 Aug 2013 10:39:04 -0500
Dzung Nguyen <dzungng89 at gmail.com> wrote:

> When I convert from image to video, it's very slow, especially if the time
> is several minutes long. This command takes almost 30'
> 
> ffmpeg -loop 1 -f image 2 -i image -q:v 1 -r 29.97 -t 438 output.mpg

You need to include the complete ffmpeg console output.
 
> Should I change the frame rate of the input image?

Default is "-r 25" for input. So you're telling ffmpeg to use "-r 25"
for input and "-r 29.97" (you should use "-r ntsc" or "-r 30000/1001"
instead of 29.97) for output resulting in duplication of frames to reach
your desired output frame rate. This may be desired in some cases. For
example you may want to input one frame per second, but you require a
specific output frame rate.

From h.reindl at thelounge.net  Wed Aug 21 21:10:44 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Wed, 21 Aug 2013 21:10:44 +0200
Subject: [FFmpeg-user] Concatenation of mpg files
In-Reply-To: <CABOWChyDij86ccscNOX+bTeiWoURxaWgHnquuZTNHz+iGryCvA@mail.gmail.com>
References: <CABOWChyDij86ccscNOX+bTeiWoURxaWgHnquuZTNHz+iGryCvA@mail.gmail.com>
Message-ID: <521510B4.8000803@thelounge.net>


Am 21.08.2013 20:58, schrieb Dzung Nguyen:
> I concat video files using cat command:
> 
> cat *.mpg >output.mpg

videos are not text files

http://ffmpeg.org/faq.html#Concatenating-using-the-concat-filter

> And the final video is not playable. How should be the encoding of *.mpg
> be? This is how I create them:
> 
> ffmpeg -f image2 -loop 1 -i input.png -q:v 0 out1.mpg


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From dzungng89 at gmail.com  Wed Aug 21 21:18:44 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 14:18:44 -0500
Subject: [FFmpeg-user] Concatenation of mpg files
In-Reply-To: <521510B4.8000803@thelounge.net>
References: <CABOWChyDij86ccscNOX+bTeiWoURxaWgHnquuZTNHz+iGryCvA@mail.gmail.com>
	<521510B4.8000803@thelounge.net>
Message-ID: <CABOWChyz1WEZTbkKp7EiEfep4su_zDc4PrtiKSnRc36-5D6v9A@mail.gmail.com>

The final video is not playable by VLC player and the movie player in
Ubuntu. It only shows the first subvideo.

Ah I forgot, the command also has time information -t:

ffmpeg -f image2 -loop 1 -r 29.97 -i resized0.png -t 60 -q:v 0 -r 29.97
0.mpg
ffmpeg -f image2 -loop 1 -r 29.97 -i resized1.png -t 60 -q:v 0 -r 29.97
1.mpg

cat *.mpg >out.mpg

I tried using concat demuxer before, but it keeps getting crashed/long time
when I concatenate 50 video files.


On Wed, Aug 21, 2013 at 2:10 PM, Reindl Harald <h.reindl at thelounge.net>wrote:

>
> Am 21.08.2013 20:58, schrieb Dzung Nguyen:
> > I concat video files using cat command:
> >
> > cat *.mpg >output.mpg
>
> videos are not text files
>
> http://ffmpeg.org/faq.html#Concatenating-using-the-concat-filter
>
> > And the final video is not playable. How should be the encoding of *.mpg
> > be? This is how I create them:
> >
> > ffmpeg -f image2 -loop 1 -i input.png -q:v 0 out1.mpg
>
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

From dzungng89 at gmail.com  Wed Aug 21 21:39:36 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 14:39:36 -0500
Subject: [FFmpeg-user] Concatenation of 50-100 files
In-Reply-To: <CABOWChznbguxmVw5eCmzWS1sqP9=cOvZ_=Pp1MgFh5Geqk2eBg@mail.gmail.com>
References: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>
	<CAPYw7P58MGYqMhpMOxoomUD0VH6bjcNQQsadEQxsJmGCq2JqqA@mail.gmail.com>
	<CABOWChznbguxmVw5eCmzWS1sqP9=cOvZ_=Pp1MgFh5Geqk2eBg@mail.gmail.com>
Message-ID: <CABOWChweSNH95Adw-gTindejMUB+XXhTT8h7V-jTXLB=jMGb+w@mail.gmail.com>

I realized that concat demuxer is the way to go. But one thing is that it's
very slow, to create a 40 min clip it would take hours. Anyway to improve
the speed?


On Wed, Aug 21, 2013 at 11:42 AM, Dzung Nguyen <dzungng89 at gmail.com> wrote:

> You're right, it's the concat demuxer. So I guess concat protocol ( or
> simply cat of files) would be more robust? I read that mpg files is
> cattable, but after cat files the final video has no time information
> (unseekable, can't play).
>
> This is how I create the subvideo:
>
> ffmpeg -loop 1 -f image2 -i image.png -q:v 0 out1.mpg
>
>
>
>
> On Wed, Aug 21, 2013 at 10:46 AM, Paul B Mahol <onemda at gmail.com> wrote:
>
>> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
>> > I'm using ffmpeg concat protocol to join many subvideo files. FFMPEG
>> keeps
>> > getting dropped packets and can't finish the process.
>> >
>> > ffmpeg -f concat -i list.txt -q:v 1 concated.mpg
>>
>> You are not using concat protocol but concat demuxer.
>>
>> Did you read documentation?
>>
>> >
>> > Should I simply cat subvideo files, or concat in group of 10 and then
>> > concat again? In the first case, which encoding can be catted as files?
>> > _______________________________________________
>> > ffmpeg-user mailing list
>> > ffmpeg-user at ffmpeg.org
>> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> >
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
>

From onemda at gmail.com  Wed Aug 21 21:40:43 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 21 Aug 2013 19:40:43 +0000
Subject: [FFmpeg-user] Concatenation of 50-100 files
In-Reply-To: <CABOWChweSNH95Adw-gTindejMUB+XXhTT8h7V-jTXLB=jMGb+w@mail.gmail.com>
References: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>
	<CAPYw7P58MGYqMhpMOxoomUD0VH6bjcNQQsadEQxsJmGCq2JqqA@mail.gmail.com>
	<CABOWChznbguxmVw5eCmzWS1sqP9=cOvZ_=Pp1MgFh5Geqk2eBg@mail.gmail.com>
	<CABOWChweSNH95Adw-gTindejMUB+XXhTT8h7V-jTXLB=jMGb+w@mail.gmail.com>
Message-ID: <CAPYw7P4LjpSpestEjzcXKcwGcGBE_KXQhNSH23E8JN92jp1gnA@mail.gmail.com>

On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> I realized that concat demuxer is the way to go. But one thing is that it's
> very slow, to create a 40 min clip it would take hours. Anyway to improve
> the speed?

Do not top post.

You can improve speed if you do not transcode.

>
>
> On Wed, Aug 21, 2013 at 11:42 AM, Dzung Nguyen <dzungng89 at gmail.com> wrote:
>
>> You're right, it's the concat demuxer. So I guess concat protocol ( or
>> simply cat of files) would be more robust? I read that mpg files is
>> cattable, but after cat files the final video has no time information
>> (unseekable, can't play).
>>
>> This is how I create the subvideo:
>>
>> ffmpeg -loop 1 -f image2 -i image.png -q:v 0 out1.mpg
>>
>>
>>
>>
>> On Wed, Aug 21, 2013 at 10:46 AM, Paul B Mahol <onemda at gmail.com> wrote:
>>
>>> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
>>> > I'm using ffmpeg concat protocol to join many subvideo files. FFMPEG
>>> keeps
>>> > getting dropped packets and can't finish the process.
>>> >
>>> > ffmpeg -f concat -i list.txt -q:v 1 concated.mpg
>>>
>>> You are not using concat protocol but concat demuxer.
>>>
>>> Did you read documentation?
>>>
>>> >
>>> > Should I simply cat subvideo files, or concat in group of 10 and then
>>> > concat again? In the first case, which encoding can be catted as
>>> > files?
>>> > _______________________________________________
>>> > ffmpeg-user mailing list
>>> > ffmpeg-user at ffmpeg.org
>>> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>> >
>>> _______________________________________________
>>> ffmpeg-user mailing list
>>> ffmpeg-user at ffmpeg.org
>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>>
>>
>>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From seandarcy2 at gmail.com  Wed Aug 21 22:04:24 2013
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Wed, 21 Aug 2013 16:04:24 -0400
Subject: [FFmpeg-user] mts > mp4 : can you just concatenate the mts files?
Message-ID: <kv36g1$221$1@ger.gmane.org>

I'm trying to generate 1 mp4 file from 10 mts files. The camcorder 
generates a new mts file every 2gigs. For the recent school play, that's 
10 mts files. Googling gives all sorts of suggestions, but no clear answer.

One post suggests just cat'ing them, and not paying any attention to 
timestamp errors. I seem to remember ffmpeg being very fussy about 
timestamps.

Or should I take them all to dv [ ffmpeg -i inN,mts -vcodec libx264 -crf 
25 -acodec copy outN.dv ] and then cat the dv files?

Or does somebody know a better way to do this?

sean


From dzungng89 at gmail.com  Wed Aug 21 22:10:38 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 15:10:38 -0500
Subject: [FFmpeg-user] mts > mp4 : can you just concatenate the mts
	files?
In-Reply-To: <kv36g1$221$1@ger.gmane.org>
References: <kv36g1$221$1@ger.gmane.org>
Message-ID: <CABOWChywfLV-nM2kvaBYexcgO0dVev+Hcxxc4=OWVF+GTmd=mw@mail.gmail.com>

On Wed, Aug 21, 2013 at 3:04 PM, Sean Darcy <seandarcy2 at gmail.com> wrote:

> I'm trying to generate 1 mp4 file from 10 mts files. The camcorder
> generates a new mts file every 2gigs. For the recent school play, that's 10
> mts files. Googling gives all sorts of suggestions, but no clear answer.
>
> One post suggests just cat'ing them, and not paying any attention to
> timestamp errors. I seem to remember ffmpeg being very fussy about
> timestamps.
>
> Or should I take them all to dv [ ffmpeg -i inN,mts -vcodec libx264 -crf
> 25 -acodec copy outN.dv ] and then cat the dv files?
>
> Or does somebody know a better way to do this?
>
> sean
>
> ______________________________**_________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/**listinfo/ffmpeg-user<http://ffmpeg.org/mailman/listinfo/ffmpeg-user>
>


Maybe using concat demuxer?

ffmpeg -f concat -i mylist.txt -c copy output

where mylist.txt hold the path to mts files

https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20media%20files

From iamanonymouscoward at gmail.com  Wed Aug 21 22:24:06 2013
From: iamanonymouscoward at gmail.com (Raghu Ram)
Date: Wed, 21 Aug 2013 13:24:06 -0700
Subject: [FFmpeg-user] FFMpeg: Webcam capture not fast enough. Ffmpeg dies
	half way (Linux)
Message-ID: <CA+4E7Qj4O_qnH76Gi-DwgzYWBUs+1J4=g9b4OXegdV6CJyN1QQ@mail.gmail.com>

Hi All,

I am trying to capture webcam output through FFMpeg (on linux inside a
virtual machine). here is the detailed question.  The capture is not
proceeding beyond 10 seconds. Can you please help.

The command I used is:

$ ffmpeg -y  -f video4linux2 -r 20 -s 160x120 -i /dev/video0 -acodec
libfaac -ab 128k  /tmp/web.avi

The most important message I am getting is:

*[video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but 153600
bytes are expected*
Information on the google for this error is scarce. I am going nuts over
this problem. Please help.

Ps: Also asked at
http://stackoverflow.com/questions/18305086/ffmpeg-webcam-capture-not-fast-enough-ffmpeg-dies-half-way-linux

*Complete message from ffmpeg:*

ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug 18 2013 09:09:13 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  configuration: --enable-libass --prefix=/opt/ffmpeg --enable-debug
--enable-libfreetype
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
[video4linux2,v4l2 @ 0x9e43fa0] The V4L2 driver changed the video from
160x120 to 320x240
[video4linux2,v4l2 @ 0x9e43fa0] The driver changed the time per frame
from 1/20 to 1/15
Input #0, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 6424.338678, bitrate: 18432 kb/s
    Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422,
320x240, 18432 kb/s, 15 fps, 15 tbr, 1000k tbn, 1000k tbc
Codec AVOption ab (set bitrate (in bits/s)) specified for output file
#0 (/tmp/web.avi) has not been used for any stream. The most likely
reason is either wrong type (e.g. a video option with no video
streams) or that it is a private option of some encoder which was not
actually used for any stream.
Output #0, avi, to '/tmp/web.avi':
  Metadata:
    ISFT            : Lavf55.12.102
    Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240,
q=2-31, 200 kb/s, 20 tbn, 20 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> mpeg4)
Press [q] to stop, [?] for help
[video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but
153600 bytes are expected
/dev/video0: Invalid data found when processing input
frame=   29 fps= 14 q=3.5 Lsize=      87kB time=00:00:01.45 bitrate=
490.0kbits/s
video:80kB audio:0kB subtitle:0 global headers:0kB muxing overhead 7.760075%
[video4linux2,v4l2 @ 0x9e43fa0] Some buffers are still owned by the
caller on close.

From dzungng89 at gmail.com  Wed Aug 21 22:39:23 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 15:39:23 -0500
Subject: [FFmpeg-user] Concatenation of 50-100 files
In-Reply-To: <CAPYw7P4LjpSpestEjzcXKcwGcGBE_KXQhNSH23E8JN92jp1gnA@mail.gmail.com>
References: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>
	<CAPYw7P58MGYqMhpMOxoomUD0VH6bjcNQQsadEQxsJmGCq2JqqA@mail.gmail.com>
	<CABOWChznbguxmVw5eCmzWS1sqP9=cOvZ_=Pp1MgFh5Geqk2eBg@mail.gmail.com>
	<CABOWChweSNH95Adw-gTindejMUB+XXhTT8h7V-jTXLB=jMGb+w@mail.gmail.com>
	<CAPYw7P4LjpSpestEjzcXKcwGcGBE_KXQhNSH23E8JN92jp1gnA@mail.gmail.com>
Message-ID: <CABOWChzhNmp58e8FMf6mogRnWercQ=Tu5-L1Yq0Kqj2TyvtXeQ@mail.gmail.com>

On Wed, Aug 21, 2013 at 2:40 PM, Paul B Mahol <onemda at gmail.com> wrote:

> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> > I realized that concat demuxer is the way to go. But one thing is that
> it's
> > very slow, to create a 40 min clip it would take hours. Anyway to improve
> > the speed?
>
> Do not top post.
>
> You can improve speed if you do not transcode.
>
> >
> >
> > On Wed, Aug 21, 2013 at 11:42 AM, Dzung Nguyen <dzungng89 at gmail.com>
> wrote:
> >
> >> You're right, it's the concat demuxer. So I guess concat protocol ( or
> >> simply cat of files) would be more robust? I read that mpg files is
> >> cattable, but after cat files the final video has no time information
> >> (unseekable, can't play).
> >>
> >> This is how I create the subvideo:
> >>
> >> ffmpeg -loop 1 -f image2 -i image.png -q:v 0 out1.mpg
> >>
> >>
> >>
> >>
> >> On Wed, Aug 21, 2013 at 10:46 AM, Paul B Mahol <onemda at gmail.com>
> wrote:
> >>
> >>> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> >>> > I'm using ffmpeg concat protocol to join many subvideo files. FFMPEG
> >>> keeps
> >>> > getting dropped packets and can't finish the process.
> >>> >
> >>> > ffmpeg -f concat -i list.txt -q:v 1 concated.mpg
> >>>
> >>> You are not using concat protocol but concat demuxer.
> >>>
> >>> Did you read documentation?
> >>>
> >>> >
> >>> > Should I simply cat subvideo files, or concat in group of 10 and then
> >>> > concat again? In the first case, which encoding can be catted as
> >>> > files?
> >>> > _______________________________________________
> >>> > ffmpeg-user mailing list
> >>> > ffmpeg-user at ffmpeg.org
> >>> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >>> >
> >>> _______________________________________________
> >>> ffmpeg-user mailing list
> >>> ffmpeg-user at ffmpeg.org
> >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >>>
> >>
> >>
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

Sorry for the top posting. To avoid transcoding, I used -c copy command:

ffmpeg -f concat -i list.txt -c copy concat.mpg

But it turns out that there's a lot of errors and ffmpeg can't concat the
files

[mpeg @ 0x21b4a40] packet too large, ignoring buffer limits to mux it
[mpeg @ 0x21b4a40] buffer underflow i=0 bufi=53885 size=133207

Is it because when I create subvideo, I used the q:v 0 options, so the
packet is too large?

ffmpeg -loop 1 -r 1 -f image2 -i image.png -q:v 0 -r 29.97 -t 300 out.mpg

From seandarcy2 at gmail.com  Wed Aug 21 22:59:09 2013
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Wed, 21 Aug 2013 16:59:09 -0400
Subject: [FFmpeg-user] on acodec copy : "codec frame size not set"
Message-ID: <kv39mm$67p$1@ger.gmane.org>

Using today's git:

ffmpeg -f concat -i mylist.txt -codec:v libx264 -crf 35 -codec:a copy 
test3.mp4
.................
[mp4 @ 0x211cb20] track 1: codec frame size is not set
Output #0, mp4, to 'test3.mp4':
   Metadata:
     encoder         : Lavf55.14.101
     Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), 
yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 30k tbn, 29.97 tbc
     Stream #0:1: Audio: ac3 ([165][0][0][0] / 0x00A5), 48000 Hz, 
stereo, 256 kb/s
Stream mapping:
   Stream #0:0 -> #0:0 (h264 -> libx264)
........

This looks like Ticket 1240: regression: "codec frame size is not set" 
using -acodec copy

But now it continues encoding. Should I do something about this warning?

sean


From onemda at gmail.com  Wed Aug 21 23:07:26 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 21 Aug 2013 21:07:26 +0000
Subject: [FFmpeg-user] Concatenation of 50-100 files
In-Reply-To: <CABOWChzhNmp58e8FMf6mogRnWercQ=Tu5-L1Yq0Kqj2TyvtXeQ@mail.gmail.com>
References: <CABOWChxZ7yYRwhTRzwBL2oBFfvrEUeSk1LTkjqAiBznBqqeEuQ@mail.gmail.com>
	<CAPYw7P58MGYqMhpMOxoomUD0VH6bjcNQQsadEQxsJmGCq2JqqA@mail.gmail.com>
	<CABOWChznbguxmVw5eCmzWS1sqP9=cOvZ_=Pp1MgFh5Geqk2eBg@mail.gmail.com>
	<CABOWChweSNH95Adw-gTindejMUB+XXhTT8h7V-jTXLB=jMGb+w@mail.gmail.com>
	<CAPYw7P4LjpSpestEjzcXKcwGcGBE_KXQhNSH23E8JN92jp1gnA@mail.gmail.com>
	<CABOWChzhNmp58e8FMf6mogRnWercQ=Tu5-L1Yq0Kqj2TyvtXeQ@mail.gmail.com>
Message-ID: <CAPYw7P5LyiGQWu6RNFQhLr31Fub8Nzd_oofJ+=_41JTEsXoFug@mail.gmail.com>

On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> On Wed, Aug 21, 2013 at 2:40 PM, Paul B Mahol <onemda at gmail.com> wrote:
>
>> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
>> > I realized that concat demuxer is the way to go. But one thing is that
>> it's
>> > very slow, to create a 40 min clip it would take hours. Anyway to
>> > improve
>> > the speed?
>>
>> Do not top post.
>>
>> You can improve speed if you do not transcode.
>>
>> >
>> >
>> > On Wed, Aug 21, 2013 at 11:42 AM, Dzung Nguyen <dzungng89 at gmail.com>
>> wrote:
>> >
>> >> You're right, it's the concat demuxer. So I guess concat protocol ( or
>> >> simply cat of files) would be more robust? I read that mpg files is
>> >> cattable, but after cat files the final video has no time information
>> >> (unseekable, can't play).
>> >>
>> >> This is how I create the subvideo:
>> >>
>> >> ffmpeg -loop 1 -f image2 -i image.png -q:v 0 out1.mpg
>> >>
>> >>
>> >>
>> >>
>> >> On Wed, Aug 21, 2013 at 10:46 AM, Paul B Mahol <onemda at gmail.com>
>> wrote:
>> >>
>> >>> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
>> >>> > I'm using ffmpeg concat protocol to join many subvideo files.
>> >>> > FFMPEG
>> >>> keeps
>> >>> > getting dropped packets and can't finish the process.
>> >>> >
>> >>> > ffmpeg -f concat -i list.txt -q:v 1 concated.mpg
>> >>>
>> >>> You are not using concat protocol but concat demuxer.
>> >>>
>> >>> Did you read documentation?
>> >>>
>> >>> >
>> >>> > Should I simply cat subvideo files, or concat in group of 10 and
>> >>> > then
>> >>> > concat again? In the first case, which encoding can be catted as
>> >>> > files?
>> >>> > _______________________________________________
>> >>> > ffmpeg-user mailing list
>> >>> > ffmpeg-user at ffmpeg.org
>> >>> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> >>> >
>> >>> _______________________________________________
>> >>> ffmpeg-user mailing list
>> >>> ffmpeg-user at ffmpeg.org
>> >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> >>>
>> >>
>> >>
>> > _______________________________________________
>> > ffmpeg-user mailing list
>> > ffmpeg-user at ffmpeg.org
>> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> >
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
> Sorry for the top posting. To avoid transcoding, I used -c copy command:
>
> ffmpeg -f concat -i list.txt -c copy concat.mpg
>
> But it turns out that there's a lot of errors and ffmpeg can't concat the
> files
>
> [mpeg @ 0x21b4a40] packet too large, ignoring buffer limits to mux it
> [mpeg @ 0x21b4a40] buffer underflow i=0 bufi=53885 size=133207
>
> Is it because when I create subvideo, I used the q:v 0 options, so the
> packet is too large?

-q:v does nothing with png, with png you can only change -compression_level.

>
> ffmpeg -loop 1 -r 1 -f image2 -i image.png -q:v 0 -r 29.97 -t 300 out.mpg
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Wed Aug 21 23:14:26 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 21 Aug 2013 21:14:26 +0000
Subject: [FFmpeg-user] FFMpeg: Webcam capture not fast enough. Ffmpeg
 dies half way (Linux)
In-Reply-To: <CA+4E7Qj4O_qnH76Gi-DwgzYWBUs+1J4=g9b4OXegdV6CJyN1QQ@mail.gmail.com>
References: <CA+4E7Qj4O_qnH76Gi-DwgzYWBUs+1J4=g9b4OXegdV6CJyN1QQ@mail.gmail.com>
Message-ID: <CAPYw7P4f8tuB_WXpxPrQV-ZSDx3yndiS-CovN-vKy8qzut9tOw@mail.gmail.com>

On 8/21/13, Raghu Ram <iamanonymouscoward at gmail.com> wrote:
> Hi All,
>
> I am trying to capture webcam output through FFMpeg (on linux inside a
> virtual machine). here is the detailed question.  The capture is not
> proceeding beyond 10 seconds. Can you please help.
>
> The command I used is:
>
> $ ffmpeg -y  -f video4linux2 -r 20 -s 160x120 -i /dev/video0 -acodec
> libfaac -ab 128k  /tmp/web.avi
>
> The most important message I am getting is:
>
> *[video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but 153600
> bytes are expected*
> Information on the google for this error is scarce. I am going nuts over
> this problem. Please help.

I have same problem, it appears error appear less frequently if I
change pixel format.

Sorry for not debuging this more.

>
> Ps: Also asked at
> http://stackoverflow.com/questions/18305086/ffmpeg-webcam-capture-not-fast-enough-ffmpeg-dies-half-way-linux
>
> *Complete message from ffmpeg:*
>
> ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Aug 18 2013 09:09:13 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
>   configuration: --enable-libass --prefix=/opt/ffmpeg --enable-debug
> --enable-libfreetype
>   libavutil      52. 40.100 / 52. 40.100
>   libavcodec     55. 19.100 / 55. 19.100
>   libavformat    55. 12.102 / 55. 12.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 82.100 /  3. 82.100
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.103 /  0. 17.103
> [video4linux2,v4l2 @ 0x9e43fa0] The V4L2 driver changed the video from
> 160x120 to 320x240
> [video4linux2,v4l2 @ 0x9e43fa0] The driver changed the time per frame
> from 1/20 to 1/15
> Input #0, video4linux2,v4l2, from '/dev/video0':
>   Duration: N/A, start: 6424.338678, bitrate: 18432 kb/s
>     Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422,
> 320x240, 18432 kb/s, 15 fps, 15 tbr, 1000k tbn, 1000k tbc
> Codec AVOption ab (set bitrate (in bits/s)) specified for output file
> #0 (/tmp/web.avi) has not been used for any stream. The most likely
> reason is either wrong type (e.g. a video option with no video
> streams) or that it is a private option of some encoder which was not
> actually used for any stream.
> Output #0, avi, to '/tmp/web.avi':
>   Metadata:
>     ISFT            : Lavf55.12.102
>     Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240,
> q=2-31, 200 kb/s, 20 tbn, 20 tbc
> Stream mapping:
>   Stream #0:0 -> #0:0 (rawvideo -> mpeg4)
> Press [q] to stop, [?] for help
> [video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but
> 153600 bytes are expected
> /dev/video0: Invalid data found when processing input
> frame=   29 fps= 14 q=3.5 Lsize=      87kB time=00:00:01.45 bitrate=
> 490.0kbits/s
> video:80kB audio:0kB subtitle:0 global headers:0kB muxing overhead
> 7.760075%
> [video4linux2,v4l2 @ 0x9e43fa0] Some buffers are still owned by the
> caller on close.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From iamanonymouscoward at gmail.com  Wed Aug 21 23:21:52 2013
From: iamanonymouscoward at gmail.com (Raghu Ram)
Date: Wed, 21 Aug 2013 14:21:52 -0700
Subject: [FFmpeg-user] FFMpeg: Webcam capture not fast enough. Ffmpeg
 dies half way (Linux)
Message-ID: <CA+4E7Qhp=VWzg1HEJyQLsXvfiCHn-UWf0FV-m8UeieJPzXRn+Q@mail.gmail.com>

Hi Paul,

No Problem. Is there any extra information you need. If it helps, I am
working on Lenovo Laptop, with Chicony webcam.

Can you please tell me how to change the pixel format?

Thanks.

On Wed, Aug 21, 2013 at 2:14 PM, Paul B Mahol <onemda at gmail.com> wrote:

> On 8/21/13, Raghu Ram <iamanonymouscoward at gmail.com> wrote:
> > Hi All,
> >
> > I am trying to capture webcam output through FFMpeg (on linux inside a
> > virtual machine). here is the detailed question.  The capture is not
> > proceeding beyond 10 seconds. Can you please help.
> >
> > The command I used is:
> >
> > $ ffmpeg -y  -f video4linux2 -r 20 -s 160x120 -i /dev/video0 -acodec
> > libfaac -ab 128k  /tmp/web.avi
> >
> > The most important message I am getting is:
> >
> > *[video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but
> 153600
> > bytes are expected*
> > Information on the google for this error is scarce. I am going nuts over
> > this problem. Please help.
>
> I have same problem, it appears error appear less frequently if I
> change pixel format.
>
> Sorry for not debuging this more.
>
> >
> > Ps: Also asked at
> >
> http://stackoverflow.com/questions/18305086/ffmpeg-webcam-capture-not-fast-enough-ffmpeg-dies-half-way-linux
> >
> > *Complete message from ffmpeg:*
> >
> > ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg
> > developers
> >   built on Aug 18 2013 09:09:13 with gcc 4.6 (Ubuntu/Linaro
> 4.6.3-1ubuntu5)
> >   configuration: --enable-libass --prefix=/opt/ffmpeg --enable-debug
> > --enable-libfreetype
> >   libavutil      52. 40.100 / 52. 40.100
> >   libavcodec     55. 19.100 / 55. 19.100
> >   libavformat    55. 12.102 / 55. 12.102
> >   libavdevice    55.  3.100 / 55.  3.100
> >   libavfilter     3. 82.100 /  3. 82.100
> >   libswscale      2.  4.100 /  2.  4.100
> >   libswresample   0. 17.103 /  0. 17.103
> > [video4linux2,v4l2 @ 0x9e43fa0] The V4L2 driver changed the video from
> > 160x120 to 320x240
> > [video4linux2,v4l2 @ 0x9e43fa0] The driver changed the time per frame
> > from 1/20 to 1/15
> > Input #0, video4linux2,v4l2, from '/dev/video0':
> >   Duration: N/A, start: 6424.338678, bitrate: 18432 kb/s
> >     Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422,
> > 320x240, 18432 kb/s, 15 fps, 15 tbr, 1000k tbn, 1000k tbc
> > Codec AVOption ab (set bitrate (in bits/s)) specified for output file
> > #0 (/tmp/web.avi) has not been used for any stream. The most likely
> > reason is either wrong type (e.g. a video option with no video
> > streams) or that it is a private option of some encoder which was not
> > actually used for any stream.
> > Output #0, avi, to '/tmp/web.avi':
> >   Metadata:
> >     ISFT            : Lavf55.12.102
> >     Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240,
> > q=2-31, 200 kb/s, 20 tbn, 20 tbc
> > Stream mapping:
> >   Stream #0:0 -> #0:0 (rawvideo -> mpeg4)
> > Press [q] to stop, [?] for help
> > [video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but
> > 153600 bytes are expected
> > /dev/video0: Invalid data found when processing input
> > frame=   29 fps= 14 q=3.5 Lsize=      87kB time=00:00:01.45 bitrate=
> > 490.0kbits/s
> > video:80kB audio:0kB subtitle:0 global headers:0kB muxing overhead
> > 7.760075%
> > [video4linux2,v4l2 @ 0x9e43fa0] Some buffers are still owned by the
> > caller on close.
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Wed Aug 21 23:27:13 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 21 Aug 2013 21:27:13 +0000
Subject: [FFmpeg-user] FFMpeg: Webcam capture not fast enough. Ffmpeg
 dies half way (Linux)
In-Reply-To: <CA+4E7Qhp=VWzg1HEJyQLsXvfiCHn-UWf0FV-m8UeieJPzXRn+Q@mail.gmail.com>
References: <CA+4E7Qhp=VWzg1HEJyQLsXvfiCHn-UWf0FV-m8UeieJPzXRn+Q@mail.gmail.com>
Message-ID: <CAPYw7P4uZQ7hsDyWhiHOzNib7wXfGs_H4d4tJmx5YR0V-gu3FQ@mail.gmail.com>

On 8/21/13, Raghu Ram <iamanonymouscoward at gmail.com> wrote:
> Hi Paul,
>
> No Problem. Is there any extra information you need. If it helps, I am
> working on Lenovo Laptop, with Chicony webcam.
>
> Can you please tell me how to change the pixel format?

See output of:

ffmpeg -h demuxer=v4l2

>
> Thanks.
>
> On Wed, Aug 21, 2013 at 2:14 PM, Paul B Mahol <onemda at gmail.com> wrote:
>
>> On 8/21/13, Raghu Ram <iamanonymouscoward at gmail.com> wrote:
>> > Hi All,
>> >
>> > I am trying to capture webcam output through FFMpeg (on linux inside a
>> > virtual machine). here is the detailed question.  The capture is not
>> > proceeding beyond 10 seconds. Can you please help.
>> >
>> > The command I used is:
>> >
>> > $ ffmpeg -y  -f video4linux2 -r 20 -s 160x120 -i /dev/video0 -acodec
>> > libfaac -ab 128k  /tmp/web.avi
>> >
>> > The most important message I am getting is:
>> >
>> > *[video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but
>> 153600
>> > bytes are expected*
>> > Information on the google for this error is scarce. I am going nuts
>> > over
>> > this problem. Please help.
>>
>> I have same problem, it appears error appear less frequently if I
>> change pixel format.
>>
>> Sorry for not debuging this more.
>>
>> >
>> > Ps: Also asked at
>> >
>> http://stackoverflow.com/questions/18305086/ffmpeg-webcam-capture-not-fast-enough-ffmpeg-dies-half-way-linux
>> >
>> > *Complete message from ffmpeg:*
>> >
>> > ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg
>> > developers
>> >   built on Aug 18 2013 09:09:13 with gcc 4.6 (Ubuntu/Linaro
>> 4.6.3-1ubuntu5)
>> >   configuration: --enable-libass --prefix=/opt/ffmpeg --enable-debug
>> > --enable-libfreetype
>> >   libavutil      52. 40.100 / 52. 40.100
>> >   libavcodec     55. 19.100 / 55. 19.100
>> >   libavformat    55. 12.102 / 55. 12.102
>> >   libavdevice    55.  3.100 / 55.  3.100
>> >   libavfilter     3. 82.100 /  3. 82.100
>> >   libswscale      2.  4.100 /  2.  4.100
>> >   libswresample   0. 17.103 /  0. 17.103
>> > [video4linux2,v4l2 @ 0x9e43fa0] The V4L2 driver changed the video from
>> > 160x120 to 320x240
>> > [video4linux2,v4l2 @ 0x9e43fa0] The driver changed the time per frame
>> > from 1/20 to 1/15
>> > Input #0, video4linux2,v4l2, from '/dev/video0':
>> >   Duration: N/A, start: 6424.338678, bitrate: 18432 kb/s
>> >     Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422,
>> > 320x240, 18432 kb/s, 15 fps, 15 tbr, 1000k tbn, 1000k tbc
>> > Codec AVOption ab (set bitrate (in bits/s)) specified for output file
>> > #0 (/tmp/web.avi) has not been used for any stream. The most likely
>> > reason is either wrong type (e.g. a video option with no video
>> > streams) or that it is a private option of some encoder which was not
>> > actually used for any stream.
>> > Output #0, avi, to '/tmp/web.avi':
>> >   Metadata:
>> >     ISFT            : Lavf55.12.102
>> >     Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240,
>> > q=2-31, 200 kb/s, 20 tbn, 20 tbc
>> > Stream mapping:
>> >   Stream #0:0 -> #0:0 (rawvideo -> mpeg4)
>> > Press [q] to stop, [?] for help
>> > [video4linux2,v4l2 @ 0x9e43fa0] The v4l2 frame is 46448 bytes, but
>> > 153600 bytes are expected
>> > /dev/video0: Invalid data found when processing input
>> > frame=   29 fps= 14 q=3.5 Lsize=      87kB time=00:00:01.45 bitrate=
>> > 490.0kbits/s
>> > video:80kB audio:0kB subtitle:0 global headers:0kB muxing overhead
>> > 7.760075%
>> > [video4linux2,v4l2 @ 0x9e43fa0] Some buffers are still owned by the
>> > caller on close.
>> > _______________________________________________
>> > ffmpeg-user mailing list
>> > ffmpeg-user at ffmpeg.org
>> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> >
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From dzungng89 at gmail.com  Thu Aug 22 00:10:53 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 17:10:53 -0500
Subject: [FFmpeg-user] Speed up overlaying process
Message-ID: <CABOWChyrMxWquWDETSV6AoHDHFOb9nmqQZ3bPU6-w5PWh_9bMg@mail.gmail.com>

I'm overlaying two videos together:

ffmpeg -i video1.mpg -i video2.mpg -filter_complex
"color=c=black:size=1280x720[base]; [0:v] setpts=PTS_STARTPTS,
scale=856x642 [left]; [1:v] setpts=PTS_STARTPTS, scale=424x318
[right]; [base][left] overlay=shortest=1:y=39[tmp1]; [tmp1][right]
overlay=shortest=1:x=856:y=201" -q:v 0 output.mpg

This takes very long time, for a final output of 1hour. Which setting
should I change to make it faster? (remove scale filter)?

From onemda at gmail.com  Thu Aug 22 00:19:41 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 21 Aug 2013 22:19:41 +0000
Subject: [FFmpeg-user] Speed up overlaying process
In-Reply-To: <CABOWChyrMxWquWDETSV6AoHDHFOb9nmqQZ3bPU6-w5PWh_9bMg@mail.gmail.com>
References: <CABOWChyrMxWquWDETSV6AoHDHFOb9nmqQZ3bPU6-w5PWh_9bMg@mail.gmail.com>
Message-ID: <CAPYw7P7zdtzLkoy7jOYEoZ_KHpgry=AuB1jv=Mk_mBT6SHyk1A@mail.gmail.com>

On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> I'm overlaying two videos together:
>
> ffmpeg -i video1.mpg -i video2.mpg -filter_complex
> "color=c=black:size=1280x720[base]; [0:v] setpts=PTS_STARTPTS,
> scale=856x642 [left]; [1:v] setpts=PTS_STARTPTS, scale=424x318
> [right]; [base][left] overlay=shortest=1:y=39[tmp1]; [tmp1][right]
> overlay=shortest=1:x=856:y=201" -q:v 0 output.mpg
>
> This takes very long time, for a final output of 1hour. Which setting
> should I change to make it faster? (remove scale filter)?

Usually one would inspect what part of filtering takes most of time.
Encoding and not scaling is probably most costly operation that you are doing.

From cehoyos at ag.or.at  Thu Aug 22 00:19:51 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 21 Aug 2013 22:19:51 +0000 (UTC)
Subject: [FFmpeg-user] Delayed sound in screencast
References: <5213C125.6060706@gmail.com>
	<loom.20130821T120936-514@post.gmane.org>
	<5214A1B6.5090807@gmail.com>
	<loom.20130821T135044-175@post.gmane.org>
	<5214B9A7.8010102@gmail.com>
Message-ID: <loom.20130822T001937-401@post.gmane.org>

Artur ??cki <alacki93 <at> gmail.com> writes:

> Now delay is smaller but sounds is "slow".

Sorry, I don't understand "slow" in this context.

Carl Eugen


From cehoyos at ag.or.at  Thu Aug 22 00:24:12 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 21 Aug 2013 22:24:12 +0000 (UTC)
Subject: [FFmpeg-user] Convert from image to video very slow
References: <CABOWChyvj7+2VMmXUrdLYF6kcVmkTYyc2a3nEubRPnFkN2wFuQ@mail.gmail.com>
Message-ID: <loom.20130822T002302-759@post.gmane.org>

Dzung Nguyen <dzungng89 <at> gmail.com> writes:

> ffmpeg -loop 1 -f image 2 -i image -q:v 1 -r 29.97 -t 438 output.mpg

Complete, uncut console output missing.

I don't think you can generally use "-qscale 1 -f mpg" 
without additional muxer options but maybe I am missing 
something.

Carl Eugen


From dzungng89 at gmail.com  Thu Aug 22 00:43:20 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 17:43:20 -0500
Subject: [FFmpeg-user] Speed up overlaying process
In-Reply-To: <CAPYw7P7zdtzLkoy7jOYEoZ_KHpgry=AuB1jv=Mk_mBT6SHyk1A@mail.gmail.com>
References: <CABOWChyrMxWquWDETSV6AoHDHFOb9nmqQZ3bPU6-w5PWh_9bMg@mail.gmail.com>
	<CAPYw7P7zdtzLkoy7jOYEoZ_KHpgry=AuB1jv=Mk_mBT6SHyk1A@mail.gmail.com>
Message-ID: <183A971D-2B84-4D1B-96AE-17C650BE2082@gmail.com>



On Aug 21, 2013, at 5:19 PM, Paul B Mahol <onemda at gmail.com> wrote:

> On 8/21/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
>> I'm overlaying two videos together:
>> 
>> ffmpeg -i video1.mpg -i video2.mpg -filter_complex
>> "color=c=black:size=1280x720[base]; [0:v] setpts=PTS_STARTPTS,
>> scale=856x642 [left]; [1:v] setpts=PTS_STARTPTS, scale=424x318
>> [right]; [base][left] overlay=shortest=1:y=39[tmp1]; [tmp1][right]
>> overlay=shortest=1:x=856:y=201" -q:v 0 output.mpg
>> 
>> This takes very long time, for a final output of 1hour. Which setting
>> should I change to make it faster? (remove scale filter)?
> 
> Usually one would inspect what part of filtering takes most of time.
> Encoding and not scaling is probably most costly operation that you are doing.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Can you please elaborate on this? Two videos have the correct size already, but without scaling filter it is displayed at wrong position.

From andrew2012 at flight.us  Thu Aug 22 03:58:24 2013
From: andrew2012 at flight.us (Andrew RT)
Date: Wed, 21 Aug 2013 21:58:24 -0400 (EDT)
Subject: [FFmpeg-user] ass subtitles
Message-ID: <1192797591.353284.1377136704952.open-xchange@email.1and1.com>


Can someone please give a clue about how to burn in subtitles with ffmpeg?

Particularly, subtitles formated with "ass"? (ass-bottom-margin, subtitle color, size, etc.)

thanks!

From dzungng89 at gmail.com  Thu Aug 22 04:06:46 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Wed, 21 Aug 2013 21:06:46 -0500
Subject: [FFmpeg-user] ass subtitles
In-Reply-To: <1192797591.353284.1377136704952.open-xchange@email.1and1.com>
References: <1192797591.353284.1377136704952.open-xchange@email.1and1.com>
Message-ID: <CABOWChyB6QAtRRWZHODKfQiFjuZGYz6O8Zps22pRNeQzD=8EwA@mail.gmail.com>

On Wed, Aug 21, 2013 at 8:58 PM, Andrew RT <andrew2012 at flight.us> wrote:

>
> Can someone please give a clue about how to burn in subtitles with ffmpeg?
>
> Particularly, subtitles formated with "ass"  (ass-bottom-margin, subtitle
> color, size, etc.)
>
> thanks!
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


ffmpeg -i video.avi -vf "ass=subtitle.ass" out.avi

You can convert from srt to ass using:

ffmpeg -i subtitle.srt subtitle.ass

From seandarcy2 at gmail.com  Wed Aug 21 22:59:47 2013
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Wed, 21 Aug 2013 16:59:47 -0400
Subject: [FFmpeg-user] mts > mp4 : can you just concatenate the mts
	files?
In-Reply-To: <CABOWChywfLV-nM2kvaBYexcgO0dVev+Hcxxc4=OWVF+GTmd=mw@mail.gmail.com>
References: <kv36g1$221$1@ger.gmane.org>
	<CABOWChywfLV-nM2kvaBYexcgO0dVev+Hcxxc4=OWVF+GTmd=mw@mail.gmail.com>
Message-ID: <52152A43.4090203@gmail.com>

On 08/21/2013 04:10 PM, Dzung Nguyen wrote:
> On Wed, Aug 21, 2013 at 3:04 PM, Sean Darcy <seandarcy2 at gmail.com> wrote:
>
>> I'm trying to generate 1 mp4 file from 10 mts files. The camcorder
>> generates a new mts file every 2gigs. For the recent school play, that's 10
>> mts files. Googling gives all sorts of suggestions, but no clear answer.
>>
>> One post suggests just cat'ing them, and not paying any attention to
>> timestamp errors. I seem to remember ffmpeg being very fussy about
>> timestamps.
>>
>> Or should I take them all to dv [ ffmpeg -i inN,mts -vcodec libx264 -crf
>> 25 -acodec copy outN.dv ] and then cat the dv files?
>>
>> Or does somebody know a better way to do this?
>>
>> sean
>>
>> ______________________________**_________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/**listinfo/ffmpeg-user<http://ffmpeg.org/mailman/listinfo/ffmpeg-user>
>>
>
>
> Maybe using concat demuxer?
>
> ffmpeg -f concat -i mylist.txt -c copy output
>
> where mylist.txt hold the path to mts files
>
> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20media%20files
>

Great. Thanks.

sean

From alacki93 at gmail.com  Thu Aug 22 12:17:24 2013
From: alacki93 at gmail.com (=?UTF-8?B?QXJ0dXIgxYHEhWNraQ==?=)
Date: Thu, 22 Aug 2013 12:17:24 +0200
Subject: [FFmpeg-user] Delayed sound in screencast
In-Reply-To: <loom.20130822T001937-401@post.gmane.org>
References: <5213C125.6060706@gmail.com>
	<loom.20130821T120936-514@post.gmane.org>
	<5214A1B6.5090807@gmail.com>
	<loom.20130821T135044-175@post.gmane.org>
	<5214B9A7.8010102@gmail.com>
	<loom.20130822T001937-401@post.gmane.org>
Message-ID: <5215E534.9040503@gmail.com>

W dniu 22.08.2013 00:19, Carl Eugen Hoyos pisze:
> Artur ??cki <alacki93 <at> gmail.com> writes:
>
>> Now delay is smaller but sounds is "slow".
> Sorry, I don't understand "slow" in this context.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

It looks for example like audio with length 10 seconds was "expanded" to
video who takes 20 seconds (I can't describe this any better).

In meanwhile I've tried many "magical" scripts but no one helped. I've
also compiled old version of ffmpeg (1.2.2) and I've the same problem.

-- 

===========
Pozdrawiam
Artur ??cki
===========



From soho123.2012 at gmail.com  Thu Aug 22 14:13:00 2013
From: soho123.2012 at gmail.com (Huang Soho)
Date: Thu, 22 Aug 2013 20:13:00 +0800
Subject: [FFmpeg-user] how to use ffplay to play RTSP/TCP stream
Message-ID: <CAFd2cZWvNMBbLg4Lkkrjwv+6LK+OB8Z6P=0DrTgU1Uwxb9abpQ@mail.gmail.com>

hi All,

As my understanding, ffplay can play RTSP/UDP stream that output from
ffserver.
That I tested, it works fine.
In advance, is it possible to play RTSP/TCP stream by ffplay ?
How to ?


Thanks!

From andrey.aleksandrovich at googlemail.com  Thu Aug 22 17:08:50 2013
From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich)
Date: Thu, 22 Aug 2013 18:08:50 +0300
Subject: [FFmpeg-user] Video Capture: how to get started?
In-Reply-To: <1377111728.9683.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1377111728.9683.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <CAPGkfYGN8LVvytbZtcCLGxFMMp28DQKSJ_AO88-suxqEnf4JgA@mail.gmail.com>

Probably, you need "Blackmagic? Gstreamer Source Plugin"
https://ubuntuforums.org/showthread.php?t=1333927

On 8/21/13, James Board <jpboard2 at yahoo.com> wrote:
> I have a BlackMagic Intensity Pro Video Capture Card.? It
> has it's own video capture software, but I'd like to use ffmpeg,
> instead.
>
> How would I get started?? I'm on a Linux machine.? What
> would an example capture command look like (assuming 1920x1080
> video, and 29.97 fps, etc)?? Also, assuming that works, how do I tell
> ffmpeg to stop capturing?
>
> Thanks
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From alacki93 at gmail.com  Thu Aug 22 18:26:33 2013
From: alacki93 at gmail.com (=?UTF-8?B?QXJ0dXIgxYHEhWNraQ==?=)
Date: Thu, 22 Aug 2013 18:26:33 +0200
Subject: [FFmpeg-user] how to use ffplay to play RTSP/TCP stream
In-Reply-To: <CAFd2cZWvNMBbLg4Lkkrjwv+6LK+OB8Z6P=0DrTgU1Uwxb9abpQ@mail.gmail.com>
References: <CAFd2cZWvNMBbLg4Lkkrjwv+6LK+OB8Z6P=0DrTgU1Uwxb9abpQ@mail.gmail.com>
Message-ID: <52163BB9.8080106@gmail.com>

W dniu 22.08.2013 14:13, Huang Soho pisze:
> hi All,
>
> As my understanding, ffplay can play RTSP/UDP stream that output from
> ffserver.
> That I tested, it works fine.
> In advance, is it possible to play RTSP/TCP stream by ffplay ?
> How to ?
>
>
> Thanks!
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Have you tried to add ?tcp to URI? For example ffplay
rtsp://domain.com/stream?tcp.

-- 

===========
Pozdrawiam
Artur ??cki
===========


From tevans.uk at googlemail.com  Thu Aug 22 18:36:35 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Thu, 22 Aug 2013 17:36:35 +0100
Subject: [FFmpeg-user] Video Capture: how to get started?
In-Reply-To: <1377111728.9683.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1377111728.9683.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1JOR-As6WHM0T+pzsRi4UBbkQH+FBpitvg6p2xV_vadEQ@mail.gmail.com>

On Wed, Aug 21, 2013 at 8:02 PM, James Board <jpboard2 at yahoo.com> wrote:
> I have a BlackMagic Intensity Pro Video Capture Card.  It
> has it's own video capture software, but I'd like to use ffmpeg,
> instead.

Most places I have seen recommend to use bmdcapture to generate a .nut
stream that you then pipe in to ffmpeg. The main issue here is that
bmdtools is based around libav and not ffmpeg, but build both static
and you won't have any problems.

>
> How would I get started?  I'm on a Linux machine.  What
> would an example capture command look like (assuming 1920x1080
> video, and 29.97 fps, etc)?  Also, assuming that works, how do I tell
> ffmpeg to stop capturing?
>

Press 'q' or send it a signal.

Cheers

Tom

From laplacezhang at 126.com  Thu Aug 22 10:38:30 2013
From: laplacezhang at 126.com (Andrew Min Chang)
Date: Thu, 22 Aug 2013 16:38:30 +0800 (CST)
Subject: [FFmpeg-user] [Generic Question]Can FFmpeg accept and decode live
 raw H.264 frame content?
Message-ID: <3e168fc1.199fc.140a52cc81f.Coremail.laplacezhang@126.com>



I am now working on a project, which receives live H.264 stream with a private protocol. What I would do are to decode this stream and display it, and meanwhile to save the stream as a movie file in a specified time duration.


Now comes to a question: Can I pass raw H.264 frame to ffmpeg in some way and get bit map pictures for each frame in live?


If the answer is YES, could you tell me what documents I should read in sequence to understand the function I need? 


Thank you at advance!


----
Andrew
2013-08-22

From jen142 at promessage.com  Thu Aug 22 18:58:44 2013
From: jen142 at promessage.com (jen142 at promessage.com)
Date: Thu, 22 Aug 2013 09:58:44 -0700
Subject: [FFmpeg-user] Converted DVDs in .mkv format lose Pause,
 FastForward, Skip, etc functions when I restream as mpeg to DLNA
In-Reply-To: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>
References: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>
Message-ID: <1377190724.6609.12950137.7EF80BA2@webmail.messagingengine.com>

On Wed, Aug 21, 2013, at 08:54 AM, jen142 at promessage.com wrote:
> Can anyone help figure out any more or different commands I need to use
> to not lose the functions when I stream the videos to the TVs like this?

This thread

	http://ffmpeg-users.933282.n4.nabble.com/Transcoded-MPEG-2-Transport-Stream-does-not-have-random-access-indicator-bit-set-td3010438.html

suggests that

"Random Access Indicator, which is very important to realize trick play
(i.e. random access, fast forward),
is not included in the stream." 

The question is asked there "how?", but no answer.

A commit thread

	[FFmpeg-cvslog] mpegtsenc: set Random Access indicator on
	keyframe start packets
	https://lists.ffmpeg.org/pipermail/ffmpeg-cvslog/2011-July/038989.html

suggests that it's already IN ffmpeg.

If it is, how can I enable the "Random Access indicator" in my stream
transcode command?  I'm not sure it's the answer, but it seems like it's
worth a try at least.

From jen142 at promessage.com  Thu Aug 22 19:52:29 2013
From: jen142 at promessage.com (jen142 at promessage.com)
Date: Thu, 22 Aug 2013 10:52:29 -0700
Subject: [FFmpeg-user] Converted DVDs in .mkv format lose Pause,
 FastForward, Skip, etc functions when I restream as mpeg to DLNA
In-Reply-To: <CAHmvugc1JmasNCjQ0WFFdcpR2bjFnCvRdrfhupwdkc2f2Suj4A@mail.gmail.com>
References: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>
	<1377190724.6609.12950137.7EF80BA2@webmail.messagingengine.com>
	<CAHmvugc1JmasNCjQ0WFFdcpR2bjFnCvRdrfhupwdkc2f2Suj4A@mail.gmail.com>
Message-ID: <1377193949.19582.12971037.3EF167CA@webmail.messagingengine.com>

On Thu, Aug 22, 2013, at 10:38 AM, Jind?ich Makovi?ka wrote:
> It doesn't need to be enabled. RAI should be set automatically on
> every starting packet of a key frame.

Thanks for clearing that up.  Then the lack of control features must be
still something else. :-/

From dzungng89 at gmail.com  Thu Aug 22 20:12:48 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Thu, 22 Aug 2013 13:12:48 -0500
Subject: [FFmpeg-user] Concatenation of mpg files
Message-ID: <CABOWChwWk6ksD56yAw8bwJ0JpLpMjnh2qh0Xgvt=RR51h3yiwQ@mail.gmail.com>

Hi,

I tried to cat in ubuntu shell mpg files, but the output file lost timing
information (can't be played, can't seek)

This is mpg video information:

 ffmpeg -i 0.mpg
ffmpeg version git-2013-08-06-a68b6ec Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 21 2013 11:58:24 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 22.100 / 55. 22.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mpeg, from '0.mpg':
  Duration: 00:02:10.00, start: 0.533367, bitrate: 2899 kb/s
    Stream #0:0[0x1e0]: Video: mpeg1video, yuv420p, 1280x720 [SAR 1:1 DAR
16:9], 104857 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 29.97 tbc

So it's mpeg1 encoded, everything is correct. The way I make this video is
using command:

ffmpeg -loop 1 -r 1 -i image.png -q:v 0 -r 29.97 -t 10 0.mpg

What should I change here in the converting from image to video, so that
the mpg files are cattable?

From jen142 at promessage.com  Thu Aug 22 21:16:20 2013
From: jen142 at promessage.com (jen142 at promessage.com)
Date: Thu, 22 Aug 2013 12:16:20 -0700
Subject: [FFmpeg-user] Converted DVDs in .mkv format lose Pause,
 FastForward, Skip, etc functions when I restream as mpeg to DLNA
In-Reply-To: <1377193949.19582.12971037.3EF167CA@webmail.messagingengine.com>
References: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>
	<1377190724.6609.12950137.7EF80BA2@webmail.messagingengine.com>
	<CAHmvugc1JmasNCjQ0WFFdcpR2bjFnCvRdrfhupwdkc2f2Suj4A@mail.gmail.com>
	<1377193949.19582.12971037.3EF167CA@webmail.messagingengine.com>
Message-ID: <1377198980.6327.13000877.2061DB7D@webmail.messagingengine.com>

I removed the Panasonic BluRay player from the picture ...

I just setup VLC as a client to my movie store.

In it I can access BOTH the native mkv files and the re-streamed ones,
through the ffmpeg cmd in Mediatomb.

In VLC, Playing the native mkv files, all the 'action' controls function
perfectly.  Also, time-progress is shown, and changes, correctly.

Otoh, if I play the ffmpeg-converted streams, all the 'action' controls
are missing/disabled. AND, time-progress just sits at "0".

The movie DOES still play though.

This says to me that this the lack of controls is NOT the 'fault' of the
Panasonic BluRay player.

From dzungng89 at gmail.com  Thu Aug 22 21:31:27 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Thu, 22 Aug 2013 14:31:27 -0500
Subject: [FFmpeg-user] Concatenation buffer underflow problem
In-Reply-To: <20130820103148.GC25866@phare.normalesup.org>
References: <CABOWChw67g29xukSkkno-QDESBbnrkJBSgqexDUA8kBfdtCWjQ@mail.gmail.com>
	<20130820103148.GC25866@phare.normalesup.org>
Message-ID: <CABOWChxZejaop9zukEjjHfU3QmadZ8FQ2nN7A6raGPD9k=UNdQ@mail.gmail.com>

On Tue, Aug 20, 2013 at 5:31 AM, Nicolas George <
nicolas.george at normalesup.org> wrote:

> Le duodi 2 fructidor, an CCXXI, Dzung Nguyen a ?crit :
> > The problem is that the 2nd command has buffer underflow problem: packet
> > too large, ignoring buffer limit. How to fix it?
>
> Your message is mostly useless, as you post nowhere near enough information
> to reproduce your problem. Your complete command lines and full console
> output are a bare minimum.
>
> As is, I can only reply two generic warnings: first, do not rely on default
> values, specify codecs and encoding parameters; second, do not use MPEG-PS,
> especially as an intermediary format.
>
> Regards,
>
> --
>   Nicolas George
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

So this is the complete output:

*1) Create subvideos:*

ffmpeg -y -loop 1 -r 1 -i resized0.png -q:v 0 -r 29.97 -t 30 0.mpg
ffmpeg version git-2013-08-06-a68b6ec Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 21 2013 11:58:24 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  configuration: --prefix=/home/andrey/ffmpeg_build
--extra-cflags=-I/home/andrey/ffmpeg_build/include
--extra-ldflags=-L/home/andrey/ffmpeg_build/lib --bindir=/home/andrey/bin
--extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac
--enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab
--enable-libfreetype
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 22.100 / 55. 22.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Input #0, image2, from 'resized0.png':
  Duration: 00:00:00.04, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: png, rgb24, 1280x720 [SAR 300:300 DAR 16:9], 25
fps, 25 tbr, 25 tbn, 25 tbc
[mpeg @ 0x277e5e0] VBV buffer size not set, muxing may fail
Output #0, mpeg, to '0.mpg':
  Metadata:
    encoder         : Lavf55.13.102
    Stream #0:0: Video: mpeg1video, yuv420p, 1280x720 [SAR 1:1 DAR 16:9],
q=2-31, 200 kb/s, 90k tbn, 29.97 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (png -> mpeg1video)
Press [q] to stop, [?] for help
frame=  900 fps=198 q=0.0 Lsize=   10620kB time=00:00:29.99
bitrate=2900.3kbits/s dup=898 drop=0
video:10570kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.475636%

*2) Concatenate using concat protocol. THIS IS WHERE THE ERROR ARE.*

 ffmpeg -y -i "concat:0.mpg|1.mpg|2.mpg|3.mpg" -c copy output.mpg

ffmpeg version git-2013-08-06-a68b6ec Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 21 2013 11:58:24 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  configuration: --prefix=/home/andrey/ffmpeg_build
--extra-cflags=-I/home/andrey/ffmpeg_build/include
--extra-ldflags=-L/home/andrey/ffmpeg_build/lib --bindir=/home/andrey/bin
--extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac
--enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab
--enable-libfreetype
  libavutil      52. 41.100 / 52. 41.100
  libavcodec     55. 22.100 / 55. 22.100
  libavformat    55. 13.102 / 55. 13.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mpeg, from 'concat:0.mpg|1.mpg|2.mpg|3.mpg':
  Duration: 00:00:00.93, start: 0.533367, bitrate: 270363 kb/s
    Stream #0:0[0x1e0]: Video: mpeg1video, yuv420p, 1280x720 [SAR 1:1 DAR
16:9], 104857 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 29.97 tbc
Output #0, mpeg, to 'output.mpg':
  Metadata:
    encoder         : Lavf55.13.102
    Stream #0:0: Video: mpeg1video, yuv420p, 1280x720 [SAR 1:1 DAR 16:9],
q=2-31, 104857 kb/s, 29.97 fps, 90k tbn, 29.97 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mpeg @ 0x317f160] buffer underflow i=0 bufi=12210 size=106615
[mpeg @ 0x317f160] packet too large, ignoring buffer limits to mux it
[mpeg @ 0x317f160] buffer underflow i=0 bufi=12210 size=106615
[mpeg @ 0x317f160] buffer underflow i=0 bufi=14239 size=106615
[mpeg @ 0x317f160] packet too large, ignoring buffer limits to mux it
[mpeg @ 0x317f160] buffer underflow i=0 bufi=14239 size=106615
[mpeg @ 0x317f160] buffer underflow i=0 bufi=16280 size=106615
[mpeg @ 0x317f160] packet too large, ignoring buffer limits to mux it
[mpeg @ 0x317f160] buffer underflow i=0 bufi=16280 size=106615
[mpeg @ 0x317f160] buffer underflow i=0 bufi=18321 size=106615
[mpeg @ 0x317f160] packet too large, ignoring buffer limits to mux it
[mpeg @ 0x317f160] buffer underflow i=0 bufi=18321 size=106615
[mpeg @ 0x317f160] buffer underflow i=0 bufi=20362 size=106615
[mpeg @ 0x317f160] packet too large, ignoring

From jen142 at promessage.com  Thu Aug 22 22:10:33 2013
From: jen142 at promessage.com (jen142 at promessage.com)
Date: Thu, 22 Aug 2013 13:10:33 -0700
Subject: [FFmpeg-user] Converted DVDs in .mkv format lose Pause,
 FastForward, Skip, etc functions when I restream as mpeg to DLNA
In-Reply-To: <1377198980.6327.13000877.2061DB7D@webmail.messagingengine.com>
References: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>
	<1377190724.6609.12950137.7EF80BA2@webmail.messagingengine.com>
	<CAHmvugc1JmasNCjQ0WFFdcpR2bjFnCvRdrfhupwdkc2f2Suj4A@mail.gmail.com>
	<1377193949.19582.12971037.3EF167CA@webmail.messagingengine.com>
	<1377198980.6327.13000877.2061DB7D@webmail.messagingengine.com>
Message-ID: <1377202233.17373.13021077.2FFF4C8D@webmail.messagingengine.com>

Continuing to try to remove variables from the situation, I'm now using
only ffmpeg/ffplay for streaming & playback.  I'm hoping someone who
knows more about this can actually help with interpreting or more
suggestions.

Since ffplay has such rudimentary motion controls -- L/R/PgUp/PgDwn --
compared to VLC or the standalone BluRay player, I'm not sure if what
follows actually tells me anything useful.

If I play the native file

	ffplay -i movie.mkv

it plays without error.

If I play the transcoded stream

	ffmpeg -i movie.mkv -f mpegts -b:v 2500k -minrate 2500k -maxrate
	2800k -r 29.97 -g 300 -bf 2 -bsf:v h264_mp4toannexb -vcodec
	mpeg2video -acodec copy - | ffplay -

it also plays without error.

In case it's useful, the output at the console in the transcoded test is


	ffmpeg -i movie.mkv -f mpegts -b:v 2500k -minrate 2500k -maxrate
	2500k -r 29.97 -g 300 -bf 2 -bsf:v h264_mp4toannexb -threads 2
	-vcodec mpeg2video -acodec copy - | ffplay -
	ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg
	developers
	...
	ffplay version 2.0.1 Copyright (c) 2003-2013 the FFmpeg
	developers
	...
	Input #0, matroska,webm, from 'movie.mkv':
	  Duration: 01:32:11.25, start: 0.000000, bitrate: 1820 kb/s
	    Chapter #0.0: start 0.000000, end 250.124867
	    Metadata:
	      title           : Chapter 1
	    ...
	    Chapter #0.27: start 6099.134700, end 6373.450500
	    Metadata:
	      title           : Chapter 28
	    Stream #0:0(eng): Video: h264 (High), yuv420p, 720x358 [SAR
	    32:27 DAR 1280:537], SAR 186:157 DAR 66960:28103, 23.98 fps,
	    23.98 tbr, 1k tbn, 47.95 tbc (default)
	    Stream #0:1(eng): Audio: aac, 48000 Hz, stereo, fltp
	    (default)
	    Stream #0:2(eng): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448
	    kb/s
	[mpeg2video @ 0x21275e0] Automatically choosing VBV buffer size
	of 224 kbyte
	Output #0, mpegts, to 'pipe:':
	  Metadata:
	    encoder         : Lavf55.12.100
	    Chapter #0.0: start 0.000000, end 250.124867
	    Metadata:
	      title           : Chapter 1
	    ...
	    Chapter #0.27: start 6099.134700, end 6373.450500
	    Metadata:
	      title           : Chapter 28
	    Stream #0:0(eng): Video: mpeg2video, yuv420p, 720x358 [SAR
	    186:157 DAR 66960:28103], q=2-31, 2500 kb/s, 90k tbn, 29.97
	    tbc (default)
	    Stream #0:1(eng): Audio: ac3, 48000 Hz, 5.1(side), 448 kb/s
	Stream mapping:
	  Stream #0:0 -> #0:0 (h264 -> mpeg2video)
	  Stream #0:2 -> #0:1 (copy)
	Press [q] to stop, [?] for help
	Input #0, mpegts, from 'pipe:':    0KB vq=    0KB sq=    0B
	f=0/0   
	  Duration: N/A, start: 1.433367, bitrate: 448 kb/s
	  Program 1 
	    Metadata:
	      service_name    : Service01
	      service_provider: FFmpeg
	    Stream #0:0[0x100]: Video: mpeg2video (Main) ([2][0][0][0] /
	    0x0002), yuv420p, 720x358 [SAR 39559:36000 DAR 221:100],
	    29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
	    Stream #0:1[0x101](eng): Audio: ac3 ([129][0][0][0] /
	    0x0081), 48000 Hz, 5.1(side), fltp, 448 kb/s
	   ...


In BOTH cases, the L/R/PgUp/PgDown controls allow chapter skips/reverse.
 But even though they seem to work, occassional -- not every -- presses
result in these messages at the console

	...
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	File is broken, keyframes not correctly marked!KB sq=    0B
	f=0/0   
	...

Does the fact that the controls work at all necessarily mean that the
problem of non-functioning action controls MUST be caused by the DLNA
server?  Seems odd as the server only pipes the input/output through the
ffmpeg command.

Or, do the "File is broken" messages indicate that there IS a problem in
the stream itself or with the transcoding by ffmpeg?

From michaeljthorpe at yahoo.com  Thu Aug 22 22:55:02 2013
From: michaeljthorpe at yahoo.com (Mike Thorpe)
Date: Thu, 22 Aug 2013 13:55:02 -0700 (PDT)
Subject: [FFmpeg-user] FFmpeg to youtube live cannot connect
Message-ID: <1377204902.13154.YahooMailNeo@web140901.mail.bf1.yahoo.com>

I am trying to do a live stream from a hardware based Epiphan Broadcaster DL encoder through FFmpeg to youtube live. ?The Epiphan hardware encoder can publish with RTSP announce but not RTMP as required by youtube. ?The FFmpeg session seems to run fine but the video can't be previewed on youtube and youtube control room reports they are not receiving data. ?Any suggestions would be appreciated. ?The uncut console output and youtubes recommended settings are below.

Thanks,
Mike

youtube recommended settings

Encoder settings
Protocol: RTMP Flash Streaming
Video codec: H.264, Main 4.1
Frame rate: 30 fps
Keyframe frequency: 2 seconds
Audio codec: AAC-LC (Audio required)
Audio sample rate: 44.1 KHz
Audio bitrate: 128 Kbps stereo

Recommended advanced settings
Pixel aspect ratio: Square
Frame types: Progressive Scan, 2 B-Frames, 1 Reference Frame
Entropy coding: CABAC

Youtube live

C:\Users\admin>C:\Utils\ffmpeg\bin\ffmpeg -i "rtsp://(encIP):554/stream.sdp" -c:v copy -c:a copy -f flv "rtmp:/
/a.rtmp.youtube.com/live2/(youtubestreamname)"
ffmpeg version N-54901-g55db06a Copyright (c) 2000-2013 the FFmpeg developers
? built on Jul 23 2013 18:05:52 with gcc 4.7.3 (GCC)
? configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfi
g --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopenco
re-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-lib
vpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
? libavutil ? ? ?52. 40.100 / 52. 40.100
? libavcodec ? ? 55. 19.100 / 55. 19.100
? libavformat ? ?55. 12.102 / 55. 12.102
? libavdevice ? ?55. ?3.100 / 55. ?3.100
? libavfilter ? ? 3. 81.102 / ?3. 81.102
? libswscale ? ? ?2. ?4.100 / ?2. ?4.100
? libswresample ? 0. 17.103 / ?0. 17.103
? libpostproc ? ?52. ?3.100 / 52. ?3.100
Input #0, rtsp, from 'rtsp://(encIP):554/stream.sdp':
? Metadata:
? ? title ? ? ? ? ? : (null)
? Duration: N/A, start: 0.053000, bitrate: N/A
? ? Stream #0:0: Audio: aac, 16000 Hz, mono, fltp
? ? Stream #0:1: Video: h264 (Main), yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], 15 fps, 30 tbr, 90k tbn, 30 tbc
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/(youtubestreamname)':
? Metadata:
? ? title ? ? ? ? ? : (null)
? ? encoder ? ? ? ? : Lavf55.12.102
? ? Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 15 fps, 1k tbn, 90k
?tbc
? ? Stream #0:1: Audio: aac ([10][0][0][0] / 0x000A), 16000 Hz, mono
Stream mapping:
? Stream #0:1 -> #0:0 (copy)
? Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
[flv @ 0000000002719140] Non-monotonous DTS in output stream 0:0; previous: 114, current: -20; changing to 114. This may
?result in incorrect timestamps in the output file.
[flv @ 0000000002719140] Non-monotonous DTS in output stream 0:0; previous: 114, current: 14; changing to 114. This may
result in incorrect timestamps in the output file.
[flv @ 0000000002719140] Non-monotonous DTS in output stream 0:0; previous: 114, current: 47; changing to 114. This may
result in incorrect timestamps in the output file.
[flv @ 0000000002719140] Non-monotonous DTS in output stream 0:0; previous: 114, current: 80; changing to 114. This may
result in incorrect timestamps in the output file.
[NULL @ 000000000271fb60] RTP: missed 119 packets
[flv @ 0000000002719140] Failed to update header with correct duration.9.0kbits/s
[flv @ 0000000002719140] Failed to update header with correct filesize.
frame= 3835 fps= 30 q=-1.0 Lsize= ? 55754kB time=00:02:08.08 bitrate=3566.0kbits/s
video:54659kB audio:987kB subtitle:0 global headers:0kB muxing overhead 0.195093%

From nickrobbins at yahoo.com  Thu Aug 22 23:57:33 2013
From: nickrobbins at yahoo.com (Nicholas Robbins)
Date: Thu, 22 Aug 2013 14:57:33 -0700 (PDT)
Subject: [FFmpeg-user] Converted DVDs in .mkv format lose Pause,
	FastForward, Skip, etc functions when I restream as mpeg to DLNA
In-Reply-To: <1377202233.17373.13021077.2FFF4C8D@webmail.messagingengine.com>
References: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>	<1377190724.6609.12950137.7EF80BA2@webmail.messagingengine.com>	<CAHmvugc1JmasNCjQ0WFFdcpR2bjFnCvRdrfhupwdkc2f2Suj4A@mail.gmail.com>	<1377193949.19582.12971037.3EF167CA@webmail.messagingengine.com>	<1377198980.6327.13000877.2061DB7D@webmail.messagingengine.com>
	<1377202233.17373.13021077.2FFF4C8D@webmail.messagingengine.com>
Message-ID: <1377208653.34746.YahooMailNeo@web160803.mail.bf1.yahoo.com>

I don't know how helpful this is, but I was having similar problems at one point. I solved them by running the mkv's through mkvmerge one time with a command like:

mkvmerge -o out.mkv?--engage no_simpleblocks in.mkv

and I think that fixed it. It takes very little time (~1min/GB.)

-Nick


----- Original Message -----
From: "jen142 at promessage.com" <jen142 at promessage.com>
To: ffmpeg-user at ffmpeg.org
Cc: 
Sent: Thursday, August 22, 2013 4:10 PM
Subject: Re: [FFmpeg-user] Converted DVDs in .mkv format lose Pause, FastForward, Skip, etc functions when I restream as mpeg to DLNA

Continuing to try to remove variables from the situation, I'm now using
only ffmpeg/ffplay for streaming & playback.? I'm hoping someone who
knows more about this can actually help with interpreting or more
suggestions.

Since ffplay has such rudimentary motion controls -- L/R/PgUp/PgDwn --
compared to VLC or the standalone BluRay player, I'm not sure if what
follows actually tells me anything useful.

If I play the native file

??? ffplay -i movie.mkv

it plays without error.

If I play the transcoded stream

??? ffmpeg -i movie.mkv -f mpegts -b:v 2500k -minrate 2500k -maxrate
??? 2800k -r 29.97 -g 300 -bf 2 -bsf:v h264_mp4toannexb -vcodec
??? mpeg2video -acodec copy - | ffplay -

it also plays without error.

In case it's useful, the output at the console in the transcoded test is


??? ffmpeg -i movie.mkv -f mpegts -b:v 2500k -minrate 2500k -maxrate
??? 2500k -r 29.97 -g 300 -bf 2 -bsf:v h264_mp4toannexb -threads 2
??? -vcodec mpeg2video -acodec copy - | ffplay -
??? ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg
??? developers
??? ...
??? ffplay version 2.0.1 Copyright (c) 2003-2013 the FFmpeg
??? developers
??? ...
??? Input #0, matroska,webm, from 'movie.mkv':
??? ? Duration: 01:32:11.25, start: 0.000000, bitrate: 1820 kb/s
??? ? ? Chapter #0.0: start 0.000000, end 250.124867
??? ? ? Metadata:
??? ? ? ? title? ? ? ? ???: Chapter 1
??? ? ? ...
??? ? ? Chapter #0.27: start 6099.134700, end 6373.450500
??? ? ? Metadata:
??? ? ? ? title? ? ? ? ???: Chapter 28
??? ? ? Stream #0:0(eng): Video: h264 (High), yuv420p, 720x358 [SAR
??? ? ? 32:27 DAR 1280:537], SAR 186:157 DAR 66960:28103, 23.98 fps,
??? ? ? 23.98 tbr, 1k tbn, 47.95 tbc (default)
??? ? ? Stream #0:1(eng): Audio: aac, 48000 Hz, stereo, fltp
??? ? ? (default)
??? ? ? Stream #0:2(eng): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448
??? ? ? kb/s
??? [mpeg2video @ 0x21275e0] Automatically choosing VBV buffer size
??? of 224 kbyte
??? Output #0, mpegts, to 'pipe:':
??? ? Metadata:
??? ? ? encoder? ? ? ???: Lavf55.12.100
??? ? ? Chapter #0.0: start 0.000000, end 250.124867
??? ? ? Metadata:
??? ? ? ? title? ? ? ? ???: Chapter 1
??? ? ? ...
??? ? ? Chapter #0.27: start 6099.134700, end 6373.450500
??? ? ? Metadata:
??? ? ? ? title? ? ? ? ???: Chapter 28
??? ? ? Stream #0:0(eng): Video: mpeg2video, yuv420p, 720x358 [SAR
??? ? ? 186:157 DAR 66960:28103], q=2-31, 2500 kb/s, 90k tbn, 29.97
??? ? ? tbc (default)
??? ? ? Stream #0:1(eng): Audio: ac3, 48000 Hz, 5.1(side), 448 kb/s
??? Stream mapping:
??? ? Stream #0:0 -> #0:0 (h264 -> mpeg2video)
??? ? Stream #0:2 -> #0:1 (copy)
??? Press [q] to stop, [?] for help
??? Input #0, mpegts, from 'pipe:':? ? 0KB vq=? ? 0KB sq=? ? 0B
??? f=0/0? 
??? ? Duration: N/A, start: 1.433367, bitrate: 448 kb/s
??? ? Program 1 
??? ? ? Metadata:
??? ? ? ? service_name? ? : Service01
??? ? ? ? service_provider: FFmpeg
??? ? ? Stream #0:0[0x100]: Video: mpeg2video (Main) ([2][0][0][0] /
??? ? ? 0x0002), yuv420p, 720x358 [SAR 39559:36000 DAR 221:100],
??? ? ? 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
??? ? ? Stream #0:1[0x101](eng): Audio: ac3 ([129][0][0][0] /
??? ? ? 0x0081), 48000 Hz, 5.1(side), fltp, 448 kb/s
??? ???...


In BOTH cases, the L/R/PgUp/PgDown controls allow chapter skips/reverse.
But even though they seem to work, occassional -- not every -- presses
result in these messages at the console

??? ...
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? File is broken, keyframes not correctly marked!KB sq=? ? 0B
??? f=0/0? 
??? ...

Does the fact that the controls work at all necessarily mean that the
problem of non-functioning action controls MUST be caused by the DLNA
server?? Seems odd as the server only pipes the input/output through the
ffmpeg command.

Or, do the "File is broken" messages indicate that there IS a problem in
the stream itself or with the transcoding by ffmpeg?
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From d-fens at 00010111.de  Fri Aug 23 00:21:09 2013
From: d-fens at 00010111.de (d-fens)
Date: Fri, 23 Aug 2013 00:21:09 +0200
Subject: [FFmpeg-user] ~50% performance difference overlaying static image
 with png series using -vf/-filter_complex
Message-ID: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>

hi,

i have a background png and a sequence of pngs with transparency which  
should overlay the background.

when i use

./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32, split  
[T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo [T2]"  
out.mp4

(taken from  
http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
i get arround 13fps for fullHD images, but using

./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex  
overlay=shortest=1 out.mp4

i get around 7fps for the same images.
I guess the background image gets re-read here but thats just a guess;  
as the second, newer version looks much more clean and simple i wonder  
whats the way to get best performance in this scenario?

greets!

From onemda at gmail.com  Fri Aug 23 01:11:42 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 22 Aug 2013 23:11:42 +0000
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
Message-ID: <CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>

On 8/22/13, d-fens <d-fens at 00010111.de> wrote:
> hi,
>
> i have a background png and a sequence of pngs with transparency which
> should overlay the background.
>
> when i use
>
> ./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32, split
> [T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo [T2]"
> out.mp4
>
> (taken from
> http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
> i get arround 13fps for fullHD images, but using
>
> ./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex
> overlay=shortest=1 out.mp4

Try removing loop and shortest.

>
> i get around 7fps for the same images.
> I guess the background image gets re-read here but thats just a guess;
> as the second, newer version looks much more clean and simple i wonder
> whats the way to get best performance in this scenario?
>
> greets!
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Fri Aug 23 01:25:42 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 22 Aug 2013 23:25:42 +0000 (UTC)
Subject: [FFmpeg-user] Delayed sound in screencast
References: <5213C125.6060706@gmail.com>
	<loom.20130821T120936-514@post.gmane.org>
	<5214A1B6.5090807@gmail.com>
	<loom.20130821T135044-175@post.gmane.org>
	<5214B9A7.8010102@gmail.com>
	<loom.20130822T001937-401@post.gmane.org>
	<5215E534.9040503@gmail.com>
Message-ID: <loom.20130823T012349-639@post.gmane.org>

Artur ??cki <alacki93 <at> gmail.com> writes:

> It looks for example like audio with length 10 seconds 
> was "expanded" to video who takes 20 seconds (I can't 
> describe this any better).

Please test the following:

$ ffmpeg -f alsa -ac 2 -i default out2.wav
$ ffmpeg -f alsa -ac 1 -i default out1.wav
How do out1.wav and out2.wav sound?

> In meanwhile I've tried many "magical" scripts but no 
> one helped.

But you didn't tell us which scripts are "magical"...

Carl Eugen


From basteon at gmail.com  Fri Aug 23 03:27:08 2013
From: basteon at gmail.com (basteon)
Date: Fri, 23 Aug 2013 01:27:08 +0000
Subject: [FFmpeg-user] Problem with playing audio
In-Reply-To: <25563E537418E342A6DDC83C793113FC100F5404@SIXPRD0410MB359.apcprd04.prod.outlook.com>
References: <25563E537418E342A6DDC83C793113FC100F5404@SIXPRD0410MB359.apcprd04.prod.outlook.com>
Message-ID: <CAEJHKkWoGKkvRB1oQ9taKOn0zL59xcXqQ5HGsrCcFuZhvAv+Vw@mail.gmail.com>

hi,
You must use
packet.pos = 0;
or try verbosive logging


On 20 August 2013 14:04, Anitha Susan Varghese
<anithasusan at tataelxsi.co.in>wrote:

> Hi
>
> Myself trying to play an audio file using libavcodec in qt5 .While trying
> to play i am unable to play the the file.
> while decoding the file using av_read_frame()
> audioStream->index=0
> and packet.stream_index is some higher nos .so avcodec_decode_audio4() is
> not getting executed.Why is it so?
>
> This is my program. My output is also given below.Please help me in
> solving this.
>
> -------------------------------------------------------------------------------------------------------------------------------------
> #include <QApplication>
> #include <QtDebug>
> #include <ao/ao.h>
>
> extern "C" {
> #include "libavutil/samplefmt.h"
> #include "libavcodec/avcodec.h"
> #include "libavutil/mathematics.h"
> #include "libavutil/avutil.h"
> #include "libavformat/avformat.h"
> #include "libswscale/swscale.h"
> #include "libavutil/dict.h"
> #include <ao/ao.h>
> #include "libavformat/version.h"
> }
> #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
> #define CODEC_TYPE_VIDEO AVMEDIA_TYPE_VIDEO
>
>
>
> void die( const char* msg)
> {
> qDebug() <<  msg ;
> }
>
> int main(int argc, char *argv[])
> {
>     QApplication app(argc, argv);
>     const char* input_filename=argv[1];
>
>     static int driver;
>     static ao_device *audio_device;
>     int i;
>     int stream_id=-1;
>     av_register_all();
>     avcodec_register_all();
>     AVFormatContext* container=NULL;
>     AVStream* audioStream=NULL;
>        //container=avformat_alloc_context();
>
>      if(avformat_open_input(&container,input_filename,NULL,NULL)<0){
>                 die("Could not open file");
>         }
>        else
>        die("File Opend");
>
>
>     if(avformat_find_stream_info(container,NULL)<0){
>                 die("Could not find file info");
>         }
>
>
>     av_dump_format(container,0,input_filename,false);
>     for(i=0; i < container->nb_streams; i++){
>
>         if(container->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO){
>                 stream_id=i;
>                 audioStream=container->streams[i];
>
>                 qWarning() <<"audioStream"<<audioStream;
>                 qWarning() <<"Stream id:"<<stream_id;
>                  break;
>                  }
>         }
>
>     if(stream_id==-1){
>         die("Could not find Audio Stream");
>     }
>
>
>     AVCodecContext *ctx=container->streams[stream_id]->codec;
>     qWarning() <<"Codec ptr:"<< ctx;
>     AVCodec *codec=avcodec_find_decoder(ctx->codec_id);
>  if(codec==NULL){
>         die("cannot find codec!");
>    }
>
>   if(avcodec_open2(ctx,codec,NULL)<0){
>         die("Codec cannot be opended!");
>    }
>    else
>         qWarning()<<"Codec Opened";
>
>
>     ao_initialize();
>     driver = ao_default_driver_id();
>     qWarning()<<"DRIVER ID" <<driver;
>    ao_sample_format sformat;
>    AVPacket dummy_packet;
>    av_read_frame(container,&dummy_packet);
>    AVSampleFormat sfmt=ctx->sample_fmt;
> //assign device sample rate depend on the input stream
>
>    qWarning()<<"Sample format"<<sfmt;
>
>    if(sfmt==AV_SAMPLE_FMT_U8){
>         sformat.bits=8;
>         qWarning()<<"Sample format"<<sformat.bits;
>    }
>    else if(sfmt==AV_SAMPLE_FMT_S16){
>         sformat.bits=16;
>         qWarning()<<"Sample format"<<sformat.bits;
>   }
>   else if(sfmt==AV_SAMPLE_FMT_S32){
>         sformat.bits=32;
>         qWarning()<<"Sample format:AV_SAMPLE_FMT_S32"<<sformat.bits;
>   }
> else if(sfmt==AV_SAMPLE_FMT_FLT){
>         sformat.bits=32;
>         qWarning()<<"Sample format:AV_SAMPLE_FMT_FLT"<<sformat.bits;
>  else if(sfmt==AV_SAMPLE_FMT_DBL){
>         sformat.bits=64;
>         qWarning()<<"Sample format"<<sformat.bits;
>   }
>   else if(sfmt==AV_SAMPLE_FMT_U8P){
>         sformat.bits=8;
>         qWarning()<<"Sample format"<<sformat.bits;
>   }
>   else if(sfmt==AV_SAMPLE_FMT_S16P){
>         sformat.bits=16;
>         qWarning()<<"Sample format"<<sformat.bits;
>   }
>
>    else if(sfmt==AV_SAMPLE_FMT_S32P){
>         sformat.bits=32;
>         qWarning()<<"Sample format:AV_SAMPLE_FMT_S32P"<<sformat.bits;
>   }
>   else if(sfmt==AV_SAMPLE_FMT_FLTP){
>         sformat.bits=32;
>         qWarning()<<"Sample format:AV_SAMPLE_FMT_FLTP"<<sformat.bits;
>   }
>  else if(sfmt==AV_SAMPLE_FMT_DBLP){
>         sformat.bits=64;
>         qWarning()<<"Sample format"<<sformat.bits;
>   }
>
>
>    sformat.channels=ctx->channels;
>
>   //sformat.channels=2;
>   qWarning()<<"Sample format channels"<<sformat.channels;
>
>   sformat.rate=ctx->sample_rate;
>   qWarning()<<"Sample format rate"<<sformat.rate;
>
>   sformat.byte_format=AO_FMT_NATIVE;
>   qWarning()<<"Sample format "<<sformat.byte_format;
>  sformat.matrix=0;
>
>  //seek back to the beginning of stream
>   av_seek_frame(container,stream_id,0,AVSEEK_FLAG_ANY);
>
>   audio_device=ao_open_live(driver,&sformat,NULL);
> //end of init AO LIB
>
> //data packet read from the stream
>   AVPacket packet;
>   av_init_packet(&packet);
>
>   int buffer_size=AVCODEC_MAX_AUDIO_FRAME_SIZE+
> FF_INPUT_BUFFER_PADDING_SIZE;;
>
>   qWarning()<<"buffersize "<<buffer_size;
>   uint8_t buffer[buffer_size];
>   packet.data=buffer;
>   packet.size =buffer_size;
>
> //frame ,where the decoded data will be written
>   AVFrame *frame=avcodec_alloc_frame();
>  int len;
>   int frameFinished=0;
>
>   qWarning()<<"CODEC"<<audioStream->codec;
>   while(av_read_frame(container,&packet)>=0)
>   {
> //        qWarning()<<audioStream->index;
>         if(packet.stream_index==audioStream->index){
>         len=avcodec_decode_audio4(ctx,frame,&frameFinished,&packet);
>         qWarning()<<"Enterd if loop";
>       //  qWarning() <<len;
>         if(frameFinished){
> //play the decoded byteis
>                 qWarning()<<"PLAYING";
>                 ao_play(audio_device,
> (char*)frame->extended_data[0],frame->linesize[0] );
>   }else{
> }
> }
>  avformat_close_input(&container);
>   ao_shutdown();
>   return app.exec();
> }
>
> ----------------------------------------------------------------------------------------------------------------------------------
> My output
> --------------------------------------------------------------------------------------------------------------------
>                     File Opend
> Input #0, ogg, from '/home/tel/Downloads/desktop-login.ogg':
>   Duration: 00:00:07.72, start: 0.000000, bitrate: 108 kb/s
>     Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp, 112 kb/s
> audioStream 0x88b3f20
> Stream id: 0
> Codec ptr: 0x88b4100
> Codec Opened
> DRIVER ID 1
> Sample format 3
> Sample format:AV_SAMPLE_FMT_FLT 32
> Sample format channels 0
> Sample format rate 0
> Sample format  4
> buffersize  192008
> CODEC 0x88b4100
>
>
>
> --------------------------------------------------------------------------------------------------------------
> Regards,
> Anitha Susan Varghese
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From basteon at gmail.com  Fri Aug 23 03:32:16 2013
From: basteon at gmail.com (basteon)
Date: Fri, 23 Aug 2013 01:32:16 +0000
Subject: [FFmpeg-user] ffmpeg 2 yum install
In-Reply-To: <SNT145-W57A6F73DE438274C2564FA96430@phx.gbl>
References: <SNT145-W928BF241879D1BDFEFE00696430@phx.gbl>
	<SNT145-W57A6F73DE438274C2564FA96430@phx.gbl>
Message-ID: <CAEJHKkUEb8NQrFMM32hJg3gXKgD1N68ArQ9z--WhmZiBUt5KHQ@mail.gmail.com>

Do you want install or make a package?
if first, that available in base repo, but sometimes packages are old.
ffmpeg-spice-devel.i686 : Development package for ffmpeg-spice
Repo        : base
Matched from:



ffmpeg-spice-libs.i686 : Libraries for ffmpeg-spice
Repo        : base
Matched from:



On 20 August 2013 08:40, Mr.Hackman 191 <mr_ax at hotmail.com> wrote:

> ffmpeg 2 yum install
>
> how?
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

From mirko.puliafito at gmail.com  Fri Aug 23 09:06:03 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Fri, 23 Aug 2013 09:06:03 +0200
Subject: [FFmpeg-user] RTMPE native support
Message-ID: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>

Hi,

I'm trying to use rtmpe (ffrtmpcrypt) with the latest version but it's not
working (I've also tested it with the 1.x). I tested it against crtmpserver
and red5 without any luck. Can anybody tell me the status of that protocol
implementation? Do I need to go on using rtmpdump libs if I want to use
this protocol?
 Mirko

From francois.visagie at gmail.com  Fri Aug 23 09:33:29 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Fri, 23 Aug 2013 09:33:29 +0200
Subject: [FFmpeg-user] Concatenation of mpg files
In-Reply-To: <CABOWChwWk6ksD56yAw8bwJ0JpLpMjnh2qh0Xgvt=RR51h3yiwQ@mail.gmail.com>
References: <CABOWChwWk6ksD56yAw8bwJ0JpLpMjnh2qh0Xgvt=RR51h3yiwQ@mail.gmail.com>
Message-ID: <005401ce9fd3$1550b650$3ff222f0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Dzung Nguyen
> Sent: 22 August 2013 20:13
> To: FFmpeg user questions
> Subject: [FFmpeg-user] Concatenation of mpg files
> 
> Hi,
> 
> I tried to cat in ubuntu shell mpg files, but the output file lost timing
> information (can't be played, can't seek)
> 
> This is mpg video information:
> 
>  ffmpeg -i 0.mpg
> ffmpeg version git-2013-08-06-a68b6ec Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Aug 21 2013 11:58:24 with gcc 4.6 (Ubuntu/Linaro
4.6.3-1ubuntu5)
>   libavutil      52. 41.100 / 52. 41.100
>   libavcodec     55. 22.100 / 55. 22.100
>   libavformat    55. 13.102 / 55. 13.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 82.100 /  3. 82.100
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.103 /  0. 17.103
>   libpostproc    52.  3.100 / 52.  3.100
> Input #0, mpeg, from '0.mpg':
>   Duration: 00:02:10.00, start: 0.533367, bitrate: 2899 kb/s
>     Stream #0:0[0x1e0]: Video: mpeg1video, yuv420p, 1280x720 [SAR 1:1 DAR
> 16:9], 104857 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 29.97 tbc
> 
> So it's mpeg1 encoded, everything is correct. The way I make this video is
> using command:
> 
> ffmpeg -loop 1 -r 1 -i image.png -q:v 0 -r 29.97 -t 10 0.mpg
> 
> What should I change here in the converting from image to video, so that
the
> mpg files are cattable?

Do I understand correctly that you're using ffmpeg to create individual mpg
files from images, and then you want to concatenate them?

If so, each individual file will have its own header information, and the
first's header information won't reflect the size/playing time of them all.
Therefore a straight-forward binary concatenation of the mpg files won't
work for most players. One way around that is to concatenate them with
ffmpeg, using the concat: protocol.

Kind regards,
Francois


From d-fens at 00010111.de  Fri Aug 23 10:09:57 2013
From: d-fens at 00010111.de (d-fens)
Date: Fri, 23 Aug 2013 10:09:57 +0200
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
	<CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
Message-ID: <20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>

Zitat von Paul B Mahol <onemda at gmail.com>:

> On 8/22/13, d-fens <d-fens at 00010111.de> wrote:
>> hi,
>>
>> i have a background png and a sequence of pngs with transparency which
>> should overlay the background.
>>
>> when i use
>>
>> ./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32, split
>> [T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo [T2]"
>> out.mp4
>>
>> (taken from
>> http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
>> i get arround 13fps for fullHD images, but using
>>
>> ./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex
>> overlay=shortest=1 out.mp4
>
> Try removing loop and shortest.


this just outputs one frame, so no success :(



From cehoyos at ag.or.at  Fri Aug 23 10:41:00 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 23 Aug 2013 08:41:00 +0000 (UTC)
Subject: [FFmpeg-user] Concatenation of mpg files
References: <CABOWChwWk6ksD56yAw8bwJ0JpLpMjnh2qh0Xgvt=RR51h3yiwQ@mail.gmail.com>
	<005401ce9fd3$1550b650$3ff222f0$@gmail.com>
Message-ID: <loom.20130823T103731-969@post.gmane.org>

Francois Visagie <francois.visagie <at> gmail.com> writes:

> > ffmpeg -loop 1 -r 1 -i image.png -q:v 0 -r 29.97 -t 10 0.mpg

> Do I understand correctly that you're using ffmpeg to 
> create individual mpg files from images, and then you 
> want to concatenate them?

> If so, each individual file will have its own header 
> information,

Program streams do not have headers.

> and the first's header information won't reflect the 
> size/playing time of them all.

> Therefore a straight-forward binary concatenation of 
> the mpg files won't work for most players.

I don't think this is correct (most players handle 
timestamp resets well - they have to if they want 
to be DVD-compatible - FFmpeg admittedly has 
problems), but in any case, it is not the main 
problem: -qscale 0 produces large frames and you 
have to tell the mpeg muxer about this (ffmpeg 
correctly warns that the output file may be 
unplayable).

Carl Eugen


From nicolas.george at normalesup.org  Fri Aug 23 11:27:49 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Fri, 23 Aug 2013 11:27:49 +0200
Subject: [FFmpeg-user] Concatenation of mpg files
In-Reply-To: <loom.20130823T103731-969@post.gmane.org>
References: <CABOWChwWk6ksD56yAw8bwJ0JpLpMjnh2qh0Xgvt=RR51h3yiwQ@mail.gmail.com>
	<005401ce9fd3$1550b650$3ff222f0$@gmail.com>
	<loom.20130823T103731-969@post.gmane.org>
Message-ID: <20130823092749.GA507@phare.normalesup.org>

Le sextidi 6 fructidor, an CCXXI, Carl Eugen Hoyos a ?crit?:
> I don't think this is correct (most players handle 
> timestamp resets well - they have to if they want 
> to be DVD-compatible - FFmpeg admittedly has 
> problems),

Actually, players that are DVD-compatible use the IFO files from the DVD
structure, not just the program streams, and the IFO files amount to a
global header. In particular, they hold a kind of seeking index.

Regards,

-- 
  Nicolas George
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From alacki93 at gmail.com  Fri Aug 23 11:45:14 2013
From: alacki93 at gmail.com (=?UTF-8?B?QXJ0dXIgxYHEhWNraQ==?=)
Date: Fri, 23 Aug 2013 11:45:14 +0200
Subject: [FFmpeg-user] Delayed sound in screencast
In-Reply-To: <loom.20130823T012349-639@post.gmane.org>
References: <5213C125.6060706@gmail.com>
	<loom.20130821T120936-514@post.gmane.org>
	<5214A1B6.5090807@gmail.com>
	<loom.20130821T135044-175@post.gmane.org>
	<5214B9A7.8010102@gmail.com>
	<loom.20130822T001937-401@post.gmane.org>
	<5215E534.9040503@gmail.com>
	<loom.20130823T012349-639@post.gmane.org>
Message-ID: <52172F2A.2080004@gmail.com>

W dniu 23.08.2013 01:25, Carl Eugen Hoyos pisze:

> Please test the following: $ ffmpeg -f alsa -ac 2 -i default out2.wav
> $ ffmpeg -f alsa -ac 1 -i default out1.wav How do out1.wav and
> out2.wav sound? 

Both files sounds good. Here are logs:

#########################################

artur at laptop:~/Wideo> ffmpeg -f alsa -ac 1 -i default out1.wav
ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug 17 2013 07:01:31 with gcc 4.7 (SUSE Linux)
  configuration: --shlibdir=/usr/lib64 --prefix=/usr
--mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared
--disable-static --enable-debug --disable-stripping
--extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2
-fstack-protector -funwind-tables -fasynchronous-unwind-tables -fPIC
-I/usr/include/gsm' --enable-gpl --enable-x11grab --enable-version3
--enable-pthreads --enable-avfilter --enable-libpulse --enable-libvpx
--enable-libopus --enable-libass --enable-libmp3lame --enable-libvorbis
--enable-libtheora --enable-libspeex --enable-libxvid --enable-libx264
--enable-libschroedinger --enable-libgsm --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-postproc --enable-libdc1394
--enable-librtmp --enable-libfreetype --enable-avresample
--enable-libtwolame --enable-libvo-aacenc
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 79.101 /  3. 79.101
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, alsa, from 'default':
  Duration: N/A, start: 1377250478.523785, bitrate: 768 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s
File 'out1.wav' already exists. Overwrite ? [y/N] y
Output #0, wav, to 'out1.wav':
  Metadata:
    ISFT            : Lavf55.12.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
mono, s16, 768 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size=    1950kB time=00:00:20.79 bitrate= 768.0kbits/s   
video:0kB audio:1950kB subtitle:0 global headers:0kB muxing overhead
0.004007%

#########################################

artur at laptop:~/Wideo> ffmpeg -f alsa -ac 2 -i default out2.wav
ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug 17 2013 07:01:31 with gcc 4.7 (SUSE Linux)
  configuration: --shlibdir=/usr/lib64 --prefix=/usr
--mandir=/usr/share/man --libdir=/usr/lib64 --enable-shared
--disable-static --enable-debug --disable-stripping
--extra-cflags='-fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2
-fstack-protector -funwind-tables -fasynchronous-unwind-tables -fPIC
-I/usr/include/gsm' --enable-gpl --enable-x11grab --enable-version3
--enable-pthreads --enable-avfilter --enable-libpulse --enable-libvpx
--enable-libopus --enable-libass --enable-libmp3lame --enable-libvorbis
--enable-libtheora --enable-libspeex --enable-libxvid --enable-libx264
--enable-libschroedinger --enable-libgsm --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-postproc --enable-libdc1394
--enable-librtmp --enable-libfreetype --enable-avresample
--enable-libtwolame --enable-libvo-aacenc
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 79.101 /  3. 79.101
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, alsa, from 'default':
  Duration: N/A, start: 1377250501.230443, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
File 'out2.wav' already exists. Overwrite ? [y/N] y
Output #0, wav, to 'out2.wav':
  Metadata:
    ISFT            : Lavf55.12.100
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo, s16, 1536 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size=    4053kB time=00:00:21.61 bitrate=1536.0kbits/s   
video:0kB audio:4053kB subtitle:0 global headers:0kB muxing overhead
0.001927%

#########################################

> But you didn't tell us which scripts are "magical"...
>

Here is example "magical" script (source
http://dl.dropboxusercontent.com/u/28788188/cast.sh )

##############################ffmpeg/avconv Screencast script###################################################################################################################
#--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
#packages needed: libavcodec-extra-xx libvorbis libx264-xx ffmpeg (ubuntu package names); libavc1394 ffmpeg gstreamer0.10-ffmpeg x264(arch packages)
#--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
#PROTIP: add keyboard shortcuts for starting the script like: /your/script/location/cast.sh and stoping shortcut with the command: killall ffmpeg (or avconv if you using that)
#--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
#script by Xpander
#--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

#date function
DATE=`which date`

#mount ramdisk (uncomment and change if you want to use ramdisk - dont forget to change directory as well)
#gksu mount -t tmpfs -o size=3096M tmpfs /tmp/ramdisk/

#package select (avconv or ffmpeg)
RECORDER=ffmpeg
#How many threads used (0 for automatic)
THREADS=0
#Resolution
RESO=1920x1080
#Xserver Display number(:0.0 is default)
XDISP=:0.0
#Audio Device
AUDIO=alsa
#Channels
CHANNELS=2
#SoundCard (pulse for pulseaudio, hw:0,1 for directly communicating with your soundcard - use aplay -l to see whats your value)
SOUNDCARD=pulse
#Frames per second
FPS=30
#Constant Rate Factor(0 is the highest quality 50 is the lowest)
CRF=20
#libx264 presets(slow, fast, superfast, ultrafast; additionaly check sudo find /usr -iname '*.ffpreset' for more)
PRESET=ultrafast
#Audio Codec (libmp3lame, pcm_s16le or libvorbis are most common)
ACODEC=libvorbis
#Directory where your video is gonna be saved.(include / at the end)
DIRECTORY=$HOME/Videos/
#File name
FILENAME=videocast`$DATE +%d%m%Y_%H.%M.%S`.mp4

#script
$RECORDER -f $AUDIO -ac $CHANNELS -i $SOUNDCARD -f x11grab -r $FPS -s $RESO -i $XDISP -vcodec libx264 -preset $PRESET -crf $CRF -acodec $ACODEC -ab 256k -f mp4 -threads $THREADS $DIRECTORY$FILENAME


#old examples
##sample:ffmpeg -f alsa -ac 2 -i hw:0,1 -f x11grab -r 30 -s 1280x1024 -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 output.avi
#sample2:avconv -f alsa -i pulse -f x11grab -r 25 -s 1920x1080 -i :0.0 -vcodec mpeg4 -b 12000k -g 300 -bf 2 -acodec libmp3lame -ab 256k Screencast.avi
#sample3:ffmpeg -threads 0 -f alsa -i pulse -f x11grab -s 1280x720 -r 30 -i :0.0+0,0 -vcodec libx264 -preset superfast -crf 16 -acodec libmp3lame -ab 256k -ar 44100 -f mp4 screencast.mp4



-- 

===========
Pozdrawiam
Artur ??cki
===========


From onemda at gmail.com  Fri Aug 23 11:59:20 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 23 Aug 2013 09:59:20 +0000
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
	<CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
	<20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>
Message-ID: <CAPYw7P5grQBk81Y_V1+yJZGXccFdQXB2Z5C5Rjgbk6mC=o-j0Q@mail.gmail.com>

On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
> Zitat von Paul B Mahol <onemda at gmail.com>:
>
>> On 8/22/13, d-fens <d-fens at 00010111.de> wrote:
>>> hi,
>>>
>>> i have a background png and a sequence of pngs with transparency which
>>> should overlay the background.
>>>
>>> when i use
>>>
>>> ./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32, split
>>> [T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo [T2]"
>>> out.mp4
>>>
>>> (taken from
>>> http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
>>> i get arround 13fps for fullHD images, but using
>>>
>>> ./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex
>>> overlay=shortest=1 out.mp4
>>
>> Try removing loop and shortest.
>
>
> this just outputs one frame, so no success :(

Than change order of streams - make bg.png 2nd.
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Fri Aug 23 12:05:35 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 23 Aug 2013 10:05:35 +0000
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
Message-ID: <CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>

On 8/23/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> Hi,
>
> I'm trying to use rtmpe (ffrtmpcrypt) with the latest version but it's not
> working (I've also tested it with the 1.x). I tested it against crtmpserver
> and red5 without any luck. Can anybody tell me the status of that protocol
> implementation? Do I need to go on using rtmpdump libs if I want to use
> this protocol?

What does not work exactly?

There is simple not enough information provided.

From d-fens at 00010111.de  Fri Aug 23 12:38:46 2013
From: d-fens at 00010111.de (d-fens)
Date: Fri, 23 Aug 2013 12:38:46 +0200
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <CAPYw7P5grQBk81Y_V1+yJZGXccFdQXB2Z5C5Rjgbk6mC=o-j0Q@mail.gmail.com>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
	<CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
	<20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>
	<CAPYw7P5grQBk81Y_V1+yJZGXccFdQXB2Z5C5Rjgbk6mC=o-j0Q@mail.gmail.com>
Message-ID: <20130823123846.Horde.qnsn9ufeoi9LvK8grPJDCQ1@aktionspotenzial.de>

Zitat von Paul B Mahol <onemda at gmail.com>:

> On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
>> Zitat von Paul B Mahol <onemda at gmail.com>:
>>
>>> On 8/22/13, d-fens <d-fens at 00010111.de> wrote:
>>>> hi,
>>>>
>>>> i have a background png and a sequence of pngs with transparency which
>>>> should overlay the background.
>>>>
>>>> when i use
>>>>
>>>> ./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32, split
>>>> [T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo [T2]"
>>>> out.mp4
>>>>
>>>> (taken from
>>>> http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
>>>> i get arround 13fps for fullHD images, but using
>>>>
>>>> ./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex
>>>> overlay=shortest=1 out.mp4
>>>
>>> Try removing loop and shortest.
>>
>>
>> this just outputs one frame, so no success :(
>
> Than change order of streams - make bg.png 2nd.

well that puts the background over the sequence, its fast (15fps  
processing) though :| but very static ...

From onemda at gmail.com  Fri Aug 23 12:51:33 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 23 Aug 2013 10:51:33 +0000
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <20130823123846.Horde.qnsn9ufeoi9LvK8grPJDCQ1@aktionspotenzial.de>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
	<CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
	<20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>
	<CAPYw7P5grQBk81Y_V1+yJZGXccFdQXB2Z5C5Rjgbk6mC=o-j0Q@mail.gmail.com>
	<20130823123846.Horde.qnsn9ufeoi9LvK8grPJDCQ1@aktionspotenzial.de>
Message-ID: <CAPYw7P7_RCvzvCgLJSQvZhNCTpx0vYW2xvUOmN+70dKZBV06BA@mail.gmail.com>

On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
> Zitat von Paul B Mahol <onemda at gmail.com>:
>
>> On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
>>> Zitat von Paul B Mahol <onemda at gmail.com>:
>>>
>>>> On 8/22/13, d-fens <d-fens at 00010111.de> wrote:
>>>>> hi,
>>>>>
>>>>> i have a background png and a sequence of pngs with transparency which
>>>>> should overlay the background.
>>>>>
>>>>> when i use
>>>>>
>>>>> ./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32, split
>>>>> [T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo [T2]"
>>>>> out.mp4
>>>>>
>>>>> (taken from
>>>>> http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
>>>>> i get arround 13fps for fullHD images, but using
>>>>>
>>>>> ./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex
>>>>> overlay=shortest=1 out.mp4
>>>>
>>>> Try removing loop and shortest.
>>>
>>>
>>> this just outputs one frame, so no success :(
>>
>> Than change order of streams - make bg.png 2nd.
>
> well that puts the background over the sequence, its fast (15fps
> processing) though :| but very static ...

I didn't benchmark but I assume -loop takes most of time as it decodes
same file over and over again.
So instead adding looping of main input inside overlay it would be
better to add generic solution:
repeat filter which would repeat each frame [0-inf] times.

> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From 89.yavuz at gmail.com  Fri Aug 23 13:06:24 2013
From: 89.yavuz at gmail.com (Mustafa Yavuz)
Date: Fri, 23 Aug 2013 14:06:24 +0300
Subject: [FFmpeg-user] Audio encoding problem
Message-ID: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>

Hi,
I would like to encode decoded frames in signed 16 bit little endian
format. By input audio is in f32le format, 1 channels and 48000 sampling
rate. When I run the code below it just lower the size of file by half,
that only number of channels affects encoding. Sampling rate that I
assigned enc_ctx->sample_rate variable has  no effect. I do not why it does
not work. I also do not want to use resampling function which are
deprecated. I have also examined and tried the code in your offical website
and it did not work for some reasons, I asked it in stackoverflow
here<http://stackoverflow.com/questions/18383442/encoding-by-using-ffmpeg-library>
.

Code is here <http://codepaste.net/rgqpc9>

Thanks..

From d-fens at 00010111.de  Fri Aug 23 13:55:37 2013
From: d-fens at 00010111.de (d-fens)
Date: Fri, 23 Aug 2013 13:55:37 +0200
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <CAPYw7P7_RCvzvCgLJSQvZhNCTpx0vYW2xvUOmN+70dKZBV06BA@mail.gmail.com>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
	<CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
	<20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>
	<CAPYw7P5grQBk81Y_V1+yJZGXccFdQXB2Z5C5Rjgbk6mC=o-j0Q@mail.gmail.com>
	<20130823123846.Horde.qnsn9ufeoi9LvK8grPJDCQ1@aktionspotenzial.de>
	<CAPYw7P7_RCvzvCgLJSQvZhNCTpx0vYW2xvUOmN+70dKZBV06BA@mail.gmail.com>
Message-ID: <20130823135537.Horde.A96VatT6jIgUSj8p9vUvGQ2@aktionspotenzial.de>

Zitat von Paul B Mahol <onemda at gmail.com>:

> On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
>> Zitat von Paul B Mahol <onemda at gmail.com>:
>>
>>> On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
>>>> Zitat von Paul B Mahol <onemda at gmail.com>:
>>>>
>>>>> On 8/22/13, d-fens <d-fens at 00010111.de> wrote:
>>>>>> hi,
>>>>>>
>>>>>> i have a background png and a sequence of pngs with transparency which
>>>>>> should overlay the background.
>>>>>>
>>>>>> when i use
>>>>>>
>>>>>> ./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32, split
>>>>>> [T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo [T2]"
>>>>>> out.mp4
>>>>>>
>>>>>> (taken from
>>>>>> http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
>>>>>> i get arround 13fps for fullHD images, but using
>>>>>>
>>>>>> ./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex
>>>>>> overlay=shortest=1 out.mp4
>>>>>
>>>>> Try removing loop and shortest.
>>>>
>>>>
>>>> this just outputs one frame, so no success :(
>>>
>>> Than change order of streams - make bg.png 2nd.
>>
>> well that puts the background over the sequence, its fast (15fps
>> processing) though :| but very static ...
>
> I didn't benchmark but I assume -loop takes most of time as it decodes
> same file over and over again.
> So instead adding looping of main input inside overlay it would be
> better to add generic solution:
> repeat filter which would repeat each frame [0-inf] times.
>
  yes, i also think the png gets re-read in a loop, maybe
http://git.videolan.org/?p=ffmpeg.git;a=commit;h=6e643239d9956ce32888d87451706ee7c8757afa speeds it up a bit, but a repeat filter would be great or loop without the overhead. But just to be clear: there is no way to archive this with current -filter_complex  
yet?

From nicolas.george at normalesup.org  Fri Aug 23 14:01:28 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Fri, 23 Aug 2013 14:01:28 +0200
Subject: [FFmpeg-user] Audio encoding problem
In-Reply-To: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>
References: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>
Message-ID: <20130823120128.GA30358@phare.normalesup.org>

Le sextidi 6 fructidor, an CCXXI, Mustafa Yavuz a ?crit?:
> I would like to encode decoded frames in signed 16 bit little endian
> format.

You forgot to specify: PCM.

> Code is here <http://codepaste.net/rgqpc9>

#    enc_ctx->sample_fmt = AV_SAMPLE_FMT_S16;
#    enc_ctx->bit_rate = 64000;
#    enc_ctx->sample_rate    = 11025;
#    enc_ctx->channel_layout = AV_CH_LAYOUT_MONO ;
#    enc_ctx->channels       = 1;

You got the wrong idea that this is asking the PCM encoder to encode at a
specific sample rate and with a specific channel count. It does not work
that way. You are saying to the encoder what you are giving it as input.
Since you are actually giving it something completely different, it does not
work at all.

You have to convert and remix the audio stream. For that, you can use
libswresample directly or libavfilter.

By the way, you will never achieve 64000 bit/s with these settings, your
bit_rate field is completely wrong.

Regards,

-- 
  Nicolas George
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From 89.yavuz at gmail.com  Fri Aug 23 14:20:45 2013
From: 89.yavuz at gmail.com (Mustafa Yavuz)
Date: Fri, 23 Aug 2013 15:20:45 +0300
Subject: [FFmpeg-user] Audio encoding problem
In-Reply-To: <20130823120128.GA30358@phare.normalesup.org>
References: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>
	<20130823120128.GA30358@phare.normalesup.org>
Message-ID: <CANK2ZX7v9Ar6+1Hp1NhUaQ-uEpxTbyxfYwBP8_m6KkN-YtLCnA@mail.gmail.com>

bit_rate is another problem for me. Is not it sample_rate *
bit_count_per_sample? Why do we need to adjust it in encoder? It also does
not change anything even I comment out that line. For your another replies,
I also thought, I need to resample it before encoding, but now I don't
understand the functionality of sample_rate, channel number parameters in
av_codec_context class, does not it resample it while encoding?


2013/8/23 Nicolas George <nicolas.george at normalesup.org>

> Le sextidi 6 fructidor, an CCXXI, Mustafa Yavuz a ?crit :
> > I would like to encode decoded frames in signed 16 bit little endian
> > format.
>
> You forgot to specify: PCM.
>
> > Code is here <http://codepaste.net/rgqpc9>
>
> #    enc_ctx->sample_fmt = AV_SAMPLE_FMT_S16;
> #    enc_ctx->bit_rate = 64000;
> #    enc_ctx->sample_rate    = 11025;
> #    enc_ctx->channel_layout = AV_CH_LAYOUT_MONO ;
> #    enc_ctx->channels       = 1;
>
> You got the wrong idea that this is asking the PCM encoder to encode at a
> specific sample rate and with a specific channel count. It does not work
> that way. You are saying to the encoder what you are giving it as input.
> Since you are actually giving it something completely different, it does
> not
> work at all.
>
> You have to convert and remix the audio stream. For that, you can use
> libswresample directly or libavfilter.
>
> By the way, you will never achieve 64000 bit/s with these settings, your
> bit_rate field is completely wrong.
>
> Regards,
>
> --
>   Nicolas George
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

From nicolas.george at normalesup.org  Fri Aug 23 14:51:37 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Fri, 23 Aug 2013 14:51:37 +0200
Subject: [FFmpeg-user] Audio encoding problem
In-Reply-To: <CANK2ZX7v9Ar6+1Hp1NhUaQ-uEpxTbyxfYwBP8_m6KkN-YtLCnA@mail.gmail.com>
References: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>
	<20130823120128.GA30358@phare.normalesup.org>
	<CANK2ZX7v9Ar6+1Hp1NhUaQ-uEpxTbyxfYwBP8_m6KkN-YtLCnA@mail.gmail.com>
Message-ID: <20130823125137.GA8018@phare.normalesup.org>

Le sextidi 6 fructidor, an CCXXI, Mustafa Yavuz a ?crit?:
> bit_rate is another problem for me. Is not it sample_rate *
> bit_count_per_sample?

In the PCM case, it is, if the bit count includes all channels. But you
probably do not need to use a pocket calculator to notice that 11025?16?1 is
not 64000.

> Why do we need to adjust it in encoder?

For PCM, you do not need to. For lossy codecs, that propose a trade between
bit rate and quality, you may want to set it to encode at fixed bit rate.

> I need to resample it before encoding, but now I don't understand the
> functionality of sample_rate, channel number parameters in
> av_codec_context class, does not it resample it while encoding?

No, it does not resample for encoding. For PCM, sample rate will not make
any difference. The channel count will make a difference, because it will
tell how many values per sample there is, but the channel layout is
irrelevant.

For other codecs, sample rate will make a difference: lossy codecs will use
frequency filters to eliminate noise that can not be heard by the human ear,
and this kind of noise does not look the same at 8000 Hz and at 48000 Hz.
And the channel layout may have an impact: there is not the same kind of
channel coupling between front and back than between left and right, and you
do not encode the sub-woofer channel the same way as the others.

> 2013/8/23 Nicolas George <nicolas.george at normalesup.org>

Please do not top-post on this mailing-list. If you do not know what it
means, look it up.

Regards,

-- 
  Nicolas George
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From onemda at gmail.com  Fri Aug 23 15:03:34 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 23 Aug 2013 13:03:34 +0000
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <20130823135537.Horde.A96VatT6jIgUSj8p9vUvGQ2@aktionspotenzial.de>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
	<CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
	<20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>
	<CAPYw7P5grQBk81Y_V1+yJZGXccFdQXB2Z5C5Rjgbk6mC=o-j0Q@mail.gmail.com>
	<20130823123846.Horde.qnsn9ufeoi9LvK8grPJDCQ1@aktionspotenzial.de>
	<CAPYw7P7_RCvzvCgLJSQvZhNCTpx0vYW2xvUOmN+70dKZBV06BA@mail.gmail.com>
	<20130823135537.Horde.A96VatT6jIgUSj8p9vUvGQ2@aktionspotenzial.de>
Message-ID: <CAPYw7P60JK+_WsV0_0=684o4w+7fPxerg=ngVwkTFjuSk_KK+w@mail.gmail.com>

On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
> Zitat von Paul B Mahol <onemda at gmail.com>:
>
>> On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
>>> Zitat von Paul B Mahol <onemda at gmail.com>:
>>>
>>>> On 8/23/13, d-fens <d-fens at 00010111.de> wrote:
>>>>> Zitat von Paul B Mahol <onemda at gmail.com>:
>>>>>
>>>>>> On 8/22/13, d-fens <d-fens at 00010111.de> wrote:
>>>>>>> hi,
>>>>>>>
>>>>>>> i have a background png and a sequence of pngs with transparency
>>>>>>> which
>>>>>>> should overlay the background.
>>>>>>>
>>>>>>> when i use
>>>>>>>
>>>>>>> ./ffmpeg -i %04d.png -vf "movie=bg.png [bg]; [in] format=rgb32,
>>>>>>> split
>>>>>>> [T1], fifo, [bg] overlay=0:0, [T2] overlay=0:0 [out]; [T1] fifo
>>>>>>> [T2]"
>>>>>>> out.mp4
>>>>>>>
>>>>>>> (taken from
>>>>>>> http://stackoverflow.com/questions/10438713/overlay-animated-images-with-transparency-over-a-static-background-image-using-f)
>>>>>>> i get arround 13fps for fullHD images, but using
>>>>>>>
>>>>>>> ./ffmpeg -loop 1 -i bg.png -i %04d.png -filter_complex
>>>>>>> overlay=shortest=1 out.mp4
>>>>>>
>>>>>> Try removing loop and shortest.
>>>>>
>>>>>
>>>>> this just outputs one frame, so no success :(
>>>>
>>>> Than change order of streams - make bg.png 2nd.
>>>
>>> well that puts the background over the sequence, its fast (15fps
>>> processing) though :| but very static ...
>>
>> I didn't benchmark but I assume -loop takes most of time as it decodes
>> same file over and over again.
>> So instead adding looping of main input inside overlay it would be
>> better to add generic solution:
>> repeat filter which would repeat each frame [0-inf] times.
>>
>   yes, i also think the png gets re-read in a loop, maybe
> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=6e643239d9956ce32888d87451706ee7c8757afa
> speeds it up a bit,

That speeds decoding of multiple frames, so should improve both cases.

> but a repeat filter would be great or loop without the
> overhead. But just to be clear: there is no way to archive this with current
> -filter_complex
> yet?

I think so. But I guess you could use 2 overlays like you did with non
filter_complex case.
(yes, extra overlay is not nice ....)

From francois.visagie at gmail.com  Fri Aug 23 15:11:23 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Fri, 23 Aug 2013 15:11:23 +0200
Subject: [FFmpeg-user] Some inputs make MP4 output sync incompatible with
	other applications
Message-ID: <000301cea002$47894400$d69bcc00$@gmail.com>

Hi All,

 

I'm batch encoding our DVDs to H.264 AAC MP4s. Two of these output MP4s have
audio that start too early in both MPC-HC and Windows Media Player, although
ffplay syncs them correctly.

 

This problem seems specific to the MP4 container; when I e.g. mux to an .mpg
container the problem doesn't occur. What's also interesting to note is that
encoding individual VOBs later in the same titlesets (i.e. which don't have
headers) also doesn't produce this problem. Both MPC-HC and WMP play all
original VOBs fine. Therefore FWIW the problem seems to be in the way that
ffmpeg interprets and then recreates header information.

 

I'm trying to prevent this MP4 desync for other applications in a way that
safely can be left in the batch process also for better-behaved inputs.

 

Specifying different MP4-type formats like ipod, mov, mp4 and psp makes no
difference. Through trial-and-error, so far either '-copypts' or the
now-deprecated '-async 1' prevents desync in other applications.

 

For reasons not related to this issue I'd prefer to continue with
'-copypts':

-          From limited testing so far, it seems '-copypts' doesn't always
produce correct results when applied across the board also to well-behaved
inputs; is that expected?

-          What other - perhaps MP4-specific - settings or other mechanisms
can I try to make MP4 sync information more compatible with other
applications, and that are safe also to use with well-behaved inputs?

 

Some examples:

 

Out of sync for other applications:

ffmpeg -i VTS_01_1.cut.VOB -filter:a aformat=channel_layouts=stereo -c:a
libvo_aacenc -c:v libx264 -preset ultrafast -b:v 1000000 zzz.mp4

 

In sync for other applications:

ffmpeg -i VTS_01_1.cut.VOB -filter:a aformat=channel_layouts=stereo -c:a
libvo_aacenc -c:v libx264 -preset ultrafast -b:v 1000000 zzz.mpg

ffmpeg -i VTS_01_1.cut.VOB -filter:a aformat=channel_layouts=stereo -c:a
libvo_aacenc -c:v libx264 -preset ultrafast -b:v 1000000 -copypts
zzzcopypts.mp4

 

Sample clip used in these examples (67 MB):

http://www.mediafire.com/download/p5ywp0aqwuuw6na/VTS_01_1.cut.VOB

 

Console output for ffmpeg -i VTS_01_1.cut.VOB -filter:a
aformat=channel_layouts=stereo -c:a libvo_aacenc -c:v libx264 -preset
ultrafast -b:v 1000000 zzz.mp4:

 

C:\Users\fvisagie\Videos\My DVDs\MainMovie\Roxette Live in Sydney -
1991\VIDEO_TS\TESTSYNCCUT>ffmpeg -i VTS_01_1.cut.VOB -filter:a
aformat=channel_layouts=stere

o -c:a libvo_aacenc -c:v libx264 -preset ultrafast -b:v 1000000 zzz.mp4

ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg
developers

  built on Aug  1 2013 18:01:57 with gcc 4.7.3 (GCC)

  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enab

le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --ena

ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp

eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --

enable-libxvid --enable-zlib

  libavutil      52. 40.100 / 52. 40.100

  libavcodec     55. 19.100 / 55. 19.100

  libavformat    55. 12.102 / 55. 12.102

  libavdevice    55.  3.100 / 55.  3.100

  libavfilter     3. 82.100 /  3. 82.100

  libswscale      2.  4.100 /  2.  4.100

  libswresample   0. 17.103 /  0. 17.103

  libpostproc    52.  3.100 / 52.  3.100

Input #0, mpeg, from 'VTS_01_1.cut.VOB':

  Duration: 00:01:12.11, start: 0.060000, bitrate: 7840 kb/s

    Stream #0:0[0x1bf]: Data: dvd_nav_packet

    Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9
DAR 4:3], max. 6000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc

    Stream #0:2[0x8a]: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536
kb/s

    Stream #0:3[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s

[libx264 @ 003c2b80] using SAR=8/9

[libx264 @ 003c2b80] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX

[libx264 @ 003c2b80] profile Constrained Baseline, level 3.0

[libx264 @ 003c2b80] 264 - core 135 r2345 f0c1c53 - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0
ref=1 deb

lock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0
me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1
chroma_qp_

offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1
interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0
keyint=250 keyint_m

in=25 scenecut=0 intra_refresh=0 rc=abr mbtree=0 bitrate=1000 ratetol=1.0
qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0

Output #0, mp4, to 'zzz.mp4':

  Metadata:

    encoder         : Lavf55.12.102

    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
720x480 [SAR 8:9 DAR 4:3], q=-1--1, 1000 kb/s, 30k tbn, 29.97 tbc

    Stream #0:1: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 48000
Hz, stereo, s16, 128 kb/s

Stream mapping:

  Stream #0:1 -> #0:0 (mpeg2video -> libx264)

  Stream #0:2 -> #0:1 (dca -> libvo_aacenc)

Press [q] to stop, [?] for help

[mpeg2video @ 003cbec0] warning: first frame is no keyframe

    Last message repeated 1 times

[mpeg2video @ 003cbec0] ac-tex damaged at 12 16=00:01:08.36
bitrate=1187.3kbits/s dup=8 drop=0

[mpeg2video @ 003cbec0] Warning MVs not available

[mpeg2video @ 003cbec0] concealing 630 DC, 630 AC, 630 MV errors in I frame

frame= 2181 fps=397 q=-1.0 Lsize=   10737kB time=00:01:12.77
bitrate=1208.6kbits/s dup=10 drop=0

video:9549kB audio:1125kB subtitle:0 global headers:0kB muxing overhead
0.580242%

[libx264 @ 003c2b80] frame I:9     Avg QP:22.00  size: 12125

[libx264 @ 003c2b80] frame P:2172  Avg QP:25.35  size:  4452

[libx264 @ 003c2b80] mb I  I16..4: 100.0%  0.0%  0.0%

[libx264 @ 003c2b80] mb P  I16..4: 21.7%  0.0%  0.0%  P16..4: 31.0%  0.0%
0.0%  0.0%  0.0%    skip:47.3%

[libx264 @ 003c2b80] final ratefactor: 25.51

[libx264 @ 003c2b80] coded y,uvDC,uvAC intra: 11.2% 48.1% 15.5% inter: 11.7%
23.9% 1.7%

[libx264 @ 003c2b80] i16 v,h,dc,p: 45% 30% 15% 10%

[libx264 @ 003c2b80] i8c dc,h,v,p: 46% 26% 19%  8%

[libx264 @ 003c2b80] kb/s:1074.91

[libvo_aacenc @ 03f07240] 1 frames left in the queue on closing

 

C:\Users\fvisagie\Videos\My DVDs\MainMovie\Roxette Live in Sydney -
1991\VIDEO_TS\TESTSYNCCUT>

 

Thanks in advance,

Francois

 

Francois Visagie

 

 

T: +27 (83) 326-4358

F: +27 (86) 511-3490

E: francois.visagie at gmail.com

 

 


From francois.visagie at gmail.com  Fri Aug 23 15:22:05 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Fri, 23 Aug 2013 15:22:05 +0200
Subject: [FFmpeg-user] Concatenation of mpg files
In-Reply-To: <loom.20130823T103731-969@post.gmane.org>
References: <CABOWChwWk6ksD56yAw8bwJ0JpLpMjnh2qh0Xgvt=RR51h3yiwQ@mail.gmail.com>	<005401ce9fd3$1550b650$3ff222f0$@gmail.com>
	<loom.20130823T103731-969@post.gmane.org>
Message-ID: <003b01cea003$c6312a60$52937f20$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos
> Sent: 23 August 2013 10:41
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] Concatenation of mpg files
> 
> I don't think this is correct

Thanks for the correction.
 


From 89.yavuz at gmail.com  Fri Aug 23 15:31:35 2013
From: 89.yavuz at gmail.com (Mustafa Yavuz)
Date: Fri, 23 Aug 2013 16:31:35 +0300
Subject: [FFmpeg-user] Audio encoding problem
In-Reply-To: <20130823125137.GA8018@phare.normalesup.org>
References: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>
	<20130823120128.GA30358@phare.normalesup.org>
	<CANK2ZX7v9Ar6+1Hp1NhUaQ-uEpxTbyxfYwBP8_m6KkN-YtLCnA@mail.gmail.com>
	<20130823125137.GA8018@phare.normalesup.org>
Message-ID: <CANK2ZX4o-5QwLQYRFmyCvU6mQjJC=KJ_1LM08QURryxdB9EB-g@mail.gmail.com>

ok, thank you for your explanations Nicolas, now I will ask a  last
question, these resampling functions are stated as deprecated. What are new
versions of them and Is there any sample codes explaining how to do it?
Thank you..

From cehoyos at ag.or.at  Fri Aug 23 15:33:44 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 23 Aug 2013 13:33:44 +0000 (UTC)
Subject: [FFmpeg-user] Some inputs make MP4 output sync incompatible
	with other applications
References: <000301cea002$47894400$d69bcc00$@gmail.com>
Message-ID: <loom.20130823T153004-714@post.gmane.org>

Francois Visagie <francois.visagie <at> gmail.com> writes:

> This problem seems specific to the MP4 container; when I 
> e.g. mux to an .mpg container the problem doesn't occur. 
> What's also interesting to note is that encoding 
> individual VOBs later in the same titlesets (i.e. which 
> don't have headers)

As said I don't think VOBs have headers.

> also doesn't produce this problem. Both MPC-HC and WMP 
> play all original VOBs fine. Therefore FWIW the problem 
> seems to be in the way that ffmpeg interprets and then 
> recreates header information.
> 
> I'm trying to prevent this MP4 desync for other 
> applications in a way that safely can be left in the 
> batch process also for better-behaved inputs.

(Unrelated to this question and not necessarily true:
I don't think this is generally possible.)

> Specifying different MP4-type formats like ipod, mov, 
> mp4 and psp makes no difference. Through 
> trial-and-error, so far either '-copypts' or the

> now-deprecated '-async 1' prevents desync in other 
> applications.

Why do you think "-async 1" is deprecated? Afair, there 
are several samples that need it.

> Out of sync for other applications:
> 
> ffmpeg -i VTS_01_1.cut.VOB -filter:a 
> aformat=channel_layouts=stereo -c:a libvo_aacenc 
> -c:v libx264 -preset ultrafast -b:v 1000000 zzz.mp4
> 
> In sync for other applications:
> 
> ffmpeg -i VTS_01_1.cut.VOB -filter:a 
> aformat=channel_layouts=stereo -c:a libvo_aacenc 
> -c:v libx264 -preset ultrafast -b:v 1000000 zzz.mpg

(Difficult to see a difference.)
There have been reports that libvo_aacenc produces worse 
results than the native encoder and should not be used.

Carl Eugen


From cehoyos at ag.or.at  Fri Aug 23 15:35:06 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 23 Aug 2013 13:35:06 +0000 (UTC)
Subject: [FFmpeg-user] Audio encoding problem
References: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>
	<20130823120128.GA30358@phare.normalesup.org>
	<CANK2ZX7v9Ar6+1Hp1NhUaQ-uEpxTbyxfYwBP8_m6KkN-YtLCnA@mail.gmail.com>
	<20130823125137.GA8018@phare.normalesup.org>
	<CANK2ZX4o-5QwLQYRFmyCvU6mQjJC=KJ_1LM08QURryxdB9EB-g@mail.gmail.com>
Message-ID: <loom.20130823T153409-752@post.gmane.org>

Mustafa Yavuz <89.yavuz <at> gmail.com> writes:

> ok, thank you for your explanations Nicolas, now I will 
> ask a last question, these resampling functions are 
> stated as deprecated. What are new versions of them and 
> Is there any sample codes explaining how to do it?

I don't think the functions that Nicolas proposed are 
deprecated, see doc/examples/resampling_audio.c

Carl Eugen


From 89.yavuz at gmail.com  Fri Aug 23 15:50:57 2013
From: 89.yavuz at gmail.com (Mustafa Yavuz)
Date: Fri, 23 Aug 2013 16:50:57 +0300
Subject: [FFmpeg-user] Audio encoding problem
In-Reply-To: <loom.20130823T153409-752@post.gmane.org>
References: <CANK2ZX6Yw4eRvYysEoN1J3yjV39r5a3yBhj6Y93ZqMXDNXz_EQ@mail.gmail.com>
	<20130823120128.GA30358@phare.normalesup.org>
	<CANK2ZX7v9Ar6+1Hp1NhUaQ-uEpxTbyxfYwBP8_m6KkN-YtLCnA@mail.gmail.com>
	<20130823125137.GA8018@phare.normalesup.org>
	<CANK2ZX4o-5QwLQYRFmyCvU6mQjJC=KJ_1LM08QURryxdB9EB-g@mail.gmail.com>
	<loom.20130823T153409-752@post.gmane.org>
Message-ID: <CANK2ZX7_bKj0hteRqZ-2G8jhL7HP8ghThCicZaGR_22SJQdmWw@mail.gmail.com>

Ok, I was talking about av_audio_resample_init, audio_resample functions
which are in avcodec.h and deprecated but yours are different functions
comes from swresample.h. Problem solved. Thanks..


2013/8/23 Carl Eugen Hoyos <cehoyos at ag.or.at>

> Mustafa Yavuz <89.yavuz <at> gmail.com> writes:
>
> > ok, thank you for your explanations Nicolas, now I will
> > ask a last question, these resampling functions are
> > stated as deprecated. What are new versions of them and
> > Is there any sample codes explaining how to do it?
>
> I don't think the functions that Nicolas proposed are
> deprecated, see doc/examples/resampling_audio.c
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From francois.visagie at gmail.com  Fri Aug 23 15:52:45 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Fri, 23 Aug 2013 15:52:45 +0200
Subject: [FFmpeg-user] Some inputs make MP4 output sync
	incompatible	with other applications
In-Reply-To: <loom.20130823T153004-714@post.gmane.org>
References: <000301cea002$47894400$d69bcc00$@gmail.com>
	<loom.20130823T153004-714@post.gmane.org>
Message-ID: <005601cea008$104e6a00$30eb3e00$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos
> Sent: 23 August 2013 15:34
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] Some inputs make MP4 output sync
> incompatible with other applications
> 
> Why do you think "-async 1" is deprecated? 

>From http://www.ffmpeg.org/ffmpeg-all.html:

'-async samples_per_second'
    ...
    This option has been deprecated. Use the aresample audio filter instead.

> > Out of sync for other applications:
> >
> > ffmpeg -i VTS_01_1.cut.VOB -filter:a
> > aformat=channel_layouts=stereo -c:a libvo_aacenc -c:v libx264 -preset
> > ultrafast -b:v 1000000 zzz.mp4
> >
> > In sync for other applications:
> >
> > ffmpeg -i VTS_01_1.cut.VOB -filter:a
> > aformat=channel_layouts=stereo -c:a libvo_aacenc -c:v libx264 -preset
> > ultrafast -b:v 1000000 zzz.mpg
> 
> (Difficult to see a difference.)

There's a very clear difference in output.

> There have been reports that libvo_aacenc produces worse results than the
> native encoder and should not be used.

By native do you mean the "experimental aac" codec below (from the Zeranoe
builds)? I.e. using it requires '-strict -2'?

C:\Users\fvisagie\Videos\My DVDs\MainMovie\Roxette Live in Sydney -
1991\VIDEO_TS\TESTSYNCCUT>ffmpeg -i VTS_01_1.cut.VOB -filter:a
aformat=channel_layouts=stere
o -c:a aac -c:v libx264 -preset ultrafast -b:v 1000000 zzz.mp4
ffmpeg version N-55159-gf118b41 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug  1 2013 18:01:57 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 19.100 / 55. 19.100
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.100 /  3. 82.100
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mpeg, from 'VTS_01_1.cut.VOB':
  Duration: 00:01:12.11, start: 0.060000, bitrate: 7840 kb/s
    Stream #0:0[0x1bf]: Data: dvd_nav_packet
    Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9
DAR 4:3], max. 6000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:2[0x8a]: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536
kb/s
    Stream #0:3[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
File 'zzz.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 002e2b60] using SAR=8/9
[libx264 @ 002e2b60] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 002e2b60] profile Constrained Baseline, level 3.0
[libx264 @ 002e2b60] 264 - core 135 r2345 f0c1c53 - H.264/MPEG-4 AVC codec -
Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0
ref=1 deb
lock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0
me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1
chroma_qp_
offset=0 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1
interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0
keyint=250 keyint_m
in=25 scenecut=0 intra_refresh=0 rc=abr mbtree=0 bitrate=1000 ratetol=1.0
qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
[aac @ 002e2f40] The encoder 'aac' is experimental but experimental codecs
are not enabled, add '-strict -2' if you want to use it.
[aac @ 002e2f40] Alternatively use the non experimental encoder
'libvo_aacenc'.


From francois.visagie at gmail.com  Fri Aug 23 16:31:25 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Fri, 23 Aug 2013 16:31:25 +0200
Subject: [FFmpeg-user] Some inputs make MP4 output sync
	incompatible	with other applications
References: <000301cea002$47894400$d69bcc00$@gmail.com>
	<loom.20130823T153004-714@post.gmane.org> 
Message-ID: <005b01cea00d$75f28490$61d78db0$@gmail.com>

> -----Original Message-----
> From: Francois Visagie [mailto:francois.visagie at gmail.com]
> Sent: 23 August 2013 15:53
> To: 'FFmpeg user questions'
> Subject: RE: [FFmpeg-user] Some inputs make MP4 output sync incompatible
> with other applications
> 
> > > Out of sync for other applications:
> > >
> > > ffmpeg -i VTS_01_1.cut.VOB -filter:a aformat=channel_layouts=stereo
> > > -c:a libvo_aacenc -c:v libx264 -preset ultrafast -b:v 1000000
> > > zzz.mp4
> > >
> > > In sync for other applications:
> > >
> > > ffmpeg -i VTS_01_1.cut.VOB -filter:a aformat=channel_layouts=stereo
> > > -c:a libvo_aacenc -c:v libx264 -preset ultrafast -b:v 1000000
> > > zzz.mpg
> >
> > (Difficult to see a difference.)
> 
> There's a very clear difference in output.
> 
> > There have been reports that libvo_aacenc produces worse results than
> > the native encoder and should not be used.
> 
> By native do you mean the "experimental aac" codec below (from the
> Zeranoe builds)? I.e. using it requires '-strict -2'?

With '-strict -2 -c:a aac' instead of '-c:a libvo_aacenc', i.e. with

ffmpeg -i VTS_01_1.cut.VOB -filter:a aformat=channel_layouts=stereo -strict
-2 -c:a aac -c:v libx264 -preset ultrafast -b:v 1000000 zzzaac.mp4

timing results are the same as before. E.g. at around 52s into the sample
clip the words "to go" appear around 650 ms before the utterance on video.


From dzungng89 at gmail.com  Fri Aug 23 17:06:41 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Fri, 23 Aug 2013 10:06:41 -0500
Subject: [FFmpeg-user] Concatenation of mpg files
In-Reply-To: <003b01cea003$c6312a60$52937f20$@gmail.com>
References: <CABOWChwWk6ksD56yAw8bwJ0JpLpMjnh2qh0Xgvt=RR51h3yiwQ@mail.gmail.com>
	<005401ce9fd3$1550b650$3ff222f0$@gmail.com>
	<loom.20130823T103731-969@post.gmane.org>
	<003b01cea003$c6312a60$52937f20$@gmail.com>
Message-ID: <CABOWChyhBKKRRV8jPixq6Xwuq42j_X9yCnZyqZt8s=+2gyJDVQ@mail.gmail.com>

On Fri, Aug 23, 2013 at 8:22 AM, Francois Visagie <
francois.visagie at gmail.com> wrote:

> > -----Original Message-----
> > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> > bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos
> > Sent: 23 August 2013 10:41
> > To: ffmpeg-user at ffmpeg.org
> > Subject: Re: [FFmpeg-user] Concatenation of mpg files
> >
> > I don't think this is correct
>
> Thanks for the correction.
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

Carl, you're right. Whenever I skip the -q:v 0 option I don't have any
buffer underflow  problem with the muxer. Because I want to keep the image
quality, how can I set the ffmpeg muxer?

From nicolas.george at normalesup.org  Fri Aug 23 17:20:28 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Fri, 23 Aug 2013 17:20:28 +0200
Subject: [FFmpeg-user] ~50% performance difference overlaying static
 image with png series using -vf/-filter_complex
In-Reply-To: <CAPYw7P7_RCvzvCgLJSQvZhNCTpx0vYW2xvUOmN+70dKZBV06BA@mail.gmail.com>
References: <20130823002109.Horde.pVzOC2RarjoCSEciMzWK9A7@aktionspotenzial.de>
	<CAPYw7P5F2oouZZ1R6=Dd_STXL_yR=L3h0nrm4ULvzyLRmrjv9Q@mail.gmail.com>
	<20130823100957.Horde.bdQSmCct5qwlUqgHTFVIGQ1@aktionspotenzial.de>
	<CAPYw7P5grQBk81Y_V1+yJZGXccFdQXB2Z5C5Rjgbk6mC=o-j0Q@mail.gmail.com>
	<20130823123846.Horde.qnsn9ufeoi9LvK8grPJDCQ1@aktionspotenzial.de>
	<CAPYw7P7_RCvzvCgLJSQvZhNCTpx0vYW2xvUOmN+70dKZBV06BA@mail.gmail.com>
Message-ID: <20130823152028.GA7472@phare.normalesup.org>

Le sextidi 6 fructidor, an CCXXI, Paul B Mahol a ?crit?:
> I didn't benchmark but I assume -loop takes most of time as it decodes
> same file over and over again.
> So instead adding looping of main input inside overlay it would be
> better to add generic solution:
> repeat filter which would repeat each frame [0-inf] times.

There are a lot of things missing in overlay to achieve subtle effects, and
this is only one of them. I will try to make a summary on the -dev
mailing-list with useful suggestions.

To implement this particular effect in the most general case, the original
solution of sandwitching the background in front of the overlay to give the
sync seems the only solution.

If the video is CFR, the following hack should eliminate most of the
overhead due to the loop effect:

-loop 1 -i bg.png -i video.mp4 lavfi '[0] fps=25 [bg] ;
[bg] [1] overlay=shortest=1'

Regards,

-- 
  Nicolas George
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From dzungng89 at gmail.com  Fri Aug 23 18:15:22 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Fri, 23 Aug 2013 11:15:22 -0500
Subject: [FFmpeg-user] Meaning of input frame rate
Message-ID: <CABOWChxQpwxgM1QwC92it8-5uVftZsQ5_2kgSbXqErosodw+fg@mail.gmail.com>

What's the meaning of input frame rate. For example, what does this command
do?

ffmpeg -loop 1 -r 1 -i image.png -q:v 0 -r 29.97 out.mpg

Instead of

ffmpeg -loop 1 -r 29.97 -i image.png -q:v 0 -r 29.97

From mirko.puliafito at gmail.com  Fri Aug 23 18:39:39 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Fri, 23 Aug 2013 18:39:39 +0200
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
Message-ID: <CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>

Hi,

just try to demux a stream, file or live, from a flash media server (I
tried with crtmpserver and red5). It only opens the TCP connection but it
doesn't start the handshake giving an invalid parameter error.

I tried with..

ffmpeg -i rtmpe://127.0.0.1/live/myStream ...

Mirko
Il giorno 23/ago/2013 12:05, "Paul B Mahol" <onemda at gmail.com> ha scritto:

> On 8/23/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> > Hi,
> >
> > I'm trying to use rtmpe (ffrtmpcrypt) with the latest version but it's
> not
> > working (I've also tested it with the 1.x). I tested it against
> crtmpserver
> > and red5 without any luck. Can anybody tell me the status of that
> protocol
> > implementation? Do I need to go on using rtmpdump libs if I want to use
> > this protocol?
>
> What does not work exactly?
>
> There is simple not enough information provided.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From jindrich.makovicka at nangu.tv  Thu Aug 22 19:38:49 2013
From: jindrich.makovicka at nangu.tv (=?ISO-8859-2?Q?Jind=F8ich_Makovi=E8ka?=)
Date: Thu, 22 Aug 2013 19:38:49 +0200
Subject: [FFmpeg-user] Converted DVDs in .mkv format lose Pause,
 FastForward, Skip, etc functions when I restream as mpeg to DLNA
In-Reply-To: <1377190724.6609.12950137.7EF80BA2@webmail.messagingengine.com>
References: <1377100446.24312.12472041.2BC23697@webmail.messagingengine.com>
	<1377190724.6609.12950137.7EF80BA2@webmail.messagingengine.com>
Message-ID: <CAHmvugc1JmasNCjQ0WFFdcpR2bjFnCvRdrfhupwdkc2f2Suj4A@mail.gmail.com>

On Thu, Aug 22, 2013 at 6:59 PM,  <jen142 at promessage.com> wrote:
> On Wed, Aug 21, 2013, at 08:54 AM, jen142 at promessage.com wrote:
>> Can anyone help figure out any more or different commands I need to use
>> to not lose the functions when I stream the videos to the TVs like this?
>
> This thread
>
>         http://ffmpeg-users.933282.n4.nabble.com/Transcoded-MPEG-2-Transport-Stream-does-not-have-random-access-indicator-bit-set-td3010438.html
>
> suggests that
>
> "Random Access Indicator, which is very important to realize trick play
> (i.e. random access, fast forward),
> is not included in the stream."
>
> The question is asked there "how?", but no answer.
>
> A commit thread
>
>         [FFmpeg-cvslog] mpegtsenc: set Random Access indicator on
>         keyframe start packets
>         https://lists.ffmpeg.org/pipermail/ffmpeg-cvslog/2011-July/038989.html
>
> suggests that it's already IN ffmpeg.
>
> If it is, how can I enable the "Random Access indicator" in my stream
> transcode command?  I'm not sure it's the answer, but it seems like it's
> worth a try at least.

It doesn't need to be enabled. RAI should be set automatically on
every starting packet of a key frame.

-- 
Jind?ich Makovi?ka

nangu.tv

From doenerbudenchef at yahoo.de  Fri Aug 23 16:49:26 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Fri, 23 Aug 2013 07:49:26 -0700 (PDT)
Subject: [FFmpeg-user] Convert progressive MXF to interlaced
In-Reply-To: <1376665654187-4660746.post@n4.nabble.com>
References: <1376665654187-4660746.post@n4.nabble.com>
Message-ID: <1377269366547-4660888.post@n4.nabble.com>

Okay, I've tried a IMX 50 ecoding first, because this is basically documented
here (1080p mxf to 1080i with -f mxf produces buffer underruns):

http://www.itbroadcastanddigitalcinema.com/ffmpeg_howto.html

It seems to work, but I have a problem with mapping the stereo audio channel
to separate mono streams. When I use the "map_channel" option, ffmpeg also
says the "This syntax is deprecated"...

C:\>ffmpeg -i test.mp4 -vf
"yadif=1:0:0,scale=720:576,pad=720:608:0:32,tinterlace=interleave_top" -r 25
-aspect 16:9 -top 1 -pix_fmt yuv422p -vcodec mpeg2video -intra_vlc 1
-non_linear_quant 1 -q:v 1 -ps 1 -qmax 3 -rc_max_vbv_use 1 -flags
+ildct+low_delay -rc_min_v
bv_use 1 -minrate 50000k -maxrate 50000k -b:v 50000k -bufsize 2000000 -g 1
-top 1 -rc_init_occupancy 2000000 -rc_buf_aggressivity 0.25 -dc 10 -map 0:0
-map 0:1 -map_channel 0.1.0:0.1 -map_channel 0.1.1:0.2 -acodec pcm_s24le -ar
48000 -ac 2 output.mxf

ffmpeg version N-54207-ge59fb3f Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jun 25 2013 21:55:00 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libbluray
--enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc
--enable-libmodplug --ena
ble-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-libschroedinger --enable-libsoxr --enable-libspeex
--enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-amrwbenc --enable-libvorbis --enable-lib
vpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 37.101 / 52. 37.101
  libavcodec     55. 17.100 / 55. 17.100
  libavformat    55. 10.100 / 55. 10.100
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 77.101 /  3. 77.101
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf55.10.100
  Duration: 00:00:26.20, start: 0.023220, bitrate: 10416 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p,
1920x1080 [SAR 1:1 DAR 16:9], 10151 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
fltp, 260 kb/s
    Metadata:
      handler_name    : SoundHandler
File 'output.mxf' already exists. Overwrite ? [y/N] y
-map_channel is forwarded to lavfi similarly to -af pan=0x4:c0=c0.
[pan @ 00000000024aece0] This syntax is deprecated. Use '|' to separate the
list items.
[pan @ 00000000024aece0] Pure channel mapping detected: 0
Output #0, mxf, to 'output.mxf':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf55.10.100
    Stream #0:0(und): Video: mpeg2video, yuv422p, 720x608 [SAR 608:405 DAR
16:9], q=2-3, 50000 kb/s, 25 tbn, 25 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg2video)
  Stream #0:1 -> #0:1 (aac -> pcm_s24le)
Press [q] to stop, [?] for help
frame=  212 fps= 36 q=1.0 Lsize=   54388kB time=00:00:08.67
bitrate=51366.1kbits/s dup=1 drop=0
video:51758kB audio:2442kB subtitle:0 global headers:0kB muxing overhead
0.345740%

Thanks,
Alex



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Convert-progressive-MXF-to-interlaced-tp4660746p4660888.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From lou at lrcd.com  Fri Aug 23 20:09:07 2013
From: lou at lrcd.com (Lou)
Date: Fri, 23 Aug 2013 10:09:07 -0800
Subject: [FFmpeg-user] Meaning of input frame rate
In-Reply-To: <CABOWChxQpwxgM1QwC92it8-5uVftZsQ5_2kgSbXqErosodw+fg@mail.gmail.com>
References: <CABOWChxQpwxgM1QwC92it8-5uVftZsQ5_2kgSbXqErosodw+fg@mail.gmail.com>
Message-ID: <1377281347.10679.13378385.06E748D6@webmail.messagingengine.com>

On Fri, Aug 23, 2013, at 08:15 AM, Dzung Nguyen wrote:
> What's the meaning of input frame rate. For example, what does this
> command
> do?
> 
> ffmpeg -loop 1 -r 1 -i image.png -q:v 0 -r 29.97 out.mpg

This will use an input frame rate of 1 frame per second. The output will
be 29.97 frames per second. ffmpeg will duplicate frames to reach your
desired output frame rate. In the opposite case, if your input -r was a
larger value than your output -r, then ffmpeg would drop frames to reach
your output frame rate. You can see the number of dropped or duped
frames at the end of your console output:

frame=  150 fps=0.0 q=0.0 Lsize=    1604kB time=00:00:04.97
bitrate=2643.0kbits/s dup=25 drop=0

The result will show each input image for a second, but your output will
still be 29.97.

Since there is no -t, -vframes, or any other option to limit the
duration or number of output frames in this command the encoding will
continue indefinitely. 

"-q:v" values of 0-2 all appear to produce the same results with this
encoder, mpeg1video, and my sample (I did not refer to any code):

$ ffmpeg -loop 1 -i 001.png -q:v 0 -t 5 -c:v mpeg1video -f md5 -
...
MD5=71fdaa434d28777735a85d4aa383d88c

$ ffmpeg -loop 1 -i 001.png -q:v 2 -t 5 -c:v mpeg1video -f md5 -
...
MD5=71fdaa434d28777735a85d4aa383d88c
 
> Instead of
> 
> ffmpeg -loop 1 -r 29.97 -i image.png -q:v 0 -r 29.97

This will use an input frame rate of 29.97 frames per second. The output
-r 29.97 is not needed since the output should inherit the input frame
rate, so declaring the same -r for both input and output is superfluous
in this case. No frames will be dropped or duplicated because there is
no difference between your input and output frame rates.

Use "-r ntsc" or "-r 30000/1001" for NTSC video frame rate.

From dzungng89 at gmail.com  Fri Aug 23 21:44:54 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Fri, 23 Aug 2013 14:44:54 -0500
Subject: [FFmpeg-user] Meaning of input frame rate
In-Reply-To: <1377281347.10679.13378385.06E748D6@webmail.messagingengine.com>
References: <CABOWChxQpwxgM1QwC92it8-5uVftZsQ5_2kgSbXqErosodw+fg@mail.gmail.com>
	<1377281347.10679.13378385.06E748D6@webmail.messagingengine.com>
Message-ID: <CABOWChyoQHmzOBed2_s3=YUHSaggfFHrSy2oZC6hEM0sWhy-UA@mail.gmail.com>

On Fri, Aug 23, 2013 at 1:09 PM, Lou <lou at lrcd.com> wrote:

> On Fri, Aug 23, 2013, at 08:15 AM, Dzung Nguyen wrote:
> > What's the meaning of input frame rate. For example, what does this
> > command
> > do?
> >
> > ffmpeg -loop 1 -r 1 -i image.png -q:v 0 -r 29.97 out.mpg
>
> This will use an input frame rate of 1 frame per second. The output will
> be 29.97 frames per second. ffmpeg will duplicate frames to reach your
> desired output frame rate. In the opposite case, if your input -r was a
> larger value than your output -r, then ffmpeg would drop frames to reach
> your output frame rate. You can see the number of dropped or duped
> frames at the end of your console output:
>
> frame=  150 fps=0.0 q=0.0 Lsize=    1604kB time=00:00:04.97
> bitrate=2643.0kbits/s dup=25 drop=0
>
> The result will show each input image for a second, but your output will
> still be 29.97.
>
> Since there is no -t, -vframes, or any other option to limit the
> duration or number of output frames in this command the encoding will
> continue indefinitely.
>
> "-q:v" values of 0-2 all appear to produce the same results with this
> encoder, mpeg1video, and my sample (I did not refer to any code):
>
> $ ffmpeg -loop 1 -i 001.png -q:v 0 -t 5 -c:v mpeg1video -f md5 -
> ...
> MD5=71fdaa434d28777735a85d4aa383d88c
>
> $ ffmpeg -loop 1 -i 001.png -q:v 2 -t 5 -c:v mpeg1video -f md5 -
> ...
> MD5=71fdaa434d28777735a85d4aa383d88c
>
> > Instead of
> >
> > ffmpeg -loop 1 -r 29.97 -i image.png -q:v 0 -r 29.97
>
> This will use an input frame rate of 29.97 frames per second. The output
> -r 29.97 is not needed since the output should inherit the input frame
> rate, so declaring the same -r for both input and output is superfluous
> in this case. No frames will be dropped or duplicated because there is
> no difference between your input and output frame rates.
>
> Use "-r ntsc" or "-r 30000/1001" for NTSC video frame rate.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


Thanks! It helps me a lot. One question, then why second command ffmpeg
-loop 1 -r 29.97 -i image.png output.mpg runs much slower than the first
one?

From dzungng89 at gmail.com  Fri Aug 23 23:22:37 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Fri, 23 Aug 2013 16:22:37 -0500
Subject: [FFmpeg-user] FFMPEG processing time
Message-ID: <CABOWChx7Gm=PXZj2T04G-pk9JYDTc2Vs-NekuV3Ba-MMcNHamA@mail.gmail.com>

I have the following script, I have run it for a day but still doesn't
finish. The goal is a one hour video

1) Images --> videos:

ffmpeg -loop 1 -r 1 image.png -r 29.97 video1.mpg

2) Concatenate allsubvideos:

cat *.mpg >concat.mpg

3) Overlay with another video:

ffmpeg -i concat.mpg -i input2.webm"\
                                    " -filter_complex
\"color=c=black:size=1280x720 [base];"\
                                    " [0:v] setpts=PTS-STARTPTS,
scale=#{size_slide} [left];"\
                                    " [1:v] setpts=PTS-STARTPTS,
scale=#{size_webcam} [right];"\
                                    " [base][left]
overlay=shortest=1:x=0:y=39[tmp1];"\
                                    " [tmp1][right]
overlay=shortest=1:x=856:y=402\" -q:v 0 overlay.mpg")

This seems unable to finish even if I let it run for a day. What settings
should I change?

From onemda at gmail.com  Fri Aug 23 23:27:10 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 23 Aug 2013 21:27:10 +0000
Subject: [FFmpeg-user] FFMPEG processing time
In-Reply-To: <CABOWChx7Gm=PXZj2T04G-pk9JYDTc2Vs-NekuV3Ba-MMcNHamA@mail.gmail.com>
References: <CABOWChx7Gm=PXZj2T04G-pk9JYDTc2Vs-NekuV3Ba-MMcNHamA@mail.gmail.com>
Message-ID: <CAPYw7P6XXaJb5QUFgLySq1duDSBs47X_eJ8QG3BLDq59qbPyVw@mail.gmail.com>

On 8/23/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> I have the following script, I have run it for a day but still doesn't
> finish. The goal is a one hour video
>
> 1) Images --> videos:
>
> ffmpeg -loop 1 -r 1 image.png -r 29.97 video1.mpg

This will never finish.

>
> 2) Concatenate allsubvideos:
>
> cat *.mpg >concat.mpg
>
> 3) Overlay with another video:
>
> ffmpeg -i concat.mpg -i input2.webm"\
>                                     " -filter_complex
> \"color=c=black:size=1280x720 [base];"\
>                                     " [0:v] setpts=PTS-STARTPTS,
> scale=#{size_slide} [left];"\
>                                     " [1:v] setpts=PTS-STARTPTS,
> scale=#{size_webcam} [right];"\
>                                     " [base][left]
> overlay=shortest=1:x=0:y=39[tmp1];"\
>                                     " [tmp1][right]
> overlay=shortest=1:x=856:y=402\" -q:v 0 overlay.mpg")
>
> This seems unable to finish even if I let it run for a day. What settings
> should I change?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From dzungng89 at gmail.com  Fri Aug 23 23:31:51 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Fri, 23 Aug 2013 16:31:51 -0500
Subject: [FFmpeg-user] FFMPEG processing time
In-Reply-To: <CAPYw7P6XXaJb5QUFgLySq1duDSBs47X_eJ8QG3BLDq59qbPyVw@mail.gmail.com>
References: <CABOWChx7Gm=PXZj2T04G-pk9JYDTc2Vs-NekuV3Ba-MMcNHamA@mail.gmail.com>
	<CAPYw7P6XXaJb5QUFgLySq1duDSBs47X_eJ8QG3BLDq59qbPyVw@mail.gmail.com>
Message-ID: <CABOWChzuP+ZXeg18cADfsUXwUbt4mVORxR9xNy5Qt+zn1aQ0vg@mail.gmail.com>

On Fri, Aug 23, 2013 at 4:27 PM, Paul B Mahol <onemda at gmail.com> wrote:

> On 8/23/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> > I have the following script, I have run it for a day but still doesn't
> > finish. The goal is a one hour video
> >
> > 1) Images --> videos:
> >
> > ffmpeg -loop 1 -r 1 image.png -r 29.97 video1.mpg
>
> This will never finish.
>
> >
> > 2) Concatenate allsubvideos:
> >
> > cat *.mpg >concat.mpg
> >
> > 3) Overlay with another video:
> >
> > ffmpeg -i concat.mpg -i input2.webm"\
> >                                     " -filter_complex
> > \"color=c=black:size=1280x720 [base];"\
> >                                     " [0:v] setpts=PTS-STARTPTS,
> > scale=#{size_slide} [left];"\
> >                                     " [1:v] setpts=PTS-STARTPTS,
> > scale=#{size_webcam} [right];"\
> >                                     " [base][left]
> > overlay=shortest=1:x=0:y=39[tmp1];"\
> >                                     " [tmp1][right]
> > overlay=shortest=1:x=856:y=402\" -q:v 0 overlay.mpg")
> >
> > This seems unable to finish even if I let it run for a day. What settings
> > should I change?
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


Sorry I miss out, there's a -t #{time} options there. Time is a variable.

From francois.visagie at gmail.com  Sat Aug 24 15:04:41 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Sat, 24 Aug 2013 15:04:41 +0200
Subject: [FFmpeg-user] Some inputs make MP4 output sync incompatible
	with other applications
Message-ID: <001901cea0ca$81fecb00$85fc6100$@gmail.com>

> -----Original Message-----
> From: Francois Visagie [mailto:francois.visagie at gmail.com]
> Sent: 23 August 2013 15:11
> To: ffmpeg-user at ffmpeg.org
> Subject: Some inputs make MP4 output sync incompatible with other
> applications
> 
> Hi All,
> 
> I'm batch encoding our DVDs to H.264 AAC MP4s. Two of these output MP4s
> have audio that start too early in both MPC-HC and Windows Media Player,
> although ffplay syncs them correctly.
> 
> I'm trying to prevent this MP4 desync for other applications in a way that
> safely can be left in the batch process also for better-behaved inputs.
> 
> Through trial-and-error, so far either '-copypts' or the now-
> deprecated '-async 1' prevents desync in other applications.
> 
> For reasons not related to this issue I'd prefer to continue with
'-copypts'

>From some quick tests '-async 1' actually seems to behave better than
'-copypts' with better-behaved inputs.

Put differently, I haven't seen obvious drawbacks so far, but what is the
potential downside to using '-async 1' across the board from a theoretical
point of view?


From frans at fransdb.nl  Sun Aug 25 14:51:33 2013
From: frans at fransdb.nl (Frans de Boer)
Date: Sun, 25 Aug 2013 14:51:33 +0200
Subject: [FFmpeg-user] Fwd: error 127
In-Reply-To: <5219FBDB.3040903@fransdb.nl>
References: <5219FBDB.3040903@fransdb.nl>
Message-ID: <5219FDD5.9050704@fransdb.nl>

Same applies for avcodec.so.55 and now it is safe to assume that it 
applies to all others as well. Intentional?

Frans


-------- Original Message --------
To: ffmpeg-user at ffmpeg.org
Subject: error 127

Ok, I try to compile ffmpeg-2.0.1 as well as the latest git repository
and when running 'make check', I get error 127 during the fft-4 test.
Meaning that libavutil.so.52 was not found.

The library is in the build tree but not yet on the system installed. So
it seems that the test assumes that the library is already installed.
Does that not defeat the purpose of testing before installing?

Regards, frans.



From frans at fransdb.nl  Sun Aug 25 14:43:07 2013
From: frans at fransdb.nl (Frans de Boer)
Date: Sun, 25 Aug 2013 14:43:07 +0200
Subject: [FFmpeg-user] error 127
Message-ID: <5219FBDB.3040903@fransdb.nl>

Ok, I try to compile ffmpeg-2.0.1 as well as the latest git repository 
and when running 'make check', I get error 127 during the fft-4 test. 
Meaning that libavutil.so.52 was not found.

The library is in the build tree but not yet on the system installed. So 
it seems that the test assumes that the library is already installed. 
Does that not defeat the purpose of testing before installing?

Regards, frans.

From cehoyos at ag.or.at  Sun Aug 25 15:10:03 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 25 Aug 2013 13:10:03 +0000 (UTC)
Subject: [FFmpeg-user] Fwd: error 127
References: <5219FBDB.3040903@fransdb.nl> <5219FDD5.9050704@fransdb.nl>
Message-ID: <loom.20130825T150436-68@post.gmane.org>

Frans de Boer <frans <at> fransdb.nl> writes:

> when running 'make check', I get error 127 during the fft-4 test.
> Meaning that libavutil.so.52 was not found.
> 
> The library is in the build tree but not yet on the system 
> installed.

Set LD_LIBRARY_PATH:
$ export LD_LIBRARY_PATH=libavutil:libavcodec: ...

Carl Eugen


From bouke at videotoolshed.com  Sun Aug 25 16:32:56 2013
From: bouke at videotoolshed.com (Bouke (VideoToolShed))
Date: Sun, 25 Aug 2013 16:32:56 +0200
Subject: [FFmpeg-user] encoding with Avid MXF as sourcefile
Message-ID: <1787869CAFDB4EB6A009965C4C7C087A@HPKANTOOR>

What is the secret recepy to do so?
For DNxHD,  ffmpeg -f dnxhd -i input yadda works (ok, it throws an error, 
but it works)
Sound works straight out of the box.

But what about other codecs?
-f prores -i avidProResFile.mxf  does not work,
for XDcamHD consolidated MXF, -f xdcamhd nor -f mpeg2video does it .

and then there are lots of more avid codecs, but these are the most 
significant to me.
Options please?


Bouke


From cehoyos at ag.or.at  Sun Aug 25 17:11:48 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 25 Aug 2013 15:11:48 +0000 (UTC)
Subject: [FFmpeg-user] encoding with Avid MXF as sourcefile
References: <1787869CAFDB4EB6A009965C4C7C087A@HPKANTOOR>
Message-ID: <loom.20130825T170531-561@post.gmane.org>

Bouke (VideoToolShed <bouke <at> videotoolshed.com> writes:

> What is the secret recepy to do so?
> For DNxHD,  ffmpeg -f dnxhd -i input yadda works

This sounds like a bug:
If "ffmpeg -i input yadda" does not work, please 
provide the sample.

> (ok, it throws an error, but it works)

Complete, uncut console output missing.

> Sound works straight out of the box.
> 
> But what about other codecs?
> -f prores -i avidProResFile.mxf  does not work,

There is no format prores that you could specify.
-f specifies file formats, in some cases you can 
specify -vcodec for the input codec, but this is 
unusual, better provide a sample if it does not 
work.

> for XDcamHD consolidated MXF, -f xdcamhd nor 
> -f mpeg2video does it .

These formats either don't exist or have other 
names ("mpegvideo" for raw MPEG-2 Video), you 
should not have to specify these codecs (I 
don't think they exist in file formats that 
need it).

> and then there are lots of more avid codecs, 
> but these are the most significant to me.
> Options please?

Samples?

Carl Eugen


From mariofutire at googlemail.com  Sun Aug 25 19:22:52 2013
From: mariofutire at googlemail.com (andrea)
Date: Sun, 25 Aug 2013 18:22:52 +0100
Subject: [FFmpeg-user] ffmpeg cannot capture h264 from webcam C920
Message-ID: <kvdeh5$2qh$1@ger.gmane.org>

Hi,

I have the same issue reported here

http://ffmpeg.org/pipermail/ffmpeg-user/2013-May/015046.html
and
http://ffmpeg.org/pipermail/ffmpeg-user/2013-July/016243.html

I have tried ffmpeg 1.2.2 on fedora and 2.0.1 on raspberry (same result).
I have only tried ffplay 1.2.2 on fedora.

if I use ffplay I get some output (a few seconds then still).



ffplay -f v4l2 -input_format h264 /dev/video0

ffplay version 1.2.2 Copyright (c) 2003-2013 the FFmpeg developers
  built on Aug  4 2013 08:23:55 with gcc 4.8.1 (GCC) 20130603 (Red Hat 4.8.1-1)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg
--incdir=/usr/include/ffmpeg --libdir=/usr/lib --mandir=/usr/share/man --arch=i686 --optflags='-O2
-g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4
-grecord-gcc-switches -m32 -march=i686 -mtune=atom -fasynchronous-unwind-tables' --enable-bzlib
--disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcelt
--enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame
--enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse
--enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2
--enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter
--enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug
--disable-stripping --shlibdir=/usr/lib --cpu=i686 --enable-runtime-cpudetect
  libavutil      52. 18.100 / 52. 18.100
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.104 / 54. 63.104
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 42.103 /  3. 42.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[video4linux2,v4l2 @ 0xb1b025c0] Estimating duration from bitrate, this may be inaccurate
Input #0, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 3512.615770, bitrate: N/A
    Stream #0:0: Video: h264 (Constrained Baseline), yuvj420p, 640x480 [SAR 1:1 DAR 4:3], -5 kb/s,
30 fps, 30 tbr, 1000k tbn, 60 tbc
3516.23 A-V:  0.000 fd=  10 aq=    0KB vq=   77KB sq=    0B f=8/8


From frans at fransdb.nl  Sun Aug 25 21:22:38 2013
From: frans at fransdb.nl (Frans de Boer)
Date: Sun, 25 Aug 2013 21:22:38 +0200
Subject: [FFmpeg-user] Fwd: error 127
In-Reply-To: <loom.20130825T150436-68@post.gmane.org>
References: <5219FBDB.3040903@fransdb.nl> <5219FDD5.9050704@fransdb.nl>
	<loom.20130825T150436-68@post.gmane.org>
Message-ID: <521A597E.7010703@fransdb.nl>

On 08/25/2013 03:10 PM, Carl Eugen Hoyos wrote:
> Frans de Boer <frans <at> fransdb.nl> writes:
>
>> when running 'make check', I get error 127 during the fft-4 test.
>> Meaning that libavutil.so.52 was not found.
>>
>> The library is in the build tree but not yet on the system
>> installed.
>
> Set LD_LIBRARY_PATH:
> $ export LD_LIBRARY_PATH=libavutil:libavcodec: ...
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
Thanks, that worked for me. But should that not be embedded in the 
testing scripts?. How many people will actually set the LD_LIBRARY_PATH 
before testing. In forgetting or not knowing this, they actually tests 
against old libraries.

I incorporated this in my scripts now, but it is rather different from 
all other programs I compiled sofar. So, it seems that the project is 
not using accepted and/or standard behavior for testing. Maybe a 
suggestion to incorporated some things in the test scripts?

Regards, Frans.


From cehoyos at ag.or.at  Mon Aug 26 00:01:43 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 25 Aug 2013 22:01:43 +0000 (UTC)
Subject: [FFmpeg-user] Fwd: error 127
References: <5219FBDB.3040903@fransdb.nl> <5219FDD5.9050704@fransdb.nl>
	<loom.20130825T150436-68@post.gmane.org>
	<521A597E.7010703@fransdb.nl>
Message-ID: <loom.20130826T000032-273@post.gmane.org>

Frans de Boer <frans <at> fransdb.nl> writes:

> How many people will actually set the LD_LIBRARY_PATH 
> before testing.

Everybody not using static libraries.

> In forgetting or not knowing this, they actually tests 
> against old libraries.

Many warnings are printed if you do this.

> I incorporated this in my scripts now, but it is rather 
> different from all other programs I compiled sofar.

Please provide a (one) example.

Carl Eugen


From cehoyos at ag.or.at  Mon Aug 26 00:07:08 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 25 Aug 2013 22:07:08 +0000 (UTC)
Subject: [FFmpeg-user] error 127
References: <5219FBDB.3040903@fransdb.nl>
Message-ID: <loom.20130826T000630-67@post.gmane.org>

Frans de Boer <frans <at> fransdb.nl> writes:

> Ok, I try to compile ffmpeg-2.0.1 as well as 
> the latest git repository and when running 
> 'make check'

Please confirm that you are setting the SAMPLES 
variable before running this test: The tests 
make not much sense without it.

Carl Eugen


From mike at orton.com  Mon Aug 26 00:41:11 2013
From: mike at orton.com (Michael Orton)
Date: Mon, 26 Aug 2013 10:41:11 +1200
Subject: [FFmpeg-user] Ticket #2892: sample file uploaded
Message-ID: <CE40E147.5C498%mike@orton.com>

Ticket #2892 (new defect) MPEG tbn value incorrectly evaluated: causes MOV
wrap to fail

Uploaded file MPEG_tbn_test.mov and explanatory MPEG_tbn_test.rtf into
upload.ffmpeg.org/incoming





From mariofutire at googlemail.com  Mon Aug 26 10:49:55 2013
From: mariofutire at googlemail.com (andrea)
Date: Mon, 26 Aug 2013 09:49:55 +0100
Subject: [FFmpeg-user] vlc does not like ffmpeg m3u8 live segments
Message-ID: <kvf4rb$ued$1@ger.gmane.org>

Hi,

I am trying to setup a live webcam stream and I use this command

capture -o | ffmpeg -re -f h264 -i - -c copy -map 0 -f segment -segment_list andrea.m3u8
-segment_list_flags +live -segment_time 5 -segment_wrap 10 -segment_list_size 5 out%03d.ts

(I need the capture program since ffmpeg cannot properly read h264 from my C920 webcam, there is an
other message in the list about that)

If I omit "-segment_list_size 5" then vlc is happy and plays continuously.
As soon as I add "-segment_list_size 5" then vlc stops with this message

    stream_filter_httplive stream error: gap in sequence numbers found: new=0 expected 14

Cant really say whose fault it is, but I did find this old bug report for ffmpeg.

https://trac.ffmpeg.org/ticket/1842

about incorrect segment numbers. I wonder if the 2 things are related.

BTW, without "-segment_list_size 5" the list .m3u8 seems to grow indefinitely. Is this correct?
It is only a text file, but eventually it will fill the hard disk.


From chronek at interia.eu  Mon Aug 26 11:08:01 2013
From: chronek at interia.eu (chronek)
Date: Mon, 26 Aug 2013 11:08:01 +0200
Subject: [FFmpeg-user] uspp filter 444p
Message-ID: <nukmceguhrpioaposwzc@iwcm>

Hello,&nbsp;I want ask if someone from developers could check uspp filter. This filter is port from mplayer and it is uniqe, cause doing full compresion/decompression based on snow codec. Snow codec support pixel format 444p so changing uspp filter to support that will not need many work.&nbsp;I asking cause it is my main filter and any more quality is appreciated (btw i am using uspp not as postprocesor filter , but avaraging filter with fixed qp, now is: source -&gt; scale (444p) -&gt; uspp (420p) -&gt; unsharp (444p) -&gt; scale (444p) -&gt; lossless (rgb24) and uspp is bottleneck)Mike

From diljit77 at gmail.com  Mon Aug 26 11:39:02 2013
From: diljit77 at gmail.com (Diljit Singh)
Date: Mon, 26 Aug 2013 15:09:02 +0530
Subject: [FFmpeg-user] Audio transcoding
Message-ID: <CAKH46aSR_ahjheHKVxs6t2DFpg5M7eHr2rGruEXN5R76t2a-PQ@mail.gmail.com>

Dear All,

Want to transcode audio in following format. Please suggest

16 bit, 8KHz, Mono- 9 db, mono, G.711 A- Law encoding

As far as sampling frequency and no of channel is concern I know the flag
used, but not for rest of the configurations

Regards
Diljit Singh

From mirko.puliafito at gmail.com  Mon Aug 26 11:42:22 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 11:42:22 +0200
Subject: [FFmpeg-user] Audio transcoding
In-Reply-To: <CAKH46aSR_ahjheHKVxs6t2DFpg5M7eHr2rGruEXN5R76t2a-PQ@mail.gmail.com>
References: <CAKH46aSR_ahjheHKVxs6t2DFpg5M7eHr2rGruEXN5R76t2a-PQ@mail.gmail.com>
Message-ID: <CAFGwXfx8wqGdH6jeorPm1tJC+VtMoGrVD431vB=pufFZhcrcyw@mail.gmail.com>

Try with:

ffmpeg -i <your_audio_source_path> -acodec pcm_alaw -ar 8000 -ac 1
<your_dest_file_path>

Mirko
2013/8/26 Diljit Singh <diljit77 at gmail.com>

> Dear All,
>
> Want to transcode audio in following format. Please suggest
>
> 16 bit, 8KHz, Mono- 9 db, mono, G.711 A- Law encoding
>
> As far as sampling frequency and no of channel is concern I know the flag
> used, but not for rest of the configurations
>
> Regards
> Diljit Singh
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From diljit77 at gmail.com  Mon Aug 26 11:56:32 2013
From: diljit77 at gmail.com (Diljit Singh)
Date: Mon, 26 Aug 2013 15:26:32 +0530
Subject: [FFmpeg-user] Audio transcoding
In-Reply-To: <CAFGwXfx8wqGdH6jeorPm1tJC+VtMoGrVD431vB=pufFZhcrcyw@mail.gmail.com>
References: <CAKH46aSR_ahjheHKVxs6t2DFpg5M7eHr2rGruEXN5R76t2a-PQ@mail.gmail.com>
	<CAFGwXfx8wqGdH6jeorPm1tJC+VtMoGrVD431vB=pufFZhcrcyw@mail.gmail.com>
Message-ID: <CAKH46aTUYRZBcRgvk4fvK=birN0p5bAoAoLvYzo3fpFwc6hQfw@mail.gmail.com>

thanks Mirko,

but how we can controls bits per sample ?

Regards
Diljit Singh


On Mon, Aug 26, 2013 at 3:12 PM, Mirko Puliafito
<mirko.puliafito at gmail.com>wrote:

> Try with:
>
> ffmpeg -i <your_audio_source_path> -acodec pcm_alaw -ar 8000 -ac 1
> <your_dest_file_path>
>
> Mirko
> 2013/8/26 Diljit Singh <diljit77 at gmail.com>
>
> > Dear All,
> >
> > Want to transcode audio in following format. Please suggest
> >
> > 16 bit, 8KHz, Mono- 9 db, mono, G.711 A- Law encoding
> >
> > As far as sampling frequency and no of channel is concern I know the flag
> > used, but not for rest of the configurations
> >
> > Regards
> > Diljit Singh
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From mirko.puliafito at gmail.com  Mon Aug 26 11:58:44 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 11:58:44 +0200
Subject: [FFmpeg-user] Audio transcoding
In-Reply-To: <CAKH46aTUYRZBcRgvk4fvK=birN0p5bAoAoLvYzo3fpFwc6hQfw@mail.gmail.com>
References: <CAKH46aSR_ahjheHKVxs6t2DFpg5M7eHr2rGruEXN5R76t2a-PQ@mail.gmail.com>
	<CAFGwXfx8wqGdH6jeorPm1tJC+VtMoGrVD431vB=pufFZhcrcyw@mail.gmail.com>
	<CAKH46aTUYRZBcRgvk4fvK=birN0p5bAoAoLvYzo3fpFwc6hQfw@mail.gmail.com>
Message-ID: <CAFGwXfyXL1cpXS1WLP_jjueB6gLKom5UjKPmD2VUmMAd_BBc6g@mail.gmail.com>

Alaw is 16 bits per sample. If you want other PCM formats take a look here:

http://trac.ffmpeg.org/wiki/audio%20types

Mirko


2013/8/26 Diljit Singh <diljit77 at gmail.com>

> thanks Mirko,
>
> but how we can controls bits per sample ?
>
> Regards
> Diljit Singh
>
>
> On Mon, Aug 26, 2013 at 3:12 PM, Mirko Puliafito
> <mirko.puliafito at gmail.com>wrote:
>
> > Try with:
> >
> > ffmpeg -i <your_audio_source_path> -acodec pcm_alaw -ar 8000 -ac 1
> > <your_dest_file_path>
> >
> > Mirko
> > 2013/8/26 Diljit Singh <diljit77 at gmail.com>
> >
> > > Dear All,
> > >
> > > Want to transcode audio in following format. Please suggest
> > >
> > > 16 bit, 8KHz, Mono- 9 db, mono, G.711 A- Law encoding
> > >
> > > As far as sampling frequency and no of channel is concern I know the
> flag
> > > used, but not for rest of the configurations
> > >
> > > Regards
> > > Diljit Singh
> > > _______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From mirko.puliafito at gmail.com  Mon Aug 26 12:07:59 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 12:07:59 +0200
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
Message-ID: <CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>

The command:

./ffmpeg -loglevel debug -i rtmpe://127.0.0.1/live/myStream_audio -acodec
copy -vcodec copy -f flv /home/mirkpul/Scrivania/test.flv

Debug level output:

ffmpeg version N-55765-g8d679bc Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 26 2013 12:05:40 with gcc 4.7 (Ubuntu/Linaro 4.7.2-2ubuntu1)
  configuration: --enable-openssl --enable-protocol=rtmpe
  libavutil      52. 42.100 / 52. 42.100
  libavcodec     55. 29.100 / 55. 29.100
  libavformat    55. 14.102 / 55. 14.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.102 /  3. 82.102
  libswscale      2.  5.100 /  2.  5.100
  libswresample   0. 17.103 /  0. 17.103
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging
level) with argument 'debug'.
Reading option '-i' ... matched as input file with argument 'rtmpe://
127.0.0.1/live/myStream_audio'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'flv'.
Reading option '/home/mirkpul/Scrivania/test.flv' ... matched as output
file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file rtmpe://127.0.0.1/live/myStream_audio
.
Successfully parsed a group of options.
Opening an input file: rtmpe://127.0.0.1/live/myStream_audio.
[rtmpe @ 0x2cec540] Handshaking...
[rtmpe @ 0x2cec540] Type answer 6
[rtmpe @ 0x2cec540] Server version 0.0.0.0
[rtmpe @ 0x2cec540] Proto = rtmpe, path = /live/myStream_audio, app = live,
fname = myStream_audio
rtmpe://127.0.0.1/live/myStream_audio: Input/output error

It seems that the server is receiving an empty message. Is anyone able to
make it work?

Mirko


2013/8/23 Mirko Puliafito <mirko.puliafito at gmail.com>

> Hi,
>
> just try to demux a stream, file or live, from a flash media server (I
> tried with crtmpserver and red5). It only opens the TCP connection but it
> doesn't start the handshake giving an invalid parameter error.
>
> I tried with..
>
> ffmpeg -i rtmpe://127.0.0.1/live/myStream ...
>
> Mirko
> Il giorno 23/ago/2013 12:05, "Paul B Mahol" <onemda at gmail.com> ha scritto:
>
>> On 8/23/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
>>
>> > Hi,
>> >
>> > I'm trying to use rtmpe (ffrtmpcrypt) with the latest version but it's
>> not
>> > working (I've also tested it with the 1.x). I tested it against
>> crtmpserver
>> > and red5 without any luck. Can anybody tell me the status of that
>> protocol
>> > implementation? Do I need to go on using rtmpdump libs if I want to use
>> > this protocol?
>>
>> What does not work exactly?
>>
>> There is simple not enough information provided.
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>

From onemda at gmail.com  Mon Aug 26 12:13:08 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 26 Aug 2013 10:13:08 +0000
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
Message-ID: <CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>

On 8/26/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> The command:
>
> ./ffmpeg -loglevel debug -i rtmpe://127.0.0.1/live/myStream_audio -acodec
> copy -vcodec copy -f flv /home/mirkpul/Scrivania/test.flv

You need to set several options.

From mirko.puliafito at gmail.com  Mon Aug 26 13:22:46 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 13:22:46 +0200
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
Message-ID: <CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>

Can you please provide me a sample?

Mirko
Il giorno 26/ago/2013 12:13, "Paul B Mahol" <onemda at gmail.com> ha scritto:

> On 8/26/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> > The command:
> >
> > ./ffmpeg -loglevel debug -i rtmpe://127.0.0.1/live/myStream_audio-acodec
> > copy -vcodec copy -f flv /home/mirkpul/Scrivania/test.flv
>
> You need to set several options.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From 89.yavuz at gmail.com  Mon Aug 26 13:35:17 2013
From: 89.yavuz at gmail.com (Mustafa Yavuz)
Date: Mon, 26 Aug 2013 14:35:17 +0300
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
Message-ID: <CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>

Does ffmpeg supports rtmpe protocol? Last time I worked with ffmpeg it did
not use to support it so I have used rtmpdump and pipe it to the ffmpeg.

Mirko I have a live stream url like that using with rtmpdump but ffmpeg
also needs these parameters like swfVfy otherwise it does not work.

"rtmpdump -v -r rtmpe://
cp195822.live.edgefcs.net:1935/live/1_arugyvor_1 at 93816 --swfVfy
http://dogus.kaltura.com.edgesuite.net/p/1076322/sp/107632200/flash/kdp3/v3.6.9/kdp3.swf
"


2013/8/26 Mirko Puliafito <mirko.puliafito at gmail.com>

> Can you please provide me a sample?
>
> Mirko
> Il giorno 26/ago/2013 12:13, "Paul B Mahol" <onemda at gmail.com> ha scritto:
>
> > On 8/26/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> > > The command:
> > >
> > > ./ffmpeg -loglevel debug -i rtmpe://
> 127.0.0.1/live/myStream_audio-acodec
> > > copy -vcodec copy -f flv /home/mirkpul/Scrivania/test.flv
> >
> > You need to set several options.
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From mirko.puliafito at gmail.com  Mon Aug 26 14:04:56 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 14:04:56 +0200
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
	<CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
Message-ID: <CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>

Hi Mustafa,

yes, it can work both with rtmpdump lib and now also natively but I've
never been able to make it work with the native. With rtmpdump lib
everything works without any other parameters. I'll take a deeper look at
the sources or waiting for Paul for the options.

Mirko


2013/8/26 Mustafa Yavuz <89.yavuz at gmail.com>

> Does ffmpeg supports rtmpe protocol? Last time I worked with ffmpeg it did
> not use to support it so I have used rtmpdump and pipe it to the ffmpeg.
>
> Mirko I have a live stream url like that using with rtmpdump but ffmpeg
> also needs these parameters like swfVfy otherwise it does not work.
>
> "rtmpdump -v -r rtmpe://
> cp195822.live.edgefcs.net:1935/live/1_arugyvor_1 at 93816 --swfVfy
>
> http://dogus.kaltura.com.edgesuite.net/p/1076322/sp/107632200/flash/kdp3/v3.6.9/kdp3.swf
> "
>
>
> 2013/8/26 Mirko Puliafito <mirko.puliafito at gmail.com>
>
> > Can you please provide me a sample?
> >
> > Mirko
> > Il giorno 26/ago/2013 12:13, "Paul B Mahol" <onemda at gmail.com> ha
> scritto:
> >
> > > On 8/26/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> > > > The command:
> > > >
> > > > ./ffmpeg -loglevel debug -i rtmpe://
> > 127.0.0.1/live/myStream_audio-acodec
> > > > copy -vcodec copy -f flv /home/mirkpul/Scrivania/test.flv
> > >
> > > You need to set several options.
> > > _______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Mon Aug 26 14:35:15 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 26 Aug 2013 12:35:15 +0000
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
	<CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
	<CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>
Message-ID: <CAPYw7P6+Qnj4vyUaERSa=mtJLbtpusiniJquB2byJs07oKq4wg@mail.gmail.com>

On 8/26/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> Hi Mustafa,
>
> yes, it can work both with rtmpdump lib and now also natively but I've
> never been able to make it work with the native. With rtmpdump lib
> everything works without any other parameters. I'll take a deeper look at
> the sources or waiting for Paul for the options.
>

Please stop top posting.

What is command you use with rtmpdump?

Note that you can configure ffmpeg to disable native rtmp* protocols.

From mirko.puliafito at gmail.com  Mon Aug 26 14:40:25 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 14:40:25 +0200
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAPYw7P6+Qnj4vyUaERSa=mtJLbtpusiniJquB2byJs07oKq4wg@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
	<CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
	<CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>
	<CAPYw7P6+Qnj4vyUaERSa=mtJLbtpusiniJquB2byJs07oKq4wg@mail.gmail.com>
Message-ID: <CAFGwXfzq_GRUgASqhDQ3_OSZcq6=36yS83DSk+oKM3QsEt_ZCA@mail.gmail.com>

I don't want to use rtmpdump but the native implementation.

ffmpeg -i rtmpe://127.0.0.1/live/myStream -acodec copy -vcodec copy -f flv
/home/mirkpul/Scrivania/test.flv

works with rtmpdump but doesn't work with native.

Mirko

From hinavinmath at gmail.com  Mon Aug 26 14:43:24 2013
From: hinavinmath at gmail.com (Navin Math)
Date: Mon, 26 Aug 2013 18:13:24 +0530
Subject: [FFmpeg-user] Stream images using ffmpeg
Message-ID: <CANHAjRxndqewUHSR-EH4Mp95+WzzY9Lmxz9n8StH4428Oce1Kg@mail.gmail.com>

Hi

My work is to stream content of camera from mobile - 1 to other another
mobile - 2. So I am planning to implement following method.
Note: Both mobile-1 and mobile-2 are not in same network. They can contact
each other through some server.

I am capturing video frames(images) from camera in mobile-1 then i will
send the current extracted image to server for ex: curenrtImage.jpg. From
server, i want to stream that image to mobile-2 through ffmpeg.

1. How to stream this image?

2. I want to know whether the method what I am following is correct or not?
or
3. Give any leads to start any new path.


Thanks
Math

From onemda at gmail.com  Mon Aug 26 14:55:09 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 26 Aug 2013 12:55:09 +0000
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAFGwXfzq_GRUgASqhDQ3_OSZcq6=36yS83DSk+oKM3QsEt_ZCA@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
	<CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
	<CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>
	<CAPYw7P6+Qnj4vyUaERSa=mtJLbtpusiniJquB2byJs07oKq4wg@mail.gmail.com>
	<CAFGwXfzq_GRUgASqhDQ3_OSZcq6=36yS83DSk+oKM3QsEt_ZCA@mail.gmail.com>
Message-ID: <CAPYw7P5dUYqnvR+wnOcVBeWMBgAhrkb2MWGFjR3HAWPBBOaaCg@mail.gmail.com>

On 8/26/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> I don't want to use rtmpdump but the native implementation.
>
> ffmpeg -i rtmpe://127.0.0.1/live/myStream -acodec copy -vcodec copy -f flv
> /home/mirkpul/Scrivania/test.flv
>
> works with rtmpdump but doesn't work with native.


You can use rtmpdump lib implementation with ffmpeg.

But you still did not provide command you use with rtmpdump.

From mirko.puliafito at gmail.com  Mon Aug 26 15:31:05 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 15:31:05 +0200
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAPYw7P5dUYqnvR+wnOcVBeWMBgAhrkb2MWGFjR3HAWPBBOaaCg@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
	<CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
	<CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>
	<CAPYw7P6+Qnj4vyUaERSa=mtJLbtpusiniJquB2byJs07oKq4wg@mail.gmail.com>
	<CAFGwXfzq_GRUgASqhDQ3_OSZcq6=36yS83DSk+oKM3QsEt_ZCA@mail.gmail.com>
	<CAPYw7P5dUYqnvR+wnOcVBeWMBgAhrkb2MWGFjR3HAWPBBOaaCg@mail.gmail.com>
Message-ID: <CAFGwXfxniuDzrEr1M-OwCO1eQCYdmpmNWNWmwBZ7qjF3wjfc2A@mail.gmail.com>

I
tried it compiling ffmpeg with rtmpdump support. Never tried rtmpdump
directly.

Mirko

From onemda at gmail.com  Mon Aug 26 15:36:23 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 26 Aug 2013 13:36:23 +0000
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAFGwXfxniuDzrEr1M-OwCO1eQCYdmpmNWNWmwBZ7qjF3wjfc2A@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
	<CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
	<CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>
	<CAPYw7P6+Qnj4vyUaERSa=mtJLbtpusiniJquB2byJs07oKq4wg@mail.gmail.com>
	<CAFGwXfzq_GRUgASqhDQ3_OSZcq6=36yS83DSk+oKM3QsEt_ZCA@mail.gmail.com>
	<CAPYw7P5dUYqnvR+wnOcVBeWMBgAhrkb2MWGFjR3HAWPBBOaaCg@mail.gmail.com>
	<CAFGwXfxniuDzrEr1M-OwCO1eQCYdmpmNWNWmwBZ7qjF3wjfc2A@mail.gmail.com>
Message-ID: <CAPYw7P5uC_hcDh2DKW8ovbfiG-0a=WwacQgWd+C_ouM=3BG9Vw@mail.gmail.com>

On 8/26/13, Mirko Puliafito <mirko.puliafito at gmail.com> wrote:
> I
> tried it compiling ffmpeg with rtmpdump support. Never tried rtmpdump
> directly.

Then it looks like to be bug, please report it.

From mirko.puliafito at gmail.com  Mon Aug 26 15:51:37 2013
From: mirko.puliafito at gmail.com (Mirko Puliafito)
Date: Mon, 26 Aug 2013 15:51:37 +0200
Subject: [FFmpeg-user] RTMPE native support
In-Reply-To: <CAPYw7P5uC_hcDh2DKW8ovbfiG-0a=WwacQgWd+C_ouM=3BG9Vw@mail.gmail.com>
References: <CAFGwXfyos5y=sJWdQ5Bj48hEVp290uhQLYRTSi0MQveAYuKE9A@mail.gmail.com>
	<CAPYw7P4kLoEY0DVWXBYPZ9XzdeeSaKY6BqOS060w027=XjEu+w@mail.gmail.com>
	<CAFGwXfwbM1hygiGPSRy09GPAPM5izC_MV2uoAvw7Nb_VWyFS-g@mail.gmail.com>
	<CAFGwXfw1LsT0N2_-VeaTyJdaHFJNAeYp0ygDtTr9mRO=m-Yuqw@mail.gmail.com>
	<CAPYw7P47c0SO0iJ7q4HOgXquAWfF2TEV_=ajysq4WA6_HVZH3Q@mail.gmail.com>
	<CAFGwXfzqqVsmZog7BPzhPEceMkSQc4ibgAfiU6aYWJExK1-gAg@mail.gmail.com>
	<CANK2ZX4RnmvRqfhZQ_02Kyo4U-TvAKeW6e6O-R9xq5cF_uL2Qw@mail.gmail.com>
	<CAFGwXfz2ahRwzfbSTB8A__6AQ=ZNqRq1QoFtbgVfrnRHEDYWDQ@mail.gmail.com>
	<CAPYw7P6+Qnj4vyUaERSa=mtJLbtpusiniJquB2byJs07oKq4wg@mail.gmail.com>
	<CAFGwXfzq_GRUgASqhDQ3_OSZcq6=36yS83DSk+oKM3QsEt_ZCA@mail.gmail.com>
	<CAPYw7P5dUYqnvR+wnOcVBeWMBgAhrkb2MWGFjR3HAWPBBOaaCg@mail.gmail.com>
	<CAFGwXfxniuDzrEr1M-OwCO1eQCYdmpmNWNWmwBZ7qjF3wjfc2A@mail.gmail.com>
	<CAPYw7P5uC_hcDh2DKW8ovbfiG-0a=WwacQgWd+C_ouM=3BG9Vw@mail.gmail.com>
Message-ID: <CAFGwXfyBbPy+pfVdNEThwNEOAXX96eu-ucT4h3LPKwo0FGG_tA@mail.gmail.com>

Ok. BTW I've just tried and this works:

./rtmpdump -v -r "rtmpe://127.0.0.1/live/myStream_audio" -o
/home/mirkpul/Scrivania/test.flv

Mirko

From diljit77 at gmail.com  Mon Aug 26 16:29:10 2013
From: diljit77 at gmail.com (Diljit Singh)
Date: Mon, 26 Aug 2013 19:59:10 +0530
Subject: [FFmpeg-user] Audio transcoding
In-Reply-To: <CAFGwXfyXL1cpXS1WLP_jjueB6gLKom5UjKPmD2VUmMAd_BBc6g@mail.gmail.com>
References: <CAKH46aSR_ahjheHKVxs6t2DFpg5M7eHr2rGruEXN5R76t2a-PQ@mail.gmail.com>
	<CAFGwXfx8wqGdH6jeorPm1tJC+VtMoGrVD431vB=pufFZhcrcyw@mail.gmail.com>
	<CAKH46aTUYRZBcRgvk4fvK=birN0p5bAoAoLvYzo3fpFwc6hQfw@mail.gmail.com>
	<CAFGwXfyXL1cpXS1WLP_jjueB6gLKom5UjKPmD2VUmMAd_BBc6g@mail.gmail.com>
Message-ID: <CAKH46aTgT1eTQQXFe-BJAHy6-pH2cZmuc4FmOca+WTiU-aSyAw@mail.gmail.com>

I tried it , but doesn't seem to be working.

Then I tried following command
ffmpeg -i SDR.wav -acodec pcm_u16be -ar 8000 -ac 1 -y SDR_final.wav

When I tried to test output file in vlc player it still shows 8 bit sample
width.

Please suggest.

Regards,
Diljit singh

On Monday, 26 August 2013, Mirko Puliafito wrote:

> Alaw is 16 bits per sample. If you want other PCM formats take a look here:
>
> http://trac.ffmpeg.org/wiki/audio%20types
>
> Mirko
>
>
> 2013/8/26 Diljit Singh <diljit77 at gmail.com <javascript:;>>
>
> > thanks Mirko,
> >
> > but how we can controls bits per sample ?
> >
> > Regards
> > Diljit Singh
> >
> >
> > On Mon, Aug 26, 2013 at 3:12 PM, Mirko Puliafito
> > <mirko.puliafito at gmail.com <javascript:;>>wrote:
> >
> > > Try with:
> > >
> > > ffmpeg -i <your_audio_source_path> -acodec pcm_alaw -ar 8000 -ac 1
> > > <your_dest_file_path>
> > >
> > > Mirko
> > > 2013/8/26 Diljit Singh <diljit77 at gmail.com <javascript:;>>
> > >
> > > > Dear All,
> > > >
> > > > Want to transcode audio in following format. Please suggest
> > > >
> > > > 16 bit, 8KHz, Mono- 9 db, mono, G.711 A- Law encoding
> > > >
> > > > As far as sampling frequency and no of channel is concern I know the
> > flag
> > > > used, but not for rest of the configurations
> > > >
> > > > Regards
> > > > Diljit Singh
> > > > _______________________________________________
> > > > ffmpeg-user mailing list
> > > > ffmpeg-user at ffmpeg.org <javascript:;>
> > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > > >
> > > _______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org <javascript:;>
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org <javascript:;>
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org <javascript:;>
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From frans at fransdb.nl  Mon Aug 26 16:49:42 2013
From: frans at fransdb.nl (Frans de Boer)
Date: Mon, 26 Aug 2013 16:49:42 +0200
Subject: [FFmpeg-user] error 127
In-Reply-To: <loom.20130826T000630-67@post.gmane.org>
References: <5219FBDB.3040903@fransdb.nl>
	<loom.20130826T000630-67@post.gmane.org>
Message-ID: <521B6B06.4050704@fransdb.nl>

On 08/26/2013 12:07 AM, Carl Eugen Hoyos wrote:
> Frans de Boer <frans <at> fransdb.nl> writes:
>
>> Ok, I try to compile ffmpeg-2.0.1 as well as
>> the latest git repository and when running
>> 'make check'
>
> Please confirm that you are setting the SAMPLES
> variable before running this test: The tests
> make not much sense without it.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
Thanks, I just did it and it passes all.

From frans at fransdb.nl  Mon Aug 26 16:49:04 2013
From: frans at fransdb.nl (Frans de Boer)
Date: Mon, 26 Aug 2013 16:49:04 +0200
Subject: [FFmpeg-user] Fwd: error 127
In-Reply-To: <loom.20130826T000032-273@post.gmane.org>
References: <5219FBDB.3040903@fransdb.nl> <5219FDD5.9050704@fransdb.nl>
	<loom.20130825T150436-68@post.gmane.org>
	<521A597E.7010703@fransdb.nl>
	<loom.20130826T000032-273@post.gmane.org>
Message-ID: <521B6AE0.70303@fransdb.nl>

On 08/26/2013 12:01 AM, Carl Eugen Hoyos wrote:
> Frans de Boer <frans <at> fransdb.nl> writes:
>
>> How many people will actually set the LD_LIBRARY_PATH
>> before testing.
>
> Everybody not using static libraries.

In hindsight this seems logic. But I can't recall ever to have problems 
when first time compiling various libraries and checking them. And no, 
no older libraries where present on the system.

>
>> In forgetting or not knowing this, they actually tests
>> against old libraries.
>
> Many warnings are printed if you do this.

Actually only one warning is given saying that not all fate test are run.

>
>> I incorporated this in my scripts now, but it is rather
>> different from all other programs I compiled sofar.
>
> Please provide a (one) example.
>
libjpeg, libxml2, libcurl etc.

They all compiled and passes the make check without prior libraries present.


> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


From soho123.2012 at gmail.com  Mon Aug 26 17:15:36 2013
From: soho123.2012 at gmail.com (Huang Soho)
Date: Mon, 26 Aug 2013 23:15:36 +0800
Subject: [FFmpeg-user] ffserver occupy a lot of CPU time while streaming
Message-ID: <CAFd2cZVhdSJBLOPb8iasRyk=CeodxvUSgjFGf4GxtirBF4TPTA@mail.gmail.com>

Hi All,


I use ffmpeg+ ffserver for a stream server.
I use "top" to check CPU usage when I use ffserver to stream output.
I can see ffserver occpuy about 67% and ffmpeg occupy about 30%
it is almost 100% when I stream out

does anyone have idea?
is there any patch for that?
I use the head master vertsion

From schultzjl at verizon.net  Mon Aug 26 20:46:52 2013
From: schultzjl at verizon.net (Jeffrey Schultz)
Date: Mon, 26 Aug 2013 14:46:52 -0400
Subject: [FFmpeg-user] I-frame only video from source input
Message-ID: <1DFF5D50-755E-41B4-827A-F9A58E2ADBE8@verizon.net>

Hi,

How can I create an I-frame only video from a source video file, and what is the most efficient set of CLI arguments for doing this? Also, would it be possible for these I-frames to maintain their original times within the output video?

Jeff

From mediastream at gmail.com  Tue Aug 27 00:46:37 2013
From: mediastream at gmail.com (Dennis)
Date: Mon, 26 Aug 2013 18:46:37 -0400
Subject: [FFmpeg-user] Watermark image, fade in and out.
Message-ID: <CAPCJWRo1y_PNJutjkLnRQfhgVTscYnu+z0HD+c1RrRRZyBDiMg@mail.gmail.com>

Is there a way to display watermark for ten seconds then fade it out?

So far this works for me, but if I add fade out it fades out the whole
image not just the watermark  -- but if I add duration it removes the
watermark after 10 sec. Not sure how command is defined as global versus
localized to watermark only.

Shows logo for 10 sec.
ffmpeg2 -i big_buck_bunny_1080p_h264.mov -i logo1.png -filter_complex
'overlay=100:main_h-overlay_h-200:enable=lte(t\,10)'  -t 20 1.mp4

Shows logo for 10 sec then fades out the whole screen with logo, so its 1/2
working (:
ffmpeg2 -i big_buck_bunny_1080p_h264.mov -i logo1.png -filter_complex
'overlay=100:main_h-overlay_h-200:enable=lte(t\,10),fade=out:0:300:alpha=1'
-t 20 1.mp4

Any advice on how to fade out logo only please?


Thank you.
Dennis.

From mediastream at gmail.com  Tue Aug 27 00:53:52 2013
From: mediastream at gmail.com (Dennis)
Date: Mon, 26 Aug 2013 18:53:52 -0400
Subject: [FFmpeg-user] I-frame only video from source input
In-Reply-To: <1DFF5D50-755E-41B4-827A-F9A58E2ADBE8@verizon.net>
References: <1DFF5D50-755E-41B4-827A-F9A58E2ADBE8@verizon.net>
Message-ID: <CAPCJWRreDD3b96BB8b=3GEFmZTc8ciV20_rti=+k9PSNpBknEw@mail.gmail.com>

used to be -intra; or -g 1 would also work.




On Mon, Aug 26, 2013 at 2:46 PM, Jeffrey Schultz <schultzjl at verizon.net>wrote:

> Hi,
>
> How can I create an I-frame only video from a source video file, and what
> is the most efficient set of CLI arguments for doing this? Also, would it
> be possible for these I-frames to maintain their original times within the
> output video?
>
> Jeff
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From schultzjl at verizon.net  Tue Aug 27 01:16:58 2013
From: schultzjl at verizon.net (Jeffrey Schultz)
Date: Mon, 26 Aug 2013 19:16:58 -0400
Subject: [FFmpeg-user] I-frame only video from source input
In-Reply-To: <CAPCJWRreDD3b96BB8b=3GEFmZTc8ciV20_rti=+k9PSNpBknEw@mail.gmail.com>
References: <1DFF5D50-755E-41B4-827A-F9A58E2ADBE8@verizon.net>
	<CAPCJWRreDD3b96BB8b=3GEFmZTc8ciV20_rti=+k9PSNpBknEw@mail.gmail.com>
Message-ID: <E7850235-9ECF-465F-9281-9EB88EA5DB03@verizon.net>

Thank you!

On Aug 26, 2013, at 6:53 PM, Dennis <mediastream at gmail.com> wrote:

> used to be -intra; or -g 1 would also work.
> 
> 
> 
> 
> On Mon, Aug 26, 2013 at 2:46 PM, Jeffrey Schultz <schultzjl at verizon.net>wrote:
> 
>> Hi,
>> 
>> How can I create an I-frame only video from a source video file, and what
>> is the most efficient set of CLI arguments for doing this? Also, would it
>> be possible for these I-frames to maintain their original times within the
>> output video?
>> 
>> Jeff
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From jp.gehrig at edeltech.ch  Tue Aug 27 11:57:28 2013
From: jp.gehrig at edeltech.ch (Jean-Pierre Gehrig)
Date: Tue, 27 Aug 2013 11:57:28 +0200
Subject: [FFmpeg-user] MJPEG Compatibility with VirtualDub / Adobe Premiere
Message-ID: <0CEAD587-6FD0-4F26-8345-2A89DB1A0C0C@edeltech.ch>

Hello,

I have a question about MJPEG encoding with ffmpeg and compatibility with VirtualDub and Adobe Premiere CS6.

ffmpeg -i input.avi -vcodec mjpeg -pix_fmt yuvj422p -an output.avi

In Adobe Premiere CS6 the video is black and VirtualDub tells me "Error: Multiple chroma blocks not supported".

Not sure if the problems are related, but has anyone successfully played back ffmpeg MJPEG files in VirtualDub or Premiere ?

Thanks!
JP

From jiachielee at live.com  Tue Aug 27 11:45:57 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 27 Aug 2013 02:45:57 -0700 (PDT)
Subject: [FFmpeg-user] Splitting at key frames?
In-Reply-To: <slk619t0dt67290v5rt0e38s55qjk0fc6v@4ax.com>
References: <slk619t0dt67290v5rt0e38s55qjk0fc6v@4ax.com>
Message-ID: <1377596757740-4660937.post@n4.nabble.com>

I?ve never tried on VP6 video before. The online storage service you used has
a fairly low download speed to me. Would you please upload it to other file
storage service such as SkyDrive or MediaFire? You can split the files into
different parts using 7-zip (free) or WinZip (Trial) or WinRAR (Trial) to
get over the file size limitation. Thank you.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Splitting-at-key-frames-tp4660798p4660937.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From onemda at gmail.com  Tue Aug 27 13:30:02 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 27 Aug 2013 11:30:02 +0000
Subject: [FFmpeg-user] MJPEG Compatibility with VirtualDub / Adobe
	Premiere
In-Reply-To: <0CEAD587-6FD0-4F26-8345-2A89DB1A0C0C@edeltech.ch>
References: <0CEAD587-6FD0-4F26-8345-2A89DB1A0C0C@edeltech.ch>
Message-ID: <CAPYw7P7B1FdavS7xhbKfekt40vFzOg35C=LOimMuCqPAKHdu3w@mail.gmail.com>

On 8/27/13, Jean-Pierre Gehrig <jp.gehrig at edeltech.ch> wrote:
> Hello,
>
> I have a question about MJPEG encoding with ffmpeg and compatibility with
> VirtualDub and Adobe Premiere CS6.
>
> ffmpeg -i input.avi -vcodec mjpeg -pix_fmt yuvj422p -an output.avi
>
> In Adobe Premiere CS6 the video is black and VirtualDub tells me "Error:
> Multiple chroma blocks not supported".
>
> Not sure if the problems are related, but has anyone successfully played
> back ffmpeg MJPEG files in VirtualDub or Premiere ?

Perhaps they do not support yuvj422p subsampling?

From sw.craftsman at gmail.com  Tue Aug 27 13:55:41 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Tue, 27 Aug 2013 17:25:41 +0530
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
Message-ID: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>

Hi,

Just realized I am seeing incorrect frame-rates for number of videos after
upgrading from 0.6.1 to 1.2 sometime back.

E.g. for below video (http://files.videohelp.com/u/211860/Video_LXF.lxf) it
used to be 29.97 fps earlier, it is 14.99 now. I have verified in other
tools like MediaInfo that the correct frame-rate is 29.97 only.

*(Output)*
----------------------------------------------------------------------------
(0.6.1)
------
C:\Users\Administrator\Desktop\FFMpeg>0.6.1\ffprobe.exe "Y:\Video_LXF.lxf"
ffprobe version N-35462-g61b1d85, Copyright (c) 2007-2011 the FFmpeg
developers
  built on Dec  5 2011 14:22:27 with gcc 4.6.2
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-ru
ntime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r
--enable-libope
ncore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm
--en
able-libmp3lame --enable-libopenjpeg --enable-librtmp
--enable-libschroedinger -
-enable-libspeex --enable-libtheora --enable-libvo-aacenc
--enable-libvo-amrwben
c --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs
--enable-
libxvid --enable-zlib
  libavutil    51. 30. 0 / 51. 30. 0
  libavcodec   53. 40. 0 / 53. 40. 0
  libavformat  53. 24. 0 / 53. 24. 0
  libavdevice  53.  4. 0 / 53.  4. 0
  libavfilter   2. 51. 0 /  2. 51. 0
  libswscale    2.  1. 0 /  2.  1. 0
  libpostproc  51.  2. 0 / 51.  2. 0
Input #0, lxf, from 'Y:\Rajiv\CTS-2219 (Frame Rate generated by CTSD7.1 for
LXF
formats is wrong)\Video_LXF.lxf':
  Duration: 00:00:16.88, start: 0.000000, bitrate: 12229 kb/s
    Stream #0:0: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p,
720x
480 [SAR 8:9 DAR 4:3], 6000 kb/s, *29.97 fps*, 29.97 tbr, 29.97 tbn, 59.94
tbc
    Stream #0:1: Audio: pcm_s16le, 48000 Hz, 8 channels, s16, 6144 kb/s
----------------------------------------------------------------------------

C:\Users\Administrator\Desktop\FFMpeg>1.2\ffprobe.exe "Y:\Video_LXF.lxf"
ffprobe version 1.2 Copyright (c) 2007-2013 the FFmpeg developers
  built on Mar 28 2013 00:34:08 with gcc 4.8.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
--enab
le-libass --enable-libbluray --enable-libcaca --enable-libfreetype
--enable-libg
sm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb
--enable-libo
pencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-li
bschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-lib
twolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enabl
e-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 18.100 / 52. 18.100
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.104 / 54. 63.104
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 42.103 /  3. 42.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
Input #0, lxf, from 'Y:\Rajiv\CTS-2219 (Frame Rate generated by CTSD7.1 for
LXF
formats is wrong)\Video_LXF.lxf':
  Duration: 00:00:16.88, start: 0.000000, bitrate: 12229 kb/s
    Stream #0:0: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p,
720x
480 [SAR 8:9 DAR 4:3], 6000 kb/s, *14.99 fps*, 29.97 tbr, 29.97 tbn, 59.94
tbc
    Stream #0:1: Audio: pcm_s16le_planar, 48000 Hz, 8 channels, s16p, 6144
kb/s
----------------------------------------------------------------------------

(2.0)
----
C:\Users\Administrator\Desktop\FFMpeg>2.0\ffprobe.exe "Y:\Video_LXF.lxf"
ffprobe version N-54835-g835eee8 Copyright (c) 2007-2013 the FFmpeg
developers
  built on Jul 20 2013 18:05:35 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
--enab
le-iconv --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
--ena
ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-l
ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libsp
eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
--enable-libvo-
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libxavs --
enable-libxvid --enable-zlib
  libavutil      52. 40.100 / 52. 40.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.102 / 55. 12.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 81.102 /  3. 81.102
  libswscale      2.  4.100 /  2.  4.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
Input #0, lxf, from 'Y:\Rajiv\CTS-2219 (Frame Rate generated by CTSD7.1 for
LXF
formats is wrong)\Video_LXF.lxf':
  Duration: 00:00:16.88, start: 0.000000, bitrate: 12229 kb/s
    Stream #0:0: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p,
720x
480 [SAR 8:9 DAR 4:3], 6000 kb/s, *14.99 fps*, 29.97 tbr, 29.97 tbn, 59.94
tbc
    Stream #0:1: Audio: pcm_s16le_planar, 48000 Hz, 8 channels, s16p, 6144
kb/s
----------------------------------------------------------------------------

Any idea ?

Thanks.

From jbennett at incomsystems.biz  Tue Aug 27 20:23:31 2013
From: jbennett at incomsystems.biz (jbennett at incomsystems.biz)
Date: Tue, 27 Aug 2013 13:23:31 -0500
Subject: [FFmpeg-user] webm video encoded with ffmpeg and streamed with
 ffserver doesn't play correctly in Firefox
Message-ID: <67f8a7989ace46d6f93629e328a4ac57@incomsystems.biz>

 

Hello all, I'm attempting to use ffmpeg to transcode live video to
webm and stream it to ffserver, and then embed the stream as html5 video
on a web site. I'm using ffmpeg 1.2.2 on my local desktop to encode, and
ffserver 1.2.1 on the server machine to stream to clients. 

I'm 99% of
the way there. Audio and video encode properly, and the video plays
correctly in Chrome and Konqueror. The strange bit is that Firefox
refuses to play my stream. It displays a single frame, but when the play
button is pressed, it seems to buffer forever, and never start playing.
There is a known firefox problem with ffmpeg encoded webm that I am
probably hitting. A negative timestamp at the beginning of a file will
cause firefox to download the entire file before playing
(https://bugzilla.mozilla.org/show_bug.cgi?id=873775 [1]). As this is a
live stream, there is no end of file to find. 

I have tried
-avoid_negative_ts 1 in the ffmpeg command, but it doesn't seem to make
a difference. 

When I use wget on the raw live stream, then use mkvinfo
-s on the captured file, I get this output: 

Track 1: audio, codec ID:
A_VORBIS, mkvmerge/mkvextract track ID: 0, language: und, channels: 1,
sampling freq: 22050
Track 2: video, codec ID: V_VP8,
mkvmerge/mkvextract track ID: 1, language: und, default duration:
33.367ms (29.970 frames/fields per second for a video track), pixel
width: 720, pixel height: 480
I frame, track 2, timecode 1377625424004
(382673:43:44.004), size 625, adler 0x92f0bc34
I frame, track 1,
timecode 1377625424011 (382673:43:44.011), size 83, adler 0xc43223a0
I
frame, track 1, timecode 1377625424034 (382673:43:44.034), size 89,
adler 0xf79f2981
P frame, track 2, timecode 1377625424038
(382673:43:44.038), size 51, adler 0x69cd1076
I frame, track 1, timecode
1377625424057 (382673:43:44.057), size 83, adler 0xb4742955
P frame,
track 2, timecode 1377625424071 (382673:43:44.071), size 51, adler
0x69cd1076
I frame, track 1, timecode 1377625424080 (382673:43:44.080),
size 85, adler 0xfda824be
I frame, track 1, timecode 1377625424103
(382673:43:44.103), size 78, adler 0xaaa52481
P frame, track 2, timecode
1377625424104 (382673:43:44.104), size 56, adler 0x8bd710d7
I frame,
track 1, timecode 1377625424127 (382673:43:44.127), size 97, adler
0x70f92be1
P frame, track 2, timecode 1377625424138 (382673:43:44.138),
size 54, adler 0x6a881093
I frame, track 1, timecode 1377625424150
(382673:43:44.150), size 88, adler 0x79042ccc
P frame, track 2, timecode
1377625424171 (382673:43:44.171), size 54, adler 0x6b84106d
I frame,
track 1, timecode 1377625424173 (382673:43:44.173), size 85, adler
0x3a4a26e7
I frame, track 1, timecode 1377625424196 (382673:43:44.196),
size 93, adler 0x4be32b07
P frame, track 2, timecode 1377625424204
(382673:43:44.204), size 55, adler 0x7c11102b
I frame, track 1, timecode
1377625424220 (382673:43:44.220), size 80, adler 0xc9b6224a
P frame,
track 2, timecode 1377625424238 (382673:43:44.238), size 51, adler
0x97d71273
I frame, track 1, timecode 1377625424243 (382673:43:44.243),
size 94, adler 0x34a22ce5
I frame, track 1, timecode 1377625424266
(382673:43:44.266), size 53, adler 0xf4dd1d7c
P frame, track 2, timecode
1377625424271 (382673:43:44.271), size 55, adler 0x7d9d10c8
I frame,
track 1, timecode 1377625424283 (382673:43:44.283), size 71, adler
0x4ec422d1
I frame, track 1, timecode 1377625424295 (382673:43:44.295),
size 79, adler 0xa6dc243f
P frame, track 2, timecode 1377625424305
(382673:43:44.305), size 51, adler 0x6ccd1086
I frame, track 1, timecode
1377625424307 (382673:43:44.307), size 80, adler 0x4d6426d8
I frame,
track 1, timecode 1377625424318 (382673:43:44.318), size 96, adler
0xaa062ec6
I frame, track 1, timecode 1377625424336 (382673:43:44.336),
size 92, adler 0xfc632ecc
P frame, track 2, timecode 1377625424338
(382673:43:44.338), size 56, adler 0x90b51136
I frame, track 1, timecode
1377625424359 (382673:43:44.359), size 90, adler 0xd1d52a36
P frame,
track 2, timecode 1377625424371 (382673:43:44.371), size 51, adler
0x6dbd108b
I frame, track 1, timecode 1377625424382 (382673:43:44.382),
size 89, adler 0xf4362e1a
I frame, track 2, timecode 1377625424405
(382673:43:44.405), size 624, adler 0xddb4e297
I frame, track 1,
timecode 1377625424405 (382673:43:44.405), size 85, adler 0x20681d9a
I
frame, track 1, timecode 1377625424428 (382673:43:44.428), size 78,
adler 0xcae9267c
P frame, track 2, timecode 1377625424438
(382673:43:44.438), size 51, adler 0x69cd1076
I frame, track 1, timecode
1377625424452 (382673:43:44.452), size 93, adler 0xc6362f19
P frame,
track 2, timecode 1377625424471 (382673:43:44.471), size 51, adler
0x69cd1076
I frame, track 1, timecode 1377625424475 (382673:43:44.475),
size 83, adler 0xd80025d4 

Obviously not ideal. Main question is what
needs to change so that Firefox will play the live stream? I'll include
my conf files and outputs. 

My html code is simply: 

<video
src="http://myServerIp:8090/test.webm" type="video/webm" controls>
 Your
browser does not support the <code>video</code> element.
</video> 

Port
8090
BindAddress 0.0.0.0
MaxHTTPConnections 30
MaxClients
20
#MaxBandwidth 2000
#CustomLog - 

<Feed feed1.ffm>
File
/tmp/feed1.ffm
FileMaxSize 1000K
ACL allow myDesktopIp
</Feed> 

<Stream
test.webm> # Output stream URL definition
 Feed feed1.ffm # Feed from
which to receive video
 Format webm 

# Audio settings
 AudioCodec
vorbis
 AudioBitRate 32 # Audio bitrate# 

# Video settings
 VideoCodec
libvpx
 VideoSize 720x480 # Video resolution 

AVOptionVideo flags
+global_header # Parameters passed to encoder
 # (same as ffmpeg
command-line parameters)
 AVOptionVideo cpu-used 0 
 AVOptionVideo qmin
10
 AVOptionVideo qmax 42
 AVOptionVideo quality good
 AVOptionAudio
flags +global_header
PreRoll 15
 StartSendOnKey
 VideoBitRate 250 #
Video bitrate
</Stream> 

ffmpeg -v debug -f v4l2 -tvstd ntsc -i
/dev/video0 -vsync 2 -f pulse -i default -avoid_negative_ts 1
http://serverip:8090/feed1.ffm
ffmpeg version 1.2.2 Copyright (c)
2000-2013 the FFmpeg developers
 built on Aug 4 2013 08:24:19 with gcc
4.8.1 (GCC) 20130603 (Red Hat 4.8.1-1)
 configuration: --prefix=/usr
--bindir=/usr/bin --datadir=/usr/share/ffmpeg
--incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man
--arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
-fexceptions -fstack-protector --param=ssp-buffer-size=4
-grecord-gcc-switches -m64 -mtune=generic' --enable-bzlib
--disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass
--enable-libcelt --enable-libdc1394 --disable-indev=jack
--enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal
--enable-libopencv --enable-libopenjpeg --enable-libopus
--enable-libpulse --enable-libschroedinger --enable-libspeex
--enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx
--enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter
--enable-postproc --enable-pthreads --disable-static --enable-shared
--enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64
--enable-runtime-cpudetect
 libavutil 52. 18.100 / 52. 18.100

libavcodec 54. 92.100 / 54. 92.100
 libavformat 54. 63.104 / 54. 63.104

libavdevice 54. 3.103 / 54. 3.103
 libavfilter 3. 42.103 / 3. 42.103

libswscale 2. 2.100 / 2. 2.100
 libswresample 0. 17.102 / 0. 17.102

libpostproc 52. 2.100 / 52. 2.100
Splitting the commandline.
Reading
option '-v' ... matched as option 'v' (set libav* logging level) with
argument 'debug'. 
Reading option '-f' ... matched as option 'f' (force
format) with argument 'v4l2'. 
Reading option '-tvstd' ... matched as
option 'tvstd' (deprecated, use -standard) with argument 'ntsc'.

Reading option '-i' ... matched as input file with argument
'/dev/video0'. 
Reading option '-vsync' ... matched as option 'vsync'
(video sync method) with argument '2'. 
Reading option '-f' ... matched
as option 'f' (force format) with argument 'pulse'. 
Reading option '-i'
... matched as input file with argument 'default'. 
Reading option
'-avoid_negative_ts' ... matched as AVOption 'avoid_negative_ts' with
argument '1'. 
Reading option 'http://serverip:8090/feed1.ffm' ...
matched as output file. 
Finished splitting the commandline. 
Parsing a
group of options: global . 
Applying option v (set libav* logging level)
with argument debug. 
Applying option tvstd (deprecated, use -standard)
with argument ntsc. 
This option is deprecated, use -standard.
Applying
option vsync (video sync method) with argument 2.
Successfully parsed a
group of options. 
Parsing a group of options: input file /dev/video0.

Applying option f (force format) with argument v4l2. 
Successfully
parsed a group of options. 
Opening an input file: /dev/video0.

[video4linux2,v4l2 @ 0x25abba0] fd:3 capabilities:85020011

[video4linux2,v4l2 @ 0x25abba0] Selecting input_channel: 0

[video4linux2,v4l2 @ 0x25abba0] input_channel: 0, input_name: S-Video

[video4linux2,v4l2 @ 0x25abba0] Querying the device for the current
frame size 
[video4linux2,v4l2 @ 0x25abba0] Setting frame size to
720x480 
[video4linux2,v4l2 @ 0x25abba0] The V4L2 driver is using the
interlaced mode 
[video4linux2,v4l2 @ 0x25abba0] Current standard: NTSC,
id: 45056, frameperiod: 1001/30000 
[video4linux2,v4l2 @ 0x25abba0] All
info found 
[video4linux2,v4l2 @ 0x25abba0] Estimating duration from
bitrate, this may be inaccurate
Input #0, video4linux2,v4l2, from
'/dev/video0':
 Duration: N/A, start: 54423.623015, bitrate: 124291
kb/s
 Stream #0:0, 1, 1/1000000: Video: rawvideo (I420 / 0x30323449),
yuv420p, 720x480, 1/1000000, 124291 kb/s, 29.97 fps, 29.97 tbr, 1000k
tbn, 1000k tbc
Successfully opened the file.
Parsing a group of options:
input file default. 
Applying option f (force format) with argument
pulse. 
Successfully parsed a group of options. 
Opening an input file:
default. 
[pulse @ 0x25ad560] All info found 
[pulse @ 0x25ad560]
Estimating duration from bitrate, this may be inaccurate
Guessed Channel
Layout for Input Stream #1.0 : stereo 
Input #1, pulse, from 'default':

Duration: N/A, start: 0.633504, bitrate: 1536 kb/s
 Stream #1:0, 1,
1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536
kb/s
Successfully opened the file.
Parsing a group of options: output
file http:/serverip:8090/feed1.ffm. 
Successfully parsed a group of
options. 
Opening an output file: http://serverip:8090/feed1.ffm. 
[ffm
@ 0x25c1ea0] Format ffm probed with size=2048 and score=101

[AVIOContext @ 0x25c00a0] Statistics: 4096 bytes read, 0 seeks

Successfully opened the file. 
[abuffer @ 0x25c75e0] Setting entry with
key 'time_base' to value '1/48000' 
[abuffer @ 0x25c75e0] Setting entry
with key 'sample_rate' to value '48000' 
[abuffer @ 0x25c75e0] Setting
entry with key 'sample_fmt' to value 's16' 
[abuffer @ 0x25c75e0]
Setting entry with key 'channel_layout' to value '0x3' 
[graph 0 input
from stream 1:0 @ 0x25ab740] tb:1/48000 samplefmt:s16 samplerate:48000
chlayout:0x3 
[aformat @ 0x25c6020] Setting entry with key 'sample_fmts'
to value 'fltp' 
[aformat @ 0x25c6020] Setting entry with key
'sample_rates' to value '22050' 
[aformat @ 0x25c6020] Setting entry
with key 'channel_layouts' to value '0x4' 
[audio format for output
stream 0:0 @ 0x25c5f20] auto-inserting filter 'auto-inserted resampler
0' between the filter 'Parsed_anull_0' and the filter 'audio format for
output stream 0:0' 
0.500000 0.500000 
[auto-inserted resampler 0 @
0x25c5560] ch:2 chl:stereo fmt:s16 r:48000Hz -> ch:1 chl:mono fmt:fltp
r:22050Hz 
[buffer @ 0x25c5ae0] Setting entry with key 'video_size' to
value '720x480' 
[buffer @ 0x25c5ae0] Setting entry with key 'pix_fmt'
to value '0' 
[buffer @ 0x25c5ae0] Setting entry with key 'time_base' to
value '1/1000000' 
[buffer @ 0x25c5ae0] Setting entry with key
'pixel_aspect' to value '0/1' 
[buffer @ 0x25c5ae0] Setting entry with
key 'sws_param' to value 'flags=2' 
[buffer @ 0x25c5ae0] Setting entry
with key 'frame_rate' to value '30000/1001' 
[graph 1 input from stream
0:0 @ 0x25c51e0] w:720 h:480 pixfmt:yuv420p tb:1/1000000 fr:30000/1001
sar:0/1 sws_param:flags=2 
[scale @ 0x25c5b80] Setting 'w' to value
'720' 
[scale @ 0x25c5b80] Setting 'h' to value '480' 
[scale @
0x25c5b80] Setting 'flags' to value '0x4' 
[scaler for output stream 0:1
@ 0x25c6cc0] w:720 h:480 flags:'0x4' interl:0 
[scaler for output stream
0:1 @ 0x25c6cc0] w:720 h:480 fmt:yuv420p sar:0/1 -> w:720 h:480
fmt:yuv420p sar:0/1 flags:0x4 
[libvpx @ 0x25c93a0] v1.2.0
[libvpx @
0x25c93a0] --target=x86_64-linux-gcc --enable-pic --disable-install-srcs
--enable-shared --prefix=/usr --libdir=/usr/lib64
[libvpx @ 0x25c93a0]
vpx_codec_enc_cfg 
[libvpx @ 0x25c93a0] generic settings 
 g_usage: 0 

g_threads: 0 
 g_profile: 0 
 g_w: 320 
 g_h: 240 
 g_timebase: {1/30} 

g_error_resilient: 0 
 g_pass: 0 
 g_lag_in_frames: 0 
[libvpx @
0x25c93a0] rate control settings 
 rc_dropframe_thresh: 0 

rc_resize_allowed: 0 
 rc_resize_up_thresh: 60 
 rc_resize_down_thresh:
30 
 rc_end_usage: 0 
 rc_twopass_stats_in: (nil)(0) 

rc_target_bitrate: 256 
[libvpx @ 0x25c93a0] quantizer settings 

rc_min_quantizer: 4 
 rc_max_quantizer: 63 
[libvpx @ 0x25c93a0] bitrate
tolerance 
 rc_undershoot_pct: 100 
 rc_overshoot_pct: 100 
[libvpx @
0x25c93a0] decoder buffer model 
 rc_buf_sz: 6000 
 rc_buf_initial_sz:
4000 
 rc_buf_optimal_sz: 5000 
[libvpx @ 0x25c93a0] 2 pass rate control
settings 
 rc_2pass_vbr_bias_pct: 50 
 rc_2pass_vbr_minsection_pct: 0 

rc_2pass_vbr_maxsection_pct: 400 
[libvpx @ 0x25c93a0] keyframing
settings 
 kf_mode: 1 
 kf_min_dist: 0 
 kf_max_dist: 128 
[libvpx @
0x25c93a0] 
[libvpx @ 0x25c93a0] vpx_codec_enc_cfg 
[libvpx @ 0x25c93a0]
generic settings 
 g_usage: 0 
 g_threads: 0 
 g_profile: 0 
 g_w: 720 

g_h: 480 
 g_timebase: {1001/30000} 
 g_error_resilient: 0 
 g_pass: 0 

g_lag_in_frames: 25 
[libvpx @ 0x25c93a0] rate control settings 

rc_dropframe_thresh: 0 
 rc_resize_allowed: 0 
 rc_resize_up_thresh: 60

 rc_resize_down_thresh: 30 
 rc_end_usage: 0 
 rc_twopass_stats_in:
(nil)(0) 
 rc_target_bitrate: 250 
[libvpx @ 0x25c93a0] quantizer
settings 
 rc_min_quantizer: 10 
 rc_max_quantizer: 42 
[libvpx @
0x25c93a0] bitrate tolerance 
 rc_undershoot_pct: 100 

rc_overshoot_pct: 100 
[libvpx @ 0x25c93a0] decoder buffer model 

rc_buf_sz: 2000 
 rc_buf_initial_sz: 1500 
 rc_buf_optimal_sz: 1666

[libvpx @ 0x25c93a0] 2 pass rate control settings 

rc_2pass_vbr_bias_pct: 50 
 rc_2pass_vbr_minsection_pct: 0

rc_2pass_vbr_maxsection_pct: 200
[libvpx @ 0x25c93a0] keyframing
settings
 kf_mode: 1
 kf_min_dist: 0
 kf_max_dist: 12
[libvpx @
0x25c93a0] 
[libvpx @ 0x25c93a0] vpx_codec_control
[libvpx @ 0x25c93a0]
VP8E_SET_CPUUSED: 3
[libvpx @ 0x25c93a0] VP8E_SET_ARNR_MAXFRAMES:
0
[libvpx @ 0x25c93a0] VP8E_SET_ARNR_STRENGTH: 3
[libvpx @ 0x25c93a0]
VP8E_SET_ARNR_TYPE: 3
[libvpx @ 0x25c93a0] VP8E_SET_NOISE_SENSITIVITY:
0
[libvpx @ 0x25c93a0] VP8E_SET_TOKEN_PARTITIONS: 0
[libvpx @ 0x25c93a0]
VP8E_SET_STATIC_THRESHOLD: 0
[libvpx @ 0x25c93a0] VP8E_SET_CQ_LEVEL:
0
[libvpx @ 0x25c93a0] Using deadline: 1000000
Output #0, ffm, to
'http://serverip:8090/feed1.ffm':
 Metadata:
 creation_time : now

encoder : Lavf54.63.104
 Stream #0:0, 0, 1/1000000: Audio: vorbis, 22050
Hz, mono, fltp, 32 kb/s
 Stream #0:1, 0, 1/1000000: Video: vp8, yuv420p,
720x480, 1001/30000, q=10-42, 250 kb/s, 1000k tbn, 29.97 tbc
Stream
mapping:
 Stream #1:0 -> #0:0 (pcm_s16le -> libvorbis)
 Stream #0:0 ->
#0:1 (rawvideo -> libvpx)
Press [q] to stop, [?] for help
Trying to
remove 320 more samples than there are in the queuetrate= 66.4kbits/s

frame= 82 fps= 36 q=0.0 Lsize= 24kB time=00:00:02.76 bitrate=
71.0kbits/s 
video:8kB audio:9kB subtitle:0 global headers:3kB muxing
overhead 18.512803%
Received signal 2: terminating. 




Links:
------
[1] https://bugzilla.mozilla.org/show_bug.cgi?id=873775

From vvubin at gmail.com  Tue Aug 27 20:19:21 2013
From: vvubin at gmail.com (Wu-bin Zhen)
Date: Tue, 27 Aug 2013 14:19:21 -0400
Subject: [FFmpeg-user] Buffer Size for Stream Copy
In-Reply-To: <CAPYw7P6Ns4XqagdVnbn3ezX-YjiL8t_hCDm=90ybfW+B1bxQsw@mail.gmail.com>
References: <CAGK10ZNKQPOjRjC=RP2h0isuudyUVQe1w2Jb50mSoy1dU2yCFg@mail.gmail.com>
	<CAGK10ZOMJ7KuvAx1A+p99VaJ=aqqzw982BuEJ-EdSPPXusLS3g@mail.gmail.com>
	<CAPYw7P6Ns4XqagdVnbn3ezX-YjiL8t_hCDm=90ybfW+B1bxQsw@mail.gmail.com>
Message-ID: <CAGK10ZPQpW=SC7uLVoV27Bj1VSkzV3Qdh=RzSGKMv0=Kys9u3w@mail.gmail.com>

On Tue, Aug 20, 2013 at 7:31 PM, Paul B Mahol <onemda at gmail.com> wrote:

> On 8/20/13, Wu-bin Zhen <vvubin at gmail.com> wrote:
> > On Fri, Aug 16, 2013 at 3:31 PM, Wu-bin Zhen <vvubin at gmail.com> wrote:
> >
> >> Hello ffmpeg,
> >>
> >> I am using ffmpeg to grab h264 video stream and aac audio stream from an
> >> internet based hardware encoder with the codec copy option, and it works
> >> pretty well. Now, I start to wonder if I can use the '-bufsize' option
> >> for
> >> the stream copy, just in case when the network is temporally slow down,
> >> or
> >> the encoder is suddenly chocking for some reason. I googled a little bit
> >> about the 'bufsize' option, and found that most people use the option
> for
> >> encoding.
> >>
> >> So my question is: Do you think using 'bufsize' option will be helpful
> >> for
> >> the stream copy? If so, what would be the recommended buffer size? My
> >> internet encoder produces video stream with 4000kbits/sec, and audio
> >> stream
> >> with 128kbits/sec.
> >>
> >> Thanks, and have a great weekend.
> >> -w
> >>
> >
> > Sorry if I didn't explain clearly. So ffmpeg works fine when I specify
> the
> > option '-bufsize', but later I realized that it actually doesn't care the
> > value of the bufsize option. The ffmpeg command below works fine without
> > complaining anything about the bufsize value:
> >
> >> ffmpeg -i http://192.168.1.224:1881/stream.ts -bufsize helloworld
> -vcodec
> > copy -acodec copy buffer_test.ts
> >
> > So I start to wonder if ffmpeg doesn't do anything about '-bufsize' for
> the
> > stream copy.
> > Please advise. Thanks
>
> As explainded in documentation and by ffmpeg itself, 'bufsize' is used
> for audio/video encoding only.
>
> -bufsize           <int>        E..VA. set ratecontrol buffer size (in
> bits) (from INT_MIN to INT_MAX) (default 0
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

Thanks for your clear explanation, so I can forget about this option for
copy streams. By the way, do you think is there a way to set buffer for the
stream copy? For my case the copy is based on the network, not from the
local hardware, so I am worry when the network or the encoder is temporally
inaccessible.

I will appreciate if you would advise any method to make the stream copy
more stable.
-w

From mediastream at gmail.com  Tue Aug 27 21:42:28 2013
From: mediastream at gmail.com (Dennis)
Date: Tue, 27 Aug 2013 15:42:28 -0400
Subject: [FFmpeg-user] Watermark image, fade in and out.
In-Reply-To: <CAPCJWRo1y_PNJutjkLnRQfhgVTscYnu+z0HD+c1RrRRZyBDiMg@mail.gmail.com>
References: <CAPCJWRo1y_PNJutjkLnRQfhgVTscYnu+z0HD+c1RrRRZyBDiMg@mail.gmail.com>
Message-ID: <CAPCJWRpyms2UViQP5zoF95kBSJQB-MOEd6sXt7fiPJVnWwcMOg@mail.gmail.com>

On Mon, Aug 26, 2013 at 6:46 PM, Dennis <mediastream at gmail.com> wrote:

>
> Is there a way to display watermark for ten seconds then fade it out?
>
> So far this works for me, but if I add fade out it fades out the whole
> image not just the watermark  -- but if I add duration it removes the
> watermark after 10 sec. Not sure how command is defined as global versus
> localized to watermark only.
>
> Shows logo for 10 sec.
> ffmpeg2 -i big_buck_bunny_1080p_h264.mov -i logo1.png -filter_complex
> 'overlay=100:main_h-overlay_h-200:enable=lte(t\,10)'  -t 20 1.mp4
>
> Shows logo for 10 sec then fades out the whole screen with logo, so its
> 1/2 working (:
> ffmpeg2 -i big_buck_bunny_1080p_h264.mov -i logo1.png -filter_complex
> 'overlay=100:main_h-overlay_h-200:enable=lte(t\,10),fade=out:0:300:alpha=1'
> -t 20 1.mp4
>

This works with 1 sec fade in, 10 sec solid overlay, 1 sec fade out:

ffmpeg -loop 1 -i logo1.png -frames:v 300 -vf
'fade=in:0:25,fade=out:275:25' -pix_fmt rgba -vcodec png LOGO.MOV
ffmpeg -i bbb.ts -vf "movie=LOGO.MOV [watermark]; [in] [watermark]
overlay=100:main_h-overlay_h-200 [out]" OUT.MP4

From onemda at gmail.com  Tue Aug 27 23:39:56 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 27 Aug 2013 21:39:56 +0000
Subject: [FFmpeg-user] Buffer Size for Stream Copy
In-Reply-To: <CAGK10ZPQpW=SC7uLVoV27Bj1VSkzV3Qdh=RzSGKMv0=Kys9u3w@mail.gmail.com>
References: <CAGK10ZNKQPOjRjC=RP2h0isuudyUVQe1w2Jb50mSoy1dU2yCFg@mail.gmail.com>
	<CAGK10ZOMJ7KuvAx1A+p99VaJ=aqqzw982BuEJ-EdSPPXusLS3g@mail.gmail.com>
	<CAPYw7P6Ns4XqagdVnbn3ezX-YjiL8t_hCDm=90ybfW+B1bxQsw@mail.gmail.com>
	<CAGK10ZPQpW=SC7uLVoV27Bj1VSkzV3Qdh=RzSGKMv0=Kys9u3w@mail.gmail.com>
Message-ID: <CAPYw7P6T-JhFoYoz09k1BnwKxHkAiNbDpZvMVjxp242PwDVySA@mail.gmail.com>

On 8/27/13, Wu-bin Zhen <vvubin at gmail.com> wrote:
> On Tue, Aug 20, 2013 at 7:31 PM, Paul B Mahol <onemda at gmail.com> wrote:
>
>> On 8/20/13, Wu-bin Zhen <vvubin at gmail.com> wrote:
>> > On Fri, Aug 16, 2013 at 3:31 PM, Wu-bin Zhen <vvubin at gmail.com> wrote:
>> >
>> >> Hello ffmpeg,
>> >>
>> >> I am using ffmpeg to grab h264 video stream and aac audio stream from
>> >> an
>> >> internet based hardware encoder with the codec copy option, and it
>> >> works
>> >> pretty well. Now, I start to wonder if I can use the '-bufsize' option
>> >> for
>> >> the stream copy, just in case when the network is temporally slow
>> >> down,
>> >> or
>> >> the encoder is suddenly chocking for some reason. I googled a little
>> >> bit
>> >> about the 'bufsize' option, and found that most people use the option
>> for
>> >> encoding.
>> >>
>> >> So my question is: Do you think using 'bufsize' option will be helpful
>> >> for
>> >> the stream copy? If so, what would be the recommended buffer size? My
>> >> internet encoder produces video stream with 4000kbits/sec, and audio
>> >> stream
>> >> with 128kbits/sec.
>> >>
>> >> Thanks, and have a great weekend.
>> >> -w
>> >>
>> >
>> > Sorry if I didn't explain clearly. So ffmpeg works fine when I specify
>> the
>> > option '-bufsize', but later I realized that it actually doesn't care
>> > the
>> > value of the bufsize option. The ffmpeg command below works fine
>> > without
>> > complaining anything about the bufsize value:
>> >
>> >> ffmpeg -i http://192.168.1.224:1881/stream.ts -bufsize helloworld
>> -vcodec
>> > copy -acodec copy buffer_test.ts
>> >
>> > So I start to wonder if ffmpeg doesn't do anything about '-bufsize' for
>> the
>> > stream copy.
>> > Please advise. Thanks
>>
>> As explainded in documentation and by ffmpeg itself, 'bufsize' is used
>> for audio/video encoding only.
>>
>> -bufsize           <int>        E..VA. set ratecontrol buffer size (in
>> bits) (from INT_MIN to INT_MAX) (default 0
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
> Thanks for your clear explanation, so I can forget about this option for
> copy streams. By the way, do you think is there a way to set buffer for the
> stream copy? For my case the copy is based on the network, not from the
> local hardware, so I am worry when the network or the encoder is temporally
> inaccessible.

I do not completly understand problem you want to solve.

>
> I will appreciate if you would advise any method to make the stream copy
> more stable.

If something is not stable then perhaps you should report bug.

From onemda at gmail.com  Tue Aug 27 23:53:05 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 27 Aug 2013 21:53:05 +0000
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
Message-ID: <CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>

On 8/27/13, Shyam Sundar <sw.craftsman at gmail.com> wrote:
> Hi,
>
> Just realized I am seeing incorrect frame-rates for number of videos after
> upgrading from 0.6.1 to 1.2 sometime back.
>
> E.g. for below video (http://files.videohelp.com/u/211860/Video_LXF.lxf) it
> used to be 29.97 fps earlier, it is 14.99 now. I have verified in other
> tools like MediaInfo that the correct frame-rate is 29.97 only.

How you know that MediaInfo is correct?

>
> *(Output)*
> ----------------------------------------------------------------------------
> (0.6.1)
> ------
> C:\Users\Administrator\Desktop\FFMpeg>0.6.1\ffprobe.exe "Y:\Video_LXF.lxf"
> ffprobe version N-35462-g61b1d85, Copyright (c) 2007-2011 the FFmpeg
> developers
>   built on Dec  5 2011 14:22:27 with gcc 4.6.2
>   configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-ru
> ntime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r
> --enable-libope
> ncore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm
> --en
> able-libmp3lame --enable-libopenjpeg --enable-librtmp
> --enable-libschroedinger -
> -enable-libspeex --enable-libtheora --enable-libvo-aacenc
> --enable-libvo-amrwben
> c --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs
> --enable-
> libxvid --enable-zlib
>   libavutil    51. 30. 0 / 51. 30. 0
>   libavcodec   53. 40. 0 / 53. 40. 0
>   libavformat  53. 24. 0 / 53. 24. 0
>   libavdevice  53.  4. 0 / 53.  4. 0
>   libavfilter   2. 51. 0 /  2. 51. 0
>   libswscale    2.  1. 0 /  2.  1. 0
>   libpostproc  51.  2. 0 / 51.  2. 0
> Input #0, lxf, from 'Y:\Rajiv\CTS-2219 (Frame Rate generated by CTSD7.1 for
> LXF
> formats is wrong)\Video_LXF.lxf':
>   Duration: 00:00:16.88, start: 0.000000, bitrate: 12229 kb/s
>     Stream #0:0: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p,
> 720x
> 480 [SAR 8:9 DAR 4:3], 6000 kb/s, *29.97 fps*, 29.97 tbr, 29.97 tbn, 59.94
> tbc
>     Stream #0:1: Audio: pcm_s16le, 48000 Hz, 8 channels, s16, 6144 kb/s
> ----------------------------------------------------------------------------
>
> C:\Users\Administrator\Desktop\FFMpeg>1.2\ffprobe.exe "Y:\Video_LXF.lxf"
> ffprobe version 1.2 Copyright (c) 2007-2013 the FFmpeg developers
>   built on Mar 28 2013 00:34:08 with gcc 4.8.0 (GCC)
>   configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-av
> isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
> --enab
> le-libass --enable-libbluray --enable-libcaca --enable-libfreetype
> --enable-libg
> sm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb
> --enable-libo
> pencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp
> --enable-li
> bschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-lib
> twolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
> --enabl
> e-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
>   libavutil      52. 18.100 / 52. 18.100
>   libavcodec     54. 92.100 / 54. 92.100
>   libavformat    54. 63.104 / 54. 63.104
>   libavdevice    54.  3.103 / 54.  3.103
>   libavfilter     3. 42.103 /  3. 42.103
>   libswscale      2.  2.100 /  2.  2.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  2.100 / 52.  2.100
> Input #0, lxf, from 'Y:\Rajiv\CTS-2219 (Frame Rate generated by CTSD7.1 for
> LXF
> formats is wrong)\Video_LXF.lxf':
>   Duration: 00:00:16.88, start: 0.000000, bitrate: 12229 kb/s
>     Stream #0:0: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p,
> 720x
> 480 [SAR 8:9 DAR 4:3], 6000 kb/s, *14.99 fps*, 29.97 tbr, 29.97 tbn, 59.94
> tbc
>     Stream #0:1: Audio: pcm_s16le_planar, 48000 Hz, 8 channels, s16p, 6144
> kb/s
> ----------------------------------------------------------------------------
>
> (2.0)
> ----
> C:\Users\Administrator\Desktop\FFMpeg>2.0\ffprobe.exe "Y:\Video_LXF.lxf"
> ffprobe version N-54835-g835eee8 Copyright (c) 2007-2013 the FFmpeg
> developers
>   built on Jul 20 2013 18:05:35 with gcc 4.7.3 (GCC)
>   configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-av
> isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
> --enab
> le-iconv --enable-libass --enable-libbluray --enable-libcaca
> --enable-libfreetyp
> e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
> --ena
> ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
> --enable-l
> ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
> --enable-libsp
> eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc
> --enable-libvo-
> amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264
> --enable-libxavs --
> enable-libxvid --enable-zlib
>   libavutil      52. 40.100 / 52. 40.100
>   libavcodec     55. 18.102 / 55. 18.102
>   libavformat    55. 12.102 / 55. 12.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 81.102 /  3. 81.102
>   libswscale      2.  4.100 /  2.  4.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  3.100 / 52.  3.100
> Input #0, lxf, from 'Y:\Rajiv\CTS-2219 (Frame Rate generated by CTSD7.1 for
> LXF
> formats is wrong)\Video_LXF.lxf':
>   Duration: 00:00:16.88, start: 0.000000, bitrate: 12229 kb/s
>     Stream #0:0: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p,
> 720x
> 480 [SAR 8:9 DAR 4:3], 6000 kb/s, *14.99 fps*, 29.97 tbr, 29.97 tbn, 59.94
> tbc
>     Stream #0:1: Audio: pcm_s16le_planar, 48000 Hz, 8 channels, s16p, 6144
> kb/s
> ----------------------------------------------------------------------------
>
> Any idea ?
>
> Thanks.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From sw.craftsman at gmail.com  Wed Aug 28 12:09:51 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Wed, 28 Aug 2013 15:39:51 +0530
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
Message-ID: <CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>

Hi Paul,

Not just Media Info, other tool like GSpot v2.70a too shows frame-rate of
this video as 29.97. I thought they were reliable.

But even if we forget other tools for the time being ... the older version
of FFProbe itself (0.6.1) shows the frame-rate as 29.97 .... contrary to
latest version which shows 14.99.

So that's what I don't understand ... why this change. Is it a regression ?

Thanks !

From onemda at gmail.com  Wed Aug 28 13:23:16 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 28 Aug 2013 11:23:16 +0000
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
Message-ID: <CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>

On 8/28/13, Shyam Sundar <sw.craftsman at gmail.com> wrote:
> Hi Paul,
>
> Not just Media Info, other tool like GSpot v2.70a too shows frame-rate of
> this video as 29.97. I thought they were reliable.
>
> But even if we forget other tools for the time being ... the older version
> of FFProbe itself (0.6.1) shows the frame-rate as 29.97 .... contrary to
> latest version which shows 14.99.
>
> So that's what I don't understand ... why this change. Is it a regression ?

Did you actually watched video?

>
> Thanks !
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From j.bakshi at unlimitedmail.org  Wed Aug 28 14:19:08 2013
From: j.bakshi at unlimitedmail.org (J. Bakshi (=?UTF-8?B?4KacLiDgpqzgppXgprjgp4A=?= ))
Date: Wed, 28 Aug 2013 17:49:08 +0530
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
Message-ID: <20130828174908.6fed1cf5@shiva.selfip.org>

Dear list,

I like to upload videos to vimeo and youtube.

Hence I need to convert .dat .vom & .avi to mp4

I have already converted .dat -> .mp4 with the following

 ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 out.mp4

and when try to play the .mp4 with ffplay I get the following error

ffplay out.mp4
 
ffplay version 0.11.1 Copyright (c) 2003-2012 the FFmpeg developers
  built on Aug 11 2012 08:15:54 with gcc 4.7.1
  configuration: --prefix=/usr --extra-cflags='-g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security ' --extra-ldflags='-Wl,-z,relro' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-nonfree --disable-decoder=libdirac --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-libvpx --enable-librtmp --enable-avfilter --enable-libfreetype --enable-libvo-aacenc --disable-decoder=amrnb --enable-libvo-amrwbenc --enable-libaacplus --libdir=/usr/lib/x86_64-linux-gnu --disable-vda --enable-libbluray --enable-libcdio --enable-gnutls --enable-frei0r --enable-openssl --enable-libass --enable-libdc1394 --disable-altivec -!
 -disable-armv5te --disable-armv6 -  libavutil      51. 54.100 / 51. 54.100
  libavcodec     54. 23.100 / 54. 23.100
  libavformat    54.  6.100 / 54.  6.100
  libavdevice    54.  0.100 / 54.  0.100
  libavfilter     2. 77.100 /  2. 77.100
  libswscale      2.  1.100 /  2.  1.100
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f280c0008c0] moov atom not found
out.mp4: Invalid data found when processing input


Though ffplay is able to play some other mp4 downloaded from youtube.

What might be the issue here ?

Suggestion of ffmpeg parameters for best compression regarding youtube and vimeo isvery much welcome.

Is winFF good for the job ?

Thanks


From soho123.2012 at gmail.com  Wed Aug 28 14:30:39 2013
From: soho123.2012 at gmail.com (Huang Soho)
Date: Wed, 28 Aug 2013 20:30:39 +0800
Subject: [FFmpeg-user] very low cpu usage when ffmpeg include "-re" option
Message-ID: <CAFd2cZWdcsFpYebFjC_bTGgwY5DAv_GK51jE+6x_GQZi3GMSUA@mail.gmail.com>

Hi All,

the command I use to stream output:
ffmpeg -sn -re -f video4linux2 -r 30 -s 1280x720 -input_format mjpeg -i
/dev/video0 -vcodec copy http://localhost:8090/feed1.ffm > /var/log/ffmpeg1
2>&1 &

I see the frame rate about 5fps only, cpu usage is about 10%,
but I set 30fps for target frame rate.
if I remove "-re" option, then the output is about 30fps. but it will
occupy a lot of cpu time (about 95%) when ffserver output the video stream.

Why "-re" option cause ffmpeg use very low cpu time?
it is very strange!
if I remove "-re" option, then ffmpeg+ffserver will occupy most of cpu time,
If I add "-re" option, then ffmpeg+ffserver will NOT use most of cput time,
Does any one have idea?
how to get the properly frame rate output and cpu usage with
ffserver+ffmpeg?

From rogat1y at gmail.com  Wed Aug 28 14:30:00 2013
From: rogat1y at gmail.com (Maxim Kozlov)
Date: Wed, 28 Aug 2013 16:30:00 +0400
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130828174908.6fed1cf5@shiva.selfip.org>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
Message-ID: <CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>

try
ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart
-f mp4 out.mp4


2013/8/28 J. Bakshi (?. ???? ) <j.bakshi at unlimitedmail.org>

> Dear list,
>
> I like to upload videos to vimeo and youtube.
>
> Hence I need to convert .dat .vom & .avi to mp4
>
> I have already converted .dat -> .mp4 with the following
>
>  ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 out.mp4
>
> and when try to play the .mp4 with ffplay I get the following error
>
> ffplay out.mp4
>
> ffplay version 0.11.1 Copyright (c) 2003-2012 the FFmpeg developers
>   built on Aug 11 2012 08:15:54 with gcc 4.7.1
>   configuration: --prefix=/usr --extra-cflags='-g -O2 -fstack-protector
> --param=ssp-buffer-size=4 -Wformat -Werror=format-security '
> --extra-ldflags='-Wl,-z,relro' --cc='ccache cc' --enable-shared
> --enable-libmp3lame --enable-gpl --enable-nonfree
> --disable-decoder=libdirac --enable-libvorbis --enable-pthreads
> --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab
> --enable-libgsm --enable-libtheora --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libx264 --enable-libspeex
> --enable-nonfree --disable-stripping --enable-libschroedinger
> --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg
> --enable-libvpx --enable-librtmp --enable-avfilter --enable-libfreetype
> --enable-libvo-aacenc --disable-decoder=amrnb --enable-libvo-amrwbenc
> --enable-libaacplus --libdir=/usr/lib/x86_64-linux-gnu --disable-vda
> --enable-libbluray --enable-libcdio --enable-gnutls --enable-frei0r
> --enable-openssl --enable-libass --enable-libdc1394 --disable-altivec -!
>  -disable-armv5te --disable-armv6 -  libavutil      51. 54.100 / 51. 54.100
>   libavcodec     54. 23.100 / 54. 23.100
>   libavformat    54.  6.100 / 54.  6.100
>   libavdevice    54.  0.100 / 54.  0.100
>   libavfilter     2. 77.100 /  2. 77.100
>   libswscale      2.  1.100 /  2.  1.100
>   libswresample   0. 15.100 /  0. 15.100
>   libpostproc    52.  0.100 / 52.  0.100
> ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f280c0008c0] moov atom not found
> out.mp4: Invalid data found when processing input
>
>
> Though ffplay is able to play some other mp4 downloaded from youtube.
>
> What might be the issue here ?
>
> Suggestion of ffmpeg parameters for best compression regarding youtube and
> vimeo isvery much welcome.
>
> Is winFF good for the job ?
>
> Thanks
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From rogat1y at gmail.com  Wed Aug 28 14:30:00 2013
From: rogat1y at gmail.com (Maxim Kozlov)
Date: Wed, 28 Aug 2013 16:30:00 +0400
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130828174908.6fed1cf5@shiva.selfip.org>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
Message-ID: <CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>

try
ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart
-f mp4 out.mp4


2013/8/28 J. Bakshi (?. ???? ) <j.bakshi at unlimitedmail.org>

> Dear list,
>
> I like to upload videos to vimeo and youtube.
>
> Hence I need to convert .dat .vom & .avi to mp4
>
> I have already converted .dat -> .mp4 with the following
>
>  ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 out.mp4
>
> and when try to play the .mp4 with ffplay I get the following error
>
> ffplay out.mp4
>
> ffplay version 0.11.1 Copyright (c) 2003-2012 the FFmpeg developers
>   built on Aug 11 2012 08:15:54 with gcc 4.7.1
>   configuration: --prefix=/usr --extra-cflags='-g -O2 -fstack-protector
> --param=ssp-buffer-size=4 -Wformat -Werror=format-security '
> --extra-ldflags='-Wl,-z,relro' --cc='ccache cc' --enable-shared
> --enable-libmp3lame --enable-gpl --enable-nonfree
> --disable-decoder=libdirac --enable-libvorbis --enable-pthreads
> --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab
> --enable-libgsm --enable-libtheora --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libx264 --enable-libspeex
> --enable-nonfree --disable-stripping --enable-libschroedinger
> --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg
> --enable-libvpx --enable-librtmp --enable-avfilter --enable-libfreetype
> --enable-libvo-aacenc --disable-decoder=amrnb --enable-libvo-amrwbenc
> --enable-libaacplus --libdir=/usr/lib/x86_64-linux-gnu --disable-vda
> --enable-libbluray --enable-libcdio --enable-gnutls --enable-frei0r
> --enable-openssl --enable-libass --enable-libdc1394 --disable-altivec -!
>  -disable-armv5te --disable-armv6 -  libavutil      51. 54.100 / 51. 54.100
>   libavcodec     54. 23.100 / 54. 23.100
>   libavformat    54.  6.100 / 54.  6.100
>   libavdevice    54.  0.100 / 54.  0.100
>   libavfilter     2. 77.100 /  2. 77.100
>   libswscale      2.  1.100 /  2.  1.100
>   libswresample   0. 15.100 /  0. 15.100
>   libpostproc    52.  0.100 / 52.  0.100
> ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f280c0008c0] moov atom not found
> out.mp4: Invalid data found when processing input
>
>
> Though ffplay is able to play some other mp4 downloaded from youtube.
>
> What might be the issue here ?
>
> Suggestion of ffmpeg parameters for best compression regarding youtube and
> vimeo isvery much welcome.
>
> Is winFF good for the job ?
>
> Thanks
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From j.bakshi at unlimitedmail.org  Wed Aug 28 14:46:54 2013
From: j.bakshi at unlimitedmail.org (J. Bakshi (=?UTF-8?B?4KacLiDgpqzgppXgprjgp4A=?= ))
Date: Wed, 28 Aug 2013 18:16:54 +0530
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
Message-ID: <20130828181654.1a880af9@shiva.selfip.org>

On Wed, 28 Aug 2013 16:30:00 +0400
Maxim Kozlov <rogat1y at gmail.com> wrote:

> try
> ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart
> -f mp4 out.mp4
> 
> 

Thanks for your response,

Tried with the parameters but get the following errors

[mp4 muxer @ 0x1349ac0] [Eval @ 0x7fffa1214d50] Undefined constant or missing '(' in 'faststart'
[mp4 muxer @ 0x1349ac0] Unable to parse option value "faststart"
[mp4 muxer @ 0x1349ac0] Error setting option movflags to value faststart.
Output #0, mp4, to 'out.mp4':
    Stream #0:0: Video: h264, yuv420p, 352x288 [SAR 178:163 DAR 1958:1467], q=-1--1, 90k tbn, 25 tbc
    Stream #0:1: Audio: aac, 44100 Hz, stereo, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mpeg1video -> libx264)
  Stream #0:1 -> #0:1 (mp2 -> libfaac)
Could not write header for output file #0 (incorrect codec parameters ?)



From j.bakshi at unlimitedmail.org  Wed Aug 28 14:46:54 2013
From: j.bakshi at unlimitedmail.org (J. Bakshi (=?UTF-8?B?4KacLiDgpqzgppXgprjgp4A=?= ))
Date: Wed, 28 Aug 2013 18:16:54 +0530
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
Message-ID: <20130828181654.1a880af9@shiva.selfip.org>

On Wed, 28 Aug 2013 16:30:00 +0400
Maxim Kozlov <rogat1y at gmail.com> wrote:

> try
> ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart
> -f mp4 out.mp4
> 
> 

Thanks for your response,

Tried with the parameters but get the following errors

[mp4 muxer @ 0x1349ac0] [Eval @ 0x7fffa1214d50] Undefined constant or missing '(' in 'faststart'
[mp4 muxer @ 0x1349ac0] Unable to parse option value "faststart"
[mp4 muxer @ 0x1349ac0] Error setting option movflags to value faststart.
Output #0, mp4, to 'out.mp4':
    Stream #0:0: Video: h264, yuv420p, 352x288 [SAR 178:163 DAR 1958:1467], q=-1--1, 90k tbn, 25 tbc
    Stream #0:1: Audio: aac, 44100 Hz, stereo, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mpeg1video -> libx264)
  Stream #0:1 -> #0:1 (mp2 -> libfaac)
Could not write header for output file #0 (incorrect codec parameters ?)



From onemda at gmail.com  Wed Aug 28 14:54:27 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 28 Aug 2013 12:54:27 +0000
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130828181654.1a880af9@shiva.selfip.org>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
	<20130828181654.1a880af9@shiva.selfip.org>
Message-ID: <CAPYw7P7p8HJd4FEGD+MwPcYCGhWiYsgJwxSwBsy8=Ba8jtMy-w@mail.gmail.com>

On 8/28/13, J. Bakshi (***. ************ ) <j.bakshi at unlimitedmail.org> wrote:
> On Wed, 28 Aug 2013 16:30:00 +0400
> Maxim Kozlov <rogat1y at gmail.com> wrote:
>
>> try
>> ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart
>> -f mp4 out.mp4
>>
>>
>
> Thanks for your response,
>
> Tried with the parameters but get the following errors

Because you use extremly old ffmpeg version.

>
> [mp4 muxer @ 0x1349ac0] [Eval @ 0x7fffa1214d50] Undefined constant or
> missing '(' in 'faststart'
> [mp4 muxer @ 0x1349ac0] Unable to parse option value "faststart"
> [mp4 muxer @ 0x1349ac0] Error setting option movflags to value faststart.
> Output #0, mp4, to 'out.mp4':
>     Stream #0:0: Video: h264, yuv420p, 352x288 [SAR 178:163 DAR 1958:1467],
> q=-1--1, 90k tbn, 25 tbc
>     Stream #0:1: Audio: aac, 44100 Hz, stereo, s16, 128 kb/s
> Stream mapping:
>   Stream #0:0 -> #0:0 (mpeg1video -> libx264)
>   Stream #0:1 -> #0:1 (mp2 -> libfaac)
> Could not write header for output file #0 (incorrect codec parameters ?)
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From j.bakshi at unlimitedmail.org  Wed Aug 28 15:14:01 2013
From: j.bakshi at unlimitedmail.org (J. Bakshi (=?UTF-8?B?4KacLiDgpqzgppXgprjgp4A=?= ))
Date: Wed, 28 Aug 2013 18:44:01 +0530
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <CAPYw7P7p8HJd4FEGD+MwPcYCGhWiYsgJwxSwBsy8=Ba8jtMy-w@mail.gmail.com>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
	<20130828181654.1a880af9@shiva.selfip.org>
	<CAPYw7P7p8HJd4FEGD+MwPcYCGhWiYsgJwxSwBsy8=Ba8jtMy-w@mail.gmail.com>
Message-ID: <20130828184401.58b4bb13@shiva.selfip.org>

On Wed, 28 Aug 2013 12:54:27 +0000
Paul B Mahol <onemda at gmail.com> wrote:

> On 8/28/13, J. Bakshi (***. ************ ) <j.bakshi at unlimitedmail.org> wrote:
> > On Wed, 28 Aug 2013 16:30:00 +0400
> > Maxim Kozlov <rogat1y at gmail.com> wrote:
> >
> >> try
> >> ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart
> >> -f mp4 out.mp4
> >>
> >>
> >
> > Thanks for your response,
> >
> > Tried with the parameters but get the following errors
> 
> Because you use extremly old ffmpeg version.
> 

Many many thanks.

I have upgrade from debian multimedia repo and it works :-)


From lucielemonnier at hotmail.fr  Wed Aug 28 17:24:24 2013
From: lucielemonnier at hotmail.fr (lucie Lemonnier)
Date: Wed, 28 Aug 2013 17:24:24 +0200
Subject: [FFmpeg-user] drop frames for real time streaming?
Message-ID: <DUB111-W111891B1422631A0212761CB94B0@phx.gbl>

Hello, 
I stream a video with multiple overlays but I can't reach 25 fps to output frame rate.
example : ffmpeg.exe -i video.mp4 -vf "movie=video1.mov [mv];movie=video2.mov [mv2];[in] crop=1920:1080:2565:0 [tin];[mv] crop=1920:1080:2565:0 [tmv];[mv2] crop=1920:1080:2565:0 [tmv2];[tin][tmv] overlay=0:0 [tout1];[tout1][tmv2] overlay=0:0" -an -vcodec mpeg4  -f rtp rtp://192.168.54.105:1234
Is it possible to drop frames to obtain a real time streaming output?
Thanks.
Lucie
 		 	   		  

From soho123.2012 at gmail.com  Wed Aug 28 18:25:20 2013
From: soho123.2012 at gmail.com (Huang Soho)
Date: Thu, 29 Aug 2013 00:25:20 +0800
Subject: [FFmpeg-user] Can FFM_PACKET_SIZE 4096 bytes be set more large?
Message-ID: <CAFd2cZXniT=6HfbX-8AZXgrStSU=hiiSpoVsHpwWRC4AU7Nq5g@mail.gmail.com>

Hi All,

FFM_PACKET_SIZE is 4096 bytes.
the size seems too small such that gets more the loop count while
read/write.
Can FFM_PACKET_SIZE be set larger than 4096 bytes?

From sw.craftsman at gmail.com  Wed Aug 28 20:09:15 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Wed, 28 Aug 2013 23:39:15 +0530
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
	<CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
Message-ID: <CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>

Yes I have watched the video.

What is the point you're trying to make ?


On Wed, Aug 28, 2013 at 4:53 PM, Paul B Mahol <onemda at gmail.com> wrote:

>
> Did you actually watched video?
>

From onemda at gmail.com  Wed Aug 28 20:13:18 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 28 Aug 2013 18:13:18 +0000
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
	<CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
	<CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>
Message-ID: <CAPYw7P4=hojo8hr7xA-NcEBDs-pObq+G-soOHiCSo-EeNa=FLA@mail.gmail.com>

On 8/28/13, Shyam Sundar <sw.craftsman at gmail.com> wrote:
> Yes I have watched the video.
>
> What is the point you're trying to make ?

And when watching at what fps it was?

>
>
> On Wed, Aug 28, 2013 at 4:53 PM, Paul B Mahol <onemda at gmail.com> wrote:
>
>>
>> Did you actually watched video?
>>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From alan at trimble.net  Thu Aug 29 00:15:17 2013
From: alan at trimble.net (Alan Trimble)
Date: Wed, 28 Aug 2013 15:15:17 -0700
Subject: [FFmpeg-user] Avid mxf to mpeg? (new user)
Message-ID: <521E7675.9030904@trimble.net>

I am attempting to convert a native Avid .mxf file to mpeg2 or some
other readable/viewable format in OSX. The original video is 1920x1080
PAL - 50hz... 1080i format.

Trying:
./ffmpeg -f dnxhd -i ./America.mxf -target mpeg2video test.mpeg

Results in:
[dnxhd @ 0x100817400] Could not find codec parameters for stream 0 
(Video: dnxhd): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' 
options
./America.mxf: could not find codec parameters

I haven't found any obvious flags for directly specifying the 
resolution/size
in the documentation. Thanks in advance for advice!
-AT

From dashing.meng at gmail.com  Thu Aug 29 02:32:19 2013
From: dashing.meng at gmail.com (littlebat)
Date: Thu, 29 Aug 2013 08:32:19 +0800
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130828174908.6fed1cf5@shiva.selfip.org>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
Message-ID: <20130829083219.ae4c502d.dashing.meng@gmail.com>

On Wed, 28 Aug 2013 17:49:08 +0530
"J. Bakshi (?. ???? )" <j.bakshi at unlimitedmail.org> wrote:

> Dear list,
> 
> I like to upload videos to vimeo and youtube.
> 
> Hence I need to convert .dat .vom & .avi to mp4

I have ever uploaded MTS file onto youtube, it will convert them into
proper format and size automatically, no need me convert it first. Have
you tried upload your file without converting? You can cut a snip to do
a test: dd if=sample-file of=small-sample-file bs=1024 count=10000

From j.bakshi at unlimitedmail.org  Thu Aug 29 09:27:37 2013
From: j.bakshi at unlimitedmail.org (J. Bakshi (=?UTF-8?B?4KacLiDgpqzgppXgprjgp4A=?= ))
Date: Thu, 29 Aug 2013 12:57:37 +0530
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130829083219.ae4c502d.dashing.meng@gmail.com>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<20130829083219.ae4c502d.dashing.meng@gmail.com>
Message-ID: <20130829125737.42ef4d56@shiva.selfip.org>


Thanks for sharing the info



On Thu, 29 Aug 2013 08:32:19 +0800
littlebat <dashing.meng at gmail.com> wrote:

> On Wed, 28 Aug 2013 17:49:08 +0530
> "J. Bakshi (?. ???? )" <j.bakshi at unlimitedmail.org> wrote:
> 
> > Dear list,
> > 
> > I like to upload videos to vimeo and youtube.
> > 
> > Hence I need to convert .dat .vom & .avi to mp4
> 
> I have ever uploaded MTS file onto youtube, it will convert them into
> proper format and size automatically, no need me convert it first. Have
> you tried upload your file without converting? You can cut a snip to do
> a test: dd if=sample-file of=small-sample-file bs=1024 count=10000
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> 

From sw.craftsman at gmail.com  Thu Aug 29 09:53:08 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Thu, 29 Aug 2013 13:23:08 +0530
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CAPYw7P4=hojo8hr7xA-NcEBDs-pObq+G-soOHiCSo-EeNa=FLA@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
	<CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
	<CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>
	<CAPYw7P4=hojo8hr7xA-NcEBDs-pObq+G-soOHiCSo-EeNa=FLA@mail.gmail.com>
Message-ID: <CA+XaXCng40XppF0vwgJBAnNVVf5uWB4ZqzGseAb-Sy72kD8PkA@mail.gmail.com>

Not sure what do you mean ... I watched it in VLC player, which doesn't
show FPS anywhere.

Am basically trying to understand the reason for results-difference in
FFProbe's own two different versions.

Thanks.

On Wed, Aug 28, 2013 at 11:43 PM, Paul B Mahol <onemda at gmail.com> wrote:


> On 8/28/13, Shyam Sundar <sw.craftsman at gmail.com> wrote:
> > Yes I have watched the video.
> >
> > What is the point you're trying to make ?
>
> And when watching at what fps it was?
>
>

From lucielemonnier at hotmail.fr  Thu Aug 29 10:09:31 2013
From: lucielemonnier at hotmail.fr (lucie Lemonnier)
Date: Thu, 29 Aug 2013 10:09:31 +0200
Subject: [FFmpeg-user] drop frames for real time streaming?
In-Reply-To: <DUB111-W111891B1422631A0212761CB94B0@phx.gbl>
References: <DUB111-W111891B1422631A0212761CB94B0@phx.gbl>
Message-ID: <DUB111-W123A1A3FD7DCDF2F693C63DB9340@phx.gbl>

Hello, 
I stream a video with multiple overlays but I can't reach 25 fps to output frame rate.
example : ffmpeg.exe -i video.mp4 -vf "movie=video1.mov [mv];movie=video2.mov [mv2];[in] crop=1920:1080:2565:0 [tin];[mv] crop=1920:1080:2565:0 [tmv];[mv2] crop=1920:1080:2565:0 [tmv2];[tin][tmv] overlay=0:0 [tout1];[tout1][tmv2] overlay=0:0" -an -vcodec mpeg4  -f rtp rtp://192.168.54.105:1234
Is it possible to drop frames to obtain a real time streaming output?
Thanks.
Lucie
 		 	   		   		 	   		  

From j.bakshi at unlimitedmail.org  Thu Aug 29 11:49:04 2013
From: j.bakshi at unlimitedmail.org (J. Bakshi (=?UTF-8?B?4KacLiDgpqzgppXgprjgp4A=?= ))
Date: Thu, 29 Aug 2013 15:19:04 +0530
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130828184401.58b4bb13@shiva.selfip.org>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
	<20130828181654.1a880af9@shiva.selfip.org>
	<CAPYw7P7p8HJd4FEGD+MwPcYCGhWiYsgJwxSwBsy8=Ba8jtMy-w@mail.gmail.com>
	<20130828184401.58b4bb13@shiva.selfip.org>
Message-ID: <20130829151904.67447577@shiva.selfip.org>

On Wed, 28 Aug 2013 18:44:01 +0530
"J. Bakshi (?. ???? )" <j.bakshi at unlimitedmail.org> wrote:

> On Wed, 28 Aug 2013 12:54:27 +0000
> Paul B Mahol <onemda at gmail.com> wrote:
> 
> > On 8/28/13, J. Bakshi (***. ************ ) <j.bakshi at unlimitedmail.org> wrote:
> > > On Wed, 28 Aug 2013 16:30:00 +0400
> > > Maxim Kozlov <rogat1y at gmail.com> wrote:
> > >
> > >> try
> > >> ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart
> > >> -f mp4 out.mp4
> > >>
> > >>
> > >
> > > Thanks for your response,
> > >
> > > Tried with the parameters but get the following errors
> > 
> > Because you use extremly old ffmpeg version.
> > 
> 
> Many many thanks.
> 
> I have upgrade from debian multimedia repo and it works :-)


The same issue :-(

I have successfully ran the command

ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags faststart -f mp4 out.mp4

and after completion when I try to play mp4 I get

[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f5dcc000920] moov atom not found0/0   
out.mp4: Invalid data found when processing input

Why again the error ?


From cehoyos at ag.or.at  Thu Aug 29 11:53:13 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 29 Aug 2013 09:53:13 +0000 (UTC)
Subject: [FFmpeg-user] Avid mxf to mpeg? (new user)
References: <521E7675.9030904@trimble.net>
Message-ID: <loom.20130829T115016-314@post.gmane.org>

Alan Trimble <alan <at> trimble.net> writes:

> ./ffmpeg -f dnxhd -i ./America.mxf -target mpeg2video test.mpeg

Complete, uncut console output missing.
What is -target mpeg2video supposed to do?

As said, if it does not work without -f dnxhd, please provide 
the input sample.

> I haven't found any obvious flags for directly specifying the 
> resolution/size in the documentation.

http://ffmpeg.org/ffmpeg-all.html#Video-size
But please understand that this cannot work for "real" videos 
like mxf.

Carl Eugen


From cehoyos at ag.or.at  Thu Aug 29 11:57:20 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 29 Aug 2013 09:57:20 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg cannot capture h264 from webcam C920
References: <kvdeh5$2qh$1@ger.gmane.org>
Message-ID: <loom.20130829T115604-277@post.gmane.org>

andrea <mariofutire <at> googlemail.com> writes:

> I have the same issue reported here
> 
> http://ffmpeg.org/pipermail/ffmpeg-user/2013-May/015046.html
> and
> http://ffmpeg.org/pipermail/ffmpeg-user/2013-July/016243.html
> 
> I have tried ffmpeg 1.2.2 on fedora and 2.0.1 on raspberry 
> (same result). I have only tried ffplay 1.2.2 on fedora.

Iirc, all users so far reported (repeatedly) that ffplay 
works fine with this camera (as does reencoding with ffmpeg).
Please test reencoding with current ffmpeg and report back.
(current != 2.0.1 !)

Carl Eugen


From doenerbudenchef at yahoo.de  Thu Aug 29 12:02:49 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Thu, 29 Aug 2013 03:02:49 -0700 (PDT)
Subject: [FFmpeg-user] Increase volume while using amerge
In-Reply-To: <CAPYw7P7Z9GLwCOkNuGkqegLavAkv3MmsEGc4Vs+T+pM7OA0usQ@mail.gmail.com>
References: <1373542433407-4659924.post@n4.nabble.com>
	<loom.20130711T140730-496@post.gmane.org>
	<1373545871656-4659926.post@n4.nabble.com>
	<1373880432667-4659981.post@n4.nabble.com>
	<CAPYw7P70EQGdMVcoBswSYABTin=VAnepLJF33fTnvDuNMaF7=A@mail.gmail.com>
	<1373884302257-4659984.post@n4.nabble.com>
	<CAPYw7P7Z9GLwCOkNuGkqegLavAkv3MmsEGc4Vs+T+pM7OA0usQ@mail.gmail.com>
Message-ID: <1377770569656-4660966.post@n4.nabble.com>

Hi all,

I am using "-filter_complex amerge,volume=15.8dB" to merge the audio streams
and increase the volume. It works pretty fine for me.

But I've noticed that amerge maps the audio to the first stream in the
container (0:0):

Stream mapping:
  Stream #0:1 (pcm_s24le) -> amerge:in0 (graph 0)
  Stream #0:2 (pcm_s24le) -> amerge:in1 (graph 0)
  volume (graph 0) -> Stream #0:0 (mp2)
  Stream #0:0 -> #0:1 (mpeg2video -> mpeg2video)

Is there a way to put the video to the first and the audio to the second
stream?

Thanks,
Alex





--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Increase-volume-while-using-amerge-tp4659924p4660966.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From onemda at gmail.com  Thu Aug 29 12:42:44 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 10:42:44 +0000
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CA+XaXCng40XppF0vwgJBAnNVVf5uWB4ZqzGseAb-Sy72kD8PkA@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
	<CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
	<CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>
	<CAPYw7P4=hojo8hr7xA-NcEBDs-pObq+G-soOHiCSo-EeNa=FLA@mail.gmail.com>
	<CA+XaXCng40XppF0vwgJBAnNVVf5uWB4ZqzGseAb-Sy72kD8PkA@mail.gmail.com>
Message-ID: <CAPYw7P6NjLAj-d9Jb7YRwzFvYseBVqK9jGk4yNEoKOu1C+_iHA@mail.gmail.com>

On 8/29/13, Shyam Sundar <sw.craftsman at gmail.com> wrote:
> Not sure what do you mean ... I watched it in VLC player, which doesn't
> show FPS anywhere.

Do you have link to sample?, so I can see what fps is actually correct one.

Please stop top-posting.

>
> Am basically trying to understand the reason for results-difference in
> FFProbe's own two different versions.
>
> Thanks.
>
> On Wed, Aug 28, 2013 at 11:43 PM, Paul B Mahol <onemda at gmail.com> wrote:
>
>
>> On 8/28/13, Shyam Sundar <sw.craftsman at gmail.com> wrote:
>> > Yes I have watched the video.
>> >
>> > What is the point you're trying to make ?
>>
>> And when watching at what fps it was?
>>
>>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From sw.craftsman at gmail.com  Thu Aug 29 12:55:37 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Thu, 29 Aug 2013 16:25:37 +0530
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CAPYw7P6NjLAj-d9Jb7YRwzFvYseBVqK9jGk4yNEoKOu1C+_iHA@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
	<CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
	<CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>
	<CAPYw7P4=hojo8hr7xA-NcEBDs-pObq+G-soOHiCSo-EeNa=FLA@mail.gmail.com>
	<CA+XaXCng40XppF0vwgJBAnNVVf5uWB4ZqzGseAb-Sy72kD8PkA@mail.gmail.com>
	<CAPYw7P6NjLAj-d9Jb7YRwzFvYseBVqK9jGk4yNEoKOu1C+_iHA@mail.gmail.com>
Message-ID: <CA+XaXCmxxBbFLNztbuE0V9zYphuxq6h9D1R3-xdZZhSF9ZsikA@mail.gmail.com>

Yeah ... it's the same as mentioned in my original post.

http://files.videohelp.com/u/211860/Video_LXF.lxf

Thanks.

From sw.craftsman at gmail.com  Thu Aug 29 13:12:32 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Thu, 29 Aug 2013 16:42:32 +0530
Subject: [FFmpeg-user] 'Duration: N/A' for some videos ?
Message-ID: <CA+XaXCmrc0bcLZAihwiiiY_3Z9sYBeSajRd-AmB-cSmmUBUeCA@mail.gmail.com>

Hi All,

I am seeing Duration as N/A for number of videos ...

-------------------------------------------------------------------------------------------------------------------
$ ./ffprobe.exe C:/Rajiv/pixeltools.atsc.cea608.mpg

ffprobe version 1.2 Copyright (c) 2007-2013 the FFmpeg developers
  built on Mar 28 2013 00:34:08 with gcc 4.8.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-libass --enable-libbluray --enable-libcaca
--enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-libtwolame
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid
--enable-zlib
  libavutil      52. 18.100 / 52. 18.100
  libavcodec     54. 92.100 / 54. 92.100
  libavformat    54. 63.104 / 54. 63.104
  libavdevice    54.  3.103 / 54.  3.103
  libavfilter     3. 42.103 /  3. 42.103
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[mpegvideo @ 000000000226ea00] Estimating duration from bitrate, this may
be inaccurate
Input #0, mpegvideo, from 'C:/Rajiv/pixeltools.atsc.cea608.mpg':
*  Duration: N/A, bitrate: N/A*
    Stream #0:0: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9 DAR
4:3], 29.97 fps, 29.97 tbr, 1200k tbn, 59.94 tbc
-------------------------------------------------------------------------------------------------------------------

It's a valid video of few seconds and plays fine.

Sample - http://files.videohelp.com/u/211860/pixeltools.atsc.cea608.mpg

Any idea ?

Thanks.

From cehoyos at ag.or.at  Thu Aug 29 14:46:15 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 29 Aug 2013 12:46:15 +0000 (UTC)
Subject: [FFmpeg-user] 'Duration: N/A' for some videos ?
References: <CA+XaXCmrc0bcLZAihwiiiY_3Z9sYBeSajRd-AmB-cSmmUBUeCA@mail.gmail.com>
Message-ID: <loom.20130829T144445-68@post.gmane.org>

Shyam Sundar <sw.craftsman <at> gmail.com> writes:

> I am seeing Duration as N/A for number of videos ...

> $ ./ffprobe.exe C:/Rajiv/pixeltools.atsc.cea608.mpg
> 
> ffprobe version 1.2

This is old.

mpeg2video only specifies a "maximum bitrate", no average 
bitrate. (The maximum bitrate is shown by newer versions.)
So the only possibility to find out the actual duration is 
to decode the video.

Carl Eugen


From onemda at gmail.com  Thu Aug 29 14:54:57 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 12:54:57 +0000
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CA+XaXCmxxBbFLNztbuE0V9zYphuxq6h9D1R3-xdZZhSF9ZsikA@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
	<CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
	<CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>
	<CAPYw7P4=hojo8hr7xA-NcEBDs-pObq+G-soOHiCSo-EeNa=FLA@mail.gmail.com>
	<CA+XaXCng40XppF0vwgJBAnNVVf5uWB4ZqzGseAb-Sy72kD8PkA@mail.gmail.com>
	<CAPYw7P6NjLAj-d9Jb7YRwzFvYseBVqK9jGk4yNEoKOu1C+_iHA@mail.gmail.com>
	<CA+XaXCmxxBbFLNztbuE0V9zYphuxq6h9D1R3-xdZZhSF9ZsikA@mail.gmail.com>
Message-ID: <CAPYw7P6jE4FcUSk3Kuj3=uDejuqi2qXk0t_4Ffzs4-xmThmuKg@mail.gmail.com>

On 8/29/13, Shyam Sundar <sw.craftsman at gmail.com> wrote:
> Yeah ... it's the same as mentioned in my original post.
>
> http://files.videohelp.com/u/211860/Video_LXF.lxf

Maybe its reported as 14.99 because it is iterlaced?

Inspecting file there are approx 30 frames per second.

From rogat1y at gmail.com  Thu Aug 29 15:04:23 2013
From: rogat1y at gmail.com (Maxim Kozlov)
Date: Thu, 29 Aug 2013 17:04:23 +0400
Subject: [FFmpeg-user] rtmp streaming to two media servers
Message-ID: <CAB-_x2Mdd--Vf1BjkVJhsd6zheddtXMy574B4gVQP38JRwDBWw@mail.gmail.com>

Hi.

I stream live video to two media servers(wowza) using such command line:
ffmpeg [input, libx264 and libfdk_aac options] -f tee "[f=flv]rtmp://
192.168.0.1/live/stream1|[f=flv]rtmp://192.168.0.2/live/stream1"

Works perfect.

But if one of servers crashes, ffmpeg  exits with message:
WriteN, RTMP send error 10053 (129 bytes)
WriteN, RTMP send error 10053 (48 bytes)
WriteN, RTMP send error 10038 (42 bytes)
av_interleaved_write_frame(): Operation not permitted

Please suggest the way to make ffmpeg reconnect to media server after
disconnection...like it doing Adobe Flaash Media Live Encoder?

ffmpeg running on windows 7
ffmpeg version N-55786-g185fc52 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 26 2013 19:39:07 with gcc 4.8.0 (GCC)
  configuration: --arch=x86 --target-os=mingw32
--cross-prefix=/mingw-w64-i686/bin/i686-w64-mingw32-
--pkg-config=pkg-config --enable-gpl --enable-libsoxr --enable-libx264
--enable-avisynth --enable-libxvid --enable-libmp3lame --enable-version3
--enable-zlib --enable-librtmp --enable-libvorbis --enable-libtheora
--enable-libspeex --enable-libopenjpeg --enable-gnutls --enable-libgsm
--enable-libfreetype --enable-fontconfig --enable-libass
--enable-libutvideo --enable-libopus --disable-w32threads --enable-frei0r
--enable-filter=frei0r --enable-libvo-aacenc --enable-bzlib
--enable-libxavs --extra-cflags=-DPTW32_STATIC_LIB
--enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libvo-amrwbenc --enable-libschroedinger --enable-libbluray
--enable-libvpx --enable-libilbc --enable-static --disable-shared
--enable-nonfree --enable-libfdk-aac --enable-runtime-cpudetect
  libavutil      52. 42.100 / 52. 42.100
  libavcodec     55. 29.100 / 55. 29.100
  libavformat    55. 14.102 / 55. 14.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.102 /  3. 82.102
  libswscale      2.  5.100 /  2.  5.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100

p.s. I hope my english not so bad and you understand me =)

From nicolas.george at normalesup.org  Thu Aug 29 15:25:14 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Thu, 29 Aug 2013 15:25:14 +0200
Subject: [FFmpeg-user] Increase volume while using amerge
In-Reply-To: <1377770569656-4660966.post@n4.nabble.com>
References: <1373542433407-4659924.post@n4.nabble.com>
	<loom.20130711T140730-496@post.gmane.org>
	<1373545871656-4659926.post@n4.nabble.com>
	<1373880432667-4659981.post@n4.nabble.com>
	<CAPYw7P70EQGdMVcoBswSYABTin=VAnepLJF33fTnvDuNMaF7=A@mail.gmail.com>
	<1373884302257-4659984.post@n4.nabble.com>
	<CAPYw7P7Z9GLwCOkNuGkqegLavAkv3MmsEGc4Vs+T+pM7OA0usQ@mail.gmail.com>
	<1377770569656-4660966.post@n4.nabble.com>
Message-ID: <20130829132514.GA27687@phare.normalesup.org>

Le duodi 12 fructidor, an CCXXI, Alex a ?crit?:
> Stream mapping:
>   Stream #0:1 (pcm_s24le) -> amerge:in0 (graph 0)
>   Stream #0:2 (pcm_s24le) -> amerge:in1 (graph 0)
>   volume (graph 0) -> Stream #0:0 (mp2)
>   Stream #0:0 -> #0:1 (mpeg2video -> mpeg2video)
> 
> Is there a way to put the video to the first and the audio to the second
> stream?

You need to give a name for the pads in your filter chain and specify the
mapping:

  -lavfi '[0:1][0:2]amerge,volume[a]' -map 0:0 -map '[a]'

Specifying the inputs you want is good practice too IMHO.

Regards,

-- 
  Nicolas George
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From onemda at gmail.com  Thu Aug 29 16:03:39 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 14:03:39 +0000
Subject: [FFmpeg-user] ffmpeg Unknown encoder 'libx264' error
In-Reply-To: <1375795324.7275.25.camel@videoathlon>
References: <1375666070.5067.41.camel@videoathlon>
	<loom.20130805T194808-897@post.gmane.org>
	<1375795324.7275.25.camel@videoathlon>
Message-ID: <CAPYw7P72bww1cr-5H99WOAOFHS2PMB2mPH01kgGjQQ_3ZGytYQ@mail.gmail.com>

On 8/6/13, Ned Haskin <feanor2112 at covad.net> wrote:
> On Mon, 2013-08-05 at 17:57 +0000, Carl Eugen Hoyos wrote:
>> Ned Haskin <feanor2112 <at> covad.net> writes:
>>
>> > $ ./configure --enable-libdc1394 --prefix=/usr
>>
>> Mostly unrelated to your original question:
>>
>> > --extra-cflags='-Wall -g '
>>
>> Please remove this.
>>
>> > --enable-gpl
>>
>> > --enable-shared
>>
>> (This is of course 100% ok, but if you are unexperienced
>> I would strongly recommend not to use it, it generally
>> has theoretical problems and the gain for you as a
>> user is not completely clear to me, in combination with
>> "--prefix=/usr", I personally consider it highly
>> dangerous and would not use it on my system.)
>>
>> > --enable-nonfree --enable-libx264
>>
>> > --enable-encoder=libx264
>>
>> This makes no sense in the configure line you posted.
> I was trying to get the encoder to compile, not sure why it wouldn't. I did
> compile ffmpeg many times before posting.

correct flag is:

--enable-libx264

>>
>> > --enable-libmp3lame --enable-libvorbis
>>
>> > --enable-libtwolame --enable-libv4l2
> I use them both on other projects. Although I see now that a mp2 encoder is
> in ffmpeg.

Please ignore what Carl says about those two.

>>
>> (I would even recommend to remove these two.)
>>
>> > --enable-pthreads
>>
>> This is the default, please remove it.
>>
>> > --enable-libfaac --enable-libxvid
>>
>> > --enable-postproc
>>
>> This also makes no sense.
> It is in the debian ffmpeg configure line.

That doesn't mean it is correct.

>>
> Got ffmpeg to compile and libx264 to work with yesterdays git. I noticed
> that --arch=amd64 didn't change ARCH. I also use
> ffmpeg to transcode input files for cinelerra-cv & mjpeg is quite slow. Or
> is that just mjpeg code?
>
> ned337 at vid:~/ffmpeg$ ./configure --arch=amd64 --enable-opencl --enable-gpl
> --enable-nonfree --enable-libx264
> --enable-libxvid --enable-libmp3lame --enable-libfaac
> install prefix            /usr/local
> source path               .
> C compiler                gcc
> ARCH                      x86 (generic)
> big-endian                no
> runtime cpu detection     yes
> yasm                      yes
> MMX enabled               yes
> MMXEXT enabled            yes
> 3DNow! enabled            yes
> 3DNow! extended enabled   yes
> SSE enabled               yes
> SSSE3 enabled             yes
> AVX enabled               yes
> FMA4 enabled              yes
> i686 features enabled     yes
> CMOV is fast              yes
> EBX available             yes
> EBP available             yes
> debug symbols             yes
> strip symbols             yes
> optimize for size         no
> optimizations             yes
> static                    yes
> shared                    no
> postprocessing support    yes
> new filter support        yes
> network support           yes
> threading support         pthreads
> safe bitstream reader     yes
> SDL support               yes
> opencl enabled            yes
> texi2html enabled         yes
> perl enabled              yes
> pod2man enabled           yes
> makeinfo enabled          yes
>
> External libraries:
> iconv			libx264			opencl
> libfaac			libxvid			zlib
> libmp3lame
>
> skip stuff: no errors
>
> License: nonfree and unredistributable
> Creating config.mak, config.h, and doc/config.texi...
> libavutil/avconfig.h is unchanged
>
> Thanks,
> Ned
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Thu Aug 29 17:13:42 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 29 Aug 2013 15:13:42 +0000 (UTC)
Subject: [FFmpeg-user] uspp filter 444p
References: <nukmceguhrpioaposwzc@iwcm>
Message-ID: <loom.20130829T171030-297@post.gmane.org>

chronek <chronek <at> interia.eu> writes:

> I want ask if someone from developers could check 
> uspp filter. This filter is port from mplayer and 
> it is uniqe, cause doing full compresion/
> decompression based on snow codec. Snow codec 
> support pixel format 444p so changing uspp filter 
> to support that will not need many work.

A patch will be welcome but since the filer was not 
ported yet, porting should be the first step imo.
If you are interested in working on the port, commit 
a2c547f could act as an example.

Carl Eugen


From thierry at lelegard.fr  Thu Aug 29 18:32:04 2013
From: thierry at lelegard.fr (Thierry Lelegard)
Date: Thu, 29 Aug 2013 18:32:04 +0200 (CEST)
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay filter
	not working
In-Reply-To: <419479807.424442289.1377792999180.JavaMail.root@spooler3-g27.priv.proxad.net>
Message-ID: <1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>

Hello all,

Sorry, yet another "subtitles" + "not working" topic ;-)

I have read the manual and a lot of posts about burning DVD subtitles into
the video stream using the overlay filter. I cannot make it work on a small
file. To make it clear, I want to remove the subtitles as an independent
stream and I want to burn/hardcode/encrust/whatever the subtitles into the
video images.

I have a small Matroska file (20 seconds) with one video stream, one audio
stream and one DVD subtitle stream:

Stream #0:0: Video: h264 (High), yuv420p, 1280x536 [SAR 1:1 DAR 160:67], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
Stream #0:1: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s (default)
Stream #0:2: Subtitle: dvd_subtitle (default)

The subtitles are correctly displayed when the file is played using vlc
and mplayer. So, I assume that the subtitles data are correct.

To make a test, I try to burn the subtitles into the video using the following
command (full output at end of mail).

ffmpeg -i f.mkv -filter_complex "[0:v][0:s]overlay" -codec:v mpeg2video -codec:a mp2 -ac 2 -y f.hardsubs.mpg

It seems the simplest possible command to do. Please don't comment on video
quality, it is just a test to see if the subtitles are burnt into the video.
The resulting video does not contain subtitles.

I tried explicit x,y placement in the overlay and other options without more luck.

Any idea? So many people report successful DVD subtitles processing using
that type of command. What could be wrong?

Thanks in advance.
-Thierry

Full command output:
$ ffmpeg -i f.mkv -filter_complex "[0:v][0:s]overlay" -codec:v mpeg2video -codec:a mp2 -ac 2 -y f.hardsubs.mpg
ffmpeg version N-55753-g88909be Copyright (c) 2000-2013 the FFmpeg developers
  built on Aug 24 2013 21:47:18 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 42.100 / 52. 42.100
  libavcodec     55. 29.100 / 55. 29.100
  libavformat    55. 14.102 / 55. 14.102
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.102 /  3. 82.102
  libswscale      2.  5.100 /  2.  5.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Input #0, matroska,webm, from 'f.mkv':
  Metadata:
    ENCODER         : Lavf55.14.102
  Duration: 00:00:23.00, start: 0.000000, bitrate: 5539 kb/s
    Chapter #0.0: start 0.000000, end 20.000000
    Stream #0:0: Video: h264 (High), yuv420p, 1280x536 [SAR 1:1 DAR 160:67], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
    Stream #0:1: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s (default)
    Stream #0:2: Subtitle: dvd_subtitle (default)
[mpeg @ 00000000026aed20] VBV buffer size not set, muxing may fail
Output #0, mpeg, to 'f.hardsubs.mpg':
  Metadata:
    encoder         : Lavf55.14.102
    Chapter #0.0: start 0.000000, end 20.000000
    Stream #0:0: Video: mpeg2video, yuv420p, 1280x536 [SAR 1:1 DAR 160:67], q=2-31, 200 kb/s, 90k tbn, 23.98 tbc
    Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s (default)
Stream mapping:
  Stream #0:0 (h264) -> overlay:main (graph 0)
  Stream #0:2 (dvdsub) -> overlay:overlay (graph 0)
  overlay (graph 0) -> Stream #0:0 (mpeg2video)
  Stream #0:1 -> #0:1 (dca -> mp2)
Press [q] to stop, [?] for help
frame=  483 fps= 80 q=31.0 Lsize=    2188kB time=00:00:20.10 bitrate= 891.6kbits/s dup=15 drop=0    
video:1866kB audio:303kB subtitle:0 global headers:0kB muxing overhead 0.877978%

From onemda at gmail.com  Thu Aug 29 18:50:54 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 16:50:54 +0000
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
References: <419479807.424442289.1377792999180.JavaMail.root@spooler3-g27.priv.proxad.net>
	<1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
Message-ID: <CAPYw7P5xOhAFOsx3LNdx+bZ9wr5Y_aU_s_8=cxkmpF79TUqj=Q@mail.gmail.com>

On 8/29/13, Thierry Lelegard <thierry at lelegard.fr> wrote:
> Hello all,
>
> Sorry, yet another "subtitles" + "not working" topic ;-)
>
> I have read the manual and a lot of posts about burning DVD subtitles into
> the video stream using the overlay filter. I cannot make it work on a small

What manual tells that for burning subtitles one should use overlay filter?

> file. To make it clear, I want to remove the subtitles as an independent
> stream and I want to burn/hardcode/encrust/whatever the subtitles into the
> video images.
>
> I have a small Matroska file (20 seconds) with one video stream, one audio
> stream and one DVD subtitle stream:
>
> Stream #0:0: Video: h264 (High), yuv420p, 1280x536 [SAR 1:1 DAR 160:67],
> 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
> Stream #0:1: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s
> (default)
> Stream #0:2: Subtitle: dvd_subtitle (default)
>
> The subtitles are correctly displayed when the file is played using vlc
> and mplayer. So, I assume that the subtitles data are correct.
>
> To make a test, I try to burn the subtitles into the video using the
> following
> command (full output at end of mail).
>
> ffmpeg -i f.mkv -filter_complex "[0:v][0:s]overlay" -codec:v mpeg2video
> -codec:a mp2 -ac 2 -y f.hardsubs.mpg

This can not work as overlay filter forks with video frames only.

>
> It seems the simplest possible command to do. Please don't comment on video
> quality, it is just a test to see if the subtitles are burnt into the
> video.
> The resulting video does not contain subtitles.
>
> I tried explicit x,y placement in the overlay and other options without more
> luck.
>
> Any idea? So many people report successful DVD subtitles processing using
> that type of command. What could be wrong?

Perhaps you should start reading documentation.

From nicolas.george at normalesup.org  Thu Aug 29 19:33:41 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Thu, 29 Aug 2013 19:33:41 +0200
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
	filter not working
In-Reply-To: <1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
References: <419479807.424442289.1377792999180.JavaMail.root@spooler3-g27.priv.proxad.net>
	<1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
Message-ID: <20130829173341.GA32063@phare.normalesup.org>

Le duodi 12 fructidor, an CCXXI, Thierry Lelegard a ?crit?:
> It seems the simplest possible command to do. Please don't comment on video
> quality, it is just a test to see if the subtitles are burnt into the video.
> The resulting video does not contain subtitles.

It should be working. I suggest three steps to understand how it fails:

First, use -loglevel debug and look at the "palette:" lines: maybe the
subtitles are completely transparent.

Second, add a showinfo filter after the subtitles input, something like
that:

  [0:s]ashowinfo[s];[0:v][s]overlay

That way, you can see if subtitles packets arrive.

Third, extract the subtitles track with mkvextract, and look at the index,
especially the size, the palette, etc.

Please ignore Paul's comment, he is mistaken.

Regards,

-- 
  Nicolas George
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From onemda at gmail.com  Thu Aug 29 19:39:23 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 17:39:23 +0000
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <20130829173341.GA32063@phare.normalesup.org>
References: <419479807.424442289.1377792999180.JavaMail.root@spooler3-g27.priv.proxad.net>
	<1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
	<20130829173341.GA32063@phare.normalesup.org>
Message-ID: <CAPYw7P6-giQDpOFtexDatQCvD83CW6k4B_HV3GSLpNH0BcpKAA@mail.gmail.com>

On 8/29/13, Nicolas George <nicolas.george at normalesup.org> wrote:
> Le duodi 12 fructidor, an CCXXI, Thierry Lelegard a ecrit :
>> It seems the simplest possible command to do. Please don't comment on
>> video
>> quality, it is just a test to see if the subtitles are burnt into the
>> video.
>> The resulting video does not contain subtitles.
>
> It should be working. I suggest three steps to understand how it fails:
>
> First, use -loglevel debug and look at the "palette:" lines: maybe the
> subtitles are completely transparent.
>
> Second, add a showinfo filter after the subtitles input, something like
> that:
>
>   [0:s]ashowinfo[s];[0:v][s]overlay
>
> That way, you can see if subtitles packets arrive.
>
> Third, extract the subtitles track with mkvextract, and look at the index,
> especially the size, the palette, etc.
>
> Please ignore Paul's comment, he is mistaken.

Its first time I heard that bitmap subtitles are supported.

From onemda at gmail.com  Thu Aug 29 19:54:31 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 17:54:31 +0000
Subject: [FFmpeg-user] uspp filter 444p
In-Reply-To: <loom.20130829T171030-297@post.gmane.org>
References: <nukmceguhrpioaposwzc@iwcm>
	<loom.20130829T171030-297@post.gmane.org>
Message-ID: <CAPYw7P6p8SF6HLt9+axQPXrH1f0gNSg0BwSrtDKBebdoLQkMBw@mail.gmail.com>

On 8/29/13, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> chronek <chronek <at> interia.eu> writes:
>
>> I want ask if someone from developers could check
>> uspp filter. This filter is port from mplayer and
>> it is uniqe, cause doing full compresion/
>> decompression based on snow codec. Snow codec
>> support pixel format 444p so changing uspp filter
>> to support that will not need many work.
>
> A patch will be welcome but since the filer was not
> ported yet, porting should be the first step imo.
> If you are interested in working on the port, commit
> a2c547f could act as an example.

Porting is far more complicated than
adding some feature to already existing filter.

>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From onemda at gmail.com  Thu Aug 29 19:56:32 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 17:56:32 +0000
Subject: [FFmpeg-user] uspp filter 444p
In-Reply-To: <nukmceguhrpioaposwzc@iwcm>
References: <nukmceguhrpioaposwzc@iwcm>
Message-ID: <CAPYw7P6GcSKjjJ+NukL++uBSa6809GeNNXyBGu1RTa1yyiPrWw@mail.gmail.com>

On 8/26/13, chronek <chronek at interia.eu> wrote:
> Hello,&nbsp;I want ask if someone from developers could check uspp filter.
> This filter is port from mplayer and it is uniqe, cause doing full
> compresion/decompression based on snow codec. Snow codec support pixel
> format 444p so changing uspp filter to support that will not need many
> work.&nbsp;I asking cause it is my main filter and any more quality is
> appreciated (btw i am using uspp not as postprocesor filter , but avaraging
> filter with fixed qp, now is: source -&gt; scale (444p) -&gt; uspp (420p)
> -&gt; unsharp (444p) -&gt; scale (444p) -&gt; lossless (rgb24) and uspp is
> bottleneck)Mike

I think you should report this as feature request on bug tracker.

From dzungng89 at gmail.com  Thu Aug 29 22:17:30 2013
From: dzungng89 at gmail.com (Dzung Nguyen)
Date: Thu, 29 Aug 2013 15:17:30 -0500
Subject: [FFmpeg-user] -Q:v 0
Message-ID: <CABOWChzG=vhzzM9e0rrFE3eMiFzY4+Q6_f0PfAMDYP-ds1iBTQ@mail.gmail.com>

I'm generating video from still images, and to keep the quality I use -q:v
0 for every intermediate processing steps. But the problem is that the
intermediate files are very large (around 1G), and it sometimes has the
buffer overflow problem. What are my alternative options?

From onemda at gmail.com  Thu Aug 29 22:53:53 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 20:53:53 +0000
Subject: [FFmpeg-user] -Q:v 0
In-Reply-To: <CABOWChzG=vhzzM9e0rrFE3eMiFzY4+Q6_f0PfAMDYP-ds1iBTQ@mail.gmail.com>
References: <CABOWChzG=vhzzM9e0rrFE3eMiFzY4+Q6_f0PfAMDYP-ds1iBTQ@mail.gmail.com>
Message-ID: <CAPYw7P5_P8w8R9Q_RhK5mfToky=drOZncCi=kv=NcvbWDbLNdA@mail.gmail.com>

On 8/29/13, Dzung Nguyen <dzungng89 at gmail.com> wrote:
> I'm generating video from still images, and to keep the quality I use -q:v
> 0 for every intermediate processing steps. But the problem is that the
> intermediate files are very large (around 1G), and it sometimes has the
> buffer overflow problem. What are my alternative options?

If you have enough fast CPU use lossless codecs for intermediate steps
and not uncompressed raw formats.

From thierry.lelegard at free.fr  Fri Aug 30 00:49:10 2013
From: thierry.lelegard at free.fr (=?ISO-8859-1?Q?Thierry_Lel=E9gard?=)
Date: Fri, 30 Aug 2013 00:49:10 +0200
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <20130829173341.GA32063@phare.normalesup.org>
References: <419479807.424442289.1377792999180.JavaMail.root@spooler3-g27.priv.proxad.net>
	<1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
	<20130829173341.GA32063@phare.normalesup.org>
Message-ID: <521FCFE6.7020208@free.fr>

Thanks for the troubleshooting steps.

I tried all three steps and they all seem to indicate that the subtitles
do exist and are not transparent. See below.

As I mentioned in my original post, both vlc and mplayer correctly display
the subtitles which are white with outline.

If you are interested, the video file (20 seconds, 15 MB) is accessible at
http://dl.free.fr/kHF1KGkZ7

Le 29/08/2013 19:33, Nicolas George a ?crit :

> Le duodi 12 fructidor, an CCXXI, Thierry Lelegard a ?crit :
>> It seems the simplest possible command to do. Please don't comment on video
>> quality, it is just a test to see if the subtitles are burnt into the video.
>> The resulting video does not contain subtitles.
> It should be working. I suggest three steps to understand how it fails:
>
> First, use -loglevel debug and look at the "palette:" lines: maybe the
> subtitles are completely transparent.

Got the following. See the full output at the end of the mail.

[dvdsub @ 00000000021b74e0] palette: 0x000000 0xf0f0f0 0xcccccc 0x999999 0x3333fa 0x1111bb 0xfa3333 0xbb1111 0x33fa33 0x11bb11 0xfafa33 0xbbbb11 0xfa33fa 0xbb11bb 0x33fafa 0x11bbbb

The subtitles are not transparent.

> Second, add a showinfo filter after the subtitles input, something like
> that:
>
>    [0:s]ashowinfo[s];[0:v][s]overlay
>
> That way, you can see if subtitles packets arrive.

I guess you mean "showinfo", not "ashowinfo", this is not audio.
Got the following. Again, see the full output at the end of the mail.

[Parsed_showinfo_0 @ 0000000000351160] n:0 pts:0 pts_time:0 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:1 pts:587 pts_time:0.587 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:2 pts:926 pts_time:0.926 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
....

So, the packets are seen by the filter chain. I am not sure how to interpret
some of those values.

> Third, extract the subtitles track with mkvextract, and look at the index,
> especially the size, the palette, etc.

$ mkvextract tracks gg.mkv 2:gg.sub
Extracting track 2 with the CodecID 'S_VOBSUB' to the file 'gg.sub'. Container format: VobSubs
Writing the VobSub index file 'gg.idx'.
Progress: 100%

The content of the index is:

# VobSub index file, v7 (do not modify this line!)
size: 1920x1080
org: 0, 0
scale: 100%, 100%
alpha: 100%
smooth: OFF
fadein/out: 0, 0
align: OFF at LEFT TOP
time offset: 0
forced subs: OFF
palette: 000000, f0f0f0, cccccc, 999999, 3333fa, 1111bb, fa3333, bb1111, 33fa33, 11bb11, fafa33, bbbb11, fa33fa, bb11bb, 33fafa, 11bbbb
custom colors: OFF, tridx: 1000, colors: 000000, 444444, 888888, cccccc

id: en, index: 0
timestamp: 00:00:00:000, filepos: 000000000
timestamp: 00:00:05:720, filepos: 000002800
timestamp: 00:00:12:200, filepos: 000004800
timestamp: 00:00:18:400, filepos: 000006800

Again, there is something to see in the subtitles.

> Please ignore Paul's comment, he is mistaken.

No problem ;) Just try Google with "ffmpeg overlay burn dvd subtitles" and you
will see many posts which indicate (after some hard tries) that the overlay
filter is successfully used to burn DVD subtitles. Concerning the manual, I
simply mean the ffmpeg manual about the overlay filter. But I agree that the
manual does not mention DVD subtitles specifically in the overlay section.

> Regards,

Full commands output:
--------------------------------------------------------------------------------------------

$ ffmpeg -loglevel debug -nostats -i gg.mkv -filter_complex "[0:v][0:s]overlay" -codec:v mpeg2video -codec:a mp2 -ac 2 -y gg.hardsubs.mpg
ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
   built on Mar 28 2013 00:34:08 with gcc 4.8.0 (GCC)
   configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
   libavutil      52. 18.100 / 52. 18.100
   libavcodec     54. 92.100 / 54. 92.100
   libavformat    54. 63.104 / 54. 63.104
   libavdevice    54.  3.103 / 54.  3.103
   libavfilter     3. 42.103 /  3. 42.103
   libswscale      2.  2.100 /  2.  2.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  2.100 / 52.  2.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set libav* logging level) with argument 'debug'.
Reading option '-nostats' ... matched as option 'stats' (print progress report during encoding) with argument 0.
Reading option '-i' ... matched as input file with argument 'gg.mkv'.
Reading option '-filter_complex' ... matched as option 'filter_complex' (create a complex filtergraph) with argument '[0:v][0:s]overlay'.
Reading option '-codec:v' ... matched as option 'codec' (codec name) with argument 'mpeg2video'.
Reading option '-codec:a' ... matched as option 'codec' (codec name) with argument 'mp2'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option 'gg.hardsubs.mpg' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set libav* logging level) with argument debug.
Applying option nostats (print progress report during encoding) with argument 0.
Applying option filter_complex (create a complex filtergraph) with argument [0:v][0:s]overlay.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file gg.mkv.
Successfully parsed a group of options.
Opening an input file: gg.mkv.
[matroska,webm @ 000000000222b900] Format matroska,webm probed with size=2048 and score=100
st:0 removing common factor 1000000 from timebase
st:1 removing common factor 1000000 from timebase
st:2 removing common factor 1000000 from timebase
[matroska,webm @ 000000000222b900] File position before avformat_find_stream_info() is 1094
[dvdsub @ 00000000021b74e0] palette: 0x000000 0xf0f0f0 0xcccccc 0x999999 0x3333fa 0x1111bb 0xfa3333 0xbb1111 0x33fa33 0x11bb11 0xfafa33 0xbbbb11 0xfa33fa 0xbb11bb 0x33fafa 0x11bbbb
[h264 @ 00000000021bae60] no picture
[dca @ 0000000002227fe0] Stream with high frequencies VQ coding
[matroska,webm @ 000000000222b900] All info found
[matroska,webm @ 000000000222b900] File position after avformat_find_stream_info() is 85246
Input #0, matroska,webm, from 'gg.mkv':
   Metadata:
     ENCODER         : Lavf55.14.102
   Duration: 00:00:23.00, start: 0.000000, bitrate: 5539 kb/s
     Chapter #0.0: start 0.000000, end 20.000000
     Stream #0:0, 3, 1/1000: Video: h264 (High), yuv420p, 1280x536 [SAR 1:1 DAR 160:67], 104271/5000000, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
     Stream #0:1, 1, 1/1000: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s (default)
     Stream #0:2, 1, 1/1000: Subtitle: dvd_subtitle (default)
Successfully opened the file.
Parsing a group of options: output file gg.hardsubs.mpg.
Applying option codec:v (codec name) with argument mpeg2video.
Applying option codec:a (codec name) with argument mp2.
Applying option ac (set number of audio channels) with argument 2.
Successfully parsed a group of options.
Opening an output file: gg.hardsubs.mpg.
[buffer @ 00000000021b8240] Setting entry with key 'video_size' to value '1280x536'
[buffer @ 00000000021b8240] Setting entry with key 'pix_fmt' to value '0'
[buffer @ 00000000021b8240] Setting entry with key 'time_base' to value '1/1000'
[buffer @ 00000000021b8240] Setting entry with key 'pixel_aspect' to value '1/1'
[buffer @ 00000000021b8240] Setting entry with key 'sws_param' to value 'flags=2'
[buffer @ 00000000021b8240] Setting entry with key 'frame_rate' to value '13978/583'
[graph 0 input from stream 0:0 @ 00000000022412a0] w:1280 h:536 pixfmt:yuv420p tb:1/1000 fr:13978/583 sar:1/1 sws_param:flags=2
[buffer @ 00000000021b7940] Setting entry with key 'video_size' to value '1920x1080'
[buffer @ 00000000021b7940] Setting entry with key 'pix_fmt' to value '30'
[buffer @ 00000000021b7940] Setting entry with key 'time_base' to value '1/1000'
[buffer @ 00000000021b7940] Setting entry with key 'pixel_aspect' to value '0/1'
[buffer @ 00000000021b7940] Setting entry with key 'sws_param' to value 'flags=2'
[graph 0 input from stream 0:2 @ 0000000002240e20] w:1920 h:1080 pixfmt:bgra tb:1/1000 fr:0/1 sar:0/1 sws_param:flags=2
Successfully opened the file.
[scale @ 0000000002229b20] Setting 'w' to value '0'
[scale @ 0000000002229b20] Setting 'h' to value '0'
[auto-inserted scaler 0 @ 0000000002240d80] w:0 h:0 flags:'' interl:0
[Parsed_overlay_0 @ 0000000002241200] auto-inserting filter 'auto-inserted scaler 0' between the filter 'graph 0 input from stream 0:2' and the filter 'Parsed_overlay_0'
[auto-inserted scaler 0 @ 0000000002240d80] w:1920 h:1080 fmt:bgra sar:0/1 -> w:1920 h:1080 fmt:yuva420p sar:0/1 flags:0x2
[Parsed_overlay_0 @ 0000000002241200] main w:1280 h:536 fmt:yuv420p overlay x:0 y:0 w:1920 h:1080 fmt:yuva420p
[Parsed_overlay_0 @ 0000000002241200] Overlay area with coordinates x1:0 y1:0 x2:1920 y2:1080 is not completely contained within the output with size 1280x536
[abuffer @ 00000000021bd4c0] Setting entry with key 'time_base' to value '1/48000'
[abuffer @ 00000000021bd4c0] Setting entry with key 'sample_rate' to value '48000'
[abuffer @ 00000000021bd4c0] Setting entry with key 'sample_fmt' to value 'fltp'
[abuffer @ 00000000021bd4c0] Setting entry with key 'channel_layout' to value '0x60f'
[graph 1 input from stream 0:1 @ 0000000002240640] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x60f
[aformat @ 00000000021bf040] Setting entry with key 'sample_fmts' to value 's16'
[aformat @ 00000000021bf040] Setting entry with key 'sample_rates' to value '44100,48000,32000,22050,24000,16000'
[aformat @ 00000000021bf040] Setting entry with key 'channel_layouts' to value '0x3'
[audio format for output stream 0:1 @ 0000000002240520] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
0.414214 0.000000 0.292893 0.000000 0.292893 0.000000
0.000000 0.414214 0.292893 0.000000 0.000000 0.292893
[auto-inserted resampler 0 @ 0000000002240400] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
[mpeg2video @ 00000000049d02e0] detected 8 logical cores
[mpeg2video @ 00000000049d02e0] intra_quant_bias = 96 inter_quant_bias = 0
[h264 @ 00000000021bae60] detected 8 logical cores
[dvdsub @ 00000000021b74e0] palette: 0x000000 0xf0f0f0 0xcccccc 0x999999 0x3333fa 0x1111bb 0xfa3333 0xbb1111 0x33fa33 0x11bb11 0xfafa33 0xbbbb11 0xfa33fa 0xbb11bb 0x33fafa 0x11bbbb
[mpeg @ 00000000021b7a00] VBV buffer size not set, muxing may fail
Output #0, mpeg, to 'gg.hardsubs.mpg':
   Metadata:
     encoder         : Lavf54.63.104
     Chapter #0.0: start 0.000000, end 20.000000
     Stream #0:0, 0, 1/90000: Video: mpeg2video, yuv420p, 1280x536 [SAR 1:1 DAR 160:67], 1001/24000, q=2-31, 200 kb/s, 90k tbn, 23.98 tbc
     Stream #0:1, 0, 1/90000: Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s (default)
Stream mapping:
   Stream #0:0 (h264) -> overlay:main (graph 0)
   Stream #0:2 (dvdsub) -> overlay:overlay (graph 0)
   overlay (graph 0) -> Stream #0:0 (mpeg2video)
   Stream #0:1 -> #0:1 (dca -> mp2)
Press [q] to stop, [?] for help
[h264 @ 00000000049614a0] no picture
[h264 @ 0000000004a78c00] no picture
[dca @ 0000000002227fe0] Stream with high frequencies VQ coding
[Parsed_overlay_0 @ 0000000002241200] Copying data in avfilter (have perms 5, need 7, reject 0)
*** 13 dup!
[mpeg @ 00000000021b7a00] First SCR: 0 First DTS: 45000
[Parsed_overlay_0 @ 0000000002241200] Copying data in avfilter (have perms 5, need 7, reject 0)
     Last message repeated 466 times
*** 1 dup!
     Last message repeated 1 times
[output stream 0:1 @ 0000000002240480] EOF on sink link output stream 0:1:default.
[output stream 0:0 @ 0000000002240be0] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
frame=  483 fps=134 q=31.0 Lsize=    2198kB time=00:00:20.10 bitrate= 895.7kbits/s dup=15 drop=0
video:1875kB audio:303kB subtitle:0 global headers:0kB muxing overhead 0.885072%
[AVIOContext @ 00000000021b92c0] Statistics: 15923228 bytes read, 0 seeks

--------------------------------------------------------------------------------------------

$ ffmpeg -i gg.mkv -filter_complex "[0:s]showinfo[s];[0:v][s]overlay" -codec:v mpeg2video -codec:a mp2 -ac 2 -y gg.hardsubs.mpg
  ffmpeg -i gg.mkv -filter_complex "[0:s]showinfo[s];[0:v][s]overlay" -codec:v mpeg2video -codec:a mp2 -ac 2 -y gg.hardsubs.mpg
ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
   built on Mar 28 2013 00:34:08 with gcc 4.8.0 (GCC)
   configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
   libavutil      52. 18.100 / 52. 18.100
   libavcodec     54. 92.100 / 54. 92.100
   libavformat    54. 63.104 / 54. 63.104
   libavdevice    54.  3.103 / 54.  3.103
   libavfilter     3. 42.103 /  3. 42.103
   libswscale      2.  2.100 /  2.  2.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  2.100 / 52.  2.100
Input #0, matroska,webm, from 'gg.mkv':
   Metadata:
     ENCODER         : Lavf55.14.102
   Duration: 00:00:23.00, start: 0.000000, bitrate: 5539 kb/s
     Chapter #0.0: start 0.000000, end 20.000000
     Stream #0:0: Video: h264 (High), yuv420p, 1280x536 [SAR 1:1 DAR 160:67], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
     Stream #0:1: Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s (default)
     Stream #0:2: Subtitle: dvd_subtitle (default)
[Parsed_overlay_1 @ 0000000000351200] Overlay area with coordinates x1:0 y1:0 x2:1920 y2:1080 is not completely contained within the output with size 1280x536
[mpeg @ 00000000049a00a0] VBV buffer size not set, muxing may fail
Output #0, mpeg, to 'gg.hardsubs.mpg':
   Metadata:
     encoder         : Lavf54.63.104
     Chapter #0.0: start 0.000000, end 20.000000
     Stream #0:0: Video: mpeg2video, yuv420p, 1280x536 [SAR 1:1 DAR 160:67], q=2-31, 200 kb/s, 90k tbn, 23.98 tbc
     Stream #0:1: Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s (default)
Stream mapping:
   Stream #0:0 (h264) -> overlay:main (graph 0)
   Stream #0:2 (dvdsub) -> showinfo (graph 0)
   overlay (graph 0) -> Stream #0:0 (mpeg2video)
   Stream #0:1 -> #0:1 (dca -> mp2)
Press [q] to stop, [?] for help
[Parsed_showinfo_0 @ 0000000000351160] n:0 pts:0 pts_time:0 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:1 pts:587 pts_time:0.587 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:2 pts:926 pts_time:0.926 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:3 pts:938 pts_time:0.938 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:4 pts:1300 pts_time:1.3 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:5 pts:1310 pts_time:1.31 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:6 pts:1673 pts_time:1.673 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:7 pts:1683 pts_time:1.683 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:8 pts:2015 pts_time:2.015 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:9 pts:2047 pts_time:2.047 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:10 pts:2388 pts_time:2.388 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:11 pts:2398 pts_time:2.398 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:12 pts:2761 pts_time:2.761 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:13 pts:2772 pts_time:2.772 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
n:14 pts:3134 pts_time:3.134 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:15 pts:3145 pts_time:3.145 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:16 pts:3508 pts_time:3.508 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:17 pts:3519 pts_time:3.519 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:18 pts:3849 pts_time:3.849 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:19 pts:3892 pts_time:3.892 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB plane_checksum:[504E28AB]
[Parsed_showinfo_0 @ 0000000000351160] n:20 pts:3993 pts_time:3.993 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:21 pts:4266 pts_time:4.266 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:22 pts:4351 pts_time:4.351 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:23 pts:4638 pts_time:4.638 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:24 pts:4724 pts_time:4.724 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:25 pts:5011 pts_time:5.011 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:26 pts:5097 pts_time:5.097 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:27 pts:5385 pts_time:5.385 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:28 pts:5470 pts_time:5.47 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
n:29 pts:5719 pts_time:5.719 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:30 pts:5720 pts_time:5.72 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:31 pts:6100 pts_time:6.1 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:32 pts:6111 pts_time:6.111 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:33 pts:6473 pts_time:6.473 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:34 pts:6484 pts_time:6.484 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:35 pts:6846 pts_time:6.846 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:36 pts:6858 pts_time:6.858 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:37 pts:7220 pts_time:7.22 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:38 pts:7230 pts_time:7.23 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:39 pts:7561 pts_time:7.561 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:40 pts:7594 pts_time:7.594 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:41 pts:7935 pts_time:7.935 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:42 pts:7945 pts_time:7.945 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
n:43 pts:8307 pts_time:8.307 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:44 pts:8318 pts_time:8.318 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:45 pts:8681 pts_time:8.681 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:46 pts:8692 pts_time:8.692 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:47 pts:9054 pts_time:9.054 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:48 pts:9065 pts_time:9.065 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:49 pts:9396 pts_time:9.396 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:50 pts:9439 pts_time:9.439 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:51 pts:9769 pts_time:9.769 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:52 pts:9812 pts_time:9.812 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:53 pts:10143 pts_time:10.143 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:54 pts:10186 pts_time:10.186 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:92935102 plane_checksum:[92935102]
[Parsed_showinfo_0 @ 0000000000351160] n:55 pts:10635 pts_time:10.635 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
n:56 pts:10722 pts_time:10.722 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:57 pts:10974 pts_time:10.974 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:58 pts:11060 pts_time:11.06 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:59 pts:11316 pts_time:11.316 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:60 pts:11433 pts_time:11.433 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:61 pts:11689 pts_time:11.689 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:62 pts:11807 pts_time:11.807 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:63 pts:12063 pts_time:12.063 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:64 pts:12148 pts_time:12.148 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:65 pts:12200 pts_time:12.2 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:66 pts:12521 pts_time:12.521 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:67 pts:12564 pts_time:12.564 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:68 pts:12894 pts_time:12.894 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:69 pts:12937 pts_time:12.937 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
n:70 pts:13268 pts_time:13.268 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:71 pts:13311 pts_time:13.311 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:72 pts:13610 pts_time:13.61 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:73 pts:13684 pts_time:13.684 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:74 pts:13983 pts_time:13.983 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:75 pts:14025 pts_time:14.025 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:76 pts:14356 pts_time:14.356 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:77 pts:14398 pts_time:14.398 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:78 pts:14729 pts_time:14.729 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:79 pts:14772 pts_time:14.772 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:80 pts:15102 pts_time:15.102 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:81 pts:15145 pts_time:15.145 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:82 pts:15443 pts_time:15.443 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:83 pts:15519 pts_time:15.519 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
n:84 pts:15817 pts_time:15.817 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:85 pts:15860 pts_time:15.86 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A1744133 plane_checksum:[A1744133]
[Parsed_showinfo_0 @ 0000000000351160] n:86 pts:16239 pts_time:16.239 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:87 pts:16244 pts_time:16.244 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:88 pts:16606 pts_time:16.606 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:89 pts:16618 pts_time:16.618 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:90 pts:16980 pts_time:16.98 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:91 pts:16990 pts_time:16.99 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:92 pts:17321 pts_time:17.321 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:93 pts:17363 pts_time:17.363 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:94 pts:17695 pts_time:17.695 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:95 pts:17737 pts_time:17.737 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
n:96 pts:18068 pts_time:18.068 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:97 pts:18110 pts_time:18.11 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:00000000 plane_checksum:[00000000]
[Parsed_showinfo_0 @ 0000000000351160] n:98 pts:18400 pts_time:18.4 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:99 pts:18483 pts_time:18.483 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:100 pts:18772 pts_time:18.772 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:101 pts:18857 pts_time:18.857 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:102 pts:19113 pts_time:19.113 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:103 pts:19241 pts_time:19.241 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:104 pts:19487 pts_time:19.487 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:105 pts:19614 pts_time:19.614 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:106 pts:19859 pts_time:19.859 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
[Parsed_showinfo_0 @ 0000000000351160] n:107 pts:19988 pts_time:19.988 pos:0 fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:A5A089F4 plane_checksum:[A5A089F4]
frame=  483 fps=121 q=31.0 Lsize=    2198kB time=00:00:20.10 bitrate= 895.7kbits/s dup=15 drop=0
video:1875kB audio:303kB subtitle:0 global headers:0kB muxing overhead 0.885072%



From onemda at gmail.com  Fri Aug 30 00:58:36 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 29 Aug 2013 22:58:36 +0000
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <521FCFE6.7020208@free.fr>
References: <419479807.424442289.1377792999180.JavaMail.root@spooler3-g27.priv.proxad.net>
	<1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
	<20130829173341.GA32063@phare.normalesup.org>
	<521FCFE6.7020208@free.fr>
Message-ID: <CAPYw7P6tga=bwTDdj4f_PMvKJZN8inX_5LYmrdtdJgWL7bDB6w@mail.gmail.com>

On 8/29/13, Thierry Lelegard <thierry.lelegard at free.fr> wrote:
> Thanks for the troubleshooting steps.
>
> I tried all three steps and they all seem to indicate that the subtitles
> do exist and are not transparent. See below.
>
> As I mentioned in my original post, both vlc and mplayer correctly display
> the subtitles which are white with outline.
>
> If you are interested, the video file (20 seconds, 15 MB) is accessible at
> http://dl.free.fr/kHF1KGkZ7
>
> Le 29/08/2013 19:33, Nicolas George a ecrit :
>
>> Le duodi 12 fructidor, an CCXXI, Thierry Lelegard a ecrit :
>>> It seems the simplest possible command to do. Please don't comment on
>>> video
>>> quality, it is just a test to see if the subtitles are burnt into the
>>> video.
>>> The resulting video does not contain subtitles.
>> It should be working. I suggest three steps to understand how it fails:
>>
>> First, use -loglevel debug and look at the "palette:" lines: maybe the
>> subtitles are completely transparent.
>
> Got the following. See the full output at the end of the mail.
>
> [dvdsub @ 00000000021b74e0] palette: 0x000000 0xf0f0f0 0xcccccc 0x999999
> 0x3333fa 0x1111bb 0xfa3333 0xbb1111 0x33fa33 0x11bb11 0xfafa33 0xbbbb11
> 0xfa33fa 0xbb11bb 0x33fafa 0x11bbbb
>
> The subtitles are not transparent.
>
>> Second, add a showinfo filter after the subtitles input, something like
>> that:
>>
>>    [0:s]ashowinfo[s];[0:v][s]overlay
>>
>> That way, you can see if subtitles packets arrive.
>
> I guess you mean "showinfo", not "ashowinfo", this is not audio.
> Got the following. Again, see the full output at the end of the mail.
>
> [Parsed_showinfo_0 @ 0000000000351160] n:0 pts:0 pts_time:0 pos:0 fmt:bgra
> sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB
> plane_checksum:[504E28AB]
> [Parsed_showinfo_0 @ 0000000000351160] n:1 pts:587 pts_time:0.587 pos:0
> fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB
> plane_checksum:[504E28AB]
> [Parsed_showinfo_0 @ 0000000000351160] n:2 pts:926 pts_time:0.926 pos:0
> fmt:bgra sar:0/0 s:1920x1080 i:P iskey:0 type:? checksum:504E28AB
> plane_checksum:[504E28AB]
> ....
>
> So, the packets are seen by the filter chain. I am not sure how to
> interpret
> some of those values.
>
>> Third, extract the subtitles track with mkvextract, and look at the
>> index,
>> especially the size, the palette, etc.
>
> $ mkvextract tracks gg.mkv 2:gg.sub
> Extracting track 2 with the CodecID 'S_VOBSUB' to the file 'gg.sub'.
> Container format: VobSubs
> Writing the VobSub index file 'gg.idx'.
> Progress: 100%
>
> The content of the index is:
>
> # VobSub index file, v7 (do not modify this line!)
> size: 1920x1080
> org: 0, 0
> scale: 100%, 100%
> alpha: 100%
> smooth: OFF
> fadein/out: 0, 0
> align: OFF at LEFT TOP
> time offset: 0
> forced subs: OFF
> palette: 000000, f0f0f0, cccccc, 999999, 3333fa, 1111bb, fa3333, bb1111,
> 33fa33, 11bb11, fafa33, bbbb11, fa33fa, bb11bb, 33fafa, 11bbbb
> custom colors: OFF, tridx: 1000, colors: 000000, 444444, 888888, cccccc
>

Well entries are really transparent.

[...]

From dashing.meng at gmail.com  Fri Aug 30 02:40:07 2013
From: dashing.meng at gmail.com (littlebat)
Date: Fri, 30 Aug 2013 08:40:07 +0800
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130829151904.67447577@shiva.selfip.org>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
	<20130828181654.1a880af9@shiva.selfip.org>
	<CAPYw7P7p8HJd4FEGD+MwPcYCGhWiYsgJwxSwBsy8=Ba8jtMy-w@mail.gmail.com>
	<20130828184401.58b4bb13@shiva.selfip.org>
	<20130829151904.67447577@shiva.selfip.org>
Message-ID: <20130830084007.ba873319.dashing.meng@gmail.com>

On Thu, 29 Aug 2013 15:19:04 +0530
"J. Bakshi (?. ???? )" <j.bakshi at unlimitedmail.org> wrote:

> On Wed, 28 Aug 2013 18:44:01 +0530
> "J. Bakshi (?. ???? )" <j.bakshi at unlimitedmail.org> wrote:
> 
> > On Wed, 28 Aug 2013 12:54:27 +0000
> > Paul B Mahol <onemda at gmail.com> wrote:
> > 
> > > On 8/28/13, J. Bakshi (***. ************ )
> > > <j.bakshi at unlimitedmail.org> wrote:
> > > > On Wed, 28 Aug 2013 16:30:00 +0400
> > > > Maxim Kozlov <rogat1y at gmail.com> wrote:
> > > >
> > > >> try
> > > >> ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264
> > > >> -movflags faststart -f mp4 out.mp4
> > > >>
> > > >>
> > > >
> > > > Thanks for your response,
> > > >
> > > > Tried with the parameters but get the following errors
> > > 
> > > Because you use extremly old ffmpeg version.
> > > 
> > 
> > Many many thanks.
> > 
> > I have upgrade from debian multimedia repo and it works :-)

Debian multimedia repo version isn't the latest version, try to compile
ffmpeg from source, if you use debian, read wiki manual:
https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide

> The same issue :-(
> 
> I have successfully ran the command
> 
> ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags
> faststart -f mp4 out.mp4
> 
> and after completion when I try to play mp4 I get
> 
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f5dcc000920] moov atom not found0/0   
> out.mp4: Invalid data found when processing input
> 
> Why again the error ?

Is original AVSEQ01.DAT played by ffplay ok? You should always post
uncutted command line and console output when you need get help in
list. And, you can also try to upload your AVSEQ01.DAT onto
youtube.com, to see if it can process your video right, if it is ok,
there must be something wrong of your converting.

From j.bakshi at unlimitedmail.org  Fri Aug 30 07:10:57 2013
From: j.bakshi at unlimitedmail.org (J. Bakshi (=?UTF-8?B?4KacLiDgpqzgppXgprjgp4A=?= ))
Date: Fri, 30 Aug 2013 10:40:57 +0530
Subject: [FFmpeg-user] ffmpeg conversion for youtube & vimeo
In-Reply-To: <20130830084007.ba873319.dashing.meng@gmail.com>
References: <20130828174908.6fed1cf5@shiva.selfip.org>
	<CAB-_x2MH68UvhLuRnMCtwd9T6wp0T5d=7_q_5mUeJ-ioykZT_Q@mail.gmail.com>
	<20130828181654.1a880af9@shiva.selfip.org>
	<CAPYw7P7p8HJd4FEGD+MwPcYCGhWiYsgJwxSwBsy8=Ba8jtMy-w@mail.gmail.com>
	<20130828184401.58b4bb13@shiva.selfip.org>
	<20130829151904.67447577@shiva.selfip.org>
	<20130830084007.ba873319.dashing.meng@gmail.com>
Message-ID: <20130830104057.618e2d61@shiva.selfip.org>

On Fri, 30 Aug 2013 08:40:07 +0800
littlebat <dashing.meng at gmail.com> wrote:


> > > 
> > > I have upgrade from debian multimedia repo and it works :-)
> 
> Debian multimedia repo version isn't the latest version, try to compile
> ffmpeg from source, if you use debian, read wiki manual:
> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
> 
> > The same issue :-(
> > 
> > I have successfully ran the command
> > 
> > ffmpeg -i AVSEQ01.DAT -acodec libfaac -vcodec libx264 -movflags
> > faststart -f mp4 out.mp4
> > 
> > and after completion when I try to play mp4 I get
> > 
> > [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f5dcc000920] moov atom not found0/0   
> > out.mp4: Invalid data found when processing input
> > 
> > Why again the error ?
> 
> Is original AVSEQ01.DAT played by ffplay ok? You should always post
> uncutted command line and console output when you need get help in
> list. And, you can also try to upload your AVSEQ01.DAT onto
> youtube.com, to see if it can process your video right, if it is ok,
> there must be something wrong of your converting.


Yes, the original DAT is working fine with ffmpeg.
I'll try your advise to upload the dat to utube.

Thanks

From gunnar.schulze at trinckle.com  Thu Aug 29 20:40:14 2013
From: gunnar.schulze at trinckle.com (Gunnar Schulze)
Date: Thu, 29 Aug 2013 20:40:14 +0200
Subject: [FFmpeg-user] FFmpeg mp4 decoding does not work for all systems
Message-ID: <521F958E.6040908@trinckle.com>

Dear FFmpeg developer team and experts,

I have a problem and I need your help:

I use FFmpeg to convert a set of images to video files of different size 
and format (for details see below). Unfortunately, FFmpeg produces mp4 
decoded files that seem to differ from mp4 video files from other 
sources (w3schools tutorial videos): on some plattforms (an older 
Android mobile phone and a Win7 PC), the FFmpeg-encoded videos are not 
shown in the browser.

I suppose the error is within my usage of the FFmpeg command line 
arguments. Maybe I have to add an audio codec although I do not need 
sound for the videos.

I collected all details of the problem like the exact command line and 
the video results on my website:

http://www.trinckle.com/videotest.php

I would be very thankful for a helpful answer to my problem.

Sincerly,
Gunnar Schulze

From chronek at interia.eu  Fri Aug 30 08:22:47 2013
From: chronek at interia.eu (Mike)
Date: Fri, 30 Aug 2013 08:22:47 +0200
Subject: [FFmpeg-user] uspp filter 444p
In-Reply-To: <CAPYw7P6GcSKjjJ+NukL++uBSa6809GeNNXyBGu1RTa1yyiPrWw@mail.gmail.com>
References: <nukmceguhrpioaposwzc@iwcm>
	<CAPYw7P6GcSKjjJ+NukL++uBSa6809GeNNXyBGu1RTa1yyiPrWw@mail.gmail.com>
Message-ID: <006401cea549$59052c70$0b0f8550$@eu>

Thanks,

 I will report this as feature request on bug tracker,

Mike

-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org]
On Behalf Of Paul B Mahol
Sent: Thursday, August 29, 2013 7:57 PM
To: FFmpeg user questions
Subject: Re: [FFmpeg-user] uspp filter 444p

On 8/26/13, chronek <chronek at interia.eu> wrote:
> Hello,&nbsp;I want ask if someone from developers could check uspp filter.
> This filter is port from mplayer and it is uniqe, cause doing full
> compresion/decompression based on snow codec. Snow codec support pixel
> format 444p so changing uspp filter to support that will not need many
> work.&nbsp;I asking cause it is my main filter and any more quality is
> appreciated (btw i am using uspp not as postprocesor filter , but
avaraging
> filter with fixed qp, now is: source -&gt; scale (444p) -&gt; uspp (420p)
> -&gt; unsharp (444p) -&gt; scale (444p) -&gt; lossless (rgb24) and uspp is
> bottleneck)Mike

I think you should report this as feature request on bug tracker.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From thierry at lelegard.fr  Fri Aug 30 09:38:58 2013
From: thierry at lelegard.fr (Thierry Lelegard)
Date: Fri, 30 Aug 2013 09:38:58 +0200 (CEST)
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <CAPYw7P6tga=bwTDdj4f_PMvKJZN8inX_5LYmrdtdJgWL7bDB6w@mail.gmail.com>
Message-ID: <272801500.425820420.1377848338905.JavaMail.root@spooler3-g27.priv.proxad.net>

> De: "Paul B Mahol" <onemda at gmail.com>

[...]

>>> First, use -loglevel debug and look at the "palette:" lines: maybe
>>> the
>>> subtitles are completely transparent.
>>
>> Got the following. See the full output at the end of the mail.
>>
>> [dvdsub @ 00000000021b74e0] palette: 0x000000 0xf0f0f0 0xcccccc 0x999999
>> 0x3333fa 0x1111bb 0xfa3333 0xbb1111 0x33fa33 0x11bb11 0xfafa33 0xbbbb11
>> 0xfa33fa 0xbb11bb 0x33fafa 0x11bbbb

[...]

>> # VobSub index file, v7 (do not modify this line!)
>> size: 1920x1080
>> org: 0, 0
>> scale: 100%, 100%
>> alpha: 100%
>> smooth: OFF
>> fadein/out: 0, 0
>> align: OFF at LEFT TOP
>> time offset: 0
>> forced subs: OFF
>> palette: 000000, f0f0f0, cccccc, 999999, 3333fa, 1111bb, fa3333, bb1111,
>> 33fa33, 11bb11, fafa33, bbbb11, fa33fa, bb11bb, 33fafa, 11bbbb
>> custom colors: OFF, tridx: 1000, colors: 000000, 444444, 888888, cccccc
>>
> 
> Well entries are really transparent.

Could you please elaborate on that? What does make you think this in the
above report? I am not sure on how to interpret the reported data.

If the subtitles are "transparent", why do vlc and mplayer display the
characters in white color with a black outline?

I also tried ffplay right now (same code base as ffmpeg I suppose).
It does display subtitles in white + black outline characters but
their size is messed up. They are much bigger than with vlc and
mplayer (roughly twice as big). Their position is also messed up
in a way which seems consistent with the wrong size (roughly shifted
left and up by twice the amount of pixels).

So, if ffplay displays "something" in the video (let's ignore the size
issue for now), which options or filters should be used with ffmpeg to
achieve the same result?

[...]

From blacktrash at gmx.net  Fri Aug 30 10:03:48 2013
From: blacktrash at gmx.net (Christian Ebert)
Date: Fri, 30 Aug 2013 09:03:48 +0100
Subject: [FFmpeg-user] FFmpeg mp4 decoding does not work for all systems
In-Reply-To: <521F958E.6040908@trinckle.com>
References: <521F958E.6040908@trinckle.com>
Message-ID: <20130830080348.GK60952@krille.blacktrash.org>

* Gunnar Schulze on Thursday, August 29, 2013 at 20:40:14 +0200
> I have a problem and I need your help:
> 
> I use FFmpeg to convert a set of images to video files of different
> size and format (for details see below). Unfortunately, FFmpeg
> produces mp4 decoded files that seem to differ from mp4 video files
> from other sources (w3schools tutorial videos): on some plattforms (an
> older Android mobile phone and a Win7 PC), the FFmpeg-encoded videos
> are not shown in the browser.
> 
> I suppose the error is within my usage of the FFmpeg command line
> arguments. Maybe I have to add an audio codec although I do not need
> sound for the videos.
> 
> I collected all details of the problem like the exact command line and
> the video results on my website:
> 
> http://www.trinckle.com/videotest.php

http://www.trinckle.com/videotest/small.mp4 is at h.264 high
profile.

You need at least -profile:v baseline and depending on the video
also -x264opts ref=1

-- 
  Was hei?t hier Dogma, ich bin Underdogma!
[ What the hell do you mean dogma, I am underdogma. ]
free movies       --->>> http://www.blacktrash.org/underdogma
http://itunes.apple.com/podcast/underdogma-movies/id363423596

From nicolas.george at normalesup.org  Fri Aug 30 10:56:43 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Fri, 30 Aug 2013 10:56:43 +0200
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <521FCFE6.7020208@free.fr>
References: <419479807.424442289.1377792999180.JavaMail.root@spooler3-g27.priv.proxad.net>
	<1644724330.424476069.1377793924863.JavaMail.root@spooler3-g27.priv.proxad.net>
	<20130829173341.GA32063@phare.normalesup.org>
	<521FCFE6.7020208@free.fr>
Message-ID: <20130830085643.GB5736@phare.normalesup.org>

Le tridi 13 fructidor, an CCXXI, Thierry Lel?gard a ?crit?:
> [graph 0 input from stream 0:0 @ 00000000022412a0] w:1280 h:536
> [graph 0 input from stream 0:2 @ 0000000002240e20] w:1920 h:1080

Ok, here is the problem: the subtitle frames are much larger than the video
frames, and the top-left corner you actually see is completely transparent.
There was a time the overlay filter would have produced an error if the
overlay frame overflows the underlying, but no longer, it is accepted.

You need to scale and/or crop the subtitles frames to get them to fit on the
video. I suspect, in this particular case, you should scale to 1280?720 and
then crop to 1280?536 to achieve the intended geometry.

Regards,

-- 
  Nicolas George
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From krueger at lesspain.de  Fri Aug 30 11:07:58 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 30 Aug 2013 11:07:58 +0200
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
Message-ID: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>

Hi,

I'm getting compile errors when building ffmpeg (current HEAD this morning) with

configure --disable-optimizations --disable-mmx --disable-stripping

I have been doing this for at least two months now, so either I am
missing something or something has changed rather recently. I have
tried a few cycles of make clean;make distclean;configure ..;make.

The errors I am getting are:

Undefined symbols for architecture x86_64:
  "_ff_add_hfyu_median_prediction_cmov", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_fdct_mmx", referenced from:
      _ff_dsputilenc_init_mmx in libavcodec.a(dsputilenc_mmx.o)
  "_ff_fdct_mmxext", referenced from:
      _ff_dsputilenc_init_mmx in libavcodec.a(dsputilenc_mmx.o)
  "_ff_fdct_sse2", referenced from:
      _ff_dsputilenc_init_mmx in libavcodec.a(dsputilenc_mmx.o)
  "_ff_idct_xvid_mmx", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_mmx_add", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_mmx_put", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_mmxext", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_mmxext_add", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_mmxext_put", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_sse2", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_sse2_add", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_idct_xvid_sse2_put", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_simple_idct_add_mmx", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_simple_idct_mmx", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
  "_ff_simple_idct_put_mmx", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
ld: symbol(s) not found for architecture x86_64
collect2: ld returned 1 exit status
make: *** [ffmpeg_g] Error 1

Does anyone have a spontaneous idea before I bisect?

BTW: Those rather crippling options are required to be able to debug
ffmpeg within my IDE.

From cehoyos at ag.or.at  Fri Aug 30 11:38:50 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 09:38:50 +0000 (UTC)
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
Message-ID: <loom.20130830T113813-270@post.gmane.org>

Robert Kr?ger <krueger <at> lesspain.de> writes:

> configure --disable-optimizations --disable-mmx --disable-stripping

> BTW: Those rather crippling options are required to be able to 
> debug ffmpeg within my IDE.

Hard to believe that --disable-stripping is needed to debug.

Carl Eugen


From sw.craftsman at gmail.com  Fri Aug 30 11:43:24 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Fri, 30 Aug 2013 15:13:24 +0530
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <CAPYw7P6jE4FcUSk3Kuj3=uDejuqi2qXk0t_4Ffzs4-xmThmuKg@mail.gmail.com>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<CAPYw7P44fxZ2iF9mXDwtFTyFTiFfxuG08qcsdjVyVO3O646k=g@mail.gmail.com>
	<CA+XaXCn5mQJT0Y_iKdvY9zUxmT9SinR3O3UANniW4VGX6_3TBw@mail.gmail.com>
	<CAPYw7P6P5X==NkgjYWtVc-AXjCnWBq5SLKUgS4CChn1T1K8E_w@mail.gmail.com>
	<CA+XaXCk8r4X6iW+ybdioLnDX0ydhR5havnz2VfjMnmg_S1Q4Fw@mail.gmail.com>
	<CAPYw7P4=hojo8hr7xA-NcEBDs-pObq+G-soOHiCSo-EeNa=FLA@mail.gmail.com>
	<CA+XaXCng40XppF0vwgJBAnNVVf5uWB4ZqzGseAb-Sy72kD8PkA@mail.gmail.com>
	<CAPYw7P6NjLAj-d9Jb7YRwzFvYseBVqK9jGk4yNEoKOu1C+_iHA@mail.gmail.com>
	<CA+XaXCmxxBbFLNztbuE0V9zYphuxq6h9D1R3-xdZZhSF9ZsikA@mail.gmail.com>
	<CAPYw7P6jE4FcUSk3Kuj3=uDejuqi2qXk0t_4Ffzs4-xmThmuKg@mail.gmail.com>
Message-ID: <CA+XaXCnpCgUjty0O=RQUamKR6dXOViSWOvK3FHXX4CMUuB7BEg@mail.gmail.com>

Am not sure why latest version shows 14.99 ...

So you're saying the correct frame-rate is 30 right ? And infact FFProbe's
own older version was also showing it as 30 only as I said previously.

So does this mean it's a regression ?

Any workaround ?

Thanks

From krueger at lesspain.de  Fri Aug 30 11:44:19 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 30 Aug 2013 11:44:19 +0200
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
In-Reply-To: <loom.20130830T113813-270@post.gmane.org>
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
	<loom.20130830T113813-270@post.gmane.org>
Message-ID: <CAEnqZEWAnanEEB-MLyKFLyKidtvFb5+SqHKoKaa1ob5bO_s+yg@mail.gmail.com>

On Fri, Aug 30, 2013 at 11:38 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> configure --disable-optimizations --disable-mmx --disable-stripping
>
>> BTW: Those rather crippling options are required to be able to
>> debug ffmpeg within my IDE.
>
> Hard to believe that --disable-stripping is needed to debug.
>

I will doublecheck and give feedback when I get it to compile again
but AFAIR without this option the debugger did not work properly.

From cehoyos at ag.or.at  Fri Aug 30 11:43:22 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 09:43:22 +0000 (UTC)
Subject: [FFmpeg-user] FFmpeg mp4 decoding does not work for all systems
References: <521F958E.6040908@trinckle.com>
Message-ID: <loom.20130830T114238-198@post.gmane.org>

Gunnar Schulze <gunnar.schulze <at> trinckle.com> writes:

> I collected all details of the problem like the exact 
> command line and the video results on my website:
> 
> http://www.trinckle.com/videotest.php

Please provide all necessary information here on the 
mailing list, external resources may disappear.

Carl Eugen


From cehoyos at ag.or.at  Fri Aug 30 11:45:32 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 09:45:32 +0000 (UTC)
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
	filter not working
References: <CAPYw7P6tga=bwTDdj4f_PMvKJZN8inX_5LYmrdtdJgWL7bDB6w@mail.gmail.com>
	<272801500.425820420.1377848338905.JavaMail.root@spooler3-g27.priv.proxad.net>
Message-ID: <loom.20130830T114444-669@post.gmane.org>

Thierry Lelegard <thierry <at> lelegard.fr> writes:

> I also tried ffplay right now (same code base as ffmpeg 
> I suppose). It does display subtitles in white + black 
> outline characters but their size is messed up.

This is ticket #679.

Carl Eugen


From onemda at gmail.com  Fri Aug 30 11:56:30 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 30 Aug 2013 09:56:30 +0000
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <272801500.425820420.1377848338905.JavaMail.root@spooler3-g27.priv.proxad.net>
References: <CAPYw7P6tga=bwTDdj4f_PMvKJZN8inX_5LYmrdtdJgWL7bDB6w@mail.gmail.com>
	<272801500.425820420.1377848338905.JavaMail.root@spooler3-g27.priv.proxad.net>
Message-ID: <CAPYw7P6HH3gbU0zHR0UkdFk37Q9+B=Nnw493MFonkSxx+ngp6g@mail.gmail.com>

On 8/30/13, Thierry Lelegard <thierry at lelegard.fr> wrote:
>> De: "Paul B Mahol" <onemda at gmail.com>
>
> [...]
>
>>>> First, use -loglevel debug and look at the "palette:" lines: maybe
>>>> the
>>>> subtitles are completely transparent.
>>>
>>> Got the following. See the full output at the end of the mail.
>>>
>>> [dvdsub @ 00000000021b74e0] palette: 0x000000 0xf0f0f0 0xcccccc 0x999999
>>> 0x3333fa 0x1111bb 0xfa3333 0xbb1111 0x33fa33 0x11bb11 0xfafa33 0xbbbb11
>>> 0xfa33fa 0xbb11bb 0x33fafa 0x11bbbb
>
> [...]
>
>>> # VobSub index file, v7 (do not modify this line!)
>>> size: 1920x1080
>>> org: 0, 0
>>> scale: 100%, 100%
>>> alpha: 100%
>>> smooth: OFF
>>> fadein/out: 0, 0
>>> align: OFF at LEFT TOP
>>> time offset: 0
>>> forced subs: OFF
>>> palette: 000000, f0f0f0, cccccc, 999999, 3333fa, 1111bb, fa3333, bb1111,
>>> 33fa33, 11bb11, fafa33, bbbb11, fa33fa, bb11bb, 33fafa, 11bbbb
>>> custom colors: OFF, tridx: 1000, colors: 000000, 444444, 888888, cccccc
>>>
>>
>> Well entries are really transparent.
>
> Could you please elaborate on that? What does make you think this in the
> above report? I am not sure on how to interpret the reported data.

Actually looking at source code those logs does not output alpha bytes at all.

>
> If the subtitles are "transparent", why do vlc and mplayer display the
> characters in white color with a black outline?
>
> I also tried ffplay right now (same code base as ffmpeg I suppose).
> It does display subtitles in white + black outline characters but
> their size is messed up. They are much bigger than with vlc and
> mplayer (roughly twice as big). Their position is also messed up
> in a way which seems consistent with the wrong size (roughly shifted
> left and up by twice the amount of pixels).
>
> So, if ffplay displays "something" in the video (let's ignore the size
> issue for now), which options or filters should be used with ffmpeg to
> achieve the same result?
>
> [...]
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From sw.craftsman at gmail.com  Fri Aug 30 12:13:54 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Fri, 30 Aug 2013 15:43:54 +0530
Subject: [FFmpeg-user] 'Duration: N/A' for some videos ?
In-Reply-To: <loom.20130829T144445-68@post.gmane.org>
References: <CA+XaXCmrc0bcLZAihwiiiY_3Z9sYBeSajRd-AmB-cSmmUBUeCA@mail.gmail.com>
	<loom.20130829T144445-68@post.gmane.org>
Message-ID: <CA+XaXCk61V_e0hJKPHx8TYetJ21yyQ85we2HKe8bXPk95chkJw@mail.gmail.com>

Hi Carl,

Bit-rate is understood ... I want to find out the 'duration' using FFProbe
like how it used to give before. The newer version also gives only the max
bit-rate but not the duration.

You mean to say now it won't be possible at all to find the duration using
FFProbe ??

Has this change been made intentionally in the newer version OR it's a
regression ?

Thanks.

From cehoyos at ag.or.at  Fri Aug 30 12:27:07 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 10:27:07 +0000 (UTC)
Subject: [FFmpeg-user] 'Duration: N/A' for some videos ?
References: <CA+XaXCmrc0bcLZAihwiiiY_3Z9sYBeSajRd-AmB-cSmmUBUeCA@mail.gmail.com>
	<loom.20130829T144445-68@post.gmane.org>
	<CA+XaXCk61V_e0hJKPHx8TYetJ21yyQ85we2HKe8bXPk95chkJw@mail.gmail.com>
Message-ID: <loom.20130830T122603-707@post.gmane.org>

Shyam Sundar <sw.craftsman <at> gmail.com> writes:

> Bit-rate is understood ... I want to find out the 
> 'duration' using FFProbe like how it used to give before.

It was wrong before.

> The newer version also gives only the max
> bit-rate but not the duration.

Because the duration is unknown:
Known are filesize and maximum duration, you have to 
decode to know the actual duration.

Carl Eugen


From ogiollae at tcd.ie  Fri Aug 30 12:36:25 2013
From: ogiollae at tcd.ie (Eoin)
Date: Fri, 30 Aug 2013 03:36:25 -0700 (PDT)
Subject: [FFmpeg-user] Sync Issue After Clipping MPEG-2 Long GoP and PCM
Message-ID: <1377858985191-4661002.post@n4.nabble.com>

Hi all,
I am working on an audio-visual speech recognition project at the moment,
and I have a bunch of ~4second MKV clips containing MPEG-2 Long GoP video
and mono PCM audio. The clips were obtained by clipping longer videos in the
same format using commands with the format: 
ffmpeg -i long_vid.mkv -vcodec copy -acodec copy -ss 00:xx:xx.xxxx -t
00:xx:xx.xxxx short_clip.mkv

After reading:
http://ffmpeg.org/pipermail/ffmpeg-user/2013-July/016291.html
and
http://ffmpeg.org/pipermail/ffmpeg-user/2013-August/016614.html , 
I now realize that my videos have probably been cut at the closest I-frames
to the times specified. This is actually not necessarily a bad thing - I
don't need frame-accurate clipping, as long as my audio and video are
in-sync, which they are when I play the clips back in VLC. 
The problem begins when I then extract the audio from the clips using: 
ffmpeg -i short_clip.mkv -vn -acodec copy audio.wav
This extracted audio seems to be out of sync with the clip it came from.
This is a problem, as the label files I obtain for speech in the audio won't
match up with the video frames I extract in Matlab. 

If I could figure out, for each clip, how much the extracted audio and video
will be out of sync, I could simply apply this offset to my label files. So,
I guess my question is: what exactly does ffmpeg do with my 1st command
above, when the input is MPEG-2 Long GoP video and PCM audio in an MKV
container? 



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Sync-Issue-After-Clipping-MPEG-2-Long-GoP-and-PCM-tp4661002.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From krueger at lesspain.de  Fri Aug 30 13:09:54 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 30 Aug 2013 13:09:54 +0200
Subject: [FFmpeg-user] 'Duration: N/A' for some videos ?
In-Reply-To: <loom.20130830T122603-707@post.gmane.org>
References: <CA+XaXCmrc0bcLZAihwiiiY_3Z9sYBeSajRd-AmB-cSmmUBUeCA@mail.gmail.com>
	<loom.20130829T144445-68@post.gmane.org>
	<CA+XaXCk61V_e0hJKPHx8TYetJ21yyQ85we2HKe8bXPk95chkJw@mail.gmail.com>
	<loom.20130830T122603-707@post.gmane.org>
Message-ID: <CAEnqZEXL5nNP-5C4ymfrCJsW-TD+imDk6+w6w=m+4chCddjgsQ@mail.gmail.com>

On Fri, Aug 30, 2013 at 12:27 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Shyam Sundar <sw.craftsman <at> gmail.com> writes:
>
>> Bit-rate is understood ... I want to find out the
>> 'duration' using FFProbe like how it used to give before.
>
> It was wrong before.
>
>> The newer version also gives only the max
>> bit-rate but not the duration.
>
> Because the duration is unknown:
> Known are filesize and maximum duration, you have to
> decode to know the actual duration.

parse the packets/headers, actually. It is possible to retrieve an
accurate duration with quite a bit of I/O but rather little CPU time
(i.e. in most practical cases a lot faster that by really decoding the
file). The thing is, I don't believe that option (parse the packets
and add their duration) is currently implemented in any of the
fftools. However, I have seen posts by people who have done that via
scripts that extract the packet durations using ffprobe and then add
those. I assume that for some formats tools like mediainfo or exiftool
take that approach but I have not verified that by looking at their
code.

@Carl: Of course you may interprete parsing headers as a subset of
decoding, I just wanted to make sure he did not get the impression he
had to decode (as in decoding image data) the whole video to get a
correct duration.

From cehoyos at ag.or.at  Fri Aug 30 13:12:31 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 11:12:31 +0000 (UTC)
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
Message-ID: <loom.20130830T131150-599@post.gmane.org>

Shyam Sundar <sw.craftsman <at> gmail.com> writes:

> E.g. for below video 
> (http://files.videohelp.com/u/211860/Video_LXF.lxf) 
> it used to be 29.97 fps earlier, it is 14.99 now.

I opened ticket #2917, thank you for the report!

Carl Eugen


From cehoyos at ag.or.at  Fri Aug 30 13:13:54 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 11:13:54 +0000 (UTC)
Subject: [FFmpeg-user] Sync Issue After Clipping MPEG-2 Long GoP and PCM
References: <1377858985191-4661002.post@n4.nabble.com>
Message-ID: <loom.20130830T131321-564@post.gmane.org>

Eoin <ogiollae <at> tcd.ie> writes:

> The problem begins when I then extract the audio 
> from the clips using: 
> ffmpeg -i short_clip.mkv -vn -acodec copy audio.wav
> This extracted audio seems to be out of sync with 
> the clip it came from.

How can a wav file be out-of-sync?

Carl Eugen


From cehoyos at ag.or.at  Fri Aug 30 13:18:51 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 11:18:51 +0000 (UTC)
Subject: [FFmpeg-user] 'Duration: N/A' for some videos ?
References: <CA+XaXCmrc0bcLZAihwiiiY_3Z9sYBeSajRd-AmB-cSmmUBUeCA@mail.gmail.com>
	<loom.20130829T144445-68@post.gmane.org>
	<CA+XaXCk61V_e0hJKPHx8TYetJ21yyQ85we2HKe8bXPk95chkJw@mail.gmail.com>
	<loom.20130830T122603-707@post.gmane.org>
	<CAEnqZEXL5nNP-5C4ymfrCJsW-TD+imDk6+w6w=m+4chCddjgsQ@mail.gmail.com>
Message-ID: <loom.20130830T131550-136@post.gmane.org>

Robert Kr?ger <krueger <at> lesspain.de> writes:

> I assume that for some formats tools like mediainfo 

[...]

> take that approach but I have not verified that by 
> looking at their code.

(Is this really true? Sounds unlikely to me.)

>  <at> Carl: Of course you may interprete parsing 
> headers as a subset of decoding, I just wanted to 
> make sure he did not get the impression he had to 
> decode (as in decoding image data) the whole 
> video to get a correct duration.

You are (most probably) right but I think answers 
should be (or at least are) based on (current) 
FFmpeg here, it is of course not wrong to mention 
what would be possible if FFmpeg would behave (in 
this case: very) differently.

Carl Eugen


From ogiollae at tcd.ie  Fri Aug 30 13:26:05 2013
From: ogiollae at tcd.ie (Eoin Padrai O'Giollain)
Date: Fri, 30 Aug 2013 12:26:05 +0100
Subject: [FFmpeg-user] Sync Issue After Clipping MPEG-2 Long GoP and PCM
In-Reply-To: <loom.20130830T131321-564@post.gmane.org>
References: <1377858985191-4661002.post@n4.nabble.com>
	<loom.20130830T131321-564@post.gmane.org>
Message-ID: <CADsUWhQu6wedvLNtDz=SjnUL-prDT9ji4Z=7rkXHCcNpCmQPAA@mail.gmail.com>

What I mean by out of sync is, when I looked at the extracted audio and the
video in Premiere, they were out of sync with one another. Note: I had to
convert the mkv into an mxf to open it in Premiere. The mxf was shorter
than the mkv and the audio clip, but when I snapped its end to the end of
the audio, the audio and video were back in sync.


On 30 August 2013 12:13, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Eoin <ogiollae <at> tcd.ie> writes:
>
> > The problem begins when I then extract the audio
> > from the clips using:
> > ffmpeg -i short_clip.mkv -vn -acodec copy audio.wav
> > This extracted audio seems to be out of sync with
> > the clip it came from.
>
> How can a wav file be out-of-sync?
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

From sw.craftsman at gmail.com  Fri Aug 30 13:36:56 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Fri, 30 Aug 2013 17:06:56 +0530
Subject: [FFmpeg-user] Incorrect frame-rate in FFProbe
In-Reply-To: <loom.20130830T131150-599@post.gmane.org>
References: <CA+XaXCk4YLnBg620cHDNrV0M6nMqsuLBJAcyDzqPexNdHLxMrQ@mail.gmail.com>
	<loom.20130830T131150-599@post.gmane.org>
Message-ID: <CA+XaXCma=-pbnnc3u=7RjX09HCm+9aR2ET6NF6fZDMKoOc8O0g@mail.gmail.com>

Thanks* Carl !!*

From thierry at lelegard.fr  Fri Aug 30 14:00:28 2013
From: thierry at lelegard.fr (Thierry Lelegard)
Date: Fri, 30 Aug 2013 14:00:28 +0200 (CEST)
Subject: [FFmpeg-user] Burning DVD subtitles into video using overlay
 filter not working
In-Reply-To: <20130830085643.GB5736@phare.normalesup.org>
Message-ID: <775412866.426484407.1377864028848.JavaMail.root@spooler3-g27.priv.proxad.net>

> De: "Nicolas George" <nicolas.george at normalesup.org>
> 
>> [graph 0 input from stream 0:0 @ 00000000022412a0] w:1280 h:536
>> [graph 0 input from stream 0:2 @ 0000000002240e20] w:1920 h:1080
> 
> Ok, here is the problem: the subtitle frames are much larger than the video
> frames, and the top-left corner you actually see is completely transparent.
> There was a time the overlay filter would have produced an error if the
> overlay frame overflows the underlying, but no longer, it is accepted.
> 
> You need to scale and/or crop the subtitles frames to get them to fit on the
> video. I suspect, in this particular case, you should scale to 1280?720 and
> then crop to 1280?536 to achieve the intended geometry.

Great, you got it Nicolas!

The following command does the job:

ffmpeg -i gg.mkv -filter_complex "[0:s]scale=width=1280:height=720,crop=w=1280:h=536:x=0:y=out_h[sub];[0:v][sub]overlay" -codec:v mpeg2video -codec:a mp2 -ac 2 -y gg.hardsubs.mpg

The subtitles are there and look nice. The width/height proportion of the
characters look fine.

I also tried without cropping, directly resizing to the video size, hence
without preserving the aspect ratio:

ffmpeg -i gg.mkv -filter_complex "[0:s]scale=width=1280:height=536[sub];[0:v][sub]overlay" -codec:v mpeg2video -codec:a mp2 -ac 2 -y gg.hardsubs.mpg

The subtitles are there but the characters are slightly vertically
compressed as expected. Interestingly, they now have exactly the same
appearance as displayed with vlc and mplayer. I suppose that both of
them simply resize the subtitle frames without preserving the aspect
ratio.

So far so good ... for this file. My original purpose is to automate
the processing of media files, making decisions based on the various
input streams. To get all characteristics of the input file, I use
ffprobe with "flat" print format. Based on these info, I generate
one or more ffmpeg commands to build the resulting media file.

Now, in the case of DVD subtitles, how can I compute the appropriate
filter? For this, I need the width and height of both the video and
dvdsub stream. However, ffprobe does not give me the size of the
dvdsub frames, only the video streams. See the example:

$ ffprobe gg.mkv -loglevel warning -show_streams -print_format flat | grep -e codec_name -e width -e height
streams.stream.0.codec_name="h264"
streams.stream.0.width=1280
streams.stream.0.height=536
streams.stream.1.codec_name="dca"
streams.stream.2.codec_name="dvdsub"

And if I look at all "stream.2" properties, I do not see anything usable:

streams.stream.2.index=2
streams.stream.2.codec_name="dvdsub"
streams.stream.2.codec_long_name="DVD subtitles"
streams.stream.2.profile="unknown"
streams.stream.2.codec_type="subtitle"
streams.stream.2.codec_time_base="1/1000"
streams.stream.2.codec_tag_string="[0][0][0][0]"
streams.stream.2.codec_tag="0x0000"
streams.stream.2.id="N/A"
streams.stream.2.r_frame_rate="0/0"
streams.stream.2.avg_frame_rate="0/0"
streams.stream.2.time_base="1/1000"
streams.stream.2.start_pts=0
streams.stream.2.start_time="0.000000"
streams.stream.2.duration_ts="N/A"
streams.stream.2.duration="N/A"
streams.stream.2.bit_rate="N/A"
streams.stream.2.nb_frames="N/A"
streams.stream.2.nb_read_frames="N/A"
streams.stream.2.nb_read_packets="N/A"
streams.stream.2.disposition.default=1
streams.stream.2.disposition.dub=0
streams.stream.2.disposition.original=0
streams.stream.2.disposition.comment=0
streams.stream.2.disposition.lyrics=0
streams.stream.2.disposition.karaoke=0
streams.stream.2.disposition.forced=0
streams.stream.2.disposition.hearing_impaired=0
streams.stream.2.disposition.visual_impaired=0
streams.stream.2.disposition.clean_effects=0
streams.stream.2.disposition.attached_pic=0

Maybe, as parameters of the scale and crop filters, I can use expressions
using the literal values of the video stream size. But I need to get more
familiar with the syntax of expressions in ffmpeg parameters.

Thanks a lot for your help,
-Thierry

PS:
> Le tridi 13 fructidor, an CCXXI, Thierry Lel?gard a ?crit :
Does anyone outside France understand the Revolutionary Calendar? :-)

From krueger at lesspain.de  Fri Aug 30 14:44:08 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 30 Aug 2013 14:44:08 +0200
Subject: [FFmpeg-user] 'Duration: N/A' for some videos ?
In-Reply-To: <loom.20130830T131550-136@post.gmane.org>
References: <CA+XaXCmrc0bcLZAihwiiiY_3Z9sYBeSajRd-AmB-cSmmUBUeCA@mail.gmail.com>
	<loom.20130829T144445-68@post.gmane.org>
	<CA+XaXCk61V_e0hJKPHx8TYetJ21yyQ85we2HKe8bXPk95chkJw@mail.gmail.com>
	<loom.20130830T122603-707@post.gmane.org>
	<CAEnqZEXL5nNP-5C4ymfrCJsW-TD+imDk6+w6w=m+4chCddjgsQ@mail.gmail.com>
	<loom.20130830T131550-136@post.gmane.org>
Message-ID: <CAEnqZEU9JNVLDjiwQBOdCS3gkQgy0nopKQN_8OZjupuQoEZPNA@mail.gmail.com>

On Fri, Aug 30, 2013 at 1:18 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> I assume that for some formats tools like mediainfo
>
> [...]
>
>> take that approach but I have not verified that by
>> looking at their code.
>
> (Is this really true? Sounds unlikely to me.)

I think so because the results looked good and e.g. for transport
stream files this is really fast but you cannot know for sure. I
certainly think it makes sense for tools which focus on metadata
extraction as it is the best thing that can be done but again, that's
just guessing from observations and what I think makes sense.

>
>>  <at> Carl: Of course you may interprete parsing
>> headers as a subset of decoding, I just wanted to
>> make sure he did not get the impression he had to
>> decode (as in decoding image data) the whole
>> video to get a correct duration.
>
> You are (most probably) right but I think answers
> should be (or at least are) based on (current)
> FFmpeg here, it is of course not wrong to mention
> what would be possible if FFmpeg would behave (in
> this case: very) differently.

That's why I mentioned the ffprobe scripting approach. IIRC there are
people who use this in production as it is currently the only way to
achieve it with FFmpeg tools (for formats that do not have the
duration in the container header).

From krueger at lesspain.de  Fri Aug 30 15:20:18 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 30 Aug 2013 15:20:18 +0200
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
In-Reply-To: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
Message-ID: <CAEnqZEUSwv7F=1y8-fcCB=M_2mEP+AxDnW96MntUNtZkU1hyrQ@mail.gmail.com>

On Fri, Aug 30, 2013 at 11:07 AM, Robert Kr?ger <krueger at lesspain.de> wrote:
> Hi,
>
> I'm getting compile errors when building ffmpeg (current HEAD this morning) with
>
> configure --disable-optimizations --disable-mmx --disable-stripping
>
> I have been doing this for at least two months now, so either I am
> missing something or something has changed rather recently. I have
> tried a few cycles of make clean;make distclean;configure ..;make.
>
> The errors I am getting are:
>
> Undefined symbols for architecture x86_64:
>   "_ff_add_hfyu_median_prediction_cmov", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_fdct_mmx", referenced from:
>       _ff_dsputilenc_init_mmx in libavcodec.a(dsputilenc_mmx.o)
>   "_ff_fdct_mmxext", referenced from:
>       _ff_dsputilenc_init_mmx in libavcodec.a(dsputilenc_mmx.o)
>   "_ff_fdct_sse2", referenced from:
>       _ff_dsputilenc_init_mmx in libavcodec.a(dsputilenc_mmx.o)
>   "_ff_idct_xvid_mmx", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_mmx_add", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_mmx_put", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_mmxext", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_mmxext_add", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_mmxext_put", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_sse2", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_sse2_add", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_idct_xvid_sse2_put", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_simple_idct_add_mmx", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_simple_idct_mmx", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
>   "_ff_simple_idct_put_mmx", referenced from:
>       _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
> ld: symbol(s) not found for architecture x86_64
> collect2: ld returned 1 exit status
> make: *** [ffmpeg_g] Error 1

it appears someone is working on this right now. A clean
clone/configure/make reduced this to:

LD ffmpeg_g
Undefined symbols for architecture x86_64:
  "_ff_add_hfyu_median_prediction_cmov", referenced from:
      _ff_dsputil_init_x86 in libavcodec.a(dsputil_init.o)
ld: symbol(s) not found for architecture x86_64
collect2: ld returned 1 exit status
make: *** [ffmpeg_g] Error 1

From jpboard2 at yahoo.com  Fri Aug 30 16:38:52 2013
From: jpboard2 at yahoo.com (James Board)
Date: Fri, 30 Aug 2013 07:38:52 -0700 (PDT)
Subject: [FFmpeg-user] Pixel Formats for libx264
Message-ID: <1377873532.71469.YahooMailNeo@web164705.mail.gq1.yahoo.com>

What pixel format, if any, does the x264 codecs convert video to before encoding?? If my source video is raw RGB (3 bytes per pixel), will the encoded video use the same pixel format?? If my source video is YUV422, will the encoded video use the same pixel format?

From pfischer at como.com  Fri Aug 30 16:45:57 2013
From: pfischer at como.com (=?utf-8?Q?Patrick_Fischer?=)
Date: Fri, 30 Aug 2013 16:45:57 +0200
Subject: [FFmpeg-user] create a mjpeg stream via ffserver with boundary
	strings
Message-ID: <zarafa.5220b025.0cba.7ff6f842171b81e8@mail.como.com>

Hello

?
I would like to create an mjpeg stream with ffserver.

?
Souce is an UDP Mpeg TS Stream

?
?
....

<Feed feed1.ffm> 
??????? # name of the file to hold the data
??????? File /tmp/feed1.ffm 
??????? # file size of the file
??????? FileMaxSize 100M
??????? Launch ffmpeg -i "udp://225.1.1.1:30120" -an -vcodec mjpeg -s 320x240 -y
</Feed> 

<Stream test.mjpeg>
??????? Feed feed1.ffm
??????? Format jpeg??????? 
??????? VideoSize 320x240
??????? VideoFrameRate 10
??????? VideoBitRate 2000
??????? VideoQMin 1
??????? VideoQMax 10
??????? NoAudio
</Stream>

# status html page
<Stream stat.html>
??????? Format status
??????? # Only allow local people to get the status
??????? ACL allow localhost
??????? ACL allow 192.168.0.0 192.168.255.255
</Stream>
?
vlc can play back the mjpeg stream.

Firefox can't play back the stream.

?
I found out that the boundary string between the jpeg images are missing.

?
?
Content-Type: multipart/x-mixed-replace; boundary=--myboundary 

?
--myboundary 

Content-Type: image/jpeg 

Content-Length: 44552 

?
How can i set ffserver off doing the boundarystring?

?
Best regards

Patrick


From onemda at gmail.com  Fri Aug 30 16:58:03 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 30 Aug 2013 14:58:03 +0000
Subject: [FFmpeg-user] Pixel Formats for libx264
In-Reply-To: <1377873532.71469.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <1377873532.71469.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P7FZGxkpi=y82PMbmRu2fmWNbybQb073NqLFxSPSBzwpg@mail.gmail.com>

On 8/30/13, James Board <jpboard2 at yahoo.com> wrote:
> What pixel format, if any, does the x264 codecs convert video to before
> encoding?  If my source video is raw RGB (3 bytes per pixel), will the
> encoded video use the same pixel format?  If my source video is YUV422, will
> the encoded video use the same pixel format?

libx246 encoder:
INPUT OUTPUT
rgb   yuv444p
yuv422 yuv422

libx264rgb encoder
INPUT OUTPUT
anything rgb24/bgr24

From krueger at lesspain.de  Fri Aug 30 17:54:34 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 30 Aug 2013 17:54:34 +0200
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
In-Reply-To: <CAEnqZEWAnanEEB-MLyKFLyKidtvFb5+SqHKoKaa1ob5bO_s+yg@mail.gmail.com>
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
	<loom.20130830T113813-270@post.gmane.org>
	<CAEnqZEWAnanEEB-MLyKFLyKidtvFb5+SqHKoKaa1ob5bO_s+yg@mail.gmail.com>
Message-ID: <CAEnqZEW00oR5wKECc7HGrh+p2nXkyRwonCe5MC-SZaV6YLaS4Q@mail.gmail.com>

On Fri, Aug 30, 2013 at 11:44 AM, Robert Kr?ger <krueger at lesspain.de> wrote:
> On Fri, Aug 30, 2013 at 11:38 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
>> Robert Kr?ger <krueger <at> lesspain.de> writes:
>>
>>> configure --disable-optimizations --disable-mmx --disable-stripping
>>
>>> BTW: Those rather crippling options are required to be able to
>>> debug ffmpeg within my IDE.
>>
>> Hard to believe that --disable-stripping is needed to debug.

You are right. It isn't.

Thanks for fixing the compilation!

From cehoyos at ag.or.at  Fri Aug 30 17:56:19 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 15:56:19 +0000 (UTC)
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
Message-ID: <loom.20130830T175549-232@post.gmane.org>

Robert Kr?ger <krueger <at> lesspain.de> writes:

> I'm getting compile errors when building ffmpeg 
> (current HEAD this morning) with
> 
> configure [...] --disable-mmx [...]

Should be fixed, thank you for the report.

Carl Eugen


From cehoyos at ag.or.at  Fri Aug 30 18:04:26 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 16:04:26 +0000 (UTC)
Subject: [FFmpeg-user] Pixel Formats for libx264
References: <1377873532.71469.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <loom.20130830T175749-587@post.gmane.org>

James Board <jpboard2 <at> yahoo.com> writes:

> What pixel format, if any, does the x264 codecs 
> convert video to before encoding?

(First of all: the "x264 codec" does no pixel format 
conversion at all, but since x264 does not accept all 
pixel formats that FFmpeg supports internally, ffmpeg 
- the application - automatically converts your input 
into a pixel format supported by the "x264 codec" if 
necessary.)

Doesn't the console output tell you?

If not, that might indicate a serious bug or at least 
some information that we could give you (on this list).

If yes, I wonder if the (correct) answer you received 
really has any advantages over simple testing:

$ ffmpeg -f rawvideo -pix_fmt rgb24 -i /dev/zero -vframes 1 out.h264
$ ffmpeg -f rawvideo -pix_fmt yuv422p -i /dev/zero -vframes 1 out.h264
$ ffmpeg -f rawvideo -pix_fmt yuva420p -i /dev/zero -vframes 1 out.h264
$ ffmpeg -f rawvideo -pix_fmt rgb24 -i /dev/zero -vcodec libx264rgb out.h264

(This is probably missing -s cif but you get the idea.)

Carl Eugen


From cehoyos at ag.or.at  Fri Aug 30 18:13:34 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 30 Aug 2013 16:13:34 +0000 (UTC)
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
	<loom.20130830T113813-270@post.gmane.org>
	<CAEnqZEWAnanEEB-MLyKFLyKidtvFb5+SqHKoKaa1ob5bO_s+yg@mail.gmail.com>
	<CAEnqZEW00oR5wKECc7HGrh+p2nXkyRwonCe5MC-SZaV6YLaS4Q@mail.gmail.com>
Message-ID: <loom.20130830T180944-62@post.gmane.org>

Robert Kr?ger <krueger <at> lesspain.de> writes:

> >>> configure --disable-optimizations --disable-mmx --disable-stripping
> >>
> >>> BTW: Those rather crippling options are required to be able to
> >>> debug ffmpeg within my IDE.
> >>
> >> Hard to believe that --disable-stripping is needed to debug.

(To elaborate: It is needed if you use --enable-shared.)

> You are right. It isn't.

Could you elaborate on why --disable-mmx and --disable-optimizations 
are needed? Both are particularly bad options if you are not 
debugging (and both may and did in the past change the 
behaviour of the library in the cases you want to debug) and 
at least for --disable-mmx it is not clear to me why it helps: 
If it were necessary, --disable-asm would probably be the 
right option (unless you debugger is pre-mmx).

Carl Eugen


From basteon at gmail.com  Sat Aug 31 00:45:41 2013
From: basteon at gmail.com (basteon)
Date: Fri, 30 Aug 2013 22:45:41 +0000
Subject: [FFmpeg-user] rtmp streaming to two media servers
In-Reply-To: <CAB-_x2Mdd--Vf1BjkVJhsd6zheddtXMy574B4gVQP38JRwDBWw@mail.gmail.com>
References: <CAB-_x2Mdd--Vf1BjkVJhsd6zheddtXMy574B4gVQP38JRwDBWw@mail.gmail.com>
Message-ID: <CAEJHKkXHmdcgjkZD3t9fW__aYQefnx9PRJiROSuQXi8SaUgG4g@mail.gmail.com>

hi,
perhaps flag -loop_output will help you.


On 29 August 2013 13:04, Maxim Kozlov <rogat1y at gmail.com> wrote:

> Hi.
>
> I stream live video to two media servers(wowza) using such command line:
> ffmpeg [input, libx264 and libfdk_aac options] -f tee "[f=flv]rtmp://
> 192.168.0.1/live/stream1|[f=flv]rtmp://192.168.0.2/live/stream1"
>
> Works perfect.
>
> But if one of servers crashes, ffmpeg  exits with message:
> WriteN, RTMP send error 10053 (129 bytes)
> WriteN, RTMP send error 10053 (48 bytes)
> WriteN, RTMP send error 10038 (42 bytes)
> av_interleaved_write_frame(): Operation not permitted
>
> Please suggest the way to make ffmpeg reconnect to media server after
> disconnection...like it doing Adobe Flaash Media Live Encoder?
>
> ffmpeg running on windows 7
> ffmpeg version N-55786-g185fc52 Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Aug 26 2013 19:39:07 with gcc 4.8.0 (GCC)
>   configuration: --arch=x86 --target-os=mingw32
> --cross-prefix=/mingw-w64-i686/bin/i686-w64-mingw32-
> --pkg-config=pkg-config --enable-gpl --enable-libsoxr --enable-libx264
> --enable-avisynth --enable-libxvid --enable-libmp3lame --enable-version3
> --enable-zlib --enable-librtmp --enable-libvorbis --enable-libtheora
> --enable-libspeex --enable-libopenjpeg --enable-gnutls --enable-libgsm
> --enable-libfreetype --enable-fontconfig --enable-libass
> --enable-libutvideo --enable-libopus --disable-w32threads --enable-frei0r
> --enable-filter=frei0r --enable-libvo-aacenc --enable-bzlib
> --enable-libxavs --extra-cflags=-DPTW32_STATIC_LIB
> --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libvo-amrwbenc --enable-libschroedinger --enable-libbluray
> --enable-libvpx --enable-libilbc --enable-static --disable-shared
> --enable-nonfree --enable-libfdk-aac --enable-runtime-cpudetect
>   libavutil      52. 42.100 / 52. 42.100
>   libavcodec     55. 29.100 / 55. 29.100
>   libavformat    55. 14.102 / 55. 14.102
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 82.102 /  3. 82.102
>   libswscale      2.  5.100 /  2.  5.100
>   libswresample   0. 17.103 /  0. 17.103
>   libpostproc    52.  3.100 / 52.  3.100
>
> p.s. I hope my english not so bad and you understand me =)
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From krueger at lesspain.de  Sat Aug 31 12:06:42 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Sat, 31 Aug 2013 12:06:42 +0200
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
In-Reply-To: <loom.20130830T180944-62@post.gmane.org>
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
	<loom.20130830T113813-270@post.gmane.org>
	<CAEnqZEWAnanEEB-MLyKFLyKidtvFb5+SqHKoKaa1ob5bO_s+yg@mail.gmail.com>
	<CAEnqZEW00oR5wKECc7HGrh+p2nXkyRwonCe5MC-SZaV6YLaS4Q@mail.gmail.com>
	<loom.20130830T180944-62@post.gmane.org>
Message-ID: <CAEnqZEWYQCxMdpQC8qjxUyzUAGUfd9LAw1oDuRnAKqmx0Vsnfw@mail.gmail.com>

On Fri, Aug 30, 2013 at 6:13 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> >>> configure --disable-optimizations --disable-mmx --disable-stripping
>> >>
>> >>> BTW: Those rather crippling options are required to be able to
>> >>> debug ffmpeg within my IDE.
>> >>
>> >> Hard to believe that --disable-stripping is needed to debug.
>
> (To elaborate: It is needed if you use --enable-shared.)
>
>> You are right. It isn't.
>
> Could you elaborate on why --disable-mmx and --disable-optimizations
> are needed? Both are particularly bad options if you are not

No, I can not, because I honestly do not know. I got the options from
a howto document on the web and as my knowledge in that area is close
to non-existant I was happy that the recipe worked.

> debugging (and both may and did in the past change the
> behaviour of the library in the cases you want to debug) and
> at least for --disable-mmx it is not clear to me why it helps:
> If it were necessary, --disable-asm would probably be the
> right option (unless you debugger is pre-mmx).

I just tried again and only --disable-asm does not work. The debugger
integrated in my IDE (Eclipse-CDT but it is gdb under the hood) does
not display variables then.  I think when I first researched I found
an article explaining where this limitation came from but I cannot
find it now.


>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From cehoyos at ag.or.at  Sat Aug 31 12:13:34 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 31 Aug 2013 10:13:34 +0000 (UTC)
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
	<loom.20130830T113813-270@post.gmane.org>
	<CAEnqZEWAnanEEB-MLyKFLyKidtvFb5+SqHKoKaa1ob5bO_s+yg@mail.gmail.com>
	<CAEnqZEW00oR5wKECc7HGrh+p2nXkyRwonCe5MC-SZaV6YLaS4Q@mail.gmail.com>
	<loom.20130830T180944-62@post.gmane.org>
	<CAEnqZEWYQCxMdpQC8qjxUyzUAGUfd9LAw1oDuRnAKqmx0Vsnfw@mail.gmail.com>
Message-ID: <loom.20130831T120955-525@post.gmane.org>

Robert Kr?ger <krueger <at> lesspain.de> writes:

> > at least for --disable-mmx it is not clear to me why it helps:
> > If it were necessary, --disable-asm would probably be the
> > right option (unless you debugger is pre-mmx).
> 
> I just tried again and only --disable-asm does not work.

That indicates that --disable-mmx is not needed.
Or am I wrong?

> The debugger integrated in my IDE (Eclipse-CDT but it is gdb 
> under the hood) does not display variables then. I think when 
> I first researched I found an article explaining where this 
> limitation came from but I cannot find it now.

Could you test --extra-cflags=-fnoomit-frame-pointer instead 
of --disable-optimizations?
Or --enable-debug=3 (2)?

Carl Eugen


From krueger at lesspain.de  Sat Aug 31 13:04:33 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Sat, 31 Aug 2013 13:04:33 +0200
Subject: [FFmpeg-user] Compile errors on OSX with current HEAD
In-Reply-To: <loom.20130831T120955-525@post.gmane.org>
References: <CAEnqZEUx4UVVKL=rf4-vFv-ty2hdz2ESRbcqa4HHuD=ctFCdLg@mail.gmail.com>
	<loom.20130830T113813-270@post.gmane.org>
	<CAEnqZEWAnanEEB-MLyKFLyKidtvFb5+SqHKoKaa1ob5bO_s+yg@mail.gmail.com>
	<CAEnqZEW00oR5wKECc7HGrh+p2nXkyRwonCe5MC-SZaV6YLaS4Q@mail.gmail.com>
	<loom.20130830T180944-62@post.gmane.org>
	<CAEnqZEWYQCxMdpQC8qjxUyzUAGUfd9LAw1oDuRnAKqmx0Vsnfw@mail.gmail.com>
	<loom.20130831T120955-525@post.gmane.org>
Message-ID: <CAEnqZEXX7X2EjB3wWWTV8=NGpKUz=6a3eq6kFzOTAQR0BPNtvQ@mail.gmail.com>

On Sat, Aug 31, 2013 at 12:13 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> > at least for --disable-mmx it is not clear to me why it helps:
>> > If it were necessary, --disable-asm would probably be the
>> > right option (unless you debugger is pre-mmx).
>>
>> I just tried again and only --disable-asm does not work.
>
> That indicates that --disable-mmx is not needed.
> Or am I wrong?

You are right.

>
>> The debugger integrated in my IDE (Eclipse-CDT but it is gdb
>> under the hood) does not display variables then. I think when
>> I first researched I found an article explaining where this
>> limitation came from but I cannot find it now.
>
> Could you test --extra-cflags=-fnoomit-frame-pointer instead
> of --disable-optimizations?
> Or --enable-debug=3 (2)?

I tried --extra-cflags=-fno-omit-frame-pointer and --enable-debug=3
separately and neither one worked.

>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From andrey.aleksandrovich at googlemail.com  Sat Aug 31 15:17:02 2013
From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich)
Date: Sat, 31 Aug 2013 16:17:02 +0300
Subject: [FFmpeg-user] Delay of one audio track makes it track played last
	among others?
Message-ID: <CAPGkfYFKTgJH60UDhQBNFgVhN0aT4KfB_gJSiJvja5rZ+oYTcQ@mail.gmail.com>

Hello,
I need to make DVD compatible mpg file. I already have video with
English track (1.mpg). And I have Rus.mp2 - Russian track, which needs
delaying (330 ms).
If I do this:
ffmpeg -y -itsoffset 0.33 -i Rus.mp2 -i 1.mpg -vcodec copy -acodec
copy -map 1:0 -map 0:0 -map 1:1 -f dvd final.vob
ffmpeg version 1.0.7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 15 2013 10:59:09 with gcc 4.6.3 (Gentoo Hardened 4.6.3
p1.13, pie-0.5.2)
  configuration: --prefix=/usr --libdir=/usr/lib64
--shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared
--cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++
--ar=x86_64-pc-linux-gnu-ar --optflags='-O2 -pipe -march=native'
--extra-cflags='-O2 -pipe -march=native' --extra-cxxflags='-O2 -pipe
-march=native' --disable-static --enable-gpl --enable-version3
--enable-postproc --enable-avfilter --enable-avresample
--disable-stripping --disable-debug --disable-doc --disable-network
--disable-vaapi --disable-runtime-cpudetect --enable-libmp3lame
--enable-libvo-amrwbenc --enable-libtheora --enable-libx264
--enable-libxvid --disable-indev=v4l2 --disable-indev=oss
--disable-indev=jack --enable-x11grab --disable-outdev=oss
--enable-libass --enable-libfreetype --enable-pthreads
--enable-libopencore-amrwb --enable-libopencore-amrnb
--enable-libvorbis --enable-libvpx --enable-libopenjpeg
--disable-altivec --disable-avx --disable-ssse3 --disable-vis
--disable-neon --cpu=host --enable-hardcoded-tables
  libavutil      51. 73.101 / 51. 73.101
  libavcodec     54. 59.100 / 54. 59.100
  libavformat    54. 29.104 / 54. 29.104
  libavdevice    54.  2.101 / 54.  2.101
  libavfilter     3. 17.100 /  3. 17.100
  libswscale      2.  1.101 /  2.  1.101
  libswresample   0. 15.100 /  0. 15.100
  libpostproc    52.  0.100 / 52.  0.100
[mp3 @ 0x503590c5e0] max_analyze_duration 5000000 reached at 5016000
[mp3 @ 0x503590c5e0] Estimating duration from bitrate, this may be
inaccurate
Input #0, mp3, from 'Rus.mp2':
  Duration: 01:23:57.88, start: 0.000000, bitrate: 128 kb/s
    Stream #0:0: Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s
[mpeg @ 0x503590cde0] max_analyze_duration 5000000 reached at 5005000
Guessed Channel Layout for  Input Stream #1.1 : stereo
Input #1, mpeg, from '1.mpg':
  Duration: 01:23:57.86, start: 0.300300, bitrate: 6921 kb/s
    Stream #1:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480
[SAR 8:9 DAR 4:3], 8000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #1:1[0xa0]: Audio: pcm_s16be, 48000 Hz, stereo, s16, 1536 kb/s
Output #0, dvd, to 'f.vob':
  Metadata:
    encoder         : Lavf54.29.104
    Stream #0:0: Video: mpeg2video, yuv420p, 720x480 [SAR 8:9 DAR
4:3], q=2-31, 8000 kb/s, 29.97 fps, 90k tbn, 29.97 tbc
    Stream #0:1: Audio: mp2, 48000 Hz, stereo, 128 kb/s
    Stream #0:2: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s
Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
  Stream #1:1 -> #0:2 (copy)
Press [q] to stop, [?] for help
frame=150986 fps=593 q=-1.0 Lsize= 4357134kB time=01:23:58.21
bitrate=7084.6kbits/s
video:3246391kB audio:1023318kB subtitle:0 global headers:0kB muxing
overhead 2.047583%

I get the final.vob file with proper delaying and mapping (1st
Russian, 2nd English) according to mediainfo. But 'ffmpeg -i' said
that Russian is second and all Players show the same (playing video ->
first is English).

Why it is happened?

From leonard at kcfchurch.org  Sat Aug 31 17:52:51 2013
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Sat, 31 Aug 2013 08:52:51 -0700
Subject: [FFmpeg-user] FFmpeg won't stay connected to Wowza server over the
	internet.
Message-ID: <CA+JzsNSKwHtswhpvnt+w+okmoMCSfc14sxd2+YYezg7hqLZ+UA@mail.gmail.com>

I've setup a new and updated version of the Amazon AMI EC2 machine image of
a Wowza server for live streaming.  I'm trying to migrate away from Adobe's
Flash Media Live Encoder (FMLE) on Windows and change to FFmpeg on Linux.
I am currently running the server version of Ubuntu with a Decklink SDI
input card using "bmdcapture" to pipe into FFmpeg.  I am using the "tee"
command to split the data to two separate instances of FFmpeg, one to
record to the local harddrive and another to rtmp stream to the remote
Wowza server.  At the moment I am using 3 named pipes to do all this.

If I setup FFmpeg to RTMP to a Wowza server running on a machine behind my
internet router, everything appears to run absolutely fine.  I've left it
running for 5 hours once (much longer than I plan to run it for
continuously) and I get no problems.  Zero, nada, zilch.  It will continue
to run for as long as I don't tell it to stop.  I've tested it connecting
to a Wowza server on Windows XP, a Wowza on a modern Mac, and a Wowza on a
virtual Ubuntu machine.

However, if I RTMP to the remote (over-the-internet) Wowza server, after
about some random period of time that is sometimes 10 minutes to as short
as 10 seconds after connecting, it just drops, with no connection specific
error messages or anything.  It does this all the time.  The only way I am
able to keep a connection is if I set the bitrate to 100K or lower, but
sometimes 200K.

Now for the kicker: FMLE works fine at 500K+ to the same remote server.  It
will stay connected, it seems, no matter what.

The big reasons why I want to migrate away from FMLE is that FMLE can't
seem to keep the audio in sync, the recording to the HDD is [teh] crap
(probably because it just saves the encoded frames from the stream instead
of remuxing to a file format), the volume is way too low, it doesn't have
an internal video resampler to get the size of the video we need (which
makes it incompatible with the Decklink card), and it absolutely will not
do 48Khz audio sample rate or AAC.

Here's a pastebin of my bash script to startup FFmpeg (with alot of
commands related to recorded file maintenance and PID based process
protection/management removed for sake of readability/simplicity.)

http://pastebin.com/2Ht3wxJb

Any help appreciated.  TIA

From naitik.p.ntech at gmail.com  Sat Aug 31 09:17:25 2013
From: naitik.p.ntech at gmail.com (Naitik patel)
Date: Sat, 31 Aug 2013 12:47:25 +0530
Subject: [FFmpeg-user] How to convert flash to ts
Message-ID: <CAA3BdmHnB2jLHmEJnGdBtV+iVFvH4Oz62D_YHoXBB64unQ27Sw@mail.gmail.com>

Hi

I have one link which live stream video in flash format. I want to convert
that in to ts format.How can i do that thing please help me for that.

regards
Naitik Patel

From XPSystemServices at yandex.ru  Sat Aug 31 13:13:09 2013
From: XPSystemServices at yandex.ru (Bill Gates)
Date: Sat, 31 Aug 2013 15:13:09 +0400
Subject: [FFmpeg-user] libmp3lame -ab 128k produces a VBR stream
Message-ID: <55CD3817E6D5449B9C03E1BC7AC078BF@nell>

Hello,

I encode in MPEG4/MP3 to AVI files with the following command:

ffmpeg -i input.avi -vcodec mpeg4 -vtag XVID -s 640x480 -qscale 4 -g 50 -bf
2 -acodec libmp3lame -ab 128k -y output.avi

The files will play fine on the PC, but not on the standalone player.

When I tried opening one of these files with VirtualDub I got the message:

"VirtualDub has detected an improper VBR audio encoding in the source AVI
file and will rewrite the header with standard CBR values during processing
for better compatibility.........."

If I do so, rewriting the file with VirtualDub by direct stream copy of both
audio and video, the result will play fine on the standalone player! So
VirtualDub did the right thing with the audio header.

Why AVI files with an mp3 audio stream produced by ffmpeg are marked in the
AVI header as VBR audio instead of CBR?

Does "-ab 128k" produce a CBR stream, or do I need to code another option in
the command-line to force CBR?

Thanks, 



From lou at lrcd.com  Sat Aug 31 18:28:52 2013
From: lou at lrcd.com (Lou)
Date: Sat, 31 Aug 2013 08:28:52 -0800
Subject: [FFmpeg-user] libmp3lame -ab 128k produces a VBR stream
In-Reply-To: <55CD3817E6D5449B9C03E1BC7AC078BF@nell>
References: <55CD3817E6D5449B9C03E1BC7AC078BF@nell>
Message-ID: <1377966532.13776.16378841.5E3ECCE6@webmail.messagingengine.com>

On Sat, Aug 31, 2013, at 03:13 AM, Bill Gates wrote:
> Hello,
> 
> I encode in MPEG4/MP3 to AVI files with the following command:
> 
> ffmpeg -i input.avi -vcodec mpeg4 -vtag XVID -s 640x480 -qscale 4 -g 50
> -bf
> 2 -acodec libmp3lame -ab 128k -y output.avi

Please include the complete ffmpeg console output.

From wimbonner at gmail.com  Sat Aug 31 20:10:31 2013
From: wimbonner at gmail.com (William C Bonner)
Date: Sat, 31 Aug 2013 11:10:31 -0700
Subject: [FFmpeg-user] FFMPeg reading from Logitech C920 generating H264
 source produces empty output file
In-Reply-To: <CALqHiHPnb5=M8TxXEoXGkF3jKtU9ySU3+3x08LbzefLFiufO6A@mail.gmail.com>
References: <CALqHiHNzHNAgCELtYSBLCU01A0R=xEZFF8v3+NNa8j_rbdmvZg@mail.gmail.com>
	<loom.20130722T104914-806@post.gmane.org>
	<CALqHiHO-4YmY2oJri-f3xg1nAT_bRbf03e3iRxXtHrZBFjbJZQ@mail.gmail.com>
	<CALqHiHPnb5=M8TxXEoXGkF3jKtU9ySU3+3x08LbzefLFiufO6A@mail.gmail.com>
Message-ID: <CALqHiHNs8sfL_Y6tpTce-Hvm7gi74=zA7Yk46gsZz7rJWkyf0A@mail.gmail.com>

I finally came up with a way of getting FFMPEG to read h.264 directly from
my Logitech C920 webcam and stream it over RTP. The parameter required
was -copyinkf (copy initial non-keyframes)

I'm not sure why this suddenly made it able to create a non-zero file when
saving to a local file, or create a working stream when pushing to another
machine, but it did.

Here's the command line that I'm currently using in my testing.

ffmpeg -s 1920x1080 -f v4l2 -vcodec h264 -i /dev/video0 -copyinkf -vcodec
copy -f rtp rtp://192.168.0.10:8090/



On Mon, Jul 22, 2013 at 2:59 PM, William C Bonner <wimbonner at gmail.com>wrote:

> Can I attach text files to this list?
>
> I just was hoping to try since the output of the report is really wide,
> and breaking it into standard email format makes it that much harder to
> read. I'm attaching both the version with the -vcodec copy and without
> because while both have lines starting "demuxer -> " and "demuxer+ffmpeg
> ->" the second also has lines "decoder ->", "encoder ->", and "muxer <-".
>
> Wim.
>
>
> On Mon, Jul 22, 2013 at 2:47 PM, William C Bonner <wimbonner at gmail.com>wrote:
>
>> I tried adding the -debug_ts as the first option, and it spit lots of
>> information back at me that I don't really understand, so I also added a
>> -report option, and here's that output.
>>
>> I wasn't sure where in the command line to try -use_wallclock_as_timestamps
>> 1 so I added it right before the -y test.mp4 but didn't notice any change,
>> so it's not in this output.
>>
>> On Mon, Jul 22, 2013 at 1:51 AM, Carl Eugen Hoyos <cehoyos at ag.or.at>wrote:
>>
>>> William C Bonner <wimbonner <at> gmail.com> writes:
>>>
>>> > frame=    0 fps=0.0 q=-1.0 Lsize=       0kB time=00:00:00.00
>>> bitrate=N/A
>>> > video:0kB audio:0kB subtitle:0 global headers:0kB muxing overhead inf%
>>> > Output file is empty, nothing was encoded
>>> > (check -ss / -t / -frames parameters if used)
>>>
>>> Use -debug_ts to show the timestamps of the incoming stream,
>>> and please try -use_wallclock_as_timestamps 1 which may
>>> help.
>>>
>>> Carl Eugen
>>>
>>> _______________________________________________
>>> ffmpeg-user mailing list
>>> ffmpeg-user at ffmpeg.org
>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>>
>>
>>
>

From wimbonner at gmail.com  Sat Aug 31 22:08:21 2013
From: wimbonner at gmail.com (William C Bonner)
Date: Sat, 31 Aug 2013 13:08:21 -0700
Subject: [FFmpeg-user] ffmpeg cannot capture h264 from webcam C920
In-Reply-To: <loom.20130829T115604-277@post.gmane.org>
References: <kvdeh5$2qh$1@ger.gmane.org>
	<loom.20130829T115604-277@post.gmane.org>
Message-ID: <CALqHiHOqoYgM3XC83EkufU3bniJQf3T=M10pqt8P_wnSCttx9g@mail.gmail.com>

I finally figured out that to use FFMPEG directly with the logitech C920
camera doing h.264 encoding, I need to use the -copyinkf flag.

ffmpeg -report -s 1920x1080 f v4l2 -vcodec h264 -i /dev/video0 -copyinkf
-vcodec copy test.mp4


On Thu, Aug 29, 2013 at 2:57 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> andrea <mariofutire <at> googlemail.com> writes:
>
> > I have the same issue reported here
> >
> > http://ffmpeg.org/pipermail/ffmpeg-user/2013-May/015046.html
> > and
> > http://ffmpeg.org/pipermail/ffmpeg-user/2013-July/016243.html
> >
> > I have tried ffmpeg 1.2.2 on fedora and 2.0.1 on raspberry
> > (same result). I have only tried ffplay 1.2.2 on fedora.
>
> Iirc, all users so far reported (repeatedly) that ffplay
> works fine with this camera (as does reencoding with ffmpeg).
> Please test reencoding with current ffmpeg and report back.
> (current != 2.0.1 !)
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Sat Aug 31 22:59:43 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 31 Aug 2013 20:59:43 +0000 (UTC)
Subject: [FFmpeg-user] FFMPeg reading from Logitech C920 generating H264
	source produces empty output file
References: <CALqHiHNzHNAgCELtYSBLCU01A0R=xEZFF8v3+NNa8j_rbdmvZg@mail.gmail.com>
	<loom.20130722T104914-806@post.gmane.org>
	<CALqHiHO-4YmY2oJri-f3xg1nAT_bRbf03e3iRxXtHrZBFjbJZQ@mail.gmail.com>
	<CALqHiHPnb5=M8TxXEoXGkF3jKtU9ySU3+3x08LbzefLFiufO6A@mail.gmail.com>
	<CALqHiHNs8sfL_Y6tpTce-Hvm7gi74=zA7Yk46gsZz7rJWkyf0A@mail.gmail.com>
Message-ID: <loom.20130831T225837-959@post.gmane.org>

William C Bonner <wimbonner <at> gmail.com> writes:

> I finally came up with a way of getting FFMPEG to read 
> h.264 directly from my Logitech C920 webcam and stream 
> it over RTP. The parameter required was -copyinkf (copy 
> initial non-keyframes)

Please provide a sample.
$ ffmpeg -f v4l2 -vcodec h264 -i /dev/video0 -copyinkf -vcodec copy out.h264

Carl Eugen


From cehoyos at ag.or.at  Sat Aug 31 23:00:27 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 31 Aug 2013 21:00:27 +0000 (UTC)
Subject: [FFmpeg-user] How to convert flash to ts
References: <CAA3BdmHnB2jLHmEJnGdBtV+iVFvH4Oz62D_YHoXBB64unQ27Sw@mail.gmail.com>
Message-ID: <loom.20130831T230003-218@post.gmane.org>

Naitik patel <naitik.p.ntech <at> gmail.com> writes:

> I have one link which live stream video in flash format. 
> I want to convert that in to ts format.

Please provide ffmpeg -i console output for your input stream.

Carl Eugen


From cehoyos at ag.or.at  Sat Aug 31 23:08:25 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 31 Aug 2013 21:08:25 +0000 (UTC)
Subject: [FFmpeg-user] FFMPeg reading from Logitech C920 generating H264
	source produces empty output file
References: <CALqHiHNzHNAgCELtYSBLCU01A0R=xEZFF8v3+NNa8j_rbdmvZg@mail.gmail.com>
	<loom.20130722T104914-806@post.gmane.org>
	<CALqHiHO-4YmY2oJri-f3xg1nAT_bRbf03e3iRxXtHrZBFjbJZQ@mail.gmail.com>
	<CALqHiHPnb5=M8TxXEoXGkF3jKtU9ySU3+3x08LbzefLFiufO6A@mail.gmail.com>
	<CALqHiHNs8sfL_Y6tpTce-Hvm7gi74=zA7Yk46gsZz7rJWkyf0A@mail.gmail.com>
Message-ID: <loom.20130831T230745-809@post.gmane.org>

William C Bonner <wimbonner <at> gmail.com> writes:

> I finally came up with a way of getting FFMPEG to 
> read h.264 directly from my Logitech C920 webcam 
> and stream it over RTP. The parameter required
> was -copyinkf (copy initial non-keyframes)

I attached a patch in ticket #2882, could you test 
if it also fixes the issue?

Carl Eugen


From wimbonner at gmail.com  Sat Aug 31 23:14:34 2013
From: wimbonner at gmail.com (William C Bonner)
Date: Sat, 31 Aug 2013 14:14:34 -0700
Subject: [FFmpeg-user] Use of the -t option with multiple output files.
Message-ID: <CALqHiHNZ3mgPig8qSo+Jad_GUT6F2=EubQb8JUEH3YxoCioepw@mail.gmail.com>

I've successfully been using ffmpeg to create multiple outputs, one a
network stream and one a local file with the following command:

ffmpeg -s 1280x720 -f v4l2 -vcodec h264 -i /dev/video0 -copyinkf -vcodec
copy -f rtp rtp://192.168.0.10:8090/ -vcodec copy test.mp4

I want it to only run for a set period of time. For testing I've been
trying to use 30 seconds. What I've not been able to figure out is where is
the appropriate place to put a "-t 30" in the command line. When I was
creating a single output file, I put it right before the name of the output
and everything worked correctly. If I duplicate the -t 30 in front of each
file things work for the first 30 seconds, but then ffmpeg stops visibly
outputting, but doesn't exit. If I put it before the -i (input) ffmpeg
doesn't stop at 30 seconds.

Any suggestions are appreciated.

Wim.

From cehoyos at ag.or.at  Sat Aug 31 23:44:04 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 31 Aug 2013 21:44:04 +0000 (UTC)
Subject: [FFmpeg-user] Use of the -t option with multiple output files.
References: <CALqHiHNZ3mgPig8qSo+Jad_GUT6F2=EubQb8JUEH3YxoCioepw@mail.gmail.com>
Message-ID: <loom.20130831T234235-984@post.gmane.org>

William C Bonner <wimbonner <at> gmail.com> writes:

> ffmpeg -s 1280x720 -f v4l2 -vcodec h264 -i /dev/video0 
> -copyinkf -vcodec copy -f rtp rtp://192.168.0.10:8090/ 
> -vcodec copy test.mp4
> 
> I want it to only run for a set period of time. For 
> testing I've been trying to use 30 seconds. What I've 
> not been able to figure out is where is the 
> appropriate place to put a "-t 30" in the command line.

That depends on your use case.

> When I was creating a single output file, I put it 
> right before the name of the output and everything 
> worked correctly.

> If I duplicate the -t 30 in front of each file things 
> work for the first 30 seconds, but then ffmpeg stops 
> visibly outputting, but doesn't exit.

Command line and complete, uncut console output missing.

> If I put it before the -i (input) ffmpeg doesn't stop 
> at 30 seconds.

Command line and complete, uncut console output missing.

Carl Eugen


From wimbonner at gmail.com  Sat Aug 31 23:53:24 2013
From: wimbonner at gmail.com (William C Bonner)
Date: Sat, 31 Aug 2013 14:53:24 -0700
Subject: [FFmpeg-user] FFMPeg reading from Logitech C920 generating H264
 source produces empty output file
In-Reply-To: <loom.20130831T230745-809@post.gmane.org>
References: <CALqHiHNzHNAgCELtYSBLCU01A0R=xEZFF8v3+NNa8j_rbdmvZg@mail.gmail.com>
	<loom.20130722T104914-806@post.gmane.org>
	<CALqHiHO-4YmY2oJri-f3xg1nAT_bRbf03e3iRxXtHrZBFjbJZQ@mail.gmail.com>
	<CALqHiHPnb5=M8TxXEoXGkF3jKtU9ySU3+3x08LbzefLFiufO6A@mail.gmail.com>
	<CALqHiHNs8sfL_Y6tpTce-Hvm7gi74=zA7Yk46gsZz7rJWkyf0A@mail.gmail.com>
	<loom.20130831T230745-809@post.gmane.org>
Message-ID: <CALqHiHOCV3F8+1T7stegUZTKYhhQD1Nb4QJ-fyz9nMvk1DmN8g@mail.gmail.com>

Here's a local file example command that creates a test.mp4 file from the
C920 without transcoding. I added -y -t 20 to simplify the testing.

ffmpeg -report -s 1280x720 -f v4l2 -vcodec h264 -i /dev/video0 -copyinkf
-vcodec copy -y -t 20 test.mp4

It sometimes produces a bunch of error messages:
[mp4 @ 0x111aac0] Non-monotonous DTS in output stream 0:0; previous:
202760, current: 136127; changing to 202761. This may result in incorrect
timestamps in the output file.

I looked at https://trac.ffmpeg.org/ticket/2882 as well as looking at the
patch submitted. (
https://trac.ffmpeg.org/attachment/ticket/2882/patchv4l2parseh264.diff)

I just now manually installed the patch in my local copy of the source and
rebuilt ffmpeg. Now things seem to work with or without the -copyinfk flag.

Here's three report files, the last being one outputing an h264 file
instead of mp4.

ffmpeg started on 2013-08-31 at 21:39:21
Report written to "ffmpeg-20130831-213921.log"
Command line:
ffmpeg -report -s 1280x720 -f v4l2 -vcodec h264 -i /dev/video0 -copyinkf
-vcodec copy -y -t 20 test.mp4
ffmpeg version N-55922-g454a11a Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 31 2013 01:46:23 with gcc 4.7.3 (Linaro GCC 4.7-2013.02-01)
20130205 (prerelease)
  configuration: --prefix=/usr --enable-gpl --enable-libx264
  libavutil      52. 43.100 / 52. 43.100
  libavcodec     55. 29.100 / 55. 29.100
  libavformat    55. 15.100 / 55. 15.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.102 /  3. 82.102
  libswscale      2.  5.100 /  2.  5.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Splitting the commandline.
Reading option '-report' ... matched as option 'report' (generate a report)
with argument '1'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '1280x720'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'v4l2'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'h264'.
Reading option '-i' ... matched as input file with argument '/dev/video0'.
Reading option '-copyinkf' ... matched as option 'copyinkf' (copy initial
non-keyframes) with argument '1'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with
argument '1'.
Reading option '-t' ... matched as option 't' (record or transcode
"duration" seconds of audio/video) with argument '20'.
Reading option 'test.mp4' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file /dev/video0.
Applying option s (set frame size (WxH or abbreviation)) with argument
1280x720.
Applying option f (force format) with argument v4l2.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument h264.
Successfully parsed a group of options.
Opening an input file: /dev/video0.
[video4linux2,v4l2 @ 0x1098020] fd:4 capabilities:84000001
[video4linux2,v4l2 @ 0x1098020] Current input_channel: 0, input_name:
Camera 1
[h264 @ 0x10989d0] Current profile doesn't provide more RBSP data in PPS,
skipping
[video4linux2,v4l2 @ 0x1098020] All info found
Input #0, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 67532.978161, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264 (Constrained Baseline),
yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], -5 kb/s, 30 fps, 30 tbr, 1
000k tbn, 60 tbc
Successfully opened the file.
Parsing a group of options: output file test.mp4.
Applying option copyinkf (copy initial non-keyframes) with argument 1.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Applying option t (record or transcode "duration" seconds of audio/video)
with argument 20.
Successfully parsed a group of options.
Opening an output file: test.mp4.
Successfully opened the file.
Output #0, mp4, to 'test.mp4':
  Metadata:
    encoder         : Lavf55.15.100
    Stream #0:0, 0, 1/1000000: Video: h264 ([33][0][0][0] / 0x0021),
yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, -5 kb/s, 30 fps,
 1000k tbn, 1000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous: 0,
current: -66915; changing to 1. This may result in incorrect
 timestamps in the output file.
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous: 1,
current: -33840; changing to 2. This may result in incorrect
 timestamps in the output file.
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous: 2,
current: -849; changing to 3. This may result in incorrect t
imestamps in the output file.
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous: 64404,
current: 3549; changing to 64405. This may result in inc
orrect timestamps in the output file.
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous: 64405,
current: 33333; changing to 64406. This may result in in
correct timestamps in the output file.
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous: 64406,
current: 63033; changing to 64407. This may result in in
correct timestamps in the output file.
frame=  121 fps= 35 q=-1.0 size=    1355kB time=00:00:03.50
bitrate=3167.2kbits/s
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous:
5891441, current: 5848004; changing to 5891442. This may result
 in incorrect timestamps in the output file.
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous:
5891442, current: 5877547; changing to 5891443. This may result
 in incorrect timestamps in the output file.
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous:
7915502, current: 7909242; changing to 7915503. This may result
 in incorrect timestamps in the output file.
[NULL @ 0x10989d0] Current profile doesn't provide more RBSP data in PPS,
skipping
[mp4 @ 0x10e2060] Non-monotonous DTS in output stream 0:0; previous:
10297665, current: 10294977; changing to 10297666. This may res
ult in incorrect timestamps in the output file.
No more output streams to write to, finishing.e=00:00:11.34
bitrate=3129.1kbits/s
frame=  348 fps= 29 q=-1.0 Lsize=    4339kB time=00:00:11.34
bitrate=3132.6kbits/s
video:4334kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.112955%
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x113a180] Statistics: 34 seeks, 375 writeouts


ffmpeg started on 2013-08-31 at 21:40:30Report written to
"ffmpeg-20130831-214030.log"
Command line:
ffmpeg -report -s 1280x720 -f v4l2 -vcodec h264 -i /dev/video0 -vcodec copy
-y -t 20 test.mp4
ffmpeg version N-55922-g454a11a Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 31 2013 01:46:23 with gcc 4.7.3 (Linaro GCC 4.7-2013.02-01)
20130205 (prerelease)
  configuration: --prefix=/usr --enable-gpl --enable-libx264
  libavutil      52. 43.100 / 52. 43.100
  libavcodec     55. 29.100 / 55. 29.100
  libavformat    55. 15.100 / 55. 15.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.102 /  3. 82.102
  libswscale      2.  5.100 /  2.  5.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Splitting the commandline.
Reading option '-report' ... matched as option 'report' (generate a report)
with argument '1'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '1280x720'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'v4l2'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'h264'.
Reading option '-i' ... matched as input file with argument '/dev/video0'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with
argument '1'.
Reading option '-t' ... matched as option 't' (record or transcode
"duration" seconds of audio/video) with argument '20'.
Reading option 'test.mp4' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file /dev/video0.
Applying option s (set frame size (WxH or abbreviation)) with argument
1280x720.
Applying option f (force format) with argument v4l2.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument h264.
Successfully parsed a group of options.
Opening an input file: /dev/video0.
[video4linux2,v4l2 @ 0x1098020] fd:4 capabilities:84000001
[video4linux2,v4l2 @ 0x1098020] Current input_channel: 0, input_name:
Camera 1
[h264 @ 0x10989d0] Current profile doesn't provide more RBSP data in PPS,
skipping
[video4linux2,v4l2 @ 0x1098020] All info found
Input #0, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 67602.272846, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264 (Constrained Baseline),
yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], -5 kb/s, 30 fps, 30 tbr, 1
000k tbn, 60 tbc
Successfully opened the file.
Parsing a group of options: output file test.mp4.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Applying option t (record or transcode "duration" seconds of audio/video)
with argument 20.
Successfully parsed a group of options.
Opening an output file: test.mp4.
Successfully opened the file.
Output #0, mp4, to 'test.mp4':
  Metadata:
    encoder         : Lavf55.15.100
    Stream #0:0, 0, 1/1000000: Video: h264 ([33][0][0][0] / 0x0021),
yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, -5 kb/s, 30 fps,
 1000k tbn, 1000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous: 0,
current: -67749; changing to 1. This may result in incorrect
 timestamps in the output file.
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous: 1,
current: -35520; changing to 2. This may result in incorrect
 timestamps in the output file.
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous: 68806,
current: 2950; changing to 68807. This may result in inc
orrect timestamps in the output file.
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous: 68807,
current: 33333; changing to 68808. This may result in in
correct timestamps in the output file.
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous: 68808,
current: 62767; changing to 68809. This may result in in
correct timestamps in the output file.
[mp4 @ 0x10 fps= 35 q=-1.0 size=    1357kB time=00:00:03.52
bitrate=3151.3kbits/s
1df0] Non-monotonous DTS in output stream 0:0; previous: 3624926, current:
3613914; changing to 3624927. This may result in incorrec
t timestamps in the output file.
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous:
5835696, current: 5828676; changing to 5835697. This may result
 in incorrect timestamps in the output file.
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous:
5859518, current: 5847887; changing to 5859519. This may result
 in incorrect timestamps in the output file.
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous:
7908361, current: 7903403; changing to 7908362. This may result
 in incorrect timestamps in the output file.
[NULL @ 0x10989d0] Current profile doesn't provide more RBSP data in PPS,
skipping
frame=  406 fps= 31 q=-1.0 size=    4858kB time=00:00:12.87
bitrate=3091.8kbits/s
frame=  498 fps= 31 q=-1.0 size=    5969kB time=00:00:15.81
bitrate=3092.2kbits/s
[mp4 @ 0x10e1df0] Non-monotonous DTS in output stream 0:0; previous:
16874078, current: 16872688; changing to 16874079. This may res
ult in incorrect timestamps in the output file.
[NULL @ 0x1 fps= 31 q=-1.0 size=    7279kB time=00:00:19.48
bitrate=3060.6kbits/s
989d0] Current profile doesn't provide more RBSP data in PPS, skipping
No more output streams to write to, finishing.e=00:00:19.92
bitrate=3078.5kbits/s
frame=  606 fps= 31 q=-1.0 Lsize=    7519kB time=00:00:19.98
bitrate=3082.1kbits/s
video:7511kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.106004%
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x1139a50] Statistics: 34 seeks, 634 writeouts

ffmpeg started on 2013-08-31 at 21:50:20
Report written to "ffmpeg-20130831-215020.log"
Command line:
ffmpeg -f v4l2 -vcodec h264 -i /dev/video0 -copyinkf -vcodec copy -report
-y -t 20 out.h264
ffmpeg version N-55922-g454a11a Copyright (c) 2000-2013 the FFmpeg
developers
  built on Aug 31 2013 01:46:23 with gcc 4.7.3 (Linaro GCC 4.7-2013.02-01)
20130205 (prerelease)
  configuration: --prefix=/usr --enable-gpl --enable-libx264
  libavutil      52. 43.100 / 52. 43.100
  libavcodec     55. 29.100 / 55. 29.100
  libavformat    55. 15.100 / 55. 15.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 82.102 /  3. 82.102
  libswscale      2.  5.100 /  2.  5.100
  libswresample   0. 17.103 /  0. 17.103
  libpostproc    52.  3.100 / 52.  3.100
Splitting the commandline.
Reading option '-f' ... matched as option 'f' (force format) with argument
'v4l2'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'h264'.
Reading option '-i' ... matched as input file with argument '/dev/video0'.
Reading option '-copyinkf' ... matched as option 'copyinkf' (copy initial
non-keyframes) with argument '1'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'copy'.
Reading option '-report' ... matched as option 'report' (generate a report)
with argument '1'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with
argument '1'.
Reading option '-t' ... matched as option 't' (record or transcode
"duration" seconds of audio/video) with argument '20'.
Reading option 'out.h264' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file /dev/video0.
Applying option f (force format) with argument v4l2.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument h264.
Successfully parsed a group of options.
Opening an input file: /dev/video0.
[video4linux2,v4l2 @ 0x1098020] fd:4 capabilities:84000001
[video4linux2,v4l2 @ 0x1098020] Current input_channel: 0, input_name:
Camera 1
[video4linux2,v4l2 @ 0x1098020] Querying the device for the current frame
size
[video4linux2,v4l2 @ 0x1098020] Setting frame size to 1280x720
[h264 @ 0x1098880] Current profile doesn't provide more RBSP data in PPS,
skipping
[video4linux2,v4l2 @ 0x1098020] All info found
Input #0, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 68191.750827, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264 (Constrained Baseline),
yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], -5 kb/s, 30 fps, 30 tbr, 1
000k tbn, 60 tbc
Successfully opened the file.
Parsing a group of options: output file out.h264.
Applying option copyinkf (copy initial non-keyframes) with argument 1.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Applying option t (record or transcode "duration" seconds of audio/video)
with argument 20.
Successfully parsed a group of options.
Opening an output file: out.h264.
Successfully opened the file.
Output #0, h264, to 'out.h264':
  Metadata:
    encoder         : Lavf55.15.100
    Stream #0:0, 0, 1/90000: Video: h264, yuvj420p, 1280x720 [SAR 1:1 DAR
16:9], q=2-31, -5 kb/s, 30 fps, 90k tbn, 30 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 6262 >= 340
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 6262 >= 3000
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 6262 >= 5653
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 125079 >= 124360
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in str
am 0: 326116 >= 325444
frame=  212 fps= 32 q=-1.0 size=    2472kB time=00:00:06.41
bitrate=3156.4kbits/s
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 713881 >= 713620
[NULL @ 0x1098880] Current profile doesn't provide more RBSP data in PPS,
skipping
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 919585 >= 917567
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 919585 >= 918262
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 1114481 >= 1114054
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 1118358 >= 1118248
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in str
am 0: 1325028 >= 1323257
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 1531328 >= 1530286
[h264 @ 0x10dded0] Application provided invalid, non monotonically
increasing dts to muxer in str
am 0: 1739153 >= 1738400
[NULL @ 0x1098880] Current profile doesn't provide more RBSP data in PPS,
skipping
No more output streams to write to, finishing.
frame=  606 fps= 31 q=-1.0 Lsize=    7512kB time=00:00:20.02
bitrate=3072.5kbits/s
video:7512kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.000000%
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x1135fc0] Statistics: 0 seeks, 608 writeouts

From neeraj01 at infosys.com  Sat Aug 31 19:29:59 2013
From: neeraj01 at infosys.com (neeraj)
Date: Sat, 31 Aug 2013 10:29:59 -0700 (PDT)
Subject: [FFmpeg-user] Error while enabling libavcodec FFMPEG
Message-ID: <1377970184349-4661028.post@n4.nabble.com>


we are getting below error while enabling libavcodec 

libavcodec/libavcodec.so: undefined reference to `x264_encoder_open_138'
collect2: ld returned 1 exit status
make: *** [ffmpeg_g] Error 1




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From arvind_kumar03 at infosys.com  Sat Aug 31 19:35:24 2013
From: arvind_kumar03 at infosys.com (arvind_kumar03)
Date: Sat, 31 Aug 2013 10:35:24 -0700 (PDT)
Subject: [FFmpeg-user] Error while enabling libavcodec FFMPEG
In-Reply-To: <1377970184349-4661028.post@n4.nabble.com>
References: <1377970184349-4661028.post@n4.nabble.com>
Message-ID: <071C8DCAAE722D4DA4EFC9959FDEB8BE40685801@BLRKECMBX11.ad.infosys.com>

This error is coming on make command of ffmpeg.

Configure is successful (below command) but make command is throwing this error.

./configure --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --enable-libmp3lame --enable-libvo_aacenc --enable-gpl --enable-libx264 --enable-shared --enable-version3

Regards,
Arvind
From: neeraj [via FFmpeg-users] [mailto:ml-node+s933282n4661028h18 at n4.nabble.com]
Sent: Saturday, August 31, 2013 1:30 PM
To: Arvind Kumar
Subject: Error while enabling libavcodec FFMPEG


we are getting below error while enabling libavcodec

libavcodec/libavcodec.so: undefined reference to `x264_encoder_open_138'
collect2: ld returned 1 exit status
make: *** [ffmpeg_g] Error 1

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