From rhleoni at yahoo.com.br  Mon Jul  1 05:13:21 2013
From: rhleoni at yahoo.com.br (Ricardo Leoni)
Date: Sun, 30 Jun 2013 20:13:21 -0700 (PDT)
Subject: [FFmpeg-user] problems with ISDB-Tb
Message-ID: <1372648401.23894.YahooMailNeo@web164501.mail.gq1.yahoo.com>

Hello,
I have some problems to open ts files that i captured with my isdb-t tuner. When i try the command: ffmpeg - i test.ts, the ffmpeg return me: "Invalid data found when processing input"
?
I think the problem is the 16 extra bytes in the file header..

Could anyone help me?

Thanks in advanced,


Ricardo Leoni
Cel.: (019) 9284-4598

From ajsinclair at gmail.com  Mon Jul  1 05:55:52 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Mon, 1 Jul 2013 13:55:52 +1000
Subject: [FFmpeg-user] Infinite loop on color bars
Message-ID: <CAOeAR17_chfS991XCAJisTdYdQz6mESMuDT=rKcrXOkXsM1Qqw@mail.gmail.com>

Does anyone know if it is possible to create an infinite loop off a test
source? This would be very useful for testing live streams

e.g.
ffmpeg -f lavfi -i smptebars -t 10000000000000  -c:v libx264 -an -f flv
rtmp://streamingserver/live/stream01

I am just make -t a large number but would like to know if there is a flag
to make this infinite.

From soho123.2012 at gmail.com  Mon Jul  1 06:26:03 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 1 Jul 2013 12:26:03 +0800
Subject: [FFmpeg-user] ffserver how to skip frame if remote client can
 not receive at moment
In-Reply-To: <CAEJHKkV_B9NJCFeY4e2Dx-XaZ2n=KK0ZVzN3=LxZmu-t6VLgsQ@mail.gmail.com>
References: <CAFd2cZVb95tvgGpaGvy=N_niJoRS3am+fZ_Ng5rQiFuVuuq1Mw@mail.gmail.com>
	<CAEJHKkV_B9NJCFeY4e2Dx-XaZ2n=KK0ZVzN3=LxZmu-t6VLgsQ@mail.gmail.com>
Message-ID: <CAFd2cZU1H+zFjRRJ+83SQaqov+qXM1zPD3XtJ8a74+8Lcmz2tg@mail.gmail.com>

2013/6/29 basteon <basteon at gmail.com>

> hi,
> you should try -vf select=1 option for that.
>
>
What is this "-vf select=1"?
Is it for ffserver ?
Is it for ffmpeg?
ffmepg I use is branch v1.2,
which version support the option?

Thanks!

From absolute01011 at gmail.com  Mon Jul  1 06:40:01 2013
From: absolute01011 at gmail.com (Nick Novia)
Date: Mon, 1 Jul 2013 00:40:01 -0400
Subject: [FFmpeg-user] problems with ISDB-Tb
In-Reply-To: <1372648401.23894.YahooMailNeo@web164501.mail.gq1.yahoo.com>
References: <1372648401.23894.YahooMailNeo@web164501.mail.gq1.yahoo.com>
Message-ID: <CAGSbOn6mFFXGk+GDqOvU=VOKRnMXeSQtc7QDWf-k9EVupjjUOA@mail.gmail.com>

stop

-Nick


On Sun, Jun 30, 2013 at 11:13 PM, Ricardo Leoni <rhleoni at yahoo.com.br>wrote:

> Hello,
> I have some problems to open ts files that i captured with my isdb-t
> tuner. When i try the command: ffmpeg - i test.ts, the ffmpeg return me:
> "Invalid data found when processing input"
>
> I think the problem is the 16 extra bytes in the file header..
>
> Could anyone help me?
>
> Thanks in advanced,
>
>
> Ricardo Leoni
> Cel.: (019) 9284-4598
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Mon Jul  1 08:17:23 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 1 Jul 2013 06:17:23 +0000 (UTC)
Subject: [FFmpeg-user] Infinite loop on color bars
References: <CAOeAR17_chfS991XCAJisTdYdQz6mESMuDT=rKcrXOkXsM1Qqw@mail.gmail.com>
Message-ID: <loom.20130701T081657-190@post.gmane.org>

Andrew Sinclair <ajsinclair <at> gmail.com> writes:

> ffmpeg -f lavfi -i smptebars -t 10000000000000  -c:v libx264 
> -an -f flv rtmp://streamingserver/live/stream01
> 
> I am just make -t a large number but would like to know if 
> there is a flag to make this infinite.

You can remove -t.

Carl Eugen


From cehoyos at ag.or.at  Mon Jul  1 08:18:09 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 1 Jul 2013 06:18:09 +0000 (UTC)
Subject: [FFmpeg-user] problems with ISDB-Tb
References: <1372648401.23894.YahooMailNeo@web164501.mail.gq1.yahoo.com>
Message-ID: <loom.20130701T081737-873@post.gmane.org>

Ricardo Leoni <rhleoni <at> yahoo.com.br> writes:

> I have some problems to open ts files that i captured 
> with my isdb-t tuner. When i try the command: 
> ffmpeg - i test.ts, the ffmpeg return me: 
> "Invalid data found when processing input"

(Complete, uncut console output missing.)

Please provide a sample.

Carl Eugen


From krueger at lesspain.de  Mon Jul  1 09:28:13 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Mon, 1 Jul 2013 09:28:13 +0200
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
Message-ID: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>

Hi,

I have files created using Apple's Compressor that are MPEG-2 files
for in 24p disguised in a 29.97 pulldown scheme. The files are OK and
work in Apple's DVD authoring solution DVD Studio Pro.

Files are detected by ffmpeg/ffprobe as follows:

ffprobe -i ~/tmp/playground/dvd-24p/Compressor-24p-NTSC.m2v -show_streams
ffprobe version N-54160-gfb0df5c Copyright (c) 2007-2013 the FFmpeg developers
  built on Jun 24 2013 11:53:38 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --disable-optimizations --disable-mmx --disable-stripping
  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
Input #0, mpegvideo, from
'/Users/krueger/tmp/playground/dvd-24p/Compressor-24p-NTSC.m2v':
  Duration: N/A, bitrate: N/A
    Stream #0:0: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 32:27
DAR 16:9], 30.25 fps, 59.94 tbr, 1200k tbn, 59.94 tbc
[STREAM]
index=0
codec_name=mpeg2video
codec_long_name=MPEG-2 video
profile=Main
codec_type=video
codec_time_base=1001/60000
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
width=720
height=480
has_b_frames=1
sample_aspect_ratio=32:27
display_aspect_ratio=16:9
pix_fmt=yuv420p
level=8
timecode=01:00:00:00
id=N/A
r_frame_rate=60000/1001
avg_frame_rate=121/4
time_base=1/1200000
start_pts=N/A
start_time=N/A
duration_ts=N/A
duration=N/A
bit_rate=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
[/STREAM]

My question is now, can I do better than this detecting the pulldown
scheme? Maybe this is rather a libav-user question if the answer is, I
have to do it programmatically, which I suspect.

(I know I should not write this but ..)

<the tool who must not be mentioned here> reports the file as (excerpt):

Frame rate                       : 23.976 fps
Scan type                        : Progressive
Scan order                       : 2:3 Pulldown

and I am looking to do the same with ffmpeg (libs or command line).

Another thing that I found noteworthy is, that when decoding the file
into still frames for testing purposes, It uses 29.97 as the output
frame rate (which AFAICS is correct if not reversing the pulldown), I
observe that there are no interlacing artefacts in the resulting
frames which I would expect if pulldown was applied. Instead frames
are repeated from time to time, which will result in the stuttering
effect well-known from watching badly converted NTSC->PAL material.
Could it be the case that ffmpeg honors the repeat flags in the stream
on a frame rather than on a field basis?

I have to add that my knowledge as far as the implications of pulldown
for the MPEG-2 bitstream are concerned, is very limited, so my
assumption may be total BS.

If this is considered a bug or you are interested in a sample, I can
submit a trac issue with a small sample file.

This is the command line and output from extracting the still frames:

ffmpeg -i ~/tmp/playground/dvd-24p/Compressor-24p-NTSC.m2v
~/tmp/playground/dvd-24p/24p-pulldown-%3d.png
ffmpeg version N-54160-gfb0df5c Copyright (c) 2000-2013 the FFmpeg developers
  built on Jun 24 2013 11:53:38 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --disable-optimizations --disable-mmx --disable-stripping
  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
Input #0, mpegvideo, from
'/Users/krueger/tmp/playground/dvd-24p/Compressor-24p-NTSC.m2v':
  Duration: N/A, bitrate: N/A
    Stream #0:0: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 32:27
DAR 16:9], 30.25 fps, 59.94 tbr, 1200k tbn, 59.94 tbc
[swscaler @ 0x7fa1ba001c00] No accelerated colorspace conversion found
from yuv420p to rgb24.
[swscaler @ 0x7fa1ba009400] No accelerated colorspace conversion found
from yuv420p to rgb24.
[swscaler @ 0x7fa1ba011c00] No accelerated colorspace conversion found
from yuv420p to rgb24.
Output #0, image2, to
'/Users/krueger/tmp/playground/dvd-24p/24p-pulldown-%3d.png':
  Metadata:
    encoder         : Lavf55.9.100
    Stream #0:0: Video: png, rgb24, 720x480 [SAR 32:27 DAR 16:9],
q=2-31, 200 kb/s, 90k tbn, 29.97 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (mpeg2video -> png)
Press [q] to stop, [?] for help
frame=   60 fps=0.0 q=0.0 Lsize=N/A time=00:00:02.00 bitrate=N/A
dup=12 drop=0
video:7894kB audio:0kB subtitle:0 global headers:0kB muxing overhead
-100.000272%

I found this in the archives:

http://permalink.gmane.org/gmane.comp.video.ffmpeg.devel/159880

But it looks as if it was never committed.

Thanks in advance,

Robert

From cehoyos at ag.or.at  Mon Jul  1 09:33:45 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 1 Jul 2013 07:33:45 +0000 (UTC)
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
References: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>
Message-ID: <loom.20130701T093313-238@post.gmane.org>

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

> My question is now, can I do better than this detecting 
> the pulldown scheme? Maybe this is rather a libav-user 
> question if the answer is, I have to do it 
> programmatically, which I suspect.
> 
> (I know I should not write this but ..)

You could add MPlayer output.

Carl Eugen


From krueger at lesspain.de  Mon Jul  1 09:50:56 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Mon, 1 Jul 2013 09:50:56 +0200
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
In-Reply-To: <loom.20130701T093313-238@post.gmane.org>
References: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>
	<loom.20130701T093313-238@post.gmane.org>
Message-ID: <CAEnqZEVKiep7-N07YXwzTbyXFoW6t=Yqoh2SMhKtZToNTrCJAg@mail.gmail.com>

On Mon, Jul 1, 2013 at 9:33 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> My question is now, can I do better than this detecting
>> the pulldown scheme? Maybe this is rather a libav-user
>> question if the answer is, I have to do it
>> programmatically, which I suspect.
>>
>> (I know I should not write this but ..)
>
> You could add MPlayer output.
>
> Carl Eugen

if it helps in diagnosing this further I will of course install
mplayer and do that. Apart from that, is there a specific reason why
Media Info is so unacceptable as yet another independent (and as all
software certainly not bugfree) source of information? From time to
time I come across cases where it does produce better output than
ffmpeg and then I find it helpful as one of several data points in
analyzing bugs etc..

Best regards,

Robert

From staalebk at ifi.uio.no  Mon Jul  1 12:06:04 2013
From: staalebk at ifi.uio.no (=?utf-8?B?U3TDpWxl?= Kristoffersen)
Date: Mon, 1 Jul 2013 12:06:04 +0200
Subject: [FFmpeg-user] h264 screenshot from multicast
Message-ID: <20130701100604.GA20160@putsch.kolbu.ws>

Hi guys, I'm trying to generate a thumbnail from a multicast stream and I'm
running into problems with h264 streams. I have written a small program to
subscribe to a multicast stream, and write out the video-pid to disk and
then run ffmpeg on it. Sometimes (1 out of 20 times perhaps) the output
will be mostly gray (as in, they are missing some or most of their
reference frames). One possible solution is to have it capture more data
and generate thumbnail for a point in time a couple of seconds into the
stream. While this works, it slows down the process a lot, and I want it to
be as fast as possible. 

Is there a way to detect that the thumbnail generated is missing some
reference data?

Here is an example file that does not work:
http://kolbu.ws/~chiller/h264fail.ts

When I run it like this:
# ffmpeg -i h264fail.ts -vframes 1 badframe.jpg
this is the output:
http://pastebin.com/KsVDqwPz
and this is the file generated:
http://kolbu.ws/~chiller/badframe.jpg

If I ask for it to decode one second first, like this:
# ffmpeg -i h264fail.ts -vframes 1 -ss 1 goodframe.jpg
this is the output:
http://pastebin.com/kwijcivf
(The exact same output from ffmpeg, but with twice the run time)
and this is the frame generated:
http://kolbu.ws/~chiller/goodframe.jpg

I can live with a couple of frames getting bad, as long as I could find a
way to detect if it was happening, but the preferred way would be to have
it decode until the frame was good, and then stop.

Anyone have any input on what I could do?

-- 
St?le Kristoffersen
staalebk at ifi.uio.no

From basteon at gmail.com  Mon Jul  1 13:39:51 2013
From: basteon at gmail.com (basteon)
Date: Mon, 1 Jul 2013 11:39:51 +0000
Subject: [FFmpeg-user] h264 screenshot from multicast
In-Reply-To: <20130701100604.GA20160@putsch.kolbu.ws>
References: <20130701100604.GA20160@putsch.kolbu.ws>
Message-ID: <CAEJHKkXWhrFEKZ-ndN9fQQzPesUCw6LoeK50QNrrNj=64TCBjA@mail.gmail.com>

Hello,
Well, with -vframes 1 you'll write one frame from this ts output and first
picture is grey. And you'd say so that sometimes happening, then it may
loose packets in the network and you must skip them with select option.
That becomes gray 20 times in minute or what period?


On 1 July 2013 10:06, St?le Kristoffersen <staalebk at ifi.uio.no> wrote:

> Hi guys, I'm trying to generate a thumbnail from a multicast stream and I'm
> running into problems with h264 streams. I have written a small program to
> subscribe to a multicast stream, and write out the video-pid to disk and
> then run ffmpeg on it. Sometimes (1 out of 20 times perhaps) the output
> will be mostly gray (as in, they are missing some or most of their
> reference frames). One possible solution is to have it capture more data
> and generate thumbnail for a point in time a couple of seconds into the
> stream. While this works, it slows down the process a lot, and I want it to
> be as fast as possible.
>
> Is there a way to detect that the thumbnail generated is missing some
> reference data?
>
> Here is an example file that does not work:
> http://kolbu.ws/~chiller/h264fail.ts
>
> When I run it like this:
> # ffmpeg -i h264fail.ts -vframes 1 badframe.jpg
> this is the output:
> http://pastebin.com/KsVDqwPz
> and this is the file generated:
> http://kolbu.ws/~chiller/badframe.jpg
>
> If I ask for it to decode one second first, like this:
> # ffmpeg -i h264fail.ts -vframes 1 -ss 1 goodframe.jpg
> this is the output:
> http://pastebin.com/kwijcivf
> (The exact same output from ffmpeg, but with twice the run time)
> and this is the frame generated:
> http://kolbu.ws/~chiller/goodframe.jpg
>
> I can live with a couple of frames getting bad, as long as I could find a
> way to detect if it was happening, but the preferred way would be to have
> it decode until the frame was good, and then stop.
>
> Anyone have any input on what I could do?
>
> --
> St?le Kristoffersen
> staalebk at ifi.uio.no
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From basteon at gmail.com  Mon Jul  1 13:42:30 2013
From: basteon at gmail.com (basteon)
Date: Mon, 1 Jul 2013 11:42:30 +0000
Subject: [FFmpeg-user] ffserver how to skip frame if remote client can
 not receive at moment
In-Reply-To: <CAFd2cZU1H+zFjRRJ+83SQaqov+qXM1zPD3XtJ8a74+8Lcmz2tg@mail.gmail.com>
References: <CAFd2cZVb95tvgGpaGvy=N_niJoRS3am+fZ_Ng5rQiFuVuuq1Mw@mail.gmail.com>
	<CAEJHKkV_B9NJCFeY4e2Dx-XaZ2n=KK0ZVzN3=LxZmu-t6VLgsQ@mail.gmail.com>
	<CAFd2cZU1H+zFjRRJ+83SQaqov+qXM1zPD3XtJ8a74+8Lcmz2tg@mail.gmail.com>
Message-ID: <CAEJHKkV4ctMWOtAPPiULWzje01GTBCHo3mZfwfS840POrbL6DQ@mail.gmail.com>

I use 0.10 and one another version I keep since 2010 supports that.


On 1 July 2013 04:26, Soho Soho123 <soho123.2012 at gmail.com> wrote:

> 2013/6/29 basteon <basteon at gmail.com>
>
> > hi,
> > you should try -vf select=1 option for that.
> >
> >
> What is this "-vf select=1"?
> Is it for ffserver ?
> Is it for ffmpeg?
> ffmepg I use is branch v1.2,
> which version support the option?
>
> Thanks!
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From staalebk at ifi.uio.no  Mon Jul  1 14:13:33 2013
From: staalebk at ifi.uio.no (=?utf-8?B?U3TDpWxl?= Kristoffersen)
Date: Mon, 1 Jul 2013 14:13:33 +0200
Subject: [FFmpeg-user] h264 screenshot from multicast
In-Reply-To: <CAEJHKkXWhrFEKZ-ndN9fQQzPesUCw6LoeK50QNrrNj=64TCBjA@mail.gmail.com>
References: <20130701100604.GA20160@putsch.kolbu.ws>
	<CAEJHKkXWhrFEKZ-ndN9fQQzPesUCw6LoeK50QNrrNj=64TCBjA@mail.gmail.com>
Message-ID: <20130701121333.GA72717@putsch.kolbu.ws>

On 2013-07-01 at 11:39, basteon wrote:
> Hello,
> Well, with -vframes 1 you'll write one frame from this ts output and first
> picture is grey. And you'd say so that sometimes happening, then it may
> loose packets in the network and you must skip them with select option.

It's not losing packets, and I know this because the CC counters are
correct.

> That becomes gray 20 times in minute or what period?

I fetch 2MB from the video PID and then point ffmpeg to it. I missed a
zero, the actual number is more like 200. So 1 out of 200 times ffmpeg
creates an image containing gray stuff. If i use -ss 1 then it always
produce good looking image.

-- 
St?le Kristoffersen
staalebk at ifi.uio.no

From stefasab at gmail.com  Mon Jul  1 19:06:17 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Mon, 1 Jul 2013 19:06:17 +0200
Subject: [FFmpeg-user] change in behavior with -vf yadif
In-Reply-To: <CAPYw7P5_kTyEDyvQ+QzWuhCQ_mXqGaKX7fN+MURm7sTvymsFLw@mail.gmail.com>
References: <1372531931.27270.YahooMailNeo@web160803.mail.bf1.yahoo.com>
	<CAPYw7P5_kTyEDyvQ+QzWuhCQ_mXqGaKX7fN+MURm7sTvymsFLw@mail.gmail.com>
Message-ID: <20130701170617.GA9226@barisone>

On date Saturday 2013-06-29 20:12:22 +0000, Paul B Mahol wrote:
> On 6/29/13, Nicholas Robbins <nickrobbins at yahoo.com> wrote:
> > Hello,
> >
> > I recently upgraded from 0.10.7 to 1.0.7.  When I did so I saw a change in
> > behavior with the -vf yadif flag.
> >
> >
> > I am using ffmpeg to converge mpeg2 interlaced video (dvd video) to x264.
> >
> >
> > The command I was using was this:
> > ffmpeg -report -v 9 -loglevel 99 -i in.mkv -c copy -vf yadif -c:v libx264
> > -preset veryfast -crf 20 /tmp/out.mkv
> >
> > With 0.10.7, I got this line in the log:    [yadif @ 0x75fdbb64a0] mode:0
> > parity:-1 auto_enable:0
> >
> >
> > With 1.0.7, yadif isn't mentioned in the log at all.
> >
> >
> > It looks, to my untrained eye that the output is deinterlaced, but I could
> > be mistaken. So this is my question:
> >
> > a) am I putting the -vf yadif option in the right place? Maybe ffmpeg got
> > pickier about where to put it?
> > b) maybe ffmpeg is now less verbose?
> 
> Yes, that message is removed.

-loglevel verbose to show it.

[...]

From onemda at gmail.com  Mon Jul  1 19:25:05 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 1 Jul 2013 17:25:05 +0000
Subject: [FFmpeg-user] change in behavior with -vf yadif
In-Reply-To: <20130701170617.GA9226@barisone>
References: <1372531931.27270.YahooMailNeo@web160803.mail.bf1.yahoo.com>
	<CAPYw7P5_kTyEDyvQ+QzWuhCQ_mXqGaKX7fN+MURm7sTvymsFLw@mail.gmail.com>
	<20130701170617.GA9226@barisone>
Message-ID: <CAPYw7P67q9Ce0oNHeZi_pcRe-v1z8G3p+nk_+u1roQ+OO5-xsg@mail.gmail.com>

On 7/1/13, Stefano Sabatini <stefasab at gmail.com> wrote:
> On date Saturday 2013-06-29 20:12:22 +0000, Paul B Mahol wrote:
>> On 6/29/13, Nicholas Robbins <nickrobbins at yahoo.com> wrote:
>> > Hello,
>> >
>> > I recently upgraded from 0.10.7 to 1.0.7.  When I did so I saw a change
>> > in
>> > behavior with the -vf yadif flag.
>> >
>> >
>> > I am using ffmpeg to converge mpeg2 interlaced video (dvd video) to
>> > x264.
>> >
>> >
>> > The command I was using was this:
>> > ffmpeg -report -v 9 -loglevel 99 -i in.mkv -c copy -vf yadif -c:v
>> > libx264
>> > -preset veryfast -crf 20 /tmp/out.mkv
>> >
>> > With 0.10.7, I got this line in the log:    [yadif @ 0x75fdbb64a0]
>> > mode:0
>> > parity:-1 auto_enable:0
>> >
>> >
>> > With 1.0.7, yadif isn't mentioned in the log at all.
>> >
>> >
>> > It looks, to my untrained eye that the output is deinterlaced, but I
>> > could
>> > be mistaken. So this is my question:
>> >
>> > a) am I putting the -vf yadif option in the right place? Maybe ffmpeg
>> > got
>> > pickier about where to put it?
>> > b) maybe ffmpeg is now less verbose?
>>
>> Yes, that message is removed.
>
> -loglevel verbose to show it.

There is nothing to show.

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

From cehoyos at ag.or.at  Tue Jul  2 02:37:55 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 2 Jul 2013 00:37:55 +0000 (UTC)
Subject: [FFmpeg-user] h264 screenshot from multicast
References: <20130701100604.GA20160@putsch.kolbu.ws>
Message-ID: <loom.20130702T023659-846@post.gmane.org>

St?le Kristoffersen <staalebk <at> ifi.uio.no> writes:

> Here is an example file that does not work:
> http://kolbu.ws/~chiller/h264fail.ts

I created ticket #2738, thank you for the sample!

Carl Eugen


From cehoyos at ag.or.at  Tue Jul  2 02:45:39 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 2 Jul 2013 00:45:39 +0000 (UTC)
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
References: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>
	<loom.20130701T093313-238@post.gmane.org>
	<CAEnqZEVKiep7-N07YXwzTbyXFoW6t=Yqoh2SMhKtZToNTrCJAg@mail.gmail.com>
Message-ID: <loom.20130702T023925-289@post.gmane.org>

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

> if it helps in diagnosing this further I will of 
> course install mplayer and do that.

MPlayer for pulldown-stream writes something like 
"Pulldown detected, changing framerate" which I 
consider helpful (in this case).

But otoh, it is a known limitation that FFmpeg does 
not handle soft-pulldown very well, so the output 
will not give a lot of information, I just wanted 
to describe a possibility to show the issue esily.

> Apart from that, is there a specific reason why
> Media Info is so unacceptable as yet another 
> independent (and as all software certainly not 
> bugfree) source of information?

It is not unacceptable, I just don't remember an 
issue / question where it was useful.
(And I incorrectly expected that you have Mplayer 
installed.)

Carl Eugen


From twear at quicinc.com  Tue Jul  2 03:07:18 2013
From: twear at quicinc.com (Wear, Tyler)
Date: Tue, 2 Jul 2013 01:07:18 +0000
Subject: [FFmpeg-user] AVCodecContext profile not being set
Message-ID: <C3A7EDDF8B714748948FE439327CBF3E15D0E0D8@NASANEXD01D.na.qualcomm.com>

So I'm using MiniDLNA, but I'm seeing the the DLNA.ORG_PN is not being set correctly for AVC_MP4_BL_CIF15_AAC_520 files.  That is mpeg4 files with AAC audio.

After some tracing in their code, I found that this is because the AVCodecContext->profile value is set to -99, which is not valid.  My understanding is that avcodec is responsible for setting this.  I disable a lot of options in ffmpeg to reduce the size as much as possible, I'm worried that I indirectly disabled the option for setting the profile...

Any help would be much appreciated.

Thanks,
Tyler

Tyler Wear | Software Engineer
Qualcomm Innovation Center, INC
Office: (858) 658-3440
Mobile: (574) 904-6777<tel:828.430.1124>
twear at quicinc.com<mailto:twear at quicinc.com>


From 550399108 at qq.com  Tue Jul  2 04:17:18 2013
From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=)
Date: Tue, 2 Jul 2013 10:17:18 +0800
Subject: [FFmpeg-user] Segmentation fault (core dumped) when run ffmpeg
Message-ID: <tencent_2D2775AA698641FB6A1FCE80@qq.com>

I use FFmpeg 1.2.1 to combiner a video and a audio together.
When I run the command like 
/usr/local/bin/ffmpeg -y -i '10806287.v' -i '10806287.a' -map 0:0 -map 1:0 -vcodec copy -acodec copy 10806287.mp4


output like this:
/usr/local/bin/ffmpeg -y -i '10806287.v' -i '10806287.a' -map 0:0 -map 1:0 -vcodec copy -acodec copy 10806287.mp4
ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  2 2013 09:13:57 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  configuration: --enable-pthreads --enable-gpl --enable-nonfree --enable-version3 --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-shared --enable-libass --enable-ffplay --enable-swscale --enable-postproc --enable-avfilter --enable-libbluray --enable-libschroedinger --enable-libspeex --enable-manpages --enable-libopenjpeg --enable-libmp3lame
  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 '10806287.v':
  Metadata:
    major_brand     : qt  
    minor_version   : 0
    compatible_brands: qt  
    creation_time   : 2013-06-26 14:47:24
  Duration: 00:02:11.54, start: 0.000000, bitrate: 958 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 480x480, 957 kb/s, 24 fps, 24 tbr, 600 tbn, 1200 tbc
    Metadata:
      rotate          : 90
      creation_time   : 2013-06-26 14:47:24
      handler_name    : Core Media Data Handler
[mp3 @ 0x1321760] max_analyze_duration 5000000 reached at 5015510 microseconds
[mp3 @ 0x1321760] Estimating duration from bitrate, this may be inaccurate
Input #1, mp3, from '10806287.a':
  Metadata:
    encoder         : Lavf53.21.0
  Duration: 00:02:11.74, start: 0.000000, bitrate: 64 kb/s
    Stream #1:0: Audio: mp3, 22050 Hz, stereo, s16p, 64 kb/s
Output #0, mp4, to '10806287.mp4':
  Metadata:
    major_brand     : qt  
    minor_version   : 0
    compatible_brands: qt  
    encoder         : Lavf54.63.104
    Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 480x480, q=2-31, 957 kb/s, 24 fps, 19200 tbn, 600 tbc
    Metadata:
      rotate          : 90
      creation_time   : 2013-06-26 14:47:24
      handler_name    : Core Media Data Handler
    Stream #0:1: Audio: mp3 (i[0][0][0] / 0x0069), 22050 Hz, stereo, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
Segmentation fault (core dumped)



but when I run this command again it printed like :
/usr/local/bin/ffmpeg -y -i '10806287.v' -i '10806287.a' -map 0:0 -map 1:0 -vcodec copy -acodec copy 10806287.mp4
ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  2 2013 09:13:57 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  configuration: --enable-pthreads --enable-gpl --enable-nonfree --enable-version3 --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libxvid --enable-shared --enable-libass --enable-ffplay --enable-swscale --enable-postproc --enable-avfilter --enable-libbluray --enable-libschroedinger --enable-libspeex --enable-manpages --enable-libopenjpeg --enable-libmp3lame
  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 '10806287.v':
  Metadata:
    major_brand     : qt  
    minor_version   : 0
    compatible_brands: qt  
    creation_time   : 2013-06-26 14:47:24
  Duration: 00:02:11.54, start: 0.000000, bitrate: 958 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 480x480, 957 kb/s, 24 fps, 24 tbr, 600 tbn, 1200 tbc
    Metadata:
      rotate          : 90
      creation_time   : 2013-06-26 14:47:24
      handler_name    : Core Media Data Handler
[mp3 @ 0x1cda760] max_analyze_duration 5000000 reached at 5015510 microseconds
[mp3 @ 0x1cda760] Estimating duration from bitrate, this may be inaccurate
Input #1, mp3, from '10806287.a':
  Metadata:
    encoder         : Lavf53.21.0
  Duration: 00:02:11.74, start: 0.000000, bitrate: 64 kb/s
    Stream #1:0: Audio: mp3, 22050 Hz, stereo, s16p, 64 kb/s
Output #0, mp4, to '10806287.mp4':
  Metadata:
    major_brand     : qt  
    minor_version   : 0
    compatible_brands: qt  
    encoder         : Lavf54.63.104
    Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 480x480, q=2-31, 957 kb/s, 24 fps, 19200 tbn, 600 tbc
    Metadata:
      rotate          : 90
      creation_time   : 2013-06-26 14:47:24
      handler_name    : Core Media Data Handler
    Stream #0:1: Audio: mp3 (i[0][0][0] / 0x0069), 22050 Hz, stereo, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 3157 fps=0.0 q=-1.0 Lsize=   16490kB time=00:02:11.73 bitrate=1025.4kbits/s    
video:15372kB audio:1029kB subtitle:0 global headers:0kB muxing overhead 0.541460%



Why the same command returns two different result?And How can I solve this problem.
Thank you!

From krueger at lesspain.de  Tue Jul  2 09:14:21 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Tue, 2 Jul 2013 09:14:21 +0200
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
In-Reply-To: <loom.20130702T023925-289@post.gmane.org>
References: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>
	<loom.20130701T093313-238@post.gmane.org>
	<CAEnqZEVKiep7-N07YXwzTbyXFoW6t=Yqoh2SMhKtZToNTrCJAg@mail.gmail.com>
	<loom.20130702T023925-289@post.gmane.org>
Message-ID: <CAEnqZEW16Mb0UXH7bicKamMV3iLmu0ihZDdSSxgGCKOJOn4sAQ@mail.gmail.com>

On Tue, Jul 2, 2013 at 2:45 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> if it helps in diagnosing this further I will of
>> course install mplayer and do that.
>
> MPlayer for pulldown-stream writes something like
> "Pulldown detected, changing framerate" which I
> consider helpful (in this case).
OK, I will try that.

>
> But otoh, it is a known limitation that FFmpeg does
> not handle soft-pulldown very well, so the output
> will not give a lot of information, I just wanted
> to describe a possibility to show the issue esily.

Thank you for that information!  So opening a feature request on Trac
and maybe offering a bounty seems like something I could do, once I
understand better what is missing.

>
>> Apart from that, is there a specific reason why
>> Media Info is so unacceptable as yet another
>> independent (and as all software certainly not
>> bugfree) source of information?
>
> It is not unacceptable, I just don't remember an
> issue / question where it was useful.
> (And I incorrectly expected that you have Mplayer
> installed.)

OK, in cases like this, it is occasionally useful when it confirms an
already existing assumption. In this case I created the file according
to Apple's guidelines for producing an NTSC-DVD from 23.967p material
and the output of MediaInfo made perfect sense, so it helped me. I
will install mplayer in any case. A more or less funny sidenote:
Quicktime does not play the pulldown file correctly either and it was
created/processed by Apple tools in every step of the workflow :-).

Thanks,

Robert

From cehoyos at ag.or.at  Tue Jul  2 11:44:34 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 2 Jul 2013 09:44:34 +0000 (UTC)
Subject: [FFmpeg-user] Segmentation fault (core dumped) when run ffmpeg
References: <tencent_2D2775AA698641FB6A1FCE80@qq.com>
Message-ID: <loom.20130702T114318-55@post.gmane.org>

??? <550399108 <at> qq.com> writes:

> ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers

Please also test current git head.

> Why the same command returns two different result?

This often indicates a hardware problem but please 
recompile with --disable-pthreads to rule out a 
race condition.

Carl Eugen


From cehoyos at ag.or.at  Tue Jul  2 11:55:09 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 2 Jul 2013 09:55:09 +0000 (UTC)
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
References: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>
	<loom.20130701T093313-238@post.gmane.org>
	<CAEnqZEVKiep7-N07YXwzTbyXFoW6t=Yqoh2SMhKtZToNTrCJAg@mail.gmail.com>
	<loom.20130702T023925-289@post.gmane.org>
	<CAEnqZEW16Mb0UXH7bicKamMV3iLmu0ihZDdSSxgGCKOJOn4sAQ@mail.gmail.com>
Message-ID: <loom.20130702T114554-233@post.gmane.org>

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

> > But otoh, it is a known limitation that FFmpeg does
> > not handle soft-pulldown very well, so the output
> > will not give a lot of information, I just wanted
> > to describe a possibility to show the issue esily.

> Quicktime does not play the pulldown file correctly 
> either and it was created/processed by Apple tools in 
  ^^^^^^
> every step of the workflow .

I realise now that what I wrote above was quite misleading:
I suspect that ffplay plays your program stream with 
(soft) pulldown fine and that ffmpeg -i file -qscale 2 out.avi 
produces a file without A/V desync.
Additionally, ffmpeg -i file -vf fps=24000/1001 -qscale 2 out.avi
produces a file with no repeated frames (I'd expect).
Everything else would be a bug that is unknown afaik.

To insert telecined frames (you called them "interlaced" 
frames in your original mails) automatically would be the 
worst thing to do imo, you can get that effect with the 
telecine filter.
I just wanted to write above that ffmpeg does not explicitly 
tell you about soft telecine and it does not allow you to 
create such streams (this is an open ticket) but using the 
libraries, it should be trivial to detect (just look at the 
timestamps). I am not convinced that another default behaviour 
will suddenly solve all problems, just run your stream through 
the usual inverse telecine chain, that should fix the output.

(Or in other words: I wonder what mediainfo says about 
streams with mixed soft and hard pulldown.)

Carl Eugen


From doenerbudenchef at yahoo.de  Tue Jul  2 12:19:45 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Tue, 2 Jul 2013 03:19:45 -0700 (PDT)
Subject: [FFmpeg-user] amerge problem with 2-pass encoding
In-Reply-To: <loom.20130628T163156-122@post.gmane.org>
References: <1372253180706-4659734.post@n4.nabble.com>
	<loom.20130626T180256-670@post.gmane.org>
	<1372331786638-4659754.post@n4.nabble.com>
	<1372333142824-4659755.post@n4.nabble.com>
	<loom.20130628T163156-122@post.gmane.org>
Message-ID: <1372760385203-4659808.post@n4.nabble.com>

This works! Thanks!

Best,
Alex



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/amerge-problem-with-2-pass-encoding-tp4659734p4659808.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From 550399108 at qq.com  Tue Jul  2 16:24:40 2013
From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=)
Date: Tue, 2 Jul 2013 22:24:40 +0800
Subject: [FFmpeg-user] =?gb18030?b?u9i4tKO6ICBTZWdtZW50YXRpb24gZmF1bHQg?=
 =?gb18030?q?=28core_dumped=29_when_run_ffmpeg?=
Message-ID: <tencent_57065B7A55D1ED6C7DBE7264@qq.com>

I tried the git one and got the same result!




------------------ ???? ------------------
???: "Carl Eugen Hoyos"<cehoyos at ag.or.at>;
????: 2013?7?2?(???) ??5:44
???: "ffmpeg-user"<ffmpeg-user at ffmpeg.org>; 

??: Re: [FFmpeg-user] Segmentation fault (core dumped) when run ffmpeg



??? <550399108 <at> qq.com> writes:

> ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers

Please also test current git head.

> Why the same command returns two different result?

This often indicates a hardware problem but please 
recompile with --disable-pthreads to rule out a 
race condition.

Carl Eugen

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

From krueger at lesspain.de  Tue Jul  2 16:29:37 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Tue, 2 Jul 2013 16:29:37 +0200
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
In-Reply-To: <loom.20130702T114554-233@post.gmane.org>
References: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>
	<loom.20130701T093313-238@post.gmane.org>
	<CAEnqZEVKiep7-N07YXwzTbyXFoW6t=Yqoh2SMhKtZToNTrCJAg@mail.gmail.com>
	<loom.20130702T023925-289@post.gmane.org>
	<CAEnqZEW16Mb0UXH7bicKamMV3iLmu0ihZDdSSxgGCKOJOn4sAQ@mail.gmail.com>
	<loom.20130702T114554-233@post.gmane.org>
Message-ID: <CAEnqZEWiTvXbOET5mp_bZqUbjqdyPkWQVoafzSmdY-DdXX0XEg@mail.gmail.com>

On Tue, Jul 2, 2013 at 11:55 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> > But otoh, it is a known limitation that FFmpeg does
>> > not handle soft-pulldown very well, so the output
>> > will not give a lot of information, I just wanted
>> > to describe a possibility to show the issue esily.
>
>> Quicktime does not play the pulldown file correctly
>> either and it was created/processed by Apple tools in
>   ^^^^^^
Oh, reading that again, I must say this was misleading. I did not mean
to say that ffplay does not play the file as that is not what I
tested.

>> every step of the workflow .
>
> I realise now that what I wrote above was quite misleading:
> I suspect that ffplay plays your program stream with
> (soft) pulldown fine and that ffmpeg -i file -qscale 2 out.avi
> produces a file without A/V desync.
> Additionally, ffmpeg -i file -vf fps=24000/1001 -qscale 2 out.avi
> produces a file with no repeated frames (I'd expect).
> Everything else would be a bug that is unknown afaik.
That's good to know and good news. I will test this.

>
> To insert telecined frames (you called them "interlaced"
> frames in your original mails) automatically would be the
> worst thing to do imo, you can get that effect with the
> telecine filter.

I am not quite sure what you mean by that. I do not want to insert
anything either. I probably did not express myself correctly.

> I just wanted to write above that ffmpeg does not explicitly
> tell you about soft telecine and it does not allow you to
> create such streams (this is an open ticket) but using the
Do you mean Ticket #2602?

> libraries, it should be trivial to detect (just look at the
> timestamps). I am not convinced that another default behaviour
OK, thanks, I will try that.

> will suddenly solve all problems, just run your stream through
Neither proposing, implying nor expecting that.

> the usual inverse telecine chain, that should fix the output.
>
> (Or in other words: I wonder what mediainfo says about
> streams with mixed soft and hard pulldown.)
I would guess nothing that makes sense. Is this combination used anywhere?

Thanks a lot! Your comments have helped me quite a bit.

Robert

From tevans.uk at googlemail.com  Tue Jul  2 16:36:40 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Tue, 2 Jul 2013 15:36:40 +0100
Subject: [FFmpeg-user]
	=?utf-8?b?5Zue5aSN77yaIFNlZ21lbnRhdGlvbiBmYXVsdCAo?=
	=?utf-8?q?core_dumped=29_when_run_ffmpeg?=
In-Reply-To: <tencent_57065B7A55D1ED6C7DBE7264@qq.com>
References: <tencent_57065B7A55D1ED6C7DBE7264@qq.com>
Message-ID: <CAFHbX1K_qP+d8jhC7z=S2uT=VNwH-EPFceJwA1yH=_uTzjZErQ@mail.gmail.com>

On Tue, Jul 2, 2013 at 3:24 PM, ??? <550399108 at qq.com> wrote:
> I tried the git one and got the same result!

Is "the same result" sometimes working, sometimes segfaulting?

If you give a program the same input, you should get the same output,
not different results. Have you tried different hardware to rule out a
hardware defect?

Cheers

Tom

From krueger at lesspain.de  Tue Jul  2 17:14:51 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Tue, 2 Jul 2013 17:14:51 +0200
Subject: [FFmpeg-user] Detecting 24 FPS pulldown in MPEG-2 DVD streams
In-Reply-To: <loom.20130702T114554-233@post.gmane.org>
References: <CAEnqZEW9AhcUy-2qjGa92F6QjgNZ=5hMG3kGmnNuMrWOk0R8rw@mail.gmail.com>
	<loom.20130701T093313-238@post.gmane.org>
	<CAEnqZEVKiep7-N07YXwzTbyXFoW6t=Yqoh2SMhKtZToNTrCJAg@mail.gmail.com>
	<loom.20130702T023925-289@post.gmane.org>
	<CAEnqZEW16Mb0UXH7bicKamMV3iLmu0ihZDdSSxgGCKOJOn4sAQ@mail.gmail.com>
	<loom.20130702T114554-233@post.gmane.org>
Message-ID: <CAEnqZEWKCne=kaQ76ki6RZYrWmS=g2F5xCT7Xdv2h_H7x+EeFg@mail.gmail.com>

On Tue, Jul 2, 2013 at 11:55 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
> Additionally, ffmpeg -i file -vf fps=24000/1001 -qscale 2 out.avi
> produces a file with no repeated frames (I'd expect).

just tested this and can confirm it works, which is great.

From johnblackjr at gmail.com  Tue Jul  2 17:18:16 2013
From: johnblackjr at gmail.com (John Black)
Date: Tue, 2 Jul 2013 09:18:16 -0600
Subject: [FFmpeg-user] HLS playback problem on Stagefright 1.2
Message-ID: <CAMLfgPyuE+X-1ZViuFf3XAK2Qsa4JVyZ8P7kn+sa1-b_4=Z8hg@mail.gmail.com>

Hi all.

I grabbed a bmw commercial MP4 from youtube and transmuxed with ffmpeg to
HLS:

ffmpeg -i bmw.mp4 -map 0 -vcodec copy -acodec copy -f segment -segment_time
10 -segment_list x -bsf h264_mp4toannexb bmw-%03d.ts

This results in 12 ts files, each 10 secs long.

This will play perfectly on VLC and QT, but on Android (using Stagefright
1.2) the video distorts badly on transition points between underlying ts
segments.  (The original MP4 plays fine.)  Other HLS streams not produced
with ffmpeg play fine on Stagefright.

I tried transcoding with libx264 as well, but I get the same problem.  It
doesn't seem to be a codec problem and I'm suspecting the problem may lie
with how ffmpeg is segmenting, or with the Stagefright player, but I'm
unsure of how to debug this.

I would be happy to follow up with any further info, eg, mpeg version
number, build parameters, -i output for mp4 and resulting ts segments, etc.

John

From fingcmo at gmail.com  Tue Jul  2 18:00:26 2013
From: fingcmo at gmail.com (Fing Cmo)
Date: Tue, 2 Jul 2013 09:00:26 -0700
Subject: [FFmpeg-user] HLS playback problem on Stagefright 1.2
In-Reply-To: <CAMLfgPyuE+X-1ZViuFf3XAK2Qsa4JVyZ8P7kn+sa1-b_4=Z8hg@mail.gmail.com>
References: <CAMLfgPyuE+X-1ZViuFf3XAK2Qsa4JVyZ8P7kn+sa1-b_4=Z8hg@mail.gmail.com>
Message-ID: <CABH_zws4jxbbZpciY5ZadZgHATHqQj9Mp5mcwjyBy8Xy6z0HRQ@mail.gmail.com>

John,

Try this and report on your successes :)
i.e. follow the way he generates his and plays it back.
http://blog.walterebert.com/creating-on-hls-video-stream-with-ffmpeg/

Ad.
.....................................



On Tue, Jul 2, 2013 at 8:18 AM, John Black <johnblackjr at gmail.com> wrote:

> Hi all.
>
> I grabbed a bmw commercial MP4 from youtube and transmuxed with ffmpeg to
> HLS:
>
> ffmpeg -i bmw.mp4 -map 0 -vcodec copy -acodec copy -f segment -segment_time
> 10 -segment_list x -bsf h264_mp4toannexb bmw-%03d.ts
>
> This results in 12 ts files, each 10 secs long.
>
> This will play perfectly on VLC and QT, but on Android (using Stagefright
> 1.2) the video distorts badly on transition points between underlying ts
> segments.  (The original MP4 plays fine.)  Other HLS streams not produced
> with ffmpeg play fine on Stagefright.
>
> I tried transcoding with libx264 as well, but I get the same problem.  It
> doesn't seem to be a codec problem and I'm suspecting the problem may lie
> with how ffmpeg is segmenting, or with the Stagefright player, but I'm
> unsure of how to debug this.
>
> I would be happy to follow up with any further info, eg, mpeg version
> number, build parameters, -i output for mp4 and resulting ts segments, etc.
>
> John
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From twear at quicinc.com  Wed Jul  3 00:30:43 2013
From: twear at quicinc.com (Wear, Tyler)
Date: Tue, 2 Jul 2013 22:30:43 +0000
Subject: [FFmpeg-user] AVCodecContext profile not being set
In-Reply-To: <C3A7EDDF8B714748948FE439327CBF3E15D0E0D8@NASANEXD01D.na.qualcomm.com>
References: <C3A7EDDF8B714748948FE439327CBF3E15D0E0D8@NASANEXD01D.na.qualcomm.com>
Message-ID: <C3A7EDDF8B714748948FE439327CBF3E15D0EE81@NASANEXD01D.na.qualcomm.com>

A little more information.

MiniDLNA is using avformat_open_input to retrieve the AVFormatContext, but when the AVCodecContext for the video stream is taken from the AVFormatContext the profile value is always -99 instead of 66 for the h.264 encoded aac 520 file.

The more I look into this the more it looks like a possible FFMPEG error. 

Thanks,
Tyler

-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Wear, Tyler
Sent: Monday, July 01, 2013 6:07 PM
To: ffmpeg-user at ffmpeg.org
Subject: [FFmpeg-user] AVCodecContext profile not being set

So I'm using MiniDLNA, but I'm seeing the the DLNA.ORG_PN is not being set correctly for AVC_MP4_BL_CIF15_AAC_520 files.  That is mpeg4 files with AAC audio.

After some tracing in their code, I found that this is because the AVCodecContext->profile value is set to -99, which is not valid.  My understanding is that avcodec is responsible for setting this.  I disable a lot of options in ffmpeg to reduce the size as much as possible, I'm worried that I indirectly disabled the option for setting the profile...

Any help would be much appreciated.

Thanks,
Tyler

Tyler Wear | Software Engineer
Qualcomm Innovation Center, INC
Office: (858) 658-3440
Mobile: (574) 904-6777<tel:828.430.1124> twear at quicinc.com<mailto:twear at quicinc.com>

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

From twear at quicinc.com  Wed Jul  3 01:03:30 2013
From: twear at quicinc.com (Wear, Tyler)
Date: Tue, 2 Jul 2013 23:03:30 +0000
Subject: [FFmpeg-user] AVCodecContext profile not being set
In-Reply-To: <C3A7EDDF8B714748948FE439327CBF3E15D0EE81@NASANEXD01D.na.qualcomm.com>
References: <C3A7EDDF8B714748948FE439327CBF3E15D0E0D8@NASANEXD01D.na.qualcomm.com>
	<C3A7EDDF8B714748948FE439327CBF3E15D0EE81@NASANEXD01D.na.qualcomm.com>
Message-ID: <C3A7EDDF8B714748948FE439327CBF3E15D0EFB5@NASANEXD01D.na.qualcomm.com>

FYI, I tested with the nothing disabled from FFMPEG and I am still seeing the issue.

Using ffmpeg 1.1

Tyler

-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Wear, Tyler
Sent: Tuesday, July 02, 2013 3:31 PM
To: FFmpeg user questions
Subject: Re: [FFmpeg-user] AVCodecContext profile not being set

A little more information.

MiniDLNA is using avformat_open_input to retrieve the AVFormatContext, but when the AVCodecContext for the video stream is taken from the AVFormatContext the profile value is always -99 instead of 66 for the h.264 encoded aac 520 file.

The more I look into this the more it looks like a possible FFMPEG error. 

Thanks,
Tyler

-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Wear, Tyler
Sent: Monday, July 01, 2013 6:07 PM
To: ffmpeg-user at ffmpeg.org
Subject: [FFmpeg-user] AVCodecContext profile not being set

So I'm using MiniDLNA, but I'm seeing the the DLNA.ORG_PN is not being set correctly for AVC_MP4_BL_CIF15_AAC_520 files.  That is mpeg4 files with AAC audio.

After some tracing in their code, I found that this is because the AVCodecContext->profile value is set to -99, which is not valid.  My understanding is that avcodec is responsible for setting this.  I disable a lot of options in ffmpeg to reduce the size as much as possible, I'm worried that I indirectly disabled the option for setting the profile...

Any help would be much appreciated.

Thanks,
Tyler

Tyler Wear | Software Engineer
Qualcomm Innovation Center, INC
Office: (858) 658-3440
Mobile: (574) 904-6777<tel:828.430.1124> twear at quicinc.com<mailto:twear at quicinc.com>

_______________________________________________
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 550399108 at qq.com  Wed Jul  3 03:15:30 2013
From: 550399108 at qq.com (=?gb18030?B?uqvI8MvJ?=)
Date: Wed, 3 Jul 2013 09:15:30 +0800
Subject: [FFmpeg-user] =?gb18030?b?u9i4tKO6ILvYuLSjuiBTZWdtZW50YXRpb24g?=
 =?gb18030?q?fault_=28core_dumped=29_when_run_ffmpeg?=
Message-ID: <tencent_244F2997404E6E3052B8FBA7@qq.com>

Hi, Tom.
'The same result' means I uninstalled the ffmpeg at version 1.2.1 and complied from the latest source from git.
When I run the command I used before.It sometimes working and sometimes segfaulting.


I complied this source at three computers and got the same result as above.


I used to run this command by ffmpeg at version r26400. It worked well.


But it can save the rotate information of the video.So I want to use a newer one.




------------------ ???? ------------------
???: "Tom Evans"<tevans.uk at googlemail.com>;
????: 2013?7?2?(???) ??10:36
???: "FFmpeg user questions"<ffmpeg-user at ffmpeg.org>; 

??: Re: [FFmpeg-user]??? Segmentation fault (core dumped) when run ffmpeg



On Tue, Jul 2, 2013 at 3:24 PM, ??? <550399108 at qq.com> wrote:
> I tried the git one and got the same result!

Is "the same result" sometimes working, sometimes segfaulting?

If you give a program the same input, you should get the same output,
not different results. Have you tried different hardware to rule out a
hardware defect?

Cheers

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

From doenerbudenchef at yahoo.de  Wed Jul  3 14:52:38 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Wed, 3 Jul 2013 05:52:38 -0700 (PDT)
Subject: [FFmpeg-user] MXF with XDCAM HD422
Message-ID: <1372855958346-4659820.post@n4.nabble.com>

Hi all,

I want to encode a prores 422 file (1080p/25) to MXF with OP1a (SMPTE 378M).
Compression should be a XDCAM HD422 (MPEG2) with 4:2:2, 50 Mbit/s and
Long-GOP. I tried the following and got several error messages:

Command:

ffmpeg -i test.mov -pix_fmt yuv422p -vcodec mpeg2video -non_linear_quant 1
-flags +ildct+ilme -top 1 -dc 10 -intra_vlc 1 -qmax 3 -lmin "1*QP2LAMBDA"
-vtag xd5c -rc_max_vbv_use 1 -rc_min_vbv_use 1 -g 12 -b:v 50000k -minrate
50000k -maxrate 50000k -bufsize 8000k -acodec pcm_s16le -ar 48000 -bf 2 -ac
2 -f mxf_d10 output.mxf

Output:

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 --
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-
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
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003be4a0] Stream #1: not enough frames to
estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003be4a0] Stream #2: not enough frames to
estimate rate; consider increasing probesize
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2013-03-18 13:06:32
  Duration: 00:03:08.96, start: 0.000000, bitrate: 184313 kb/s
    Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le,
1920x1080, 182769 kb/s, SAR 1920:1920 DAR 16:9, 25 fps, 25 tbr, 25 tbn, 25
tbc
    Metadata:
      creation_time   : 2013-03-18 13:06:32
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:01
    Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz,
stereo, s16, 1536 kb/s
    Metadata:
      creation_time   : 2013-03-18 13:06:32
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:01
    Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      creation_time   : 2013-03-18 13:18:27
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:01
File 'output.mxf' already exists. Overwrite ? [y/N] y
Output #0, mxf_d10, to 'output.mxf':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    encoder         : Lavf55.10.100
    Stream #0:0(eng): Video: mpeg2video (xd5c / 0x63356478), yuv422p,
1920x1080 [SAR 1:1 DAR 16:9], q=2-3, 50000 kb/s, 25 tbn, 25 tbc
    Metadata:
      creation_time   : 2013-03-18 13:06:32
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:01
    Stream #0:1(eng): Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    Metadata:
      creation_time   : 2013-03-18 13:06:32
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:01
Stream mapping:
  Stream #0:0 -> #0:0 (prores -> mpeg2video)
  Stream #0:1 -> #0:1 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
cannot fill d-10 video packete=   32999kB time=00:00:05.24
bitrate=51589.3kbits/s
cannot fill d-10 video packete=   41614kB time=00:00:06.04
bitrate=56440.7kbits/s
cannot fill d-10 video packete=   41971kB time=00:00:06.40
bitrate=53722.3kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 17 times
cannot fill d-10 video packete= 1001931kB time=00:02:12.04
bitrate=62161.6kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 8 times
cannot fill d-10 video packete= 1012526kB time=00:02:13.28
bitrate=62234.5kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
cannot fill d-10 video packete= 1014452kB time=00:02:13.56
bitrate=62222.1kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 10 times
cannot fill d-10 video packete= 1023922kB time=00:02:14.52
bitrate=62354.8kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 1 times
cannot fill d-10 video packete= 1024994kB time=00:02:14.84
bitrate=62271.9kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 10 times
cannot fill d-10 video packete= 1034857kB time=00:02:15.88
bitrate=62390.0kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
cannot fill d-10 video packete= 1035996kB time=00:02:16.20
bitrate=62311.9kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 10 times
cannot fill d-10 video packete= 1045977kB time=00:02:17.32
bitrate=62399.1kbits/s
cannot fill d-10 video packete= 1047529kB time=00:02:17.64
bitrate=62346.4kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 8 times
cannot fill d-10 video packete= 1057005kB time=00:02:18.80
bitrate=62384.6kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 1 times
cannot fill d-10 video packete= 1059597kB time=00:02:19.12
bitrate=62393.8kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 8 times
cannot fill d-10 video packete= 1068043kB time=00:02:19.88
bitrate=62549.4kbits/s
cannot fill d-10 video packete= 1068377kB time=00:02:20.20
bitrate=62426.1kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 14 times
cannot fill d-10 video packete= 1079824kB time=00:02:21.60
bitrate=62471.2kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 13 times
cannot fill d-10 video packete= 1090319kB time=00:02:22.64
bitrate=62618.4kbits/s
cannot fill d-10 video packete= 1091266kB time=00:02:23.00
bitrate=62515.0kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 10 times
cannot fill d-10 video packete= 1101488kB time=00:02:24.28
bitrate=62540.8kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 6 times
cannot fill d-10 video packete= 1107846kB time=00:02:24.60
bitrate=62762.6kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 3 times
cannot fill d-10 video packete= 1113141kB time=00:02:25.44
bitrate=62698.4kbits/s
cannot fill d-10 video packete= 1114526kB time=00:02:25.80
bitrate=62621.4kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 9 times
cannot fill d-10 video packete= 1124328kB time=00:02:27.00
bitrate=62656.4kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 12 times
cannot fill d-10 video packete= 1135578kB time=00:02:28.20
bitrate=62771.0kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
cannot fill d-10 video packete= 1137198kB time=00:02:28.56
bitrate=62708.1kbits/s
[mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
    Last message repeated 9 times
frame= 4457 fps= 17 q=1.0 Lsize= 1366084kB time=00:02:58.24
bitrate=62785.9kbits/s


Thanks,
Alex







--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/MXF-with-XDCAM-HD422-tp4659820.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From krueger at lesspain.de  Wed Jul  3 17:50:54 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Wed, 3 Jul 2013 17:50:54 +0200
Subject: [FFmpeg-user] MXF with XDCAM HD422
In-Reply-To: <1372855958346-4659820.post@n4.nabble.com>
References: <1372855958346-4659820.post@n4.nabble.com>
Message-ID: <CAEnqZEXVWoX9MQ075+iuBpxE2-kkGZbjE4+r662RHiixB0bhsg@mail.gmail.com>

On Wed, Jul 3, 2013 at 2:52 PM, Alex <doenerbudenchef at yahoo.de> wrote:
> Hi all,
>
> I want to encode a prores 422 file (1080p/25) to MXF with OP1a (SMPTE 378M).
> Compression should be a XDCAM HD422 (MPEG2) with 4:2:2, 50 Mbit/s and
> Long-GOP. I tried the following and got several error messages:
>
> Command:
>
> ffmpeg -i test.mov -pix_fmt yuv422p -vcodec mpeg2video -non_linear_quant 1
> -flags +ildct+ilme -top 1 -dc 10 -intra_vlc 1 -qmax 3 -lmin "1*QP2LAMBDA"
> -vtag xd5c -rc_max_vbv_use 1 -rc_min_vbv_use 1 -g 12 -b:v 50000k -minrate
> 50000k -maxrate 50000k -bufsize 8000k -acodec pcm_s16le -ar 48000 -bf 2 -ac
> 2 -f mxf_d10 output.mxf
>
> Output:
>
> 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 --
> 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-
> 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
> [mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003be4a0] Stream #1: not enough frames to
> estimate rate; consider increasing probesize
> [mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003be4a0] Stream #2: not enough frames to
> estimate rate; consider increasing probesize
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov':
>   Metadata:
>     major_brand     : qt
>     minor_version   : 537199360
>     compatible_brands: qt
>     creation_time   : 2013-03-18 13:06:32
>   Duration: 00:03:08.96, start: 0.000000, bitrate: 184313 kb/s
>     Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le,
> 1920x1080, 182769 kb/s, SAR 1920:1920 DAR 16:9, 25 fps, 25 tbr, 25 tbn, 25
> tbc
>     Metadata:
>       creation_time   : 2013-03-18 13:06:32
>       handler_name    : Apple Alias Data Handler
>       timecode        : 00:00:00:01
>     Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz,
> stereo, s16, 1536 kb/s
>     Metadata:
>       creation_time   : 2013-03-18 13:06:32
>       handler_name    : Apple Alias Data Handler
>       timecode        : 00:00:00:01
>     Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
>     Metadata:
>       creation_time   : 2013-03-18 13:18:27
>       handler_name    : Apple Alias Data Handler
>       timecode        : 00:00:00:01
> File 'output.mxf' already exists. Overwrite ? [y/N] y
> Output #0, mxf_d10, to 'output.mxf':
>   Metadata:
>     major_brand     : qt
>     minor_version   : 537199360
>     compatible_brands: qt
>     encoder         : Lavf55.10.100
>     Stream #0:0(eng): Video: mpeg2video (xd5c / 0x63356478), yuv422p,
> 1920x1080 [SAR 1:1 DAR 16:9], q=2-3, 50000 kb/s, 25 tbn, 25 tbc
>     Metadata:
>       creation_time   : 2013-03-18 13:06:32
>       handler_name    : Apple Alias Data Handler
>       timecode        : 00:00:00:01
>     Stream #0:1(eng): Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
>     Metadata:
>       creation_time   : 2013-03-18 13:06:32
>       handler_name    : Apple Alias Data Handler
>       timecode        : 00:00:00:01
> Stream mapping:
>   Stream #0:0 -> #0:0 (prores -> mpeg2video)
>   Stream #0:1 -> #0:1 (pcm_s16le -> pcm_s16le)
> Press [q] to stop, [?] for help
> cannot fill d-10 video packete=   32999kB time=00:00:05.24
> bitrate=51589.3kbits/s
> cannot fill d-10 video packete=   41614kB time=00:00:06.04
> bitrate=56440.7kbits/s
> cannot fill d-10 video packete=   41971kB time=00:00:06.40
> bitrate=53722.3kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 17 times
> cannot fill d-10 video packete= 1001931kB time=00:02:12.04
> bitrate=62161.6kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 8 times
> cannot fill d-10 video packete= 1012526kB time=00:02:13.28
> bitrate=62234.5kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
> cannot fill d-10 video packete= 1014452kB time=00:02:13.56
> bitrate=62222.1kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 10 times
> cannot fill d-10 video packete= 1023922kB time=00:02:14.52
> bitrate=62354.8kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 1 times
> cannot fill d-10 video packete= 1024994kB time=00:02:14.84
> bitrate=62271.9kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 10 times
> cannot fill d-10 video packete= 1034857kB time=00:02:15.88
> bitrate=62390.0kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
> cannot fill d-10 video packete= 1035996kB time=00:02:16.20
> bitrate=62311.9kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 10 times
> cannot fill d-10 video packete= 1045977kB time=00:02:17.32
> bitrate=62399.1kbits/s
> cannot fill d-10 video packete= 1047529kB time=00:02:17.64
> bitrate=62346.4kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 8 times
> cannot fill d-10 video packete= 1057005kB time=00:02:18.80
> bitrate=62384.6kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 1 times
> cannot fill d-10 video packete= 1059597kB time=00:02:19.12
> bitrate=62393.8kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 8 times
> cannot fill d-10 video packete= 1068043kB time=00:02:19.88
> bitrate=62549.4kbits/s
> cannot fill d-10 video packete= 1068377kB time=00:02:20.20
> bitrate=62426.1kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 14 times
> cannot fill d-10 video packete= 1079824kB time=00:02:21.60
> bitrate=62471.2kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 13 times
> cannot fill d-10 video packete= 1090319kB time=00:02:22.64
> bitrate=62618.4kbits/s
> cannot fill d-10 video packete= 1091266kB time=00:02:23.00
> bitrate=62515.0kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 10 times
> cannot fill d-10 video packete= 1101488kB time=00:02:24.28
> bitrate=62540.8kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 6 times
> cannot fill d-10 video packete= 1107846kB time=00:02:24.60
> bitrate=62762.6kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 3 times
> cannot fill d-10 video packete= 1113141kB time=00:02:25.44
> bitrate=62698.4kbits/s
> cannot fill d-10 video packete= 1114526kB time=00:02:25.80
> bitrate=62621.4kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 9 times
> cannot fill d-10 video packete= 1124328kB time=00:02:27.00
> bitrate=62656.4kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 12 times
> cannot fill d-10 video packete= 1135578kB time=00:02:28.20
> bitrate=62771.0kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
> cannot fill d-10 video packete= 1137198kB time=00:02:28.56
> bitrate=62708.1kbits/s
> [mxf_d10 @ 000000000249e7a0] cannot fill d-10 video packet
>     Last message repeated 9 times
> frame= 4457 fps= 17 q=1.0 Lsize= 1366084kB time=00:02:58.24
> bitrate=62785.9kbits/s
>

Why the "-f mxf_d10" in there? I must admit, I was not aware of this
muxer but D10 is an SD format and certainly not XDCAM. I just looked
at your output briefly but the error messages contained a lot of
"d-10" stuff. I'd try again just without the "-f mxf_d10".

HTH,

Robert

From grovher at gmail.com  Wed Jul  3 18:55:32 2013
From: grovher at gmail.com (grovher)
Date: Wed, 3 Jul 2013 09:55:32 -0700 (PDT)
Subject: [FFmpeg-user] how to add more than 1 stream of audio to a TS
 filesetting the language?
In-Reply-To: <CAKuDJW20mr1ya7bKtLmwTf9r-U_pKrAYozL_7Ft5gE+F__cOfg@mail.gmail.com>
References: <CAKuDJW3iX0jbq0gwMXr-T1uEHZLS-qSBDnj3yud5zEFdHJcA5A@mail.gmail.com>
	<F7BB948CF403AA4092626040F236AAEE13FA65@bbcxues46.national.core.bbc.co.uk>
	<CAKuDJW2UCy0U=agKn=gENwgck5nk4bcq-256ZgofxNjJLUQt9w@mail.gmail.com>
	<loom.20120621T095304-386@post.gmane.org>
	<CAKuDJW20mr1ya7bKtLmwTf9r-U_pKrAYozL_7Ft5gE+F__cOfg@mail.gmail.com>
Message-ID: <1372870532784-4659822.post@n4.nabble.com>

how to create Elementary Stream (.es) or generate paquet elementary Stream
(P.E.S.) of video and audio separately (x264 and AAC), from videoTdT.mp4?

information:
ffmpeg -i videoTdT.mp4
------------------------------------------------------------------------------------------------------------------------ 
ffmpeg version git-2013-07-01-b791a08 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul  1 2013 17:22:27 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
  configuration: --prefix=/home/grovher/ffmpeg_build
--extra-cflags=-I/home/grovher/ffmpeg_build/include
--extra-ldflags=-L/home/grovher/ffmpeg_build/lib --bindir=/home/grovher/bin
--extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac
--enable-libx264 --enable-nonfree
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.100 / 55. 18.100
  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 'videoTdT.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: isomavc1mp42
    creation_time   : 2010-04-07 18:43:22
  Duration: 00:02:09.00, start: 0.000000, bitrate: 2102 kb/s
    Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
fltp, 110 kb/s
    Metadata:
      creation_time   : 2010-04-07 18:43:22
      handler_name    : (C) 2007 Google Inc. v08.13.2007.
    Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], 1989 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95
tbc
    Metadata:
      creation_time   : 2010-04-07 18:43:23
      handler_name    : (C) 2007 Google Inc. v08.13.2007.
------------------------------------------------------------------------------------------------------------------------

can you help?





--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/how-to-add-more-than-1-stream-of-audio-to-a-TS-file-setting-the-language-tp4651132p4659822.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From oliver at fromme.com  Wed Jul  3 16:44:30 2013
From: oliver at fromme.com (Oliver Fromme)
Date: Wed, 3 Jul 2013 16:44:30 +0200 (CEST)
Subject: [FFmpeg-user] Hardcoding subtitles
Message-ID: <201307031444.r63EiU0R058599@haluter.fromme.com>

I'm trying to "hardcode" forced subtitles into a video, so they are
always displayed, independent of the player's subtitle support.
In the end I have been successful, but it was kind of an adventure.
I'm trying to describe the problems that I encountered, maybe this
can help to make ffmpeg better.

My ffmpeg version is a recent git snapshot, just a few days old.

These are my input files:  Video and audio are in an MPEG2 PS file
that comes from a DVD (via "mplayer -dumpstream").  The subtitles
are in VOBSUB format (I think ffmpeg calls this "dvdsub"), i.e.
.idx + .sub files.  These have been retrieved from the same DVD
with "mencoder -vobsubout".

The first problem was that ffmpeg didn't want to read the VOBSUB
files directly, no matter what I tried.  It seemed to be unable
to detect the format automatically, and I was unable to find out
how to specify it on the command line.  I probably tried more
than 20 variantions, without success.  However, it's not a big
deal:  I just wrapped the VOBSUB stream into an MKV container
(using mkvmerge), and ffmpeg happily accepted it.  So, my input
files are now the MPEG2 PS file (same as before) and an MKV file
containing just the VOBSUB subtitle stream.

According to the excellent manual page, the overlay filter can
be used for hardcoding subtitles, so I tried that.  The filter
specification was simple:

-filter_complex "[0:v][1:s]overlay"

where file 0 is the MPEG2 PS and file 1 is the MKV file with the
VOBSUB stream.  If it matters (I don't think it does), here's
the complete command line:

ffmpeg1 -loglevel warning -stats -i dvdstream-3.mpg -i subtitles-3-1.mkv \
        -filter_complex "[0:v][1:s]overlay" \
        -c:v libx264 -preset veryslow -tune film -crf 18 -level 4.1 \
        -codec:a copy -map 0:a -sn dvdstream-3-vid.mkv

Basically, that command worked and created a video with hardcoded
subtitles.  However, there was one small problem:  The duration
of the last subtitle was "infinite", i.e. the very last subtitle
stayed on the screen until end of the video.  It was visible for
several minutes including the whole end credits of the movie.
Not nice.  So I went back to the manual page and re-read the
section about the overlay filter.

I noticed the "repeatlast" option of this filter, but the last
sentence said something like "A value of 0 ... is enabled by
default", so I thought that 0 was the default.  But I didn't
look closely enough ...  Upon further inspection, it's just the
other way around.  Of course that's my mistake, but maybe the
manual page could be made a little clearer?  Something like
"The default value is 1" would probably be clear enough.

Anyway, I tried to disable that flag:

-filter_complex "[0:v][1:s]overlay=repeatlast=0"

It did make a difference, but not what I expected:  Now the last
subtitle didn't appear at all!  I think this is a bug, although
I wouldn't rule out the possibility that I did something wrong
again.  I tried with a different DVD, but same result:  When
repeatlast=0 is specified, the last VOBSUB subtitle is omitted
completely.

So, now I pondered over a workaround for that problem.
I skimmed over some examples in the manual page that gave me an
Idea:  I could try to use the gte() function to forcefully end
the last subtitle at the desired time stamp.

First I used mplayer to look for the exact position where the
last subtitle was supposed to end.  In this case that was at
t = 3375 (that's 56 minutes and 15 seconds).  I reverted the
"repeatlast" flag to the default so the last subtitle was
visible, and used gte() to cancel it at the proper time:

-filter_complex "[0:v][1:s]overlay=x='if(gt(t,3375),NAN,x)'"

The idee was:  If the time is beyond 3375, set the x value of
the subtitle frames to NAN (effectively disabling them),
otherwise keep them at their original x value.  At least I
thought that's what the x variable meant.

Obviously I was wrong, because the resulting video did not
contain *any* subtitles at all.  No subtitles anywhere.
Either I misunderstood the description of the x variable in
the manual page (and in this case the manual page should be
clarified), or there's another bug.  In my next attempt I
replaced the x variable with 0, which should be the default
for x anyway:

-filter_complex "[0:v][1:s]overlay=x='if(gt(t,3375),NAN,0)'"

Finally, that produced the expected result, and I'm happy now.

Having to find out the ending time stamp for the last subtitle
is a little cumbersome and can't be done automatically (I don't
know a CLI tool that's able to display durations or ending
time stamps from VOBSUB streams).  Therefore I would appreciate
if the bug with repeatlast=0 could be fixed, so my workaround
won't be necessary anymore.

Unfortunately I can't easily upload sample files, because the
full movie is too large, and I don't have appropriate software
to cut a short clip from it that also retains the subtitles
while keeping them in sync with the movie.

Best regards
   Oliver

-- 

From compuvision.sr at gmail.com  Wed Jul  3 21:50:11 2013
From: compuvision.sr at gmail.com (Albert Scholtalbers)
Date: Wed, 3 Jul 2013 16:50:11 -0300
Subject: [FFmpeg-user] Hardcoding subtitles
In-Reply-To: <201307031444.r63EiU0R058599@haluter.fromme.com>
References: <201307031444.r63EiU0R058599@haluter.fromme.com>
Message-ID: <CAECKL3q_FpoW6UqEnzy=ciKMfGX+BbJEYZPYcopd2XCD2X5AWA@mail.gmail.com>

>In the end I have been successful, but it was kind of an adventure.
>I'm trying to describe the problems that I encountered, maybe this
> can help to make ffmpeg better.

Thank you verymuch for this post! Maybe time to create a Wiki page.
It will help me a lot to encode other video's with hardcodes subs.

Yours sincerely,

Albert

From onemda at gmail.com  Wed Jul  3 21:57:28 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 3 Jul 2013 19:57:28 +0000
Subject: [FFmpeg-user] Hardcoding subtitles
In-Reply-To: <201307031444.r63EiU0R058599@haluter.fromme.com>
References: <201307031444.r63EiU0R058599@haluter.fromme.com>
Message-ID: <CAPYw7P4atHAHY=sKM7=30uDdVWwheusEzsE9DVLv27Q3ibZuWw@mail.gmail.com>

On 7/3/13, Oliver Fromme <oliver at fromme.com> wrote:
> I'm trying to "hardcode" forced subtitles into a video, so they are
> always displayed, independent of the player's subtitle support.
> In the end I have been successful, but it was kind of an adventure.
> I'm trying to describe the problems that I encountered, maybe this
> can help to make ffmpeg better.
>
> My ffmpeg version is a recent git snapshot, just a few days old.
>
> These are my input files:  Video and audio are in an MPEG2 PS file
> that comes from a DVD (via "mplayer -dumpstream").  The subtitles
> are in VOBSUB format (I think ffmpeg calls this "dvdsub"), i.e.
> .idx + .sub files.  These have been retrieved from the same DVD
> with "mencoder -vobsubout".
>
> The first problem was that ffmpeg didn't want to read the VOBSUB
> files directly, no matter what I tried.  It seemed to be unable
> to detect the format automatically, and I was unable to find out
> how to specify it on the command line.  I probably tried more
> than 20 variantions, without success.  However, it's not a big
> deal:  I just wrapped the VOBSUB stream into an MKV container

Yes, it is big deal, if you want bug fixed, provide sample and open bug report.

You don't need to upload full file. Just enough so bug can be reproduced.

> (using mkvmerge), and ffmpeg happily accepted it.  So, my input
> files are now the MPEG2 PS file (same as before) and an MKV file
> containing just the VOBSUB subtitle stream.
>
> According to the excellent manual page, the overlay filter can
> be used for hardcoding subtitles, so I tried that.  The filter
> specification was simple:
>
> -filter_complex "[0:v][1:s]overlay"
>
> where file 0 is the MPEG2 PS and file 1 is the MKV file with the
> VOBSUB stream.  If it matters (I don't think it does), here's
> the complete command line:
>
> ffmpeg1 -loglevel warning -stats -i dvdstream-3.mpg -i subtitles-3-1.mkv \
>         -filter_complex "[0:v][1:s]overlay" \
>         -c:v libx264 -preset veryslow -tune film -crf 18 -level 4.1 \
>         -codec:a copy -map 0:a -sn dvdstream-3-vid.mkv
>
> Basically, that command worked and created a video with hardcoded
> subtitles.  However, there was one small problem:  The duration
> of the last subtitle was "infinite", i.e. the very last subtitle
> stayed on the screen until end of the video.  It was visible for
> several minutes including the whole end credits of the movie.
> Not nice.  So I went back to the manual page and re-read the
> section about the overlay filter.
>
> I noticed the "repeatlast" option of this filter, but the last
> sentence said something like "A value of 0 ... is enabled by
> default", so I thought that 0 was the default.  But I didn't
> look closely enough ...  Upon further inspection, it's just the
> other way around.  Of course that's my mistake, but maybe the
> manual page could be made a little clearer?  Something like
> "The default value is 1" would probably be clear enough.
>
> Anyway, I tried to disable that flag:
>
> -filter_complex "[0:v][1:s]overlay=repeatlast=0"
>
> It did make a difference, but not what I expected:  Now the last
> subtitle didn't appear at all!  I think this is a bug, although
> I wouldn't rule out the possibility that I did something wrong
> again.  I tried with a different DVD, but same result:  When
> repeatlast=0 is specified, the last VOBSUB subtitle is omitted
> completely.
>
> So, now I pondered over a workaround for that problem.
> I skimmed over some examples in the manual page that gave me an
> Idea:  I could try to use the gte() function to forcefully end
> the last subtitle at the desired time stamp.
>
> First I used mplayer to look for the exact position where the
> last subtitle was supposed to end.  In this case that was at
> t = 3375 (that's 56 minutes and 15 seconds).  I reverted the
> "repeatlast" flag to the default so the last subtitle was
> visible, and used gte() to cancel it at the proper time:
>
> -filter_complex "[0:v][1:s]overlay=x='if(gt(t,3375),NAN,x)'"
>
> The idee was:  If the time is beyond 3375, set the x value of
> the subtitle frames to NAN (effectively disabling them),
> otherwise keep them at their original x value.  At least I
> thought that's what the x variable meant.
>
> Obviously I was wrong, because the resulting video did not
> contain *any* subtitles at all.  No subtitles anywhere.
> Either I misunderstood the description of the x variable in
> the manual page (and in this case the manual page should be
> clarified), or there's another bug.  In my next attempt I
> replaced the x variable with 0, which should be the default
> for x anyway:
>
> -filter_complex "[0:v][1:s]overlay=x='if(gt(t,3375),NAN,0)'"
>
> Finally, that produced the expected result, and I'm happy now.
>
> Having to find out the ending time stamp for the last subtitle
> is a little cumbersome and can't be done automatically (I don't
> know a CLI tool that's able to display durations or ending
> time stamps from VOBSUB streams).  Therefore I would appreciate
> if the bug with repeatlast=0 could be fixed, so my workaround
> won't be necessary anymore.
>
> Unfortunately I can't easily upload sample files, because the
> full movie is too large, and I don't have appropriate software
> to cut a short clip from it that also retains the subtitles
> while keeping them in sync with the movie.

From johnblackjr at gmail.com  Thu Jul  4 03:54:06 2013
From: johnblackjr at gmail.com (John Black)
Date: Wed, 3 Jul 2013 19:54:06 -0600
Subject: [FFmpeg-user] Muxing private stream into/from mp4 and mpegts
Message-ID: <CAMLfgPxL-TsyGJK6GwDUvWCVzkp24pF=OjOM-u5x68_JbMdEeg@mail.gmail.com>

Suppose I have a new track (that is neither audio nor video nor subtitles)
and I want to mux into an existing mp4 file or extract from an mp4 file.
 Or I want to mux into/from mpegts.

Can ffmpeg do this for me without source mods?  I'm going to guess the
answer is "no" but thought there might be some obscure feature I haven't
run across.

Thanks in advance!

John

From horsley1953 at gmail.com  Thu Jul  4 05:00:53 2013
From: horsley1953 at gmail.com (Tom Horsley)
Date: Wed, 3 Jul 2013 23:00:53 -0400
Subject: [FFmpeg-user] Arbitrarily correct packets?
Message-ID: <20130703230053.3a998b6d@zooty>

I keep fighting with files downloaded from my TiVo. Some
combination of horrible things broadcasters do to the
stream on the way to my TiVo and horrible things the
TiVo itself does when converting from the TS to PS
mpg often leaves the audio and video screwed up in wondrous
ways.

I'm just wondering: Since I happen to know this is all
NTSC format 29.97 fps video, can I force ffmpeg to
utterly re-write the packet info with nice clean
alternating fixed length video and audio frames
while I'm doing the (2-pass) conversion to H.264?

If I can, is there any chance it might actually work
and leave the audio and video in sync?

Just curious.

From cehoyos at ag.or.at  Thu Jul  4 11:00:41 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 4 Jul 2013 09:00:41 +0000 (UTC)
Subject: [FFmpeg-user] Arbitrarily correct packets?
References: <20130703230053.3a998b6d@zooty>
Message-ID: <loom.20130704T110020-956@post.gmane.org>

Tom Horsley <horsley1953 <at> gmail.com> writes:

> Some combination of horrible things broadcasters do to the
> stream on the way to my TiVo and horrible things the
> TiVo itself does when converting from the TS to PS
> mpg often leaves the audio and video screwed up in wondrous
> ways.

Please provide a sample.

Carl Eugen


From cehoyos at ag.or.at  Thu Jul  4 11:08:55 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 4 Jul 2013 09:08:55 +0000 (UTC)
Subject: [FFmpeg-user] Hardcoding subtitles
References: <201307031444.r63EiU0R058599@haluter.fromme.com>
Message-ID: <loom.20130704T110739-136@post.gmane.org>

Oliver Fromme <oliver <at> fromme.com> writes:

> Basically, that command worked and created a video with hardcoded
> subtitles.  However, there was one small problem:  The duration
> of the last subtitle was "infinite"

Did you test -fix_sub_duration ?

Carl Eugen


From oliver at fromme.com  Thu Jul  4 12:36:53 2013
From: oliver at fromme.com (Oliver Fromme)
Date: Thu, 4 Jul 2013 12:36:53 +0200 (CEST)
Subject: [FFmpeg-user] Hardcoding subtitles
In-Reply-To: <CAPYw7P4atHAHY=sKM7=30uDdVWwheusEzsE9DVLv27Q3ibZuWw@mail.gmail.com>
Message-ID: <201307041036.r64AarN0062109@haluter.fromme.com>

Paul B Mahol wrote:
 > On 7/3/13, Oliver Fromme <oliver at fromme.com> wrote:
 > > [...]
 > > The first problem was that ffmpeg didn't want to read the VOBSUB
 > > files directly, no matter what I tried.  It seemed to be unable
 > > to detect the format automatically, and I was unable to find out
 > > how to specify it on the command line.  I probably tried more
 > > than 20 variantions, without success.  However, it's not a big
 > > deal:  I just wrapped the VOBSUB stream into an MKV container
 > 
 > Yes, it is big deal, if you want bug fixed, provide sample and open bug report.

It doesn't appear to be a bug, I think.  It seems that reading
"raw" VOBSUB .idx files is simply not supported.  Also, it's
not mentioned anywhere in the manual page or in the wiki.

Best regards
   Oliver

-- 

From oliver at fromme.com  Thu Jul  4 13:05:01 2013
From: oliver at fromme.com (Oliver Fromme)
Date: Thu, 4 Jul 2013 13:05:01 +0200 (CEST)
Subject: [FFmpeg-user] Hardcoding subtitles
In-Reply-To: <loom.20130704T110739-136@post.gmane.org>
Message-ID: <201307041105.r64B51dG062186@haluter.fromme.com>

Carl Eugen Hoyos wrote:
 > Oliver Fromme <oliver <at> fromme.com> writes:
 > 
 > > Basically, that command worked and created a video with hardcoded
 > > subtitles.  However, there was one small problem:  The duration
 > > of the last subtitle was "infinite"
 > 
 > Did you test -fix_sub_duration ?

Yes, I did.  I'm sorry I forgot to mention it.

In fact -fix_sub_duration made things worse:  The *first*
subtitle frame stayed on the screen until the end of the
movie.  None of the other subtitles ever appeared, only the
first one.

Best regards
   Oliver

-- 

From nicolas.george at normalesup.org  Thu Jul  4 15:54:11 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Thu, 4 Jul 2013 15:54:11 +0200
Subject: [FFmpeg-user] Hardcoding subtitles
In-Reply-To: <loom.20130704T110739-136@post.gmane.org>
References: <201307031444.r63EiU0R058599@haluter.fromme.com>
	<loom.20130704T110739-136@post.gmane.org>
Message-ID: <20130704135411.GA24164@phare.normalesup.org>

Le sextidi 16 messidor, an CCXXI, Carl Eugen Hoyos a ?crit?:
> Did you test -fix_sub_duration ?

Please stop with -fix_sub_duration; -fix_sub_duration is for cases where the
duration is not known. That happens for DVB and PGS subtitles inside MPEG
containers. On the other hand, DVD subtitles have their duration embedded
and Matroska format always has duration for subtitles packets, so
-fix_sub_duration is doubly wrong in this case. It would still be unneeded
reading from VOBSUB directly.

Regards,

-- 
  Nicolas George
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From krueger at lesspain.de  Thu Jul  4 16:02:09 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 4 Jul 2013 16:02:09 +0200
Subject: [FFmpeg-user] Support for Prores4444 12bit files
Message-ID: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>

Hi,

Apple's whitepaper states that Prores4444 has "up to 12bit" color
depth. There are no Apple tools that I know of that have a switch to
produce such a 12bit beast (their professional transcoder "Compressor"
has the options "16.7 million colors" or "more than 16.7 million
colors" which is useless in that case as IMO that translates to 8bit
and more than 8 bit).

Now there exist recorders which explicitly claim to support 10bit
_and_ 12bit Prores4444 (see
http://www.sounddevices.com/products/pix240i/sample-files/) and I
downloaded those and tested transcoding with ffmpeg and the output
looks fine. Since the pixel formats used by ffmpeg for Prores are 10
bit only, I am wondering if there is a way to test if the color depth
is really 12bit and thus reduced in any transcoding.

Could anyone with more intimate knowledge of the decoder(s) shed some
light on this? Is the current decoder reducing bit depth of these
files when decoding and if so are there plans to add support for a
pixel format that allows for 12 bit?

Is there a way of testing such a file if it is 10 or 12bit? Apart from
the alpha channel (which in this case is not there) neither the header
documentation at
http://wiki.multimedia.cx/index.php?title=Apple_ProRes nor the source
code reveal anything that would be suitable for making that
distinction? Or is the formulation "up to 12bit" simply wrong and
Prores4444 is always 12bit?

Best regards,

Robert

P.S.: I am not talking about 10 bit files with a special gamma
function like Arri Log-C or Sony S-Log but real 12bits of depth.

From cehoyos at ag.or.at  Thu Jul  4 16:10:21 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 4 Jul 2013 14:10:21 +0000 (UTC)
Subject: [FFmpeg-user] Support for Prores4444 12bit files
References: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>
Message-ID: <loom.20130704T160955-124@post.gmane.org>

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

> Could anyone with more intimate knowledge of the 
> decoder(s) shed some light on this?

Wouldn't this need somebody with intimate knowledge 
of the encoders?

Carl Eugen


From doenerbudenchef at yahoo.de  Thu Jul  4 16:27:09 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Thu, 4 Jul 2013 07:27:09 -0700 (PDT)
Subject: [FFmpeg-user] MXF with XDCAM HD422
In-Reply-To: <CAEnqZEXVWoX9MQ075+iuBpxE2-kkGZbjE4+r662RHiixB0bhsg@mail.gmail.com>
References: <1372855958346-4659820.post@n4.nabble.com>
	<CAEnqZEXVWoX9MQ075+iuBpxE2-kkGZbjE4+r662RHiixB0bhsg@mail.gmail.com>
Message-ID: <1372948029015-4659835.post@n4.nabble.com>

Works fine. Thanks.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/MXF-with-XDCAM-HD422-tp4659820p4659835.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From krueger at lesspain.de  Thu Jul  4 16:50:42 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 4 Jul 2013 16:50:42 +0200
Subject: [FFmpeg-user] Support for Prores4444 12bit files
In-Reply-To: <loom.20130704T160955-124@post.gmane.org>
References: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>
	<loom.20130704T160955-124@post.gmane.org>
Message-ID: <CAEnqZEXfeRgKzK01BVDy3XiPwmjsEfca8ri=-ku8kALJ98NMgw@mail.gmail.com>

On Thu, Jul 4, 2013 at 4:10 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> Could anyone with more intimate knowledge of the
>> decoder(s) shed some light on this?
>
> Wouldn't this need somebody with intimate knowledge
> of the encoders?

You may be right but I thought maybe someone who knows about what's
actually in the bitstream might also know enough to answer the
questions. Lets just hope one of either side comments.

Robert

From onemda at gmail.com  Thu Jul  4 17:25:07 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 4 Jul 2013 15:25:07 +0000
Subject: [FFmpeg-user] Support for Prores4444 12bit files
In-Reply-To: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>
References: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>
Message-ID: <CAPYw7P6=oVMs6QVGD-9PWuiCXrbz7d0F1K3CD6QTpEjU=ovVhQ@mail.gmail.com>

On 7/4/13, Robert Krueger <krueger at lesspain.de> wrote:
> Hi,
>
> Apple's whitepaper states that Prores4444 has "up to 12bit" color
> depth. There are no Apple tools that I know of that have a switch to
> produce such a 12bit beast (their professional transcoder "Compressor"
> has the options "16.7 million colors" or "more than 16.7 million
> colors" which is useless in that case as IMO that translates to 8bit
> and more than 8 bit).
>
> Now there exist recorders which explicitly claim to support 10bit
> _and_ 12bit Prores4444 (see
> http://www.sounddevices.com/products/pix240i/sample-files/) and I
> downloaded those and tested transcoding with ffmpeg and the output
> looks fine. Since the pixel formats used by ffmpeg for Prores are 10
> bit only, I am wondering if there is a way to test if the color depth
> is really 12bit and thus reduced in any transcoding.
>
> Could anyone with more intimate knowledge of the decoder(s) shed some
> light on this? Is the current decoder reducing bit depth of these
> files when decoding and if so are there plans to add support for a
> pixel format that allows for 12 bit?
>
> Is there a way of testing such a file if it is 10 or 12bit? Apart from
> the alpha channel (which in this case is not there) neither the header
> documentation at
> http://wiki.multimedia.cx/index.php?title=Apple_ProRes nor the source
> code reveal anything that would be suitable for making that
> distinction? Or is the formulation "up to 12bit" simply wrong and
> Prores4444 is always 12bit?
>
> Best regards,
>
> Robert
>
> P.S.: I am not talking about 10 bit files with a special gamma
> function like Arri Log-C or Sony S-Log but real 12bits of depth.

I think you are confused. There is no 12bit prores.

Why would 12 bit prores file be smaller than 10 bit one?

From krueger at lesspain.de  Thu Jul  4 17:59:59 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 4 Jul 2013 17:59:59 +0200
Subject: [FFmpeg-user] Support for Prores4444 12bit files
In-Reply-To: <CAPYw7P6=oVMs6QVGD-9PWuiCXrbz7d0F1K3CD6QTpEjU=ovVhQ@mail.gmail.com>
References: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>
	<CAPYw7P6=oVMs6QVGD-9PWuiCXrbz7d0F1K3CD6QTpEjU=ovVhQ@mail.gmail.com>
Message-ID: <CAEnqZEWjZMJjdd_XQST8H5Kvk6jqrOTB7nmPZdHxO-zMxMTUpQ@mail.gmail.com>

On Thu, Jul 4, 2013 at 5:25 PM, Paul B Mahol <onemda at gmail.com> wrote:
> On 7/4/13, Robert Krueger <krueger at lesspain.de> wrote:
>> Hi,
>>
>> Apple's whitepaper states that Prores4444 has "up to 12bit" color
>> depth. There are no Apple tools that I know of that have a switch to
>> produce such a 12bit beast (their professional transcoder "Compressor"
>> has the options "16.7 million colors" or "more than 16.7 million
>> colors" which is useless in that case as IMO that translates to 8bit
>> and more than 8 bit).
>>
>> Now there exist recorders which explicitly claim to support 10bit
>> _and_ 12bit Prores4444 (see
>> http://www.sounddevices.com/products/pix240i/sample-files/) and I
>> downloaded those and tested transcoding with ffmpeg and the output
>> looks fine. Since the pixel formats used by ffmpeg for Prores are 10
>> bit only, I am wondering if there is a way to test if the color depth
>> is really 12bit and thus reduced in any transcoding.
>>
>> Could anyone with more intimate knowledge of the decoder(s) shed some
>> light on this? Is the current decoder reducing bit depth of these
>> files when decoding and if so are there plans to add support for a
>> pixel format that allows for 12 bit?
>>
>> Is there a way of testing such a file if it is 10 or 12bit? Apart from
>> the alpha channel (which in this case is not there) neither the header
>> documentation at
>> http://wiki.multimedia.cx/index.php?title=Apple_ProRes nor the source
>> code reveal anything that would be suitable for making that
>> distinction? Or is the formulation "up to 12bit" simply wrong and
>> Prores4444 is always 12bit?
>>
>> Best regards,
>>
>> Robert
>>
>> P.S.: I am not talking about 10 bit files with a special gamma
>> function like Arri Log-C or Sony S-Log but real 12bits of depth.
>
> I think you are confused. There is no 12bit prores.

OK, so your theory is:
- Apple's official whitepaper is bullshit
- http://www.sounddevices.com/products/pix240i/specs/ + a lot of
articles about it are fraud or just bullshit

Definitely possible though it does not sound too likely and i am
trying to find out. But you are correct. I am indeed confused.

>
> Why would 12 bit prores file be smaller than 10 bit one?

File size for a video file with a de-factor VBR codec would always
depend on a lot of factors
1) the exact motive
2) noise of the camera under the conditions used to record each clip
3) exposure time of the camera
4) the way bitrate control is implemented/tuned for the given codec/setting

If those were exactly the same for both clips, then I would say, you
are right and it cannot be.

From onemda at gmail.com  Thu Jul  4 18:29:46 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 4 Jul 2013 16:29:46 +0000
Subject: [FFmpeg-user] Support for Prores4444 12bit files
In-Reply-To: <CAEnqZEWjZMJjdd_XQST8H5Kvk6jqrOTB7nmPZdHxO-zMxMTUpQ@mail.gmail.com>
References: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>
	<CAPYw7P6=oVMs6QVGD-9PWuiCXrbz7d0F1K3CD6QTpEjU=ovVhQ@mail.gmail.com>
	<CAEnqZEWjZMJjdd_XQST8H5Kvk6jqrOTB7nmPZdHxO-zMxMTUpQ@mail.gmail.com>
Message-ID: <CAPYw7P5Ccf8-QjdRSaU5xPDXqwGQzs27OjCpveEgRjPmi8dQXw@mail.gmail.com>

On 7/4/13, Robert Krueger <krueger at lesspain.de> wrote:
> On Thu, Jul 4, 2013 at 5:25 PM, Paul B Mahol <onemda at gmail.com> wrote:
>> On 7/4/13, Robert Krueger <krueger at lesspain.de> wrote:
>>> Hi,
>>>
>>> Apple's whitepaper states that Prores4444 has "up to 12bit" color
>>> depth. There are no Apple tools that I know of that have a switch to
>>> produce such a 12bit beast (their professional transcoder "Compressor"
>>> has the options "16.7 million colors" or "more than 16.7 million
>>> colors" which is useless in that case as IMO that translates to 8bit
>>> and more than 8 bit).
>>>
>>> Now there exist recorders which explicitly claim to support 10bit
>>> _and_ 12bit Prores4444 (see
>>> http://www.sounddevices.com/products/pix240i/sample-files/) and I
>>> downloaded those and tested transcoding with ffmpeg and the output
>>> looks fine. Since the pixel formats used by ffmpeg for Prores are 10
>>> bit only, I am wondering if there is a way to test if the color depth
>>> is really 12bit and thus reduced in any transcoding.
>>>
>>> Could anyone with more intimate knowledge of the decoder(s) shed some
>>> light on this? Is the current decoder reducing bit depth of these
>>> files when decoding and if so are there plans to add support for a
>>> pixel format that allows for 12 bit?
>>>
>>> Is there a way of testing such a file if it is 10 or 12bit? Apart from
>>> the alpha channel (which in this case is not there) neither the header
>>> documentation at
>>> http://wiki.multimedia.cx/index.php?title=Apple_ProRes nor the source
>>> code reveal anything that would be suitable for making that
>>> distinction? Or is the formulation "up to 12bit" simply wrong and
>>> Prores4444 is always 12bit?
>>>
>>> Best regards,
>>>
>>> Robert
>>>
>>> P.S.: I am not talking about 10 bit files with a special gamma
>>> function like Arri Log-C or Sony S-Log but real 12bits of depth.
>>
>> I think you are confused. There is no 12bit prores.
>
> OK, so your theory is:
> - Apple's official whitepaper is bullshit
> - http://www.sounddevices.com/products/pix240i/specs/ + a lot of
> articles about it are fraud or just bullshit
>
> Definitely possible though it does not sound too likely and i am
> trying to find out. But you are correct. I am indeed confused.

Looking at both obfuscated decoders there is possibility
prores can store any bit depth.

If they decode real 12bit as 10bit and output is same as given by
other decoders that claim they support real 12bit than I'm
in wrong universe.

>
>>
>> Why would 12 bit prores file be smaller than 10 bit one?
>
> File size for a video file with a de-factor VBR codec would always
> depend on a lot of factors
> 1) the exact motive
> 2) noise of the camera under the conditions used to record each clip
> 3) exposure time of the camera
> 4) the way bitrate control is implemented/tuned for the given codec/setting
>
> If those were exactly the same for both clips, then I would say, you
> are right and it cannot be.

There are still many open questions left.

From krueger at lesspain.de  Thu Jul  4 19:10:18 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 4 Jul 2013 19:10:18 +0200
Subject: [FFmpeg-user] Support for Prores4444 12bit files
In-Reply-To: <CAPYw7P5Ccf8-QjdRSaU5xPDXqwGQzs27OjCpveEgRjPmi8dQXw@mail.gmail.com>
References: <CAEnqZEWv5mTrogDJZQDyXRUP_qy=UQyH08zHS=BFJ4eMCbmGxg@mail.gmail.com>
	<CAPYw7P6=oVMs6QVGD-9PWuiCXrbz7d0F1K3CD6QTpEjU=ovVhQ@mail.gmail.com>
	<CAEnqZEWjZMJjdd_XQST8H5Kvk6jqrOTB7nmPZdHxO-zMxMTUpQ@mail.gmail.com>
	<CAPYw7P5Ccf8-QjdRSaU5xPDXqwGQzs27OjCpveEgRjPmi8dQXw@mail.gmail.com>
Message-ID: <CAEnqZEUC05qBJsZ_swEZ=tn7rXh+cqoqQPpZdMNtfC-cP0uHLA@mail.gmail.com>

On Jul 4, 2013 6:30 PM, "Paul B Mahol" <onemda at gmail.com> wrote:
>
> On 7/4/13, Robert Krueger <krueger at lesspain.de> wrote:
> > On Thu, Jul 4, 2013 at 5:25 PM, Paul B Mahol <onemda at gmail.com> wrote:
> >> On 7/4/13, Robert Krueger <krueger at lesspain.de> wrote:
> >>> Hi,
> >>>
> >>> Apple's whitepaper states that Prores4444 has "up to 12bit" color
> >>> depth. There are no Apple tools that I know of that have a switch to
> >>> produce such a 12bit beast (their professional transcoder "Compressor"
> >>> has the options "16.7 million colors" or "more than 16.7 million
> >>> colors" which is useless in that case as IMO that translates to 8bit
> >>> and more than 8 bit).
> >>>
> >>> Now there exist recorders which explicitly claim to support 10bit
> >>> _and_ 12bit Prores4444 (see
> >>> http://www.sounddevices.com/products/pix240i/sample-files/) and I
> >>> downloaded those and tested transcoding with ffmpeg and the output
> >>> looks fine. Since the pixel formats used by ffmpeg for Prores are 10
> >>> bit only, I am wondering if there is a way to test if the color depth
> >>> is really 12bit and thus reduced in any transcoding.
> >>>
> >>> Could anyone with more intimate knowledge of the decoder(s) shed some
> >>> light on this? Is the current decoder reducing bit depth of these
> >>> files when decoding and if so are there plans to add support for a
> >>> pixel format that allows for 12 bit?
> >>>
> >>> Is there a way of testing such a file if it is 10 or 12bit? Apart from
> >>> the alpha channel (which in this case is not there) neither the header
> >>> documentation at
> >>> http://wiki.multimedia.cx/index.php?title=Apple_ProRes nor the source
> >>> code reveal anything that would be suitable for making that
> >>> distinction? Or is the formulation "up to 12bit" simply wrong and
> >>> Prores4444 is always 12bit?
> >>>
> >>> Best regards,
> >>>
> >>> Robert
> >>>
> >>> P.S.: I am not talking about 10 bit files with a special gamma
> >>> function like Arri Log-C or Sony S-Log but real 12bits of depth.
> >>
> >> I think you are confused. There is no 12bit prores.
> >
> > OK, so your theory is:
> > - Apple's official whitepaper is bullshit
> > - http://www.sounddevices.com/products/pix240i/specs/ + a lot of
> > articles about it are fraud or just bullshit
> >
> > Definitely possible though it does not sound too likely and i am
> > trying to find out. But you are correct. I am indeed confused.
>
> Looking at both obfuscated decoders there is possibility
> prores can store any bit depth.
>
> If they decode real 12bit as 10bit and output is same as given by
> other decoders that claim they support real 12bit than I'm
> in wrong universe.
>
I am looking for a way to really compare the decoder output because
visually it is not possible to distinguish 10 bit from 12bit (unless you
manipulate it in a color grading application that is known to use the full
12bit). I probably have to decode them using libavcodec and using apple
apis (where you can decode it to a16bit pixel format) and then compare the
numbers if I don't find an easier solution. Furthermore I am trying to get
a sample from a more official (preferably Apple) source. There is always
the possibility that the sample on the site is not really 12bit because
someone messed up.

From stefasab at gmail.com  Fri Jul  5 12:52:17 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Fri, 5 Jul 2013 12:52:17 +0200
Subject: [FFmpeg-user] Hardcoding subtitles
In-Reply-To: <201307031444.r63EiU0R058599@haluter.fromme.com>
References: <201307031444.r63EiU0R058599@haluter.fromme.com>
Message-ID: <20130705105217.GN19933@barisone>

On date Wednesday 2013-07-03 16:44:30 +0200, Oliver Fromme wrote:
> I'm trying to "hardcode" forced subtitles into a video, so they are
> always displayed, independent of the player's subtitle support.
> In the end I have been successful, but it was kind of an adventure.
> I'm trying to describe the problems that I encountered, maybe this
> can help to make ffmpeg better.
> 
> My ffmpeg version is a recent git snapshot, just a few days old.
> 
> These are my input files:  Video and audio are in an MPEG2 PS file
> that comes from a DVD (via "mplayer -dumpstream").  The subtitles
> are in VOBSUB format (I think ffmpeg calls this "dvdsub"), i.e.
> .idx + .sub files.  These have been retrieved from the same DVD
> with "mencoder -vobsubout".
> 
> The first problem was that ffmpeg didn't want to read the VOBSUB
> files directly, no matter what I tried.  It seemed to be unable
> to detect the format automatically, and I was unable to find out
> how to specify it on the command line.  I probably tried more
> than 20 variantions, without success.  However, it's not a big
> deal:  I just wrapped the VOBSUB stream into an MKV container
> (using mkvmerge), and ffmpeg happily accepted it.  So, my input
> files are now the MPEG2 PS file (same as before) and an MKV file
> containing just the VOBSUB subtitle stream.
> 
> According to the excellent manual page, the overlay filter can
> be used for hardcoding subtitles, so I tried that.  The filter
> specification was simple:
> 
> -filter_complex "[0:v][1:s]overlay"
> 
> where file 0 is the MPEG2 PS and file 1 is the MKV file with the
> VOBSUB stream.  If it matters (I don't think it does), here's
> the complete command line:
> 
> ffmpeg1 -loglevel warning -stats -i dvdstream-3.mpg -i subtitles-3-1.mkv \
>         -filter_complex "[0:v][1:s]overlay" \
>         -c:v libx264 -preset veryslow -tune film -crf 18 -level 4.1 \
>         -codec:a copy -map 0:a -sn dvdstream-3-vid.mkv
> 
> Basically, that command worked and created a video with hardcoded
> subtitles.  However, there was one small problem:  The duration
> of the last subtitle was "infinite", i.e. the very last subtitle
> stayed on the screen until end of the video.  It was visible for
> several minutes including the whole end credits of the movie.
> Not nice.  So I went back to the manual page and re-read the
> section about the overlay filter.
> 
> I noticed the "repeatlast" option of this filter, but the last
> sentence said something like "A value of 0 ... is enabled by
> default", so I thought that 0 was the default.  But I didn't
> look closely enough ...  Upon further inspection, it's just the
> other way around.  Of course that's my mistake, but maybe the
> manual page could be made a little clearer?  Something like
> "The default value is 1" would probably be clear enough.

Changed.

> Anyway, I tried to disable that flag:
> 
> -filter_complex "[0:v][1:s]overlay=repeatlast=0"
> 
> It did make a difference, but not what I expected:  Now the last
> subtitle didn't appear at all!  I think this is a bug, although
> I wouldn't rule out the possibility that I did something wrong
> again.  I tried with a different DVD, but same result:  When
> repeatlast=0 is specified, the last VOBSUB subtitle is omitted
> completely.

If this is a bug, we need a sample and a command to reproduce the
error. Please consider to file a bug report on trac.

> 
> So, now I pondered over a workaround for that problem.
> I skimmed over some examples in the manual page that gave me an
> Idea:  I could try to use the gte() function to forcefully end
> the last subtitle at the desired time stamp.
> 
> First I used mplayer to look for the exact position where the
> last subtitle was supposed to end.  In this case that was at
> t = 3375 (that's 56 minutes and 15 seconds).  I reverted the
> "repeatlast" flag to the default so the last subtitle was
> visible, and used gte() to cancel it at the proper time:
> 

> -filter_complex "[0:v][1:s]overlay=x='if(gt(t,3375),NAN,x)'"

The problem is that x is NAN when evaluating this expression, so the
result is always NAN. Or in other words, recursive definitions are not
supported (and probably meaningless).

> 
> The idee was:  If the time is beyond 3375, set the x value of
> the subtitle frames to NAN (effectively disabling them),
> otherwise keep them at their original x value.  At least I
> thought that's what the x variable meant.


> 
> Obviously I was wrong, because the resulting video did not
> contain *any* subtitles at all.  No subtitles anywhere.
> Either I misunderstood the description of the x variable in
> the manual page (and in this case the manual page should be
> clarified), or there's another bug.  In my next attempt I
> replaced the x variable with 0, which should be the default
> for x anyway:
> 
> -filter_complex "[0:v][1:s]overlay=x='if(gt(t,3375),NAN,0)'"
> 
> Finally, that produced the expected result, and I'm happy now.

Even better, you could use the enable option (check timeline section
in the manual) to avoid x/NAN hacks:
enable=gt(t,3375)

[...]

From salmjuh at hotmail.com  Fri Jul  5 12:58:57 2013
From: salmjuh at hotmail.com (juha s.)
Date: Fri, 5 Jul 2013 13:58:57 +0300
Subject: [FFmpeg-user] rawvideo -> libx264
Message-ID: <DUB123-W3224477D40EDB56446A70EAB7D0@phx.gbl>

Hi
I have a DecLinks SDI capture card and I try to transcode to h264. It seems that this is not supported:
---------------------bmdcapture -m 2 -C 1 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -re -i - -copyts -vcodec libx264  -vprofile main -preset medium -crf 20  -an -b:v 1100k -maxrate 1300k -bufsize 1500k   -y  /tmp/test.mp4ffmpeg version git-2012-10-02-c785b6d Copyright (c) 2000-2012 the FFmpeg developers  built on Jul  5 2013 13:28:23 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)  configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx  libavutil      51. 73.101 / 51. 73.101  libavcodec     54. 62.100 / 54. 62.100  libavformat    54. 29.105 / 54. 29.105  libavdevice    54.  3.100 / 54.  3.100  libavfilter     3. 19.100 /  3. 19.100  libswscale      2.  1.101 /  2.  1.101  libswresample   0. 16.100 /  0. 16.100  libpostproc    52.  1.100 / 52.  1.100Frame received (#1) - No input signal detected - Frames dropped 1 - Total dropped 1[nut @ 0x1aed280] Unknown codec tag '0x10445350' for stream number 1[nut @ 0x1aed280] Could not find codec parameters for stream 1 (Audio: none (PSD[16] / 0x10445350), 48000 Hz, 2 channels): unknown codecConsider increasing the value for the 'analyzeduration' and 'probesize' options[nut @ 0x1aed280] Estimating duration from bitrate, this may be inaccurateGuessed Channel Layout for  Input Stream #0.1 : stereoInput #0, nut, from 'pipe:':  Metadata:    encoder         : Lavf54.20.3  Duration: N/A, start: 0.000000, bitrate: N/A    Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x576, 25 fps, 25 tbr, 25 tbn, 25 tbc    Stream #0:1: Audio: none (PSD[16] / 0x10445350), 48000 Hz, stereox264 [error]: main profile doesn't support 4:2:2[libx264 @ 0x1afe460] Error setting profile main.[libx264 @ 0x1afe460] Possible profiles: baseline main high high10 high422 high444Output #0, mp4, to '/tmp/test.mp4':  Metadata:    encoder         : Lavf54.20.3    Stream #0:0: Video: h264, yuv422p, 720x576, q=-1--1, 1100 kb/s, 90k tbn, 25 tbcStream mapping:  Stream #0:0 -> #0:0 (rawvideo -> libx264)Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
---------------------------------------
Any ideas?
br,jii

 		 	   		  

From yildiz.abdullah at yahoo.com  Fri Jul  5 11:33:27 2013
From: yildiz.abdullah at yahoo.com (Abdullah YILDIZ)
Date: Fri, 5 Jul 2013 02:33:27 -0700 (PDT)
Subject: [FFmpeg-user] error running configure: libx264 not found
Message-ID: <1373016807.22026.YahooMailNeo@web120505.mail.ne1.yahoo.com>

When I run the configure script, it gives the following error:

./configure --enable-cross-compile --cross-prefix=${CCPREFIX} --arch=armel --target-os=linux

--prefix=/path/to/raspberrypi-ffmpeg/

--enable-gpl --enable-libx264 --enable-nonfree --enable-libaacplus

--extra-cflags='-I/path/to/raspberrypi-ffmpeg/include' --extra-ldflags='-L/path/to/raspberrypi-ffmpeg/lib'



ERROR: libx264 not found

libx264.a is located at /path/to/raspberrypi-ffmpeg/lib/



abdullah

From lou at lrcd.com  Fri Jul  5 22:50:13 2013
From: lou at lrcd.com (Lou)
Date: Fri, 5 Jul 2013 12:50:13 -0800
Subject: [FFmpeg-user] rawvideo -> libx264
In-Reply-To: <DUB123-W3224477D40EDB56446A70EAB7D0@phx.gbl>
References: <DUB123-W3224477D40EDB56446A70EAB7D0@phx.gbl>
Message-ID: <20130705125013.69b017fc@lrcd.com>

On Fri, 5 Jul 2013 13:58:57 +0300
juha s. <salmjuh at hotmail.com> wrote:

> Hi
> I have a DecLinks SDI capture card and I try to transcode to h264. It seems that this is not supported:
...

Your output was too hard to read with the lack of line breaks.

> [error]: main profile doesn't support 4:2:2

This is the error message you missed. Your output is yuv422p. Is this
what you want?

Either remove "-vprofile main", or add "-pix_fmt yuv420p" (or "-vf
format=yuv420p").

Note that "-crf 20" and "-b:v 1100k" are mutually exclusive.

From h.reindl at thelounge.net  Fri Jul  5 22:51:04 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Fri, 05 Jul 2013 22:51:04 +0200
Subject: [FFmpeg-user] error running configure: libx264 not found
In-Reply-To: <1373016807.22026.YahooMailNeo@web120505.mail.ne1.yahoo.com>
References: <1373016807.22026.YahooMailNeo@web120505.mail.ne1.yahoo.com>
Message-ID: <51D731B8.1060109@thelounge.net>


Am 05.07.2013 11:33, schrieb Abdullah YILDIZ:
> When I run the configure script, it gives the following error:
> 
> ./configure --enable-cross-compile --cross-prefix=${CCPREFIX} --arch=armel --target-os=linux
> 
> --prefix=/path/to/raspberrypi-ffmpeg/
> 
> --enable-gpl --enable-libx264 --enable-nonfree --enable-libaacplus
> 
> --extra-cflags='-I/path/to/raspberrypi-ffmpeg/include' --extra-ldflags='-L/path/to/raspberrypi-ffmpeg/lib'
> 
> ERROR: libx264 not found
> 
> libx264.a is located at /path/to/raspberrypi-ffmpeg/lib/

i guess you want to link x264 static
use the configure options below to make itself independent of shared libraries

that did for me the trick, some of them are added since the latest x264
because it stopped to work as before

 --enable-static \
 --disable-avs \
 --disable-cli \
 --disable-ffms \
 --disable-gpac \
 --disable-lavf \
 --disable-opencl \
 --disable-swscale

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From lou at lrcd.com  Fri Jul  5 23:23:08 2013
From: lou at lrcd.com (Lou)
Date: Fri, 5 Jul 2013 13:23:08 -0800
Subject: [FFmpeg-user] error running configure: libx264 not found
In-Reply-To: <1373016807.22026.YahooMailNeo@web120505.mail.ne1.yahoo.com>
References: <1373016807.22026.YahooMailNeo@web120505.mail.ne1.yahoo.com>
Message-ID: <20130705132308.40b3f814@lrcd.com>

On Fri, 5 Jul 2013 02:33:27 -0700 (PDT)
Abdullah YILDIZ <yildiz.abdullah at yahoo.com> wrote:

> When I run the configure script, it gives the following error:
> 
> ./configure --enable-cross-compile --cross-prefix=${CCPREFIX} --arch=armel --target-os=linux
> 
> --prefix=/path/to/raspberrypi-ffmpeg/
> 
> --enable-gpl --enable-libx264 --enable-nonfree --enable-libaacplus
> 
> --extra-cflags='-I/path/to/raspberrypi-ffmpeg/include' --extra-ldflags='-L/path/to/raspberrypi-ffmpeg/lib'
> 
> 
> 
> ERROR: libx264 not found
> 
> libx264.a is located at /path/to/raspberrypi-ffmpeg/lib/

Try adding "--extra-libs=-ldl".

From jdevelop at gmail.com  Sat Jul  6 00:48:33 2013
From: jdevelop at gmail.com (Eugene Dzhurinsky)
Date: Fri, 5 Jul 2013 18:48:33 -0400
Subject: [FFmpeg-user] Distinguish between muxer and demuxer optopms
Message-ID: <20130705224833.GA28253@devbox.ma.labs.intellij.net>

Hello!

The "loop" option is available for both image2pipe demuxer and gif muxer. I
want to change behavior of GIF muxer so if negative value is passed - then
GIF98a header is not written into GIF file. But this conflicts with demuxer
option and allowed value range (0 .. 1)

I have 2 choices - either add new option, or distinguish muxer/demuxer options
at command line - and this is the preferred way.

Please advice.

Thanks!

-- 
Eugene N Dzhurinsky
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From pdchen at gmail.com  Sat Jul  6 01:47:53 2013
From: pdchen at gmail.com (Peter Chen)
Date: Fri, 5 Jul 2013 19:47:53 -0400
Subject: [FFmpeg-user] Set start time offset in ts video segment output from
	mp4 movie clip input
Message-ID: <CAL22iBhJ8pRMGrrMzqKFdwLM83SyL5718C9pw9mC6zpJy6Bpsg@mail.gmail.com>

I'm converting an mp4 movie clip into a ts video segment using the
following command:

   > ffmpeg -i input-segment-4.mp4 -vcodec copy -acodec copy -qscale 0
-vbsf h264_mp4toannexb -y -r 24 -vf transpose=1 -f
mpegts output-segment-4.ts

This ts video segment will be part of an HLS (HTTP Live Streaming) video
stream.

The ffmpeg conversion works great.

I might not be asking the following question using the proper terminology,
but here's what I'd like to do. I'd like the output file, called
output-segment-4.ts in the ffmpeg command above, to have a start time
offset of, say, 40 seconds. So, in an HLS video stream that contains these
output ts segments, output-segment-0.ts would have a start time of 0
seconds, output-segment-1.ts would have a start time of 10 seconds, etc...

Currently, if I'm reading the output of ffprobe correctly, every ts file
shows a start time offset of about 1.4 seconds. Below is the output of
"ffprobe output-segment-4.ts". I'm basically trying to
manipulate output-segment-4.ts so "start: 1.421678" becomes "start: 40.0"
in the line "Duration: 00:00:09.95, start: 1.421678, bitrate: 1718 kb/s".

In summary, how do I set that start field?

Thanks for your help.

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

> ffprobe output-segment-4.ts

ffprobe version N-54474-g9f89beb Copyright (c) 2007-2013 the FFmpeg
developers
  built on Jul  5 2013 07:00:23 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration:
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.101 / 55. 18.101
  libavformat    55. 11.100 / 55. 11.100
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 78.100 /  3. 78.100
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
Input #0, mpegts, from 'output-segment-4.ts':
  Duration: 00:00:09.95, start: 1.421678, bitrate: 1718 kb/s
  Program 1
    Metadata:
      service_name    : Service01
      service_provider: FFmpeg
    Stream #0:0[0x100]: Video: h264 (Baseline) ([27][0][0][0] / 0x001B),
yuv420p, 710x400, 29.97 tbr, 90k tbn, 180k tbc
    Stream #0:1[0x101](und): Audio: aac ([15][0][0][0] / 0x000F), 44100 Hz,
stereo, fltp, 110 kb/s

From lost.alternative at gmail.com  Sat Jul  6 18:52:57 2013
From: lost.alternative at gmail.com (J. Scott Edwards)
Date: Sat, 6 Jul 2013 10:52:57 -0600
Subject: [FFmpeg-user] Is it possible to just change the fps flags?
Message-ID: <CAPajkzOQ49SBkB_y3Hzi8Qxi30FTxjMTsUUs3H7+OYSzpMJb9w@mail.gmail.com>

Hello,

i have several videos that were created from film, one film frame to one
video frame, so it plays too fast (29.97 fps).  i have tried:

 ffmpeg -i input.avi -r 24 output.avi

But that re-encodes the video and the result is horribly pixelated.  It
seems all i really want to do is change the flag that says what the fps is
to playback, and leave all of the video frames as is.

While searching for a solution it appears that both the container and the
codec have a frame rate, is there some way to change both of them to 24
without changing anything else?

Thanks!

From cehoyos at ag.or.at  Sat Jul  6 18:57:21 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 6 Jul 2013 16:57:21 +0000 (UTC)
Subject: [FFmpeg-user] Is it possible to just change the fps flags?
References: <CAPajkzOQ49SBkB_y3Hzi8Qxi30FTxjMTsUUs3H7+OYSzpMJb9w@mail.gmail.com>
Message-ID: <loom.20130706T185523-523@post.gmane.org>

J. Scott Edwards <lost.alternative <at> gmail.com> writes:

> i have several videos that were created from film, one film 
> frame to one video frame, so it plays too fast (29.97 fps).

Could you elaborate / command line and complete, uncut 
console output missing.
(This may be a serious bug but I need more information to be 
able to reproduce.)

>  ffmpeg -i input.avi -r 24 output.avi

$ ffmpeg -i input.avi -vf fps=24000/1001 -qscale 2 out.avi

Carl Eugen


From lost.alternative at gmail.com  Sat Jul  6 19:52:14 2013
From: lost.alternative at gmail.com (J. Scott Edwards)
Date: Sat, 6 Jul 2013 11:52:14 -0600
Subject: [FFmpeg-user] Is it possible to just change the fps flags?
In-Reply-To: <loom.20130706T185523-523@post.gmane.org>
References: <CAPajkzOQ49SBkB_y3Hzi8Qxi30FTxjMTsUUs3H7+OYSzpMJb9w@mail.gmail.com>
	<loom.20130706T185523-523@post.gmane.org>
Message-ID: <CAPajkzOzZghG9TGyypyK65qt6Fn-cTyaSPhicFEy0bK=w6dBcw@mail.gmail.com>

Hi Carl,

i don't know that it is a bug at all...  well, maybe that the output
is so terribly pixelated.

i assume that this command:

ffmpeg -i input.avi -r 24 output.avi

is trying to resample the frames so that there are 24 of them in every
second instead of 29.97, but the video is still playing at the same
speed.  I.E. the input file is 10:27 and the output file is 10:27.

But that is not what I want.  I want it to leave the frames exactly
the way they are now but only play 24 of them every second so that the
video runs slower.  I.E. instead of being 10:27 long the video is
13:02.

The command line output is below.

Thanks,
  -scott


$ time ffmpeg -i drag_races.avi -r 24 drag_races-24fps.avi
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 17:00:59 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.
[mpeg4 @ 0x8d75940] Invalid and inefficient vfw-avi packed B frames detected
Input #0, avi, from 'drag_races.avi':
  Metadata:
    encoder         : MEncoder 2:1.0~rc1-0ubuntu13.2
  Duration: 00:10:27.59, start: 0.000000, bitrate: 6615 kb/s
    Stream #0.0: Video: mpeg4 (Advanced Simple Profile), yuv420p,
720x480 [PAR 1:1 DAR 3:2], PAR 8:9 DAR 4:3, 29.97 tbr, 29.97 tbn,
29.97 tbc
[buffer @ 0x8d74f60] w:720 h:480 pixfmt:yuv420p
Output #0, avi, to 'drag_races-24fps.avi':
  Metadata:
    ISFT            : Lavf53.21.1
    Stream #0.0: Video: mpeg4, yuv420p, 720x480 [PAR 8:9 DAR 4:3],
q=2-31, 200 kb/s, 24 tbn, 24 tbc
Stream mapping:
  Stream #0.0 -> #0.0
Press ctrl-c to stop encoding
[mpeg4 @ 0x8d75940] Invalid and inefficient vfw-avi packed B frames detected
frame=15062 fps=223 q=31.0 Lsize=   72816kB time=627.62 bitrate=
950.4kbits/s dup=0 drop=3747
video:72450kB audio:0kB global headers:0kB muxing overhead 0.505254%

real    1m8.118s
user    1m0.412s
sys    0m5.992s

On Sat, Jul 6, 2013 at 10:57 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> J. Scott Edwards <lost.alternative <at> gmail.com> writes:
>
>> i have several videos that were created from film, one film
>> frame to one video frame, so it plays too fast (29.97 fps).
>
> Could you elaborate / command line and complete, uncut
> console output missing.
> (This may be a serious bug but I need more information to be
> able to reproduce.)
>
>>  ffmpeg -i input.avi -r 24 output.avi
>
> $ ffmpeg -i input.avi -vf fps=24000/1001 -qscale 2 out.avi
>
> 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 Jul  6 23:01:57 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 6 Jul 2013 21:01:57 +0000 (UTC)
Subject: [FFmpeg-user] Is it possible to just change the fps flags?
References: <CAPajkzOQ49SBkB_y3Hzi8Qxi30FTxjMTsUUs3H7+OYSzpMJb9w@mail.gmail.com>
	<loom.20130706T185523-523@post.gmane.org>
	<CAPajkzOzZghG9TGyypyK65qt6Fn-cTyaSPhicFEy0bK=w6dBcw@mail.gmail.com>
Message-ID: <loom.20130706T225839-174@post.gmane.org>

J. Scott Edwards <lost.alternative <at> gmail.com> writes:

> i don't know that it is a bug at all...  well, maybe 
> that the output is so terribly pixelated.
> 
> i assume that this command:
> 
> ffmpeg -i input.avi -r 24 output.avi

Honestly: Did you read my mail?

[...]

> I want it to leave the frames exactly the way they are 
> now but only play 24 of them every second so that the
> video runs slower.  I.E. instead of being 10:27 long 
> the video is 13:02.

The setpts filter supports this, it needs reencoding though.

> ffmpeg version 0.8.6-4:0.8.6-0ubuntu0.12.04.1

This is not FFmpeg but an intentionally broken version that 
contains several hundred bugs not present in FFmpeg, including
some security relevant issues, please understand that we 
cannot support it here.
See http://ffmpeg.org/download.html for supported versions.

Please do not top-post here, Carl Eugen


From Stephen_Hamer_UK at yahoo.co.uk  Sun Jul  7 00:22:39 2013
From: Stephen_Hamer_UK at yahoo.co.uk (Yahoo! mail)
Date: Sat, 06 Jul 2013 23:22:39 +0100
Subject: [FFmpeg-user] FFmpeg fails to install with opus enabled in Ubuntu
	12.04
Message-ID: <51D898AF.9010608@yahoo.co.uk>

Hi,

A noob running Ubuntu 12.04 on a samsung N110 netbook. Tried to install 
ffmeg using the instructions here:

http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide

Although I actually began with:

|sudo apt-get remove ffmpeg x264 libav-tools libvpx-dev libx264-dev
|

as suggested here:

http://askubuntu.com/questions/265371/please-give-me-an-ffmpeg-tutorial-that-works-on-ubuntu-12-04

Query: should I have removed my existing ffmpeg?

I then moved on to the instructions in the ffmpeg wiki, above,. The 
installs of Yasm, x264, fdk-aac, libmp3lame, libopus and libvpx all 
completed without a hitch [actually that's not quite correct: the final 
command in each command-block tended not to run; it was left dangling in 
the terminal, and I had to execute it with "Enter". The opus-install was 
left with ". ~/.profile" dangling, which I also treated as a command]

But the final install of ffmpeg broke down. Here is the terminal o/p:

|azed at azed-N110:~/ffmpeg_sources$ cd ffmpeg |

|azed at azed-N110:~/ffmpeg_sources/ffmpeg$ 
PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" |

|azed at azed-N110:~/ffmpeg_sources/ffmpeg$ ./configure 
--prefix="$HOME/ffmpeg_build" \ |

|> --extra-cflags="-I$HOME/ffmpeg_build/include" 
--extra-ldflags="-L$HOME/ffmpeg_build/lib" \ |

|> --bindir="$HOME/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 |

|ERROR: opus 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. |

|azed at azed-N110:~/ffmpeg_sources/ffmpeg$ make |

|Makefile:2: config.mak: No such file or directory |

|Makefile:48: /common.mak: No such file or directory |

|Makefile:91: /libavutil/Makefile: No such file or directory |

|Makefile:91: /library.mak: No such file or directory |

|Makefile:168: /doc/Makefile: No such file or directory |

|Makefile:169: /tests/Makefile: No such file or directory |

|make: *** No rule to make target `/tests/Makefile'. Stop. |

|azed at azed-N110:~/ffmpeg_sources/ffmpeg$ make install |

|Makefile:2: config.mak: No such file or directory |

|Makefile:48: /common.mak: No such file or directory |

|Makefile:91: /libavutil/Makefile: No such file or directory |

|Makefile:91: /library.mak: No such file or directory |

|Makefile:168: /doc/Makefile: No such file or directory |

|Makefile:169: /tests/Makefile: No such file or directory |

|make: *** No rule to make target `/tests/Makefile'. Stop. |

|azed at azed-N110:~/ffmpeg_sources/ffmpeg$ make distclean |

|Makefile:2: config.mak: No such file or directory |

|Makefile:48: /common.mak: No such file or directory |

|Makefile:91: /libavutil/Makefile: No such file or directory |

|Makefile:91: /library.mak: No such file or directory |

|Makefile:168: /doc/Makefile: No such file or directory |

|Makefile:169: /tests/Makefile: No such file or directory |

|make: *** No rule to make target `/tests/Makefile'. Stop. |

|azed at azed-N110:~/ffmpeg_sources/ffmpeg$ hash -r |


Back to me. I noticed that the first 3 commands in the install-ffmpeg 
command-block had run ok, and that I was still in the ~/ffmpeg folder, 
so I ran the fourth command, removing "|--enable-libopus|". This ran ok; 
so I then ran the remaining commands as a block, and these also ran ok. 
So I did, finally, get my ffmpeg install, albeit without opus.

azed at azed-N110:~$ ffmpeg 2>&1 | head -n1
ffmpeg version git-2013-07-06-7f25050 Copyright (c) 2000-2013 the FFmpeg 
developers

Or did I? This is what I really want to know. Have I simply installed 
ffmpeg without opus?

And, if so, how do I get rid of the opus software I have installed (I 
wasn't planning to use opus, and don't know why I tried to install it).

There seems to be a problem compiling (is that the word?) ffmpeg with 
opus in Ubuntu 12.04. I am really writing this e-mail to bring this to 
the attention of the ffmpeg developers - hope it doesn't sound like carping

Regards

Stephen Hamer

PS: sorry, I forgot to make a copy of the config.log of the failed 
ffmpeg install, and it got overwritten (I think) in the course of the 
successful install.

From joe at livetechkeynotes.com  Sun Jul  7 04:55:57 2013
From: joe at livetechkeynotes.com (Joe de Max)
Date: Sun, 7 Jul 2013 03:55:57 +0100
Subject: [FFmpeg-user] RTMP streaming dropped frames stuttering audio
Message-ID: <003BCC8D-0B1B-465D-99DD-B6DE91905E0F@livetechkeynotes.com>

Hey all,

I'm using FFmpeg on my Mac to stream the output of my TriCaster 8000 (essentially transcoding the 100Mb/s network output the TriCaster can provide and transcoding it to x264 for RTMP streaming). I am doing 2 outputs using the syntax below. All seems to work well until I do something like pan a camera or use a transition on the TriCaster (Alien Hand!) at which point the stream will pause, drop a bunch of frames and eventually pick back up  and play the past few seconds really quickly. I have tested outputting to 2 FLV files instead of the 2 RTMP streams and those files end up fine with no stuttering or dropped frames so it is not the TriCaster that is dropping frames or the speed of the network between the TriCaster and the Mac. I have 12 megabits up and I am only attempting to use about 2.5Mbps between the 2 streams so I'm really not sure where the problem lies...

Thanks!

Joe

ffmpeg -i http://192.168.1.112:49480/NewTekNetworkSend.mpg -s 1280x720 -r 50 -c:v libx264 -preset veryfast -profile:v high  -b:v 700k -strict -2 -c:a aac -b:a 192k -f flv - | 
ffmpeg -f flv -i - -c copy -f flv rtmp://server1  -c copy -f flv rtmp://server2

From lou at lrcd.com  Sun Jul  7 09:24:19 2013
From: lou at lrcd.com (Lou)
Date: Sat, 06 Jul 2013 23:24:19 -0800
Subject: [FFmpeg-user] FFmpeg fails to install with opus enabled in
 Ubuntu 12.04
In-Reply-To: <51D898AF.9010608@yahoo.co.uk>
References: <51D898AF.9010608@yahoo.co.uk>
Message-ID: <1373181859.29162.140661252724525.347E2ABC@webmail.messagingengine.com>

On Sat, Jul 6, 2013, at 02:22 PM, Yahoo! mail wrote:

> Query: should I have removed my existing ffmpeg?

You didn't need remove anything since the guide on ffmpeg.org does not
perform a system installation, so it can co-exist with any repository
packages. If nothing depended on the "ffmpeg" package then it doesn't
really matter, and if anything did you can simply just reinstall them
with your favorite package manager.

Side note: the so-called "ffmpeg" package in the repository is not
actually from the FFmpeg project:
http://stackoverflow.com/q/9477115/1109017

> So I did, finally, get my ffmpeg install, albeit without opus.
> 
> azed at azed-N110:~$ ffmpeg 2>&1 | head -n1
> ffmpeg version git-2013-07-06-7f25050 Copyright (c) 2000-2013 the FFmpeg 
> developers
> 
> Or did I? This is what I really want to know.

Yes.

> Have I simply installed ffmpeg without opus?

Unknown since you did not include the line that shows all of the
configure options. Alternatively you could run "ffmpeg -encoders | grep
opus".

> And, if so, how do I get rid of the opus software I have installed (I 
> wasn't planning to use opus, and don't know why I tried to install it).

Although you ran "make install", and assuming opus compilation was
successful, it didn't install anything to your system (it "installed" to
~/ffmpeg_build) so it won't do anything if you leave it. If you want to
get rid of all traces of just the opus stuff one method is:

$ cd ~/ffmpeg_sources/opus
$ make uninstall
$ rm -rf ~/ffmpeg_sources/opus

> There seems to be a problem compiling (is that the word?) ffmpeg with 
> opus in Ubuntu 12.04. I am really writing this e-mail to bring this to 
> the attention of the ffmpeg developers - hope it doesn't sound like
> carping

Thanks. It's certainly possible that something is incorrect or
incomplete, and I admit that I may not have tested all releases that the
guide claims to support.

From bbutscheidt at yahoo.de  Sun Jul  7 10:22:14 2013
From: bbutscheidt at yahoo.de (Bernd Butscheidt)
Date: Sun, 7 Jul 2013 09:22:14 +0100 (BST)
Subject: [FFmpeg-user] Is it possible to just change the fps flags?
In-Reply-To: <loom.20130706T225839-174@post.gmane.org>
References: <CAPajkzOQ49SBkB_y3Hzi8Qxi30FTxjMTsUUs3H7+OYSzpMJb9w@mail.gmail.com>
	<loom.20130706T185523-523@post.gmane.org>
	<CAPajkzOzZghG9TGyypyK65qt6Fn-cTyaSPhicFEy0bK=w6dBcw@mail.gmail.com>
	<loom.20130706T225839-174@post.gmane.org>
Message-ID: <1373185334.85170.YahooMailNeo@web133205.mail.ir2.yahoo.com>



----- Original Message -----
> From: Carl Eugen Hoyos <cehoyos at ag.or.at>
> 
>>  I want it to leave the frames exactly the way they are 
>>  now but only play 24 of them every second so that the
>>  video runs slower.? I.E. instead of being 10:27 long 
>>  the video is 13:02.
> 
> The setpts filter supports this, it needs reencoding though.

There was an effort to supply a patched version of ffmpeg doing this trick without reencoding (if the video format used supports this at all ...)
http://forum.doom9.org/showthread.php?t=152419

But what I see it is not maintained anymore. Still, there is a windows binary you can try.

From svnpenn at gmail.com  Sun Jul  7 09:59:30 2013
From: svnpenn at gmail.com (svnpenn)
Date: Sun, 7 Jul 2013 00:59:30 -0700 (PDT)
Subject: [FFmpeg-user] Remux to a "brand mp42" MP4 file
In-Reply-To: <loom.20130214T194915-724@post.gmane.org>
References: <66567361A4EC46149E5014E23C83D4CE@vasonote>
	<loom.20130214T194915-724@post.gmane.org>
Message-ID: <1373183970642-4659859.post@n4.nabble.com>

> Could you explain your use-case, so we would
> know what to test?
> (Why is mp41 not godd enough?)

Carl, I believe my issue is also caused by bad "atoms" being written by
FFmpeg

http://superuser.com/q/616484/ffmpeg-corrupts-itunes-files

GOOD
compatible_brands: M4A mp42isom

BAD
compatible_brands: isomiso2



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Remux-to-a-brand-mp42-MP4-file-tp4657305p4659859.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From fragabr at gmail.com  Sun Jul  7 10:56:33 2013
From: fragabr at gmail.com (=?UTF-8?B?RMOibmllbA==?= Fraga)
Date: Sun, 7 Jul 2013 05:56:33 -0300
Subject: [FFmpeg-user] Trivial: ffmpeg configure doesn't detect libva path
Message-ID: <krbafq$d7s$1@ger.gmane.org>

	I have libva installed in /usr/xorg/lib and when I configure
ffmpeg it doesn't detect libva.

	The solution is to move libva to /usr/local/lib. The problem is
that if I do that "vainfo" doesn't work anymore.

	Is there a way to force ffmpeg's configure to use libva
from /usr/xorg/lib ?

	I searched for some configure options, but I couldn't found a
specific one. 

	Thanks.

-- 
Linux 3.10.0: Unicycling Gorilla
http://www.youtube.com/DanielFragaBR
http://www.libertarios.org.br



From fragabr at gmail.com  Sun Jul  7 11:00:38 2013
From: fragabr at gmail.com (=?UTF-8?B?RMOibmllbA==?= Fraga)
Date: Sun, 7 Jul 2013 06:00:38 -0300
Subject: [FFmpeg-user] Trivial: ffmpeg configure doesn't detect libva
	path (SOLVED)
References: <krbafq$d7s$1@ger.gmane.org>
Message-ID: <krbanf$d7s$2@ger.gmane.org>

On Sun, 7 Jul 2013 05:56:33 -0300
D?niel Fraga <fragabr at gmail.com> wrote:

> 	Is there a way to force ffmpeg's configure to use libva
> from /usr/xorg/lib ?

	Nevermind! The following solved the issue. It wouldn't search
for va.h inside /usr/xorg/include:

CPPFLAGS=-I/usr/xorg/include

-- 
Linux 3.10.0: Unicycling Gorilla
http://www.youtube.com/DanielFragaBR
http://www.libertarios.org.br



From cehoyos at ag.or.at  Sun Jul  7 13:23:20 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 7 Jul 2013 11:23:20 +0000 (UTC)
Subject: [FFmpeg-user] Trivial: ffmpeg configure doesn't detect libva
	path
References: <krbafq$d7s$1@ger.gmane.org>
Message-ID: <loom.20130707T132202-366@post.gmane.org>

D?niel Fraga <fragabr <at> gmail.com> writes:

> Is there a way to force ffmpeg's configure to use libva
> from /usr/xorg/lib ?

Did you try --extra-cflags=-I/usr/xorg/include ?
This is supported, other solutions may brake without notice 
or may brake compilation.

Carl Eugen


From cehoyos at ag.or.at  Sun Jul  7 13:28:17 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 7 Jul 2013 11:28:17 +0000 (UTC)
Subject: [FFmpeg-user] Is it possible to just change the fps flags?
References: <CAPajkzOQ49SBkB_y3Hzi8Qxi30FTxjMTsUUs3H7+OYSzpMJb9w@mail.gmail.com>
	<loom.20130706T185523-523@post.gmane.org>
	<CAPajkzOzZghG9TGyypyK65qt6Fn-cTyaSPhicFEy0bK=w6dBcw@mail.gmail.com>
	<loom.20130706T225839-174@post.gmane.org>
	<1373185334.85170.YahooMailNeo@web133205.mail.ir2.yahoo.com>
Message-ID: <loom.20130707T132714-582@post.gmane.org>

Bernd Butscheidt <bbutscheidt <at> yahoo.de> writes:

> >>  I want it to leave the frames exactly the way they are 
> >>  now but only play 24 of them every second so that the
> >>  video runs slower.? I.E. instead of being 10:27 long 
> >>  the video is 13:02.
> > 
> > The setpts filter supports this, it needs reencoding though.
> 
> There was an effort to supply a patched version of ffmpeg 
> doing this trick without reencoding

I have a suspicion it actually works with current FFmpeg 
as well, I just never tested it (and I am unsure if 
this feature isn't actually a bug).

Carl Eugen


From cehoyos at ag.or.at  Sun Jul  7 13:32:50 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 7 Jul 2013 11:32:50 +0000 (UTC)
Subject: [FFmpeg-user] Remux to a "brand mp42" MP4 file
References: <66567361A4EC46149E5014E23C83D4CE@vasonote>
	<loom.20130214T194915-724@post.gmane.org>
	<1373183970642-4659859.post@n4.nabble.com>
Message-ID: <loom.20130707T133020-977@post.gmane.org>

svnpenn <svnpenn <at> gmail.com> writes:

> > Could you explain your use-case, so we would
> > know what to test?
> > (Why is mp41 not godd enough?)
> 
> Carl, I believe my issue is also caused by bad 
> "atoms" being written by FFmpeg

Command line and complete, uncut console output missing.

Please test -f ipod


Carl Eugen


From cehoyos at ag.or.at  Sun Jul  7 13:26:15 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 7 Jul 2013 11:26:15 +0000 (UTC)
Subject: [FFmpeg-user] RTMP streaming dropped frames stuttering audio
References: <003BCC8D-0B1B-465D-99DD-B6DE91905E0F@livetechkeynotes.com>
Message-ID: <loom.20130707T132338-809@post.gmane.org>

Joe de Max <joe <at> livetechkeynotes.com> writes:

> ffmpeg -i http://192.168.1.112:49480/NewTekNetworkSend.mpg 

> -s 1280x720 -r 50 -c:v libx264 -preset veryfast 

(You may want to test ultrafast.)

> -profile:v high  -b:v 700k -strict -2 -c:a aac 
> -b:a 192k -f flv - | ffmpeg -f flv -i - -c copy 
> -f flv rtmp://server1  -c copy -f flv rtmp://server2


(Complete, uncut console output missing.)

Sorry if I miss something but don't you agree that using 
two ffmpeg instances must produce an overhead?
I didn't test but I would expect that using only one 
instance saves a few cycles (which may or may not be 
enough to fix your problem).

Carl Eugen


From nixscripter at gmail.com  Sun Jul  7 20:47:00 2013
From: nixscripter at gmail.com (Dan Harrison)
Date: Sun, 7 Jul 2013 13:47:00 -0500
Subject: [FFmpeg-user] RTMP stream URL truncation
Message-ID: <CAF9111UGwU5mSccmDHwA2PeyDT_siZ1W5ehv3pr12WTWaOrkyQ@mail.gmail.com>

Hello, list.

I'm trying to use FFmpeg to save an RTMP stream from the internet (i.e. a
server I don't control). It claims it cannot find the stream. When I look
at verbose output, however, the problem is clear:

$ ffmpeg -loglevel verbose -i "rtmp://foo.net/cfx/st/mp4:BAR_BAZ/file.mp4"
out.mp4
...
HandleInvoke, received result for method call
<createStream>
SendPlay, seekTime=0, stopTime=0, sending play: mp4:file

Invoking
play
...

Notice that "file.mp4" got its suffix dropped, even though it was part of
the stream's name. As a result, the server replies it cannot find the
stream:

...
(object begin)
Property: <Name:              level, STRING:    error>
Property: <Name:               code, STRING:
NetStream.Play.StreamNotFound>
Property: <Name:        description, STRING:    Failed to play file; stream
not found.>
Property: <Name:            details, STRING:    file>
Property: <Name:           clientid, STRING:    xlaQOJIn>
(object end)
...

I'm guessing that FFmpeg used the librtmp library the same way rtmpdump
does, because it does the same thing (verbose output not shown is identical
to the above):

$ rtmpdump -r "rtmp://foo.net/cfx/st/mp4:BAR_BAZ/file.mp4" -o out.mp4
(c) 2010 Andrej Stepanchuk, Howard Chu, The Flvstreamer Team; license: GPL
Connecting ...
INFO: Connected...
...
ERROR: Closing connection: NetStream.Play.StreamNotFound

And yet, when I use mplayer or mencoder -- which also uses librtmp -- it
decides not to drop the suffix, and works correctly:

$ mplayer -msglevel all=6 "rtmp://foo.net/cfx/st/mp4:BAR_BAZ/file.mp4"
...
[rtmp @ 0x7f33ea814060]Handshaking...
[rtmp @ 0x7f33ea814060]Type answer 3
[rtmp @ 0x7f33ea814060]Server version 3.5.7.1
[rtmp @ 0x7f33ea814060]Proto = rtmp, path = /cfx/st/mp4:BAR_BAZ/file.mp4,
app = cfx/st, fname = mp4:BAR_BAZ/file.mp4
[rtmp @ 0x7f33ea814060]Server bandwidth = 1310720
[rtmp @ 0x7f33ea814060]Client bandwidth = 2500000
[rtmp @ 0x7f33ea814060]Creating stream...
[rtmp @ 0x7f33ea814060]Sending play command for 'mp4:BAR_BAZ/file.mp4'
[rtmp @ 0x7f33ea814060]New incoming chunk size = 4096
STREAM: [ffmpeg] rtmp://foo.net/cfx/st/mp4:BAR_BAZ/file.mp4
...

In fact, as you can see, mplayer uses libavcodec to decode the stream! So I
know the code all works once this little problem gets solved.

How do I get FFmpeg to tack my suffix back on?
--
Dan Harrison

From fragabr at gmail.com  Mon Jul  8 03:49:44 2013
From: fragabr at gmail.com (=?UTF-8?B?RMOibmllbA==?= Fraga)
Date: Sun, 7 Jul 2013 22:49:44 -0300
Subject: [FFmpeg-user] Trivial: ffmpeg configure doesn't detect libva
	path
References: <krbafq$d7s$1@ger.gmane.org>
	<loom.20130707T132202-366@post.gmane.org>
Message-ID: <krd5ri$vmm$1@ger.gmane.org>

On Sun, 7 Jul 2013 11:23:20 +0000 (UTC)
Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Did you try --extra-cflags=-I/usr/xorg/include ?
> This is supported, other solutions may brake without notice 
> or may brake compilation.

	That works! Thanks Carl!

-- 
Linux 3.10.0: Unicycling Gorilla
http://www.youtube.com/DanielFragaBR
http://www.libertarios.org.br



From me at robpitt.me.uk  Mon Jul  8 07:43:53 2013
From: me at robpitt.me.uk (me at robpitt.me.uk)
Date: Mon, 8 Jul 2013 06:43:53 +0100
Subject: [FFmpeg-user] Record microphone on android
Message-ID: <847d645e-537e-46ba-b16e-e829f437930b.maildroid@localhost>

Hello, 

I can record the screen as video with

ffmpeg -f fbdev -r 25 -i /dev/graphics/fb0 -y -vcodec mpeg4 -b 2000k -s 480x853 new.mp4

I would like to record the microphone as the audio track, how do I accomplish this? 
Sent from my android device.

From salmjuh at hotmail.com  Mon Jul  8 09:36:41 2013
From: salmjuh at hotmail.com (Jii)
Date: Mon, 8 Jul 2013 00:36:41 -0700 (PDT)
Subject: [FFmpeg-user] rawvideo -> libx264
In-Reply-To: <20130705125013.69b017fc@lrcd.com>
References: <DUB123-W3224477D40EDB56446A70EAB7D0@phx.gbl>
	<20130705125013.69b017fc@lrcd.com>
Message-ID: <1373269001066-4659870.post@n4.nabble.com>

Hi

And thanks fo this, it worked nice. One more problem capturing this
SDI-signal. There is also embedded audio in the SDI but ffmpeg can't handle
this. So do I need codec or is there something else to do with this one?


br,
juha



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/rawvideo-libx264-tp4659842p4659870.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From salmjuh at hotmail.com  Mon Jul  8 11:56:05 2013
From: salmjuh at hotmail.com (juha s.)
Date: Mon, 8 Jul 2013 12:56:05 +0300
Subject: [FFmpeg-user] Capturin SDI / No audio support
Message-ID: <DUB123-W10FB24A4BDE9187F24676CAB780@phx.gbl>

Hi
I'm succesfully now can capture video from Blackmagic's Duo card. SD- and HD-SDI bouth working great. But it seems that ffmpeg doesn't support embedded PCM audio:
-------------------  SNIP  ----------------------------
 bmdcapture -m 2 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -re -i - -copyts -r 50 -s 640x360 -aspect 16:9 -vcodec libx264  -vprofile main -preset medium -crf 20   -b:v 1100k -maxrate 1300k -bufsize 1500k  -pix_fmt yuv420p  -y  /tmp/test.mp4ffmpeg version git-2012-10-02-c785b6d Copyright (c) 2000-2012 the FFmpeg developers  built on Jul  5 2013 13:28:23 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)  configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab --enable-libvpx  libavutil      51. 73.101 / 51. 73.101  libavcodec     54. 62.100 / 54. 62.100  libavformat    54. 29.105 / 54. 29.105  libavdevice    54.  3.100 / 54.  3.100  libavfilter     3. 19.100 /  3. 19.100  libswscale      2.  1.101 /  2.  1.101  libswresample   0. 16.100 /  0. 16.100  libpostproc    52.  1.100 / 52.  1.100[nut @ 0x2273280] Unknown codec tag '0x10445350' for stream number 1[nut @ 0x2273280] Could not find codec parameters for stream 1 (Audio: none (PSD[16] / 0x10445350), 48000 Hz, 2 channels): unknown codecConsider increasing the value for the 'analyzeduration' and 'probesize' options[nut @ 0x2273280] Estimating duration from bitrate, this may be inaccurateGuessed Channel Layout for  Input Stream #0.1 : stereoInput #0, nut, from 'pipe:':  Metadata:    encoder         : Lavf54.20.3  Duration: N/A, start: 0.000000, bitrate: N/A    Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x576, 25 fps, 25 tbr, 25 tbn, 25 tbc    Stream #0:1: Audio: none (PSD[16] / 0x10445350), 48000 Hz, stereo[graph 1 input from stream 0:1 @ 0x22eeca0] Invalid sample format '(null)'Error opening filters!
-------------------  SNIP  ----------------------------

So please, every help would be highly appreciate. 
Br,Jii

 		 	   		  

From barsnick at gmx.net  Mon Jul  8 19:49:23 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Mon, 8 Jul 2013 19:49:23 +0200
Subject: [FFmpeg-user] Capturin SDI / No audio support
In-Reply-To: <DUB123-W10FB24A4BDE9187F24676CAB780@phx.gbl>
References: <DUB123-W10FB24A4BDE9187F24676CAB780@phx.gbl>
Message-ID: <20130708174923.GA14280@sunshine.barsnick.net>

(Wow, I actually had to insert manual line breaks to read this.)

On Mon, Jul 08, 2013 at 12:56:05 +0300, juha s. wrote:

> bmdcapture -m 2 -C 0 -A 2 -V 4 -F nut -f pipe:1 | ffmpeg -re -i - -copyts -r 50 -s 640x360 -aspect 16:9 -vcodec libx264  -vprofile main -preset medium -crf 20   -b:v 1100k -maxrate 1300k -bufsize 1500k  -pix_fmt yuv420p  -y  /tmp/test.mp4
> ffmpeg version git-2012-10-02-c785b6d Copyright (c) 2000-2012 the FFmpeg developers

This is quite old. Could you try a recent version, i.e. latest snapshot
or git?

> [nut @ 0x2273280] Unknown codec tag '0x10445350' for stream number 1
> [nut @ 0x2273280] Could not find codec parameters for stream 1 (Audio: none (PSD[16] / 0x10445350), 48000 Hz, 2 channels): unknown codec

I have seen indications on the net that this should be detected as
"pcm_s16le" instead of "none", if I'm not totally mistaken. You might
want to try forcing that. But first, get a newer version of ffmpeg and
try that. :-)

Moritz

From cehoyos at ag.or.at  Mon Jul  8 23:17:04 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 8 Jul 2013 21:17:04 +0000 (UTC)
Subject: [FFmpeg-user] [FFmpeg user] Frame drops with file in ffplay
References: <CAHss8oGTwe+iDd+kLYbfu--rFFGPf-bo4XC1cdWAfcxkRCr0ow@mail.gmail.com>
	<loom.20130515T174803-761@post.gmane.org>
Message-ID: <loom.20130708T231647-244@post.gmane.org>

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

> Massimo Battistel <battistel <at> gmail.com> writes:
> 
> > http://www.sendspace.com/file/xq8q4v
> > 
> > there is a frame drop problem in ffplay.
> 
> I opened ticket #2574 for the regression.

Should be fixed, thank you for the report!

Carl Eugen


From bobm-ffmpeg at burner.com  Mon Jul  8 23:16:53 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Mon, 08 Jul 2013 15:16:53 -0600
Subject: [FFmpeg-user] A change in -timecode ?
Message-ID: <51DB2C45.8040004@burner.com>

Updated to a more recent Windows build of ffmpeg and have been getting
something new when using the -timecode switch making Quicktimes:

  Codec AVOption gop_timecode (MPEG GOP Timecode in hh:mm:ss[:;.]ff
  format) specified for output file #0 (D:\13HD2002H_ProRes.mov) 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.

but I'm sort of not deciphering the meaning, and further confused by the
fact that it all seems to work anyway -- I get the desired timecode in
my output movie as before.

Is there some other method I should be using, and/or what is this
warning (?) really trying to tell me?


Complete console:

ffmpeg.exe -i "D:\13HD2000H.mov" -timecode 01:00:00:00 -top 1 -flags
+ildct -codec:v prores_ks -profile:v hq -pix_fmt yuv422p10le -vf
colormatrix=bt601:bt709 -threads 8 -codec:a pcm_s16le -filter_complex
asetnsamples=n=16384:p=0 "D:\13HD2000H_ProRes.mov"

ffmpeg version N-54277-gff130d7 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jun 30 2013 01:25:12 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. 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
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce240] Stream #1: not enough
frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce240] Stream #2: not enough
frames to estimate rate; consider increasing probesize
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\13HD2000H.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2013-07-08 19:30:37
  Duration: 00:00:30.03, start: 0.000000, bitrate: 993856 kb/s
    Stream #0:0(eng): Video: qtrle (rle  / 0x20656C72), rgb24,
1920x1080, 991547 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 30k tbn,
30k tbc
    Metadata:
      creation_time   : 2013-07-08 19:30:37
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
    Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz,
stereo, s32, 2304 kb/s
    Metadata:
      creation_time   : 2013-07-08 19:30:37
      handler_name    : Apple Alias Data Handler
    Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
    Metadata:
      creation_time   : 2013-07-08 19:31:33
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
Codec AVOption gop_timecode (MPEG GOP Timecode in hh:mm:ss[:;.]ff
format) specified for output file #0 (D:\13HD2000H_ProRes.mov) 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, mov, to 'D:\13HD2000H_ProRes.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    timecode        : 01:00:00:00
    encoder         : Lavf55.10.100
    Stream #0:0: Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo,
s16, 1536 kb/s
    Stream #0:1(eng): Video: prores (prores_ks) (apch / 0x68637061),
yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31,200 kb/s, 30k tbn,
29.97 tbc
    Metadata:
      creation_time   : 2013-07-08 19:30:37
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
Stream mapping:
  Stream #0:1 (pcm_s24le) -> asetnsamples (graph 0)
  asetnsamples (graph 0) -> Stream #0:0 (pcm_s16le)
  Stream #0:0 -> #0:1 (qtrle -> prores_ks)
Press [q] to stop, [?] for help
frame=   56 fps=6.6 q=0.0 size=   38900kB time=00:00:01.86
bitrate=170546.9kbits/s

From cehoyos at ag.or.at  Mon Jul  8 23:20:22 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 8 Jul 2013 21:20:22 +0000 (UTC)
Subject: [FFmpeg-user] Capturin SDI / No audio support
References: <DUB123-W10FB24A4BDE9187F24676CAB780@phx.gbl>
Message-ID: <loom.20130708T231937-679@post.gmane.org>

juha s. <salmjuh <at> hotmail.com> writes:

> it seems that ffmpeg doesn't support embedded PCM audio:
> -------------------  SNIP  ----------------------------
>  bmdcapture -m 2 -C 0 -A 2 -V 4 -F nut

Is the problem reproducible if you save this stream 
to a file?
If yes, please provide a sample.

Carl Eugen


From cehoyos at ag.or.at  Mon Jul  8 23:25:45 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 8 Jul 2013 21:25:45 +0000 (UTC)
Subject: [FFmpeg-user] A change in -timecode ?
References: <51DB2C45.8040004@burner.com>
Message-ID: <loom.20130708T232242-289@post.gmane.org>

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

>     timecode        : 01:00:00:00

Do I understand correctly that the output file 
is correct (as far as timecode is concerned) 
and that the message is the only bad thing?

(My problem is that I simply fail to understand 
what a "timecode" could be, so I typically don't 
understand the numerous reports connected to it.)

Iiuc, the message appears because ffmpeg (the 
executable) translates "-timecode" to "-gop_timecode" 
but neither encoder nor muxer react on the later 
option (but they apparently react to another 
setting from ffmpeg).

Carl Eugen


From bobm-ffmpeg at burner.com  Mon Jul  8 23:47:35 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Mon, 08 Jul 2013 15:47:35 -0600
Subject: [FFmpeg-user] A change in -timecode ?
In-Reply-To: <loom.20130708T232242-289@post.gmane.org>
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
Message-ID: <51DB3377.8060003@burner.com>

On 7/8/2013 3:25 PM, Carl Eugen Hoyos wrote:

> Do I understand correctly that the output file 
> is correct (as far as timecode is concerned) 
> and that the message is the only bad thing?

Correct.

> Iiuc, the message appears because ffmpeg (the 
> executable) translates "-timecode" to "-gop_timecode" 
> but neither encoder nor muxer react on the later 
> option (but they apparently react to another 
> setting from ffmpeg).

Hmmm.  I guess I'll just ignore it for now, until it stops working :)
It's just a warning that I've never seen before in prior versions and I
wondered what had changed.


From cehoyos at ag.or.at  Tue Jul  9 02:14:53 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 9 Jul 2013 00:14:53 +0000 (UTC)
Subject: [FFmpeg-user] A change in -timecode ?
References: <51DB2C45.8040004@burner.com>
Message-ID: <loom.20130709T021352-375@post.gmane.org>

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

> Updated to a more recent Windows build of ffmpeg 
> and have been getting something new when using 
> the -timecode switch making Quicktimes:
> 
>   Codec AVOption gop_timecode (MPEG GOP Timecode 
>   in hh:mm:ss[:;.]ff format) specified for 
>   output file #0 (D:\13HD2002H_ProRes.mov) has 
>   not been used for any stream.

I opened ticket #2762, thank you for the report!

Carl Eugen


From jam at tigger.ws  Tue Jul  9 04:26:01 2013
From: jam at tigger.ws (James Linder)
Date: Tue, 9 Jul 2013 10:26:01 +0800
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <loom.20130708T232242-289@post.gmane.org>
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
Message-ID: <DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>

Some years ago I setup a record-v4l-camera-from-interface with audio on line in using transcode. Perfect, but at kernel 2.6.32

I modernized the recording system only to find transcode still uses VIDIOC_S_PARM ioctls that were removed from kernel cica 2.6.33

I use ffmpeg now. All good except the audio is not synced. I apply -itsoffset and all is good for an hour or two then the recording hesitates every 5 or so seconds for say 100mS. I've looked at top, vmstat, iostat even gnome-system-monitor and can see nothing that would be causing it.

ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 -i /dev/video0 -vtag divx -s svga -r pal -q:v 5 file.avi -s svga -f rawvideo - | ffplay -f rawvideo -i - -s svga

There is not any swap or obvious unusual disk activity.

Any ideas welcome
James

PS
I tried avconv instead of ffmpeg and the stuttering ceased (joy, rapture) but I cannot sync audio (sad)
avplay stutters where ffplay does not

From thefatalaccidents at gmail.com  Mon Jul  8 22:02:03 2013
From: thefatalaccidents at gmail.com (fatalaccidents)
Date: Mon, 8 Jul 2013 13:02:03 -0700 (PDT)
Subject: [FFmpeg-user] Problem with white-washed images
Message-ID: <1373313723432-4659873.post@n4.nabble.com>

????????????Hey guys, 

I have been having a problem with converting multiple jpegs into an mp4. 
The problem seems to be with the colorspaces, or at least in the way that
most video players play back certain mp4s with certain colorspaces.  I have
posted on this without being able to resolve it, but I thought giving more
information might help me come to a working conclusion.  The images I start
with are grey, and an identify -verbose of one looks like: 

[code] 
Image: GA1_264000_b3.jpg
  Format: JPEG (Joint Photographic Experts Group JFIF format)
  Class: PseudoClass
  Geometry: 1800x1800+0+0
  Resolution: 72x72
  Print size: 25x25
  Units: PixelsPerInch
  Type: Grayscale
  Endianess: Undefined
  Colorspace: Gray
  Depth: 8-bit
  Channel depth:
    gray: 8-bit
  Channel statistics:
    Gray:
      min: 0 (0)
      max: 255 (1)
      mean: 211.31 (0.828667)
      standard deviation: 92.5697 (0.363019)
      kurtosis: 1.30688
      skewness: -1.79377
  Colors: 256
  Histogram: 
  Colormap:
  Rendering intent: Undefined
  Gamma: 1
  Background color: gray(255)
  Border color: gray(223)
  Matte color: gray(189)
  Transparent color: gray(0)
  Interlace: None
  Intensity: Undefined
  Compose: Over
  Page geometry: 1800x1800+0+0
  Dispose: Undefined
  Iterations: 0
  Compression: JPEG
  Quality: 100
  Orientation: Undefined
  Properties:
    date:create: 2013-07-02T10:27:12-05:00
    date:modify: 2013-05-26T11:16:54-05:00
    jpeg:colorspace: 1
    jpeg:sampling-factor: 1x1
    signature:
abeb35b2067e33de189df94bb0f1b973aa8c9ce47a70ebdd3da71661abd4c6fc
  Artifacts:
    filename: GA1_264000_b3.jpg
    verbose: true
  Tainted: True
  Filesize: 285KB
  Number pixels: 3.24M
  Pixels per second: 29.45MB
[/code]


I took the Histogram and Colormap out for space, but the colors go from 0 to
255.  I think somewhere along the lines it is being compressed into a
smaller scale, and white washing the resulting video.  FFplay will play the
video with the correct colors, but every other player messes it up.  Has
anyone encountered this problem before?   I've been trying to fix it for
quite some time now.

The command: ffmpeg -f image2 -r 10 -pattern_type glob -i '*.jpg' -c:v
libx264 out.mp4   gives this output:
[code]
ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
  built on May 29 2013 17:51:44 with gcc 4.2.1 (GCC) (Apple Inc. build 5666)
(dot 3)
  configuration: --prefix=/opt/local --enable-swscale --enable-avfilter
--enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora
--enable-libschroedinger --enable-libopenjpeg --enable-libmodplug
--enable-libvpx --enable-libspeex --enable-libfreetype
--mandir=/opt/local/share/man --enable-shared --enable-pthreads
--cc=/usr/bin/gcc-4.2 --enable-gpl --enable-postproc --enable-libx264
--enable-libxvid --arch=x86_64 --enable-yasm
  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, image2, from '*.jpg':
  Duration: 00:00:12.10, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: mjpeg, gray, 1800x1800 [SAR 72:72 DAR 1:1], 10 fps,
10 tbr, 10 tbn, 10 tbc
[libx264 @ 0x102009e00] using SAR=1/1
[libx264 @ 0x102009e00] using cpu capabilities: none!
[libx264 @ 0x102009e00] profile High, level 5.0
[libx264 @ 0x102009e00] 264 - core 129 - 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=10 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 'out.mp4':
  Metadata:
    encoder         : Lavf54.63.104
    Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuvj420p, 1800x1800
[SAR 1:1 DAR 1:1], q=-1--1, 10240 tbn, 10 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (mjpeg -> libx264)
Press [q] to stop, [?] for help
frame=  121 fps=3.0 q=-1.0 Lsize=    1955kB time=00:00:11.90
bitrate=1345.7kbits/s    
video:1953kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.107075%
[libx264 @ 0x102009e00] frame I:2     Avg QP:15.70  size:108097
[libx264 @ 0x102009e00] frame P:51    Avg QP:19.14  size: 30949
[libx264 @ 0x102009e00] frame B:68    Avg QP:20.66  size:  3003
[libx264 @ 0x102009e00] consecutive B-frames: 23.1%  3.3%  7.4% 66.1%
[libx264 @ 0x102009e00] mb I  I16..4: 66.7% 19.1% 14.2%
[libx264 @ 0x102009e00] mb P  I16..4: 13.6%  8.7%  2.3%  P16..4:  7.1%  0.7% 
0.7%  0.0%  0.0%    skip:66.8%
[libx264 @ 0x102009e00] mb B  I16..4:  0.1%  0.2%  0.1%  B16..8:  3.0%  0.2% 
0.1%  direct: 1.6%  skip:94.7%  L0:37.0% L1:58.6% BI: 4.4%
[libx264 @ 0x102009e00] 8x8 transform intra:33.5% inter:65.0%
[libx264 @ 0x102009e00] coded y,uvDC,uvAC intra: 29.2% 0.0% 0.0% inter: 4.4%
0.0% 0.0%
[libx264 @ 0x102009e00] i16 v,h,dc,p: 94%  3%  1%  2%
[libx264 @ 0x102009e00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu:  9% 23% 25%  6%  8% 
5% 11%  4%  9%
[libx264 @ 0x102009e00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 13% 26% 23%  6%  8% 
5% 10%  3%  6%
[libx264 @ 0x102009e00] i8c dc,h,v,p: 100%  0%  0%  0%
[libx264 @ 0x102009e00] Weighted P-Frames: Y:47.1% UV:0.0%
[libx264 @ 0x102009e00] ref P L0: 48.0% 33.3% 10.3%  3.3%  5.1%
[libx264 @ 0x102009e00] ref B L0: 78.2% 17.5%  4.3%
[libx264 @ 0x102009e00] ref B L1: 94.6%  5.4%
[libx264 @ 0x102009e00] kb/s:1321.53
[/code]

When played with mplayer, I'm getting this output: 
[code]
MPlayer 1.1-4.2.1 (C) 2000-2012 MPlayer Team
192 audio & 400 video codecs

Playing out.mp4.
libavformat version 54.6.101 (internal)
libavformat file format detected.
[lavf] stream 0: video (h264), -vid 0
VIDEO:  [H264]  1800x1800  24bpp  10.000 fps  1322.0 kbps (161.4 kbyte/s)
Clip info:
 major_brand: isom
 minor_version: 512
 compatible_brands: isomiso2avc1mp41
 encoder: Lavf54.63.104
Load subtitles in ./
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
libavcodec version 54.23.100 (internal)
Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264)
==========================================================================
Audio: no sound
Starting playback...
Unsupported PixelFormat 61
Unsupported PixelFormat 53
Unsupported PixelFormat 81
Could not find matching colorspace - retrying with -vf scale...
Opening video filter: [scale]
Movie-Aspect is 1.00:1 - prescaling to correct movie aspect.
[swscaler @ 0x100b41620]BICUBIC scaler, from yuv420p to yuyv422 using MMX2
VO: [corevideo] 1800x1800 => 1800x1800 Packed YUY2 
[/code]

If I capture the first image from the video and identify it, it is much
different in terms of its histogram than the first jpeg.
Any help would be greatly, greatly appreciated. 

Thanks!



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Problem-with-white-washed-images-tp4659873.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From cehoyos at ag.or.at  Tue Jul  9 09:44:57 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 9 Jul 2013 07:44:57 +0000 (UTC)
Subject: [FFmpeg-user] Comments about audio sync invited
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
Message-ID: <loom.20130709T094054-726@post.gmane.org>

James Linder <jam <at> tigger.ws> writes:

> ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 
> -i /dev/video0 -vtag divx -s svga -r pal -q:v 5 
> file.avi -s svga -f rawvideo - | 
> ffplay -f rawvideo -i - -s svga

Complete, uncut console output missing.

Do I understand correctly that the output file out.avi 
shows problems or did you only see problems in the 
ffplay output?
If this is about the output file, is the problem also 
reproducible with ffmpeg alone (without the pipe to 
ffplay)?

I never tested pulse but it was suggested repeatedly 
not to use it, try alsa instead.

Please understand that thread-hijacking is considered 
rude here (as is top-posting), Carl Eugen


From cehoyos at ag.or.at  Tue Jul  9 09:48:32 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 9 Jul 2013 07:48:32 +0000 (UTC)
Subject: [FFmpeg-user] Problem with white-washed images
References: <1373313723432-4659873.post@n4.nabble.com>
Message-ID: <loom.20130709T094555-920@post.gmane.org>

fatalaccidents <thefatalaccidents <at> gmail.com> writes:

> I have been having a problem with converting 
> multiple jpegs into an mp4. 

> The images I start with are grey

Please test current git head, I believe this 
bug was fixed yesterday.

If you can confirm the bug is fixed, consider 
posting on ffmpeg-devel, some developers claim 
that the old behaviour is expected and desirable 
and don't want this fixed in 1.2.2

Carl Eugen


From nichot20 at yahoo.com  Tue Jul  9 10:35:52 2013
From: nichot20 at yahoo.com (Tim Nicholson)
Date: Tue, 09 Jul 2013 09:35:52 +0100
Subject: [FFmpeg-user] A change in -timecode ?
In-Reply-To: <51DB2C45.8040004@burner.com>
References: <51DB2C45.8040004@burner.com>
Message-ID: <51DBCB68.1050801@yahoo.com>

On 08/07/13 22:16, Bob Maple wrote:
> Updated to a more recent Windows build of ffmpeg and have been getting
> something new when using the -timecode switch making Quicktimes:
> 
>   Codec AVOption gop_timecode (MPEG GOP Timecode in hh:mm:ss[:;.]ff
>   format) specified for output file #0 (D:\13HD2002H_ProRes.mov) 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.
> 
> but I'm sort of not deciphering the meaning, and further confused by the
> fact that it all seems to work anyway -- I get the desired timecode in
> my output movie as before.
> 
> Is there some other method I should be using, and/or what is this
> warning (?) really trying to tell me?

This is a recent and somewhat confusing message.

The problem seems to be that the '-timecode' option is used for adding
both .mov timecode streams that can be used in any mov with any codec,
and also for setting an mpeg stream timecode in a variety of muxers. The
warning is happening because you are coding to a non mpeg stream, but
have asked for timecode  and so the MPEG GOP timecode side of the code
is telling you there is no valid stream to add the timecode too. However
since you are trying to add a mov wrapper timecode this is not an issue.

IMHO this is a bug, as the warning is incorrect in this situation.


-- 
Tim

From jam at tigger.ws  Tue Jul  9 13:41:43 2013
From: jam at tigger.ws (James Linder)
Date: Tue, 9 Jul 2013 19:41:43 +0800
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <loom.20130709T094054-726@post.gmane.org>
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
	<loom.20130709T094054-726@post.gmane.org>
Message-ID: <70EF77B2-4254-417D-8D2B-4849B793A5D9@tigger.ws>


On 09/07/2013, at 3:44 PM, Carl Eugen Hoyos wrote:

> James Linder <jam <at> tigger.ws> writes:
> 
>> ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 
>> -i /dev/video0 -vtag divx -s svga -r pal -q:v 5 
>> file.avi -s svga -f rawvideo - | 
>> ffplay -f rawvideo -i - -s svga
> 
> Complete, uncut console output missing.

dvr at dvr:~> ffmpeg -f pulse -i default -f video4linux2 -i /dev/video0 -vtag divx -s svga -q:v 5 /home/dvr/Tue.avi
ffmpeg version 1.0.6 Copyright (c) 2000-2013 the FFmpeg developers
  built on Apr 11 2013 18:03:32 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 -g -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      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
[pulse @ 0xc352a0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, pulse, from 'default':
  Duration: N/A, start: 0.636342, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
[video4linux2,v4l2 @ 0xc32520] Estimating duration from bitrate, this may be inaccurate
Input #1, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 1373359581.412447, bitrate: 23040 kb/s
    Stream #1:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 320x240, 23040 kb/s, 25 tbr, 1000k tbn, 25 tbc
Output #0, avi, to '/home/dvr/Tue.avi':
  Metadata:
    ISFT            : Lavf54.29.104
    Stream #0:0: Video: mpeg4 (divx / 0x78766964), yuv420p, 800x600, q=2-31, 200 kb/s, 25 tbn, 25 tbc
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16
Stream mapping:
  Stream #1:0 -> #0:0 (rawvideo -> mpeg4)
  Stream #0:0 -> #0:1 (pcm_s16le -> libmp3lame)
Press [q] to stop, [?] for help
frame=    7 fps=0.0 q=5.0 size=     769kB time=00:00:01.17 bitrate=5354.8kbits/s
frame=   14 fps= 14 q=5.0 size=    1264kB time=00:00:01.17 bitrate=8805.9kbits/s
frame=   20 fps= 13 q=5.0 size=    1412kB time=00:00:02.16 bitrate=5354.8kbits/s
frame=   27 fps= 13 q=5.0 size=    1572kB time=00:00:02.16 bitrate=5963.2kbits/s
frame=   35 fps= 14 q=5.0 size=    2147kB time=00:00:03.12 bitrate=5637.6kbits/s
frame=   42 fps= 14 q=5.0 size=    2578kB time=00:00:03.16 bitrate=6683.8kbits/s
frame=   49 fps= 14 q=5.0 size=    2993kB time=00:00:03.93 bitrate=6228.6kbits/s
frame=   56 fps= 14 q=5.0 size=    3151kB time=00:00:04.16 bitrate=6204.8kbits/s
frame=   63 fps= 14 q=5.0 size=    3412kB time=00:00:04.94
[snip]
[avi @ 0x6ae480] st:0 PTS: 352982 DTS: 352982 < 353004 invalid, clipping
[avi @ 0x6ae480] st:0 PTS: 352983 DTS: 352983 < 353005 invalid, clipping
frame=210999 fps= 25 q=5.0 size= 1650925kB time=02:21:12.14 bitrate=1596.3kbits/st:0 PTS: 352984 DTS: 352984 < 353006 invalid, clipping
[avi @ 0x6ae480] st:0 PTS: 352985 DTS: 352985 < 353007 invalid, clipping
[avi @ 0x6ae480] st:0 PTS: 352986 DTS: 352986 < 353008 invalid, clipping
[avi @ 0x6ae480] st:0 PTS: 352987 DTS: 352987 < 353009 invalid, clipping
frame=210999 fps= 25 q=5.0 Lsize= 1652659kB time=02:21:12.24 bitrate=1598.0kbits/s    
video:1505976kB audio:132379kB subtitle:0 global headers:0kB muxing overhead 0.873043%
Received signal 2: terminating 

> Do I understand correctly that the output file out.avi 
> shows problems or did you only see problems in the 
> ffplay output?

Both live ffplay display and the file show the same pauses

> If this is about the output file, is the problem also 
> reproducible with ffmpeg alone (without the pipe to 
> ffplay)?

Hmmm without the ffplay display all looks good. Only tried for 2 hours but usually the problem has occurred by now.
Will do an overnight test.
But I do need a live stream for my users. I'll try (again, the first was a diaster) to build the git-head to get tee and try that with ffplay or even mplayer. I watch cpu usage at 50% odd.

> I never tested pulse but it was suggested repeatedly 
> not to use it, try alsa instead.

I did try with alsa but exactly same as with pulse.
Except for all the bad things about pulse, it is very nice. In particular I want an audio level meter for my users, pulse does that nicely alsa does not do so.

> Please understand that thread-hijacking is considered 
> rude here (as is top-posting), Carl Eugen

All my list mail is by digest, except ffmpeg-users so even though I thought the headers were clean, it was my flight-deck-finger-trouble. I'm sorry.

James

From cehoyos at ag.or.at  Tue Jul  9 14:00:45 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 9 Jul 2013 12:00:45 +0000 (UTC)
Subject: [FFmpeg-user] Comments about audio sync invited
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
	<loom.20130709T094054-726@post.gmane.org>
	<70EF77B2-4254-417D-8D2B-4849B793A5D9@tigger.ws>
Message-ID: <loom.20130709T135126-473@post.gmane.org>

James Linder <jam <at> tigger.ws> writes:

> ffmpeg version 1.0.6

This is old, please update to current git head.

> Input #0, pulse, from 'default':
>   Duration: N/A, start: 0.636342, bitrate: 1536 kb/s

> Input #1, video4linux2,v4l2, from '/dev/video0':
>   Duration: N/A, start: 1373359581.412447, bitrate: 23040 kb/s

> [avi at 0x6ae480] st:0 PTS: 352982 DTS: 
> 352982 < 353004 invalid, clipping

These are the culprits (note the different start 
times apart from the warning).

Please try -use_wallclock_as_timestamps (it is 
possible that this only works with alsa).

[...]

> > is the problem also reproducible with ffmpeg 
> > alone (without the pipe to ffplay)?
> 
> Hmmm without the ffplay display all looks good.

Maybe somebody who knows more about pipes can comment.

> I watch cpu usage at 50% odd.

What happens if you significantly reduce the load by 
using a much smaller resolution?

Carl Eugen


From cehoyos at ag.or.at  Tue Jul  9 14:01:58 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 9 Jul 2013 12:01:58 +0000 (UTC)
Subject: [FFmpeg-user] A change in -timecode ?
References: <51DB2C45.8040004@burner.com>
Message-ID: <loom.20130709T140110-878@post.gmane.org>

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

> Updated to a more recent Windows build of ffmpeg 
> and have been getting something new when using 
> the -timecode switch making Quicktimes:
> 
>   Codec AVOption gop_timecode (MPEG GOP Timecode 
>   in hh:mm:ss[:;.]ff format) specified for 
>   output file #0 (D:\13HD2002H_ProRes.mov) has 
>   not been used for any stream.

Should be fixed, thank you for the report!

Carl Eugen


From soho123.2012 at gmail.com  Tue Jul  9 15:07:27 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 9 Jul 2013 21:07:27 +0800
Subject: [FFmpeg-user] HELP :ffplay CANOT play h264 video + pcm_S16LE audio
	rtp stream
Message-ID: <CAFd2cZWh0EHepSghXrvKZ-OJ=cGz3BOJpNbs62vGyjqrZ9s-Zw@mail.gmail.com>

Hi all,

I got the big trouble about :
ffplay can not play play h264 video + pcm_S16LE audio rtp stream.
originally, ffplay can play video stream if no audio.
after add audio part , then ffplay can not play
Does anyone have ideas?
the configuration of ffserver.conf:
======================================================
Port 8090
RTSPPort 5554
<Feed feed1.ffm>
File /tmp/feed1.ffm FileMaxSize 5M
ACL allow 127.0.0.1
</Feed>
<Stream test1-rtsp.mpg>
Feed feed1.ffm
Format rtp
AVOptionVideo flags +global_header
VideoSize 1280x720
VideoFrameRate 30
VideoCodec libx264
AVOptionAudio flags +global_header
AudioCodec pcm_s16le
AudioChannels 2
AudioSampleRate 48000
</Stream>
======================================================
the detail log :
======================================================
ffserver version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  9 2013 15:51:03 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 --dis
able-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --d
isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
--bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
--enable-libx2
64 --extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,copy,format,fps
,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
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
Tue Jul  9 20:57:52 2013 FFserver started.
ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  9 2013 15:51:03 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 --dis
able-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --d
isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
--bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
--enable-libx2
64 --extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,copy,format,fps
,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
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
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set libav*
logging level) with argument 'debu
g'.
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 a
rgument '30'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '1280x7
20'.
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 '-i' ... matched as input file with argument 'hw:0'.
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 '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with a
rgument 'copy'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with a
rgument '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 libav* 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
1280x720.
Successfully parsed a group of options.
Opening an input file: /dev/video1.
[video4linux2,v4l2 @ 0x69e240] fd:3 capabilities:4000001
[video4linux2,v4l2 @ 0x69e240] Selecting input_channel: 0
[video4linux2,v4l2 @ 0x69e240] input_channel: 0, input_name: Camera 1
[video4linux2,v4l2 @ 0x69e240] Setting time per frame to 1/30
[h264 @ 0x69ede0] Using externally provided dimensions
[h264 @ 0x69ede0] Frame num gap 18 16
[h264 @ 0x69ede0] Frame num gap 21 19
[h264 @ 0x69ede0] Frame num gap 27 25
[video4linux2,v4l2 @ 0x69e240] All info found
[video4linux2,v4l2 @ 0x69e240] Estimating duration from bitrate, this may
be inaccurate
Input #0, video4linux2,v4l2, from '/dev/video1':
  Duration: N/A, start: 24.690000, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 1280x720, 1/2000000,
-5 kb/s, 30 fps, 30 tbr, 1000k t
bn, 2000k tbc
Successfully opened the file.
Parsing a group of options: input file hw:0.
Applying option f (force format) with argument alsa.
Successfully parsed a group of options.
Opening an input file: hw:0.
[alsa @ 0x837ba0] codec_id ==NONE
[alsa @ 0x837ba0] All info found
[alsa @ 0x837ba0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #1.0 : stereo
Input #1, alsa, from 'hw:0':
  Duration: N/A, start: 1373403477.887958, 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://localhost:8090/feed1.ffm.
Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
Applying option ac (set number of audio channels) with argument 2.
Applying option acodec (force audio codec ('copy' to copy stream)) with
argument copy.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Successfully parsed a group of options.
Opening an output file: http://localhost:8090/feed1.ffm.
[ffm @ 0x842e40] Format ffm probed with size=2048 and score=101
[AVIOContext @ 0x842700] Statistics: 4096 bytes read, 0 seeks
Tue Jul  9 20:57:57 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         : Lavf54.63.104
    Stream #0:0, 0, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s
    Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 1280x720, 1/1000000,
q=2-31, -5 kb/s, 30 fps, 1000k t
bn, 1000k tbc
Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=   23 fps=0.0 q=-1.0 size=     136kB time=00:00:02.37 bitrate=
470.1kbits/s
    frame=   38 fps= 37 q=-1.0 size=     272kB time=00:00:02.87 bitrate=
776.4kbits/s
    frame=   53 fps= 35 q=-1.0 size=     388kB time=00:00:03.37 bitrate=
943.2kbits/s
    frame=   68 fps= 34 q=-1.0 size=     532kB time=00:00:03.87
bitrate=1126.1kbits/s
    frame=   83 fps= 33 q=-1.0 size=     740kB time=00:00:04.37
bitrate=1387.2kbits/s
    frame=   98 fps= 32 q=-1.0 size=     972kB time=00:00:04.87
bitrate=1635.0kbits/s
    frame=  113 fps= 32 q=-1.0 size=    1192kB time=00:00:05.37
bitrate=1818.4kbits/s
    frame=  128 fps= 32 q=-1.0 size=    1416kB time=00:00:05.87
bitrate=1976.1kbits/s
    frame=  143 fps= 32 q=-1.0 size=    1632kB time=00:00:06.37
bitrate=2098.8kbits/s
    frame=  158 fps= 31 q=-1.0 size=    1864kB time=00:00:06.87
bitrate=2222.7kbits/s
    frame=  173 fps= 31 q=-1.0 size=    2068kB time=00:00:07.37
bitrate=2298.7kbits/s
    frame=  188 fps= 31 q=-1.0 size=    2276kB time=00:00:07.87
bitrate=2369.1kbits/s
    frame=  203 fps= 31 q=-1.0 size=    2484kB time=00:00:08.37
bitrate=2431.2kbits/s
    frame=  218 fps= 31 q=-1.0 size=    2692kB time=00:00:08.87
bitrate=2486.2kbits/s
    frame=  233 fps= 31 q=-1.0 size=    2904kB time=00:00:09.37
bitrate=2538.9kbits/s
    frame=  248 fps= 31 q=-1.0 size=    3116kB time=00:00:09.87
bitrate=2586.2kbits/s
    frame=  263 fps= 31 q=-1.0 size=    3328kB time=00:00:10.37
bitrate=2629.0kbits/s
    frame=  278 fps= 31 q=-1.0 size=    3544kB time=00:00:10.87
bitrate=2670.9kbits/s
    frame=  293 fps= 31 q=-1.0 size=    3756kB time=00:00:11.37
bitrate=2706.2kbits/s
    frame=  308 fps= 31 q=-1.0 size=    3968kB time=00:00:11.87
bitrate=2738.5kbits/s
    frame=  323 fps= 31 q=-1.0 size=    4180kB time=00:00:12.37
bitrate=2768.2kbits/s
    Tue Jul  9 20:58:08 2013 192.168.1.3:34550 - - "PLAY
test1-rtsp.mpg/streamid=0 RTP/UDP"
Tue Jul  9 20:58:08 2013 192.168.1.3:34552 - - "PLAY
test1-rtsp.mpg/streamid=1 RTP/UDP"
frame=  338 fps= 31 q=-1.0 size=    4308kB time=00:00:12.87
bitrate=2742.1kbits/s
    frame=  353 fps= 31 q=-1.0 size=    4424kB time=00:00:13.37
bitrate=2710.7kbits/s
    frame=  368 fps= 31 q=-1.0 size=    4536kB time=00:00:13.87
bitrate=2679.1kbits/s
    frame=  383 fps= 31 q=-1.0 size=    4652kB time=00:00:14.37
bitrate=2652.0kbits/s
    frame=  398 fps= 31 q=-1.0 size=    4764kB time=00:00:14.87
bitrate=2624.5kbits/s
    frame=  413 fps= 31 q=-1.0 size=    4876kB time=00:00:15.37
bitrate=2598.8kbits/s
    ALSA buffer xrun.
Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]cannot find FFM syncword
Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
frame=  428 fps= 31 q=-1.0 size=    5280kB time=00:00:15.87
bitrate=2725.5kbits/s
    frame=  443 fps= 30 q=-1.0 size=    5396kB time=00:00:16.37
bitrate=2700.3kbits/s
    frame=  458 fps= 30 q=-1.0 size=    5512kB time=00:00:16.87
bitrate=2676.6kbits/s
    frame=  473 fps= 30 q=-1.0 size=    5624kB time=00:00:17.37
bitrate=2652.4kbits/s
    find Key
frame=  488 fps= 30 q=-1.0 size=    5748kB time=00:00:17.87
bitrate=2635.0kbits/s
    ALSA buffer xrun.
frame=  503 fps= 30 q=-1.0 size=    6196kB time=00:00:18.37
bitrate=2763.1kbits/s
    ALSA buffer xrun.
frame=  509 fps= 26 q=-1.0 size=    6224kB time=00:00:18.57
bitrate=2745.7kbits/s
    frame=  520 fps= 23 q=-1.0 size=    6776kB time=00:00:21.37
bitrate=2597.5kbits/s
    ALSA buffer xrun.
frame=  537 fps= 22 q=-1.0 size=    6856kB time=00:00:24.47
bitrate=2295.2kbits/s
    ALSA buffer xrun.
frame=  561 fps= 22 q=-1.0 size=    7040kB time=00:00:27.30
bitrate=2112.5kbits/s
    frame=  576 fps= 22 q=-1.0 size=    7092kB time=00:00:27.80
bitrate=2089.8kbits/s
    frame=  591 fps= 22 q=-1.0 size=    7144kB time=00:00:28.30
bitrate=2068.0kbits/s
    frame=  606 fps= 22 q=-1.0 size=    7200kB time=00:00:28.80
bitrate=2048.0kbits/s
    frame=  621 fps= 23 q=-1.0 size=    7252kB time=00:00:29.30
bitrate=2027.6kbits/s
    ALSA buffer xrun.
frame=  636 fps= 23 q=-1.0 size=    7572kB time=00:00:29.80
bitrate=2081.5kbits/s
    frame=  652 fps= 23 q=-1.0 size=    7808kB time=00:00:30.34
bitrate=2108.2kbits/s
    frame=  667 fps= 23 q=-1.0 size=    8020kB time=00:00:30.84
bitrate=2130.3kbits/s
    frame=  682 fps= 23 q=-1.0 size=    8132kB time=00:00:31.34
bitrate=2125.6kbits/s
    frame=  697 fps= 23 q=-1.0 size=    8244kB time=00:00:31.84
bitrate=2121.1kbits/s
    frame=  712 fps= 23 q=-1.0 size=    8360kB time=00:00:32.34
bitrate=2117.7kbits/s
    frame=  726 fps= 23 q=-1.0 size=    8472kB time=00:00:32.80
bitrate=2115.9kbits/s
    frame=  741 fps= 24 q=-1.0 size=    8584kB time=00:00:33.30
bitrate=2111.7kbits/s
    ALSA buffer xrun.
frame=  756 fps= 24 q=-1.0 size=    8996kB time=00:00:33.80
bitrate=2180.3kbits/s
    frame=  771 fps= 24 q=-1.0 size=    9108kB time=00:00:34.30
bitrate=2175.3kbits/s
    frame=  786 fps= 24 q=-1.0 size=    9220kB time=00:00:34.80
bitrate=2170.4kbits/s
    frame=  802 fps= 24 q=-1.0 size=    9352kB time=00:00:35.34
bitrate=2167.8kbits/s
    frame=  816 fps= 24 q=-1.0 size=    9460kB time=00:00:35.80
bitrate=2164.7kbits/s
    frame=  831 fps= 24 q=-1.0 size=    9576kB time=00:00:36.30
bitrate=2161.1kbits/s
    ALSA buffer xrun.
frame=  846 fps= 24 q=-1.0 size=   10008kB time=00:00:36.80
bitrate=2227.9kbits/s
    frame=  861 fps= 24 q=-1.0 size=   10120kB time=00:00:37.30
bitrate=2222.6kbits/s
    frame=  876 fps= 24 q=-1.0 size=   10232kB time=00:00:37.80
bitrate=2217.5kbits/s
    frame=  891 fps= 24 q=-1.0 size=   10348kB time=00:00:38.30
bitrate=2213.3kbits/s
    Tue Jul  9 20:58:34 2013 192.168.1.3 - - [PLAY] "rtsp://
192.168.1.254:5554/test1-rtsp.mpg/ RTSP/1.0" 200 1
030
frame=  907 fps= 25 q=-1.0 size=   10460kB time=00:00:38.84
bitrate=2206.2kbits/s
    frame=  922 fps= 25 q=-1.0 size=   10572kB time=00:00:39.34
bitrate=2201.5kbits/s
    ALSA buffer xrun.
[ffm @ 0x842940] st:0 PTS: 37891681 DTS: 37891681 < 37892211 invalid,
clipping
frame=  937 fps= 25 q=-1.0 size=   10984kB time=00:00:39.84
bitrate=2258.6kbits/s
    frame=  952 fps= 25 q=-1.0 size=   11096kB time=00:00:40.34
bitrate=2253.3kbits/s
    qframe=  967 fps= 25 q=-1.0 size=   11208kB time=00:00:40.84
bitrate=2248.2kbits/s
    frame=  982 fps= 25 q=-1.0 size=   11324kB time=00:00:41.34
bitrate=2244.0kbits/s
    frame=  997 fps= 25 q=-1.0 size=   11444kB time=00:00:41.84
bitrate=2240.7kbits/s
    frame= 1012 fps= 25 q=-1.0 size=   11560kB time=00:00:42.34
bitrate=2236.6kbits/s
    ALSA buffer xrun.
frame= 1027 fps= 25 q=-1.0 size=   11988kB time=00:00:42.84
bitrate=2292.4kbits/s
    ALSA buffer xrun.
frame= 1029 fps= 23 q=-1.0 size=   12004kB time=00:00:42.90
bitrate=2292.2kbits/s
    frame= 1052 fps= 23 q=-1.0 size=   12572kB time=00:00:46.90
bitrate=2195.9kbits/s
    frame= 1067 fps= 23 q=-1.0 size=   12624kB time=00:00:47.40
bitrate=2181.8kbits/s

=============================================================================
ffplay log:
ffplay -loglevel debug rtsp://192.168.1.254:5554/test1-rtsp.mpg
ffplay version N-54362-ge0be3cb Copyright (c) 2003-2013 the FFmpeg
developers
  built on Jul  2 2013 22:15:59 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. 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
SDP:nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Title
c=IN IP4 0.0.0.0
t=0 0
a=tool:libavformat 54.63.104
m=audio 0 RTP/AVP 96
b=AS:1536
a=control:streamid=0
m=video 0 RTP/AVP 97
b=AS:-5
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=1;
sprop-parameter-sets=Z0JAH5ZUAoAtyA==,aM44gA==;
profile-level-id=42401F
a=control:streamid=1

[rtsp @ 02a64500] video codec set to: h264
[NULL @ 02a679e0] RTP Packetization Mode: 1
[NULL @ 02a679e0] Extradata set to 02a67ea0 (size: 22)!
[NULL @ 02a679e0] RTP Profile IDC: 42 Profile IOP: 40 Level: 1f
hello state=0 0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

From soho123.2012 at gmail.com  Tue Jul  9 15:12:22 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 9 Jul 2013 21:12:22 +0800
Subject: [FFmpeg-user] HELP :ffplay CANOT play h264 video + pcm_S16LE
	audio rtp stream
In-Reply-To: <CAFd2cZWh0EHepSghXrvKZ-OJ=cGz3BOJpNbs62vGyjqrZ9s-Zw@mail.gmail.com>
References: <CAFd2cZWh0EHepSghXrvKZ-OJ=cGz3BOJpNbs62vGyjqrZ9s-Zw@mail.gmail.com>
Message-ID: <CAFd2cZWqcgBhVPhO_0F2FegSY-ZcGCf-odpb2Xc_3DdW2MuQqQ@mail.gmail.com>

Hi All,

the command line of ffmpeg I used:
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format h264
-i /dev/video1 -f alsa -i hw:0 -ar 48000 -ac 2 -acodec copy -vcodec copy
http://localhost:8090/feed1.ffm

It is very appreciated for any input!!

2013/7/9 Soho Soho123 <soho123.2012 at gmail.com>

> Hi all,
>
> I got the big trouble about :
> ffplay can not play play h264 video + pcm_S16LE audio rtp stream.
> originally, ffplay can play video stream if no audio.
> after add audio part , then ffplay can not play
> Does anyone have ideas?
> the configuration of ffserver.conf:
> ======================================================
> Port 8090
> RTSPPort 5554
> <Feed feed1.ffm>
> File /tmp/feed1.ffm FileMaxSize 5M
> ACL allow 127.0.0.1
> </Feed>
> <Stream test1-rtsp.mpg>
> Feed feed1.ffm
> Format rtp
> AVOptionVideo flags +global_header
> VideoSize 1280x720
> VideoFrameRate 30
> VideoCodec libx264
> AVOptionAudio flags +global_header
> AudioCodec pcm_s16le
> AudioChannels 2
> AudioSampleRate 48000
> </Stream>
> ======================================================
> the detail log :
> ======================================================
> ffserver version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
>   built on Jul  9 2013 15:51:03 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 --dis
> able-ffplay --disable-postproc --disable-runtime-cpudetect
> --disable-swscale-alpha --disable-mipsdspr1 --d
> isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
> --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
> --enable-libx2
> 64 --extra-cflags=-I/ShareLib_Install/bin/../include
> --extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
> --enable-filter='aformat,aresample,copy,format,fps
> ,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
> 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
> Tue Jul  9 20:57:52 2013 FFserver started.
> ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
>   built on Jul  9 2013 15:51:03 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 --dis
> able-ffplay --disable-postproc --disable-runtime-cpudetect
> --disable-swscale-alpha --disable-mipsdspr1 --d
> isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
> --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
> --enable-libx2
> 64 --extra-cflags=-I/ShareLib_Install/bin/../include
> --extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
> --enable-filter='aformat,aresample,copy,format,fps
> ,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
> 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
> Splitting the commandline.
> Reading option '-loglevel' ... matched as option 'loglevel' (set libav*
> logging level) with argument 'debu
> g'.
> 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 a
> rgument '30'.
> Reading option '-s' ... matched as option 's' (set frame size (WxH or
> abbreviation)) with argument '1280x7
> 20'.
> 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 '-i' ... matched as input file with argument 'hw:0'.
> 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 '-acodec' ... matched as option 'acodec' (force audio codec
> ('copy' to copy stream)) with a
> rgument 'copy'.
> Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
> ('copy' to copy stream)) with a
> rgument '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 libav* 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
> 1280x720.
> Successfully parsed a group of options.
> Opening an input file: /dev/video1.
> [video4linux2,v4l2 @ 0x69e240] fd:3 capabilities:4000001
> [video4linux2,v4l2 @ 0x69e240] Selecting input_channel: 0
> [video4linux2,v4l2 @ 0x69e240] input_channel: 0, input_name: Camera 1
> [video4linux2,v4l2 @ 0x69e240] Setting time per frame to 1/30
> [h264 @ 0x69ede0] Using externally provided dimensions
> [h264 @ 0x69ede0] Frame num gap 18 16
> [h264 @ 0x69ede0] Frame num gap 21 19
> [h264 @ 0x69ede0] Frame num gap 27 25
> [video4linux2,v4l2 @ 0x69e240] All info found
> [video4linux2,v4l2 @ 0x69e240] Estimating duration from bitrate, this may
> be inaccurate
> Input #0, video4linux2,v4l2, from '/dev/video1':
>   Duration: N/A, start: 24.690000, bitrate: N/A
>     Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 1280x720, 1/2000000,
> -5 kb/s, 30 fps, 30 tbr, 1000k t
> bn, 2000k tbc
> Successfully opened the file.
> Parsing a group of options: input file hw:0.
> Applying option f (force format) with argument alsa.
> Successfully parsed a group of options.
> Opening an input file: hw:0.
> [alsa @ 0x837ba0] codec_id ==NONE
> [alsa @ 0x837ba0] All info found
> [alsa @ 0x837ba0] Estimating duration from bitrate, this may be inaccurate
> Guessed Channel Layout for  Input Stream #1.0 : stereo
> Input #1, alsa, from 'hw:0':
>   Duration: N/A, start: 1373403477.887958, 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://localhost:8090/feed1.ffm.
> Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
> Applying option ac (set number of audio channels) with argument 2.
> Applying option acodec (force audio codec ('copy' to copy stream)) with
> argument copy.
> Applying option vcodec (force video codec ('copy' to copy stream)) with
> argument copy.
> Successfully parsed a group of options.
> Opening an output file: http://localhost:8090/feed1.ffm.
> [ffm @ 0x842e40] Format ffm probed with size=2048 and score=101
> [AVIOContext @ 0x842700] Statistics: 4096 bytes read, 0 seeks
> Tue Jul  9 20:57:57 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         : Lavf54.63.104
>     Stream #0:0, 0, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, 1536
> kb/s
>     Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 1280x720, 1/1000000,
> q=2-31, -5 kb/s, 30 fps, 1000k t
> bn, 1000k tbc
> Stream mapping:
>   Stream #1:0 -> #0:0 (copy)
>   Stream #0:0 -> #0:1 (copy)
> Press [q] to stop, [?] for help
> frame=   23 fps=0.0 q=-1.0 size=     136kB time=00:00:02.37 bitrate=
> 470.1kbits/s
>     frame=   38 fps= 37 q=-1.0 size=     272kB time=00:00:02.87 bitrate=
> 776.4kbits/s
>     frame=   53 fps= 35 q=-1.0 size=     388kB time=00:00:03.37 bitrate=
> 943.2kbits/s
>     frame=   68 fps= 34 q=-1.0 size=     532kB time=00:00:03.87
> bitrate=1126.1kbits/s
>     frame=   83 fps= 33 q=-1.0 size=     740kB time=00:00:04.37
> bitrate=1387.2kbits/s
>     frame=   98 fps= 32 q=-1.0 size=     972kB time=00:00:04.87
> bitrate=1635.0kbits/s
>     frame=  113 fps= 32 q=-1.0 size=    1192kB time=00:00:05.37
> bitrate=1818.4kbits/s
>     frame=  128 fps= 32 q=-1.0 size=    1416kB time=00:00:05.87
> bitrate=1976.1kbits/s
>     frame=  143 fps= 32 q=-1.0 size=    1632kB time=00:00:06.37
> bitrate=2098.8kbits/s
>     frame=  158 fps= 31 q=-1.0 size=    1864kB time=00:00:06.87
> bitrate=2222.7kbits/s
>     frame=  173 fps= 31 q=-1.0 size=    2068kB time=00:00:07.37
> bitrate=2298.7kbits/s
>     frame=  188 fps= 31 q=-1.0 size=    2276kB time=00:00:07.87
> bitrate=2369.1kbits/s
>     frame=  203 fps= 31 q=-1.0 size=    2484kB time=00:00:08.37
> bitrate=2431.2kbits/s
>     frame=  218 fps= 31 q=-1.0 size=    2692kB time=00:00:08.87
> bitrate=2486.2kbits/s
>     frame=  233 fps= 31 q=-1.0 size=    2904kB time=00:00:09.37
> bitrate=2538.9kbits/s
>     frame=  248 fps= 31 q=-1.0 size=    3116kB time=00:00:09.87
> bitrate=2586.2kbits/s
>     frame=  263 fps= 31 q=-1.0 size=    3328kB time=00:00:10.37
> bitrate=2629.0kbits/s
>     frame=  278 fps= 31 q=-1.0 size=    3544kB time=00:00:10.87
> bitrate=2670.9kbits/s
>     frame=  293 fps= 31 q=-1.0 size=    3756kB time=00:00:11.37
> bitrate=2706.2kbits/s
>     frame=  308 fps= 31 q=-1.0 size=    3968kB time=00:00:11.87
> bitrate=2738.5kbits/s
>     frame=  323 fps= 31 q=-1.0 size=    4180kB time=00:00:12.37
> bitrate=2768.2kbits/s
>     Tue Jul  9 20:58:08 2013 192.168.1.3:34550 - - "PLAY
> test1-rtsp.mpg/streamid=0 RTP/UDP"
> Tue Jul  9 20:58:08 2013 192.168.1.3:34552 - - "PLAY
> test1-rtsp.mpg/streamid=1 RTP/UDP"
> frame=  338 fps= 31 q=-1.0 size=    4308kB time=00:00:12.87
> bitrate=2742.1kbits/s
>     frame=  353 fps= 31 q=-1.0 size=    4424kB time=00:00:13.37
> bitrate=2710.7kbits/s
>     frame=  368 fps= 31 q=-1.0 size=    4536kB time=00:00:13.87
> bitrate=2679.1kbits/s
>     frame=  383 fps= 31 q=-1.0 size=    4652kB time=00:00:14.37
> bitrate=2652.0kbits/s
>     frame=  398 fps= 31 q=-1.0 size=    4764kB time=00:00:14.87
> bitrate=2624.5kbits/s
>     frame=  413 fps= 31 q=-1.0 size=    4876kB time=00:00:15.37
> bitrate=2598.8kbits/s
>     ALSA buffer xrun.
> Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
> Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]cannot find FFM syncword
> Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
> frame=  428 fps= 31 q=-1.0 size=    5280kB time=00:00:15.87
> bitrate=2725.5kbits/s
>     frame=  443 fps= 30 q=-1.0 size=    5396kB time=00:00:16.37
> bitrate=2700.3kbits/s
>     frame=  458 fps= 30 q=-1.0 size=    5512kB time=00:00:16.87
> bitrate=2676.6kbits/s
>     frame=  473 fps= 30 q=-1.0 size=    5624kB time=00:00:17.37
> bitrate=2652.4kbits/s
>     find Key
> frame=  488 fps= 30 q=-1.0 size=    5748kB time=00:00:17.87
> bitrate=2635.0kbits/s
>     ALSA buffer xrun.
> frame=  503 fps= 30 q=-1.0 size=    6196kB time=00:00:18.37
> bitrate=2763.1kbits/s
>     ALSA buffer xrun.
> frame=  509 fps= 26 q=-1.0 size=    6224kB time=00:00:18.57
> bitrate=2745.7kbits/s
>     frame=  520 fps= 23 q=-1.0 size=    6776kB time=00:00:21.37
> bitrate=2597.5kbits/s
>     ALSA buffer xrun.
> frame=  537 fps= 22 q=-1.0 size=    6856kB time=00:00:24.47
> bitrate=2295.2kbits/s
>     ALSA buffer xrun.
> frame=  561 fps= 22 q=-1.0 size=    7040kB time=00:00:27.30
> bitrate=2112.5kbits/s
>     frame=  576 fps= 22 q=-1.0 size=    7092kB time=00:00:27.80
> bitrate=2089.8kbits/s
>     frame=  591 fps= 22 q=-1.0 size=    7144kB time=00:00:28.30
> bitrate=2068.0kbits/s
>     frame=  606 fps= 22 q=-1.0 size=    7200kB time=00:00:28.80
> bitrate=2048.0kbits/s
>     frame=  621 fps= 23 q=-1.0 size=    7252kB time=00:00:29.30
> bitrate=2027.6kbits/s
>     ALSA buffer xrun.
> frame=  636 fps= 23 q=-1.0 size=    7572kB time=00:00:29.80
> bitrate=2081.5kbits/s
>     frame=  652 fps= 23 q=-1.0 size=    7808kB time=00:00:30.34
> bitrate=2108.2kbits/s
>     frame=  667 fps= 23 q=-1.0 size=    8020kB time=00:00:30.84
> bitrate=2130.3kbits/s
>     frame=  682 fps= 23 q=-1.0 size=    8132kB time=00:00:31.34
> bitrate=2125.6kbits/s
>     frame=  697 fps= 23 q=-1.0 size=    8244kB time=00:00:31.84
> bitrate=2121.1kbits/s
>     frame=  712 fps= 23 q=-1.0 size=    8360kB time=00:00:32.34
> bitrate=2117.7kbits/s
>     frame=  726 fps= 23 q=-1.0 size=    8472kB time=00:00:32.80
> bitrate=2115.9kbits/s
>     frame=  741 fps= 24 q=-1.0 size=    8584kB time=00:00:33.30
> bitrate=2111.7kbits/s
>     ALSA buffer xrun.
> frame=  756 fps= 24 q=-1.0 size=    8996kB time=00:00:33.80
> bitrate=2180.3kbits/s
>     frame=  771 fps= 24 q=-1.0 size=    9108kB time=00:00:34.30
> bitrate=2175.3kbits/s
>     frame=  786 fps= 24 q=-1.0 size=    9220kB time=00:00:34.80
> bitrate=2170.4kbits/s
>     frame=  802 fps= 24 q=-1.0 size=    9352kB time=00:00:35.34
> bitrate=2167.8kbits/s
>     frame=  816 fps= 24 q=-1.0 size=    9460kB time=00:00:35.80
> bitrate=2164.7kbits/s
>     frame=  831 fps= 24 q=-1.0 size=    9576kB time=00:00:36.30
> bitrate=2161.1kbits/s
>     ALSA buffer xrun.
> frame=  846 fps= 24 q=-1.0 size=   10008kB time=00:00:36.80
> bitrate=2227.9kbits/s
>     frame=  861 fps= 24 q=-1.0 size=   10120kB time=00:00:37.30
> bitrate=2222.6kbits/s
>     frame=  876 fps= 24 q=-1.0 size=   10232kB time=00:00:37.80
> bitrate=2217.5kbits/s
>     frame=  891 fps= 24 q=-1.0 size=   10348kB time=00:00:38.30
> bitrate=2213.3kbits/s
>     Tue Jul  9 20:58:34 2013 192.168.1.3 - - [PLAY] "rtsp://
> 192.168.1.254:5554/test1-rtsp.mpg/ RTSP/1.0" 200 1
> 030
> frame=  907 fps= 25 q=-1.0 size=   10460kB time=00:00:38.84
> bitrate=2206.2kbits/s
>     frame=  922 fps= 25 q=-1.0 size=   10572kB time=00:00:39.34
> bitrate=2201.5kbits/s
>     ALSA buffer xrun.
> [ffm @ 0x842940] st:0 PTS: 37891681 DTS: 37891681 < 37892211 invalid,
> clipping
> frame=  937 fps= 25 q=-1.0 size=   10984kB time=00:00:39.84
> bitrate=2258.6kbits/s
>     frame=  952 fps= 25 q=-1.0 size=   11096kB time=00:00:40.34
> bitrate=2253.3kbits/s
>     qframe=  967 fps= 25 q=-1.0 size=   11208kB time=00:00:40.84
> bitrate=2248.2kbits/s
>     frame=  982 fps= 25 q=-1.0 size=   11324kB time=00:00:41.34
> bitrate=2244.0kbits/s
>     frame=  997 fps= 25 q=-1.0 size=   11444kB time=00:00:41.84
> bitrate=2240.7kbits/s
>     frame= 1012 fps= 25 q=-1.0 size=   11560kB time=00:00:42.34
> bitrate=2236.6kbits/s
>     ALSA buffer xrun.
> frame= 1027 fps= 25 q=-1.0 size=   11988kB time=00:00:42.84
> bitrate=2292.4kbits/s
>     ALSA buffer xrun.
> frame= 1029 fps= 23 q=-1.0 size=   12004kB time=00:00:42.90
> bitrate=2292.2kbits/s
>     frame= 1052 fps= 23 q=-1.0 size=   12572kB time=00:00:46.90
> bitrate=2195.9kbits/s
>     frame= 1067 fps= 23 q=-1.0 size=   12624kB time=00:00:47.40
> bitrate=2181.8kbits/s
>
> =============================================================================
> ffplay log:
> ffplay -loglevel debug rtsp://192.168.1.254:5554/test1-rtsp.mpg
> ffplay version N-54362-ge0be3cb Copyright (c) 2003-2013 the FFmpeg
> developers
>   built on Jul  2 2013 22:15:59 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. 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
> SDP:nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
> v=0
> o=- 0 0 IN IP4 127.0.0.1
> s=No Title
> c=IN IP4 0.0.0.0
> t=0 0
> a=tool:libavformat 54.63.104
> m=audio 0 RTP/AVP 96
> b=AS:1536
> a=control:streamid=0
> m=video 0 RTP/AVP 97
> b=AS:-5
> a=rtpmap:97 H264/90000
> a=fmtp:97 packetization-mode=1;
> sprop-parameter-sets=Z0JAH5ZUAoAtyA==,aM44gA==;
> profile-level-id=42401F
> a=control:streamid=1
>
> [rtsp @ 02a64500] video codec set to: h264
> [NULL @ 02a679e0] RTP Packetization Mode: 1
> [NULL @ 02a679e0] Extradata set to 02a67ea0 (size: 22)!
> [NULL @ 02a679e0] RTP Profile IDC: 42 Profile IOP: 40 Level: 1f
> hello state=0 0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
>     nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
>
>
>
>
>
>
>

From salmjuh at hotmail.com  Tue Jul  9 13:06:53 2013
From: salmjuh at hotmail.com (Jii)
Date: Tue, 9 Jul 2013 04:06:53 -0700 (PDT)
Subject: [FFmpeg-user] Capturin SDI / No audio support
In-Reply-To: <1373356151053-4659882.post@n4.nabble.com>
References: <DUB123-W10FB24A4BDE9187F24676CAB780@phx.gbl>
	<20130708174923.GA14280@sunshine.barsnick.net>
	<1373356151053-4659882.post@n4.nabble.com>
Message-ID: <1373368013785-4659885.post@n4.nabble.com>

Ok, so now I have workin capture.

But somehow every time I starting recording, this decklink's card gives me
color bars for few frames.

So is there a way how to tell ffmpeg that skip first 10 frames and after
that start record the stream. 

Now I use this command:


bmdcapture -m 2 -C 0 -A 2 -V 4 -F nut -f pipe:1 |ffmpeg -re -i - -copyts -r
25 -aspect 16:9 -copyts -vcodec libx264  -vprofile main -preset medium -crf
17  -s 640x360 -b:a 196k -ar 44100 -b:v 1100k -maxrate 1300k -bufsize 1500k 
-pix_fmt yuv420p  -y /tmp/testfile.mp4


Thanks for your help advanced.
Ans if you have ideas how to make better quolity I'm pleased to hear any
advices.

Br,
Jii



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From salmjuh at hotmail.com  Tue Jul  9 09:49:11 2013
From: salmjuh at hotmail.com (Jii)
Date: Tue, 9 Jul 2013 00:49:11 -0700 (PDT)
Subject: [FFmpeg-user] Capturin SDI / No audio support
In-Reply-To: <20130708174923.GA14280@sunshine.barsnick.net>
References: <DUB123-W10FB24A4BDE9187F24676CAB780@phx.gbl>
	<20130708174923.GA14280@sunshine.barsnick.net>
Message-ID: <1373356151053-4659882.post@n4.nabble.com>

Hi Moritz and thanks for the hint. 

New version did the trick and thanks  for you help :-)


Br,
jii



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From thefatalaccidents at gmail.com  Tue Jul  9 17:53:36 2013
From: thefatalaccidents at gmail.com (fatalaccidents)
Date: Tue, 9 Jul 2013 08:53:36 -0700 (PDT)
Subject: [FFmpeg-user] Problem with white-washed images
In-Reply-To: <loom.20130709T094555-920@post.gmane.org>
References: <1373313723432-4659873.post@n4.nabble.com>
	<loom.20130709T094555-920@post.gmane.org>
Message-ID: <1373385216360-4659891.post@n4.nabble.com>

This indeed fix my problem.  I can play the movies properly in vlc player and
quicktime.  Although mplayer still has the line "[swscaler @
0x100b41620]BICUBIC scaler, from yuv420p to yuyv422 using MMX2 " which I
think it is making it reproduce this white-washed video.  I don't need to
run them with mplayer though, so it is a trivial matter for me. 

Thanks for your help.



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From bobm-ffmpeg at burner.com  Tue Jul  9 21:50:24 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Tue, 09 Jul 2013 13:50:24 -0600
Subject: [FFmpeg-user] A change in -timecode ?
In-Reply-To: <loom.20130709T140110-878@post.gmane.org>
References: <51DB2C45.8040004@burner.com>
	<loom.20130709T140110-878@post.gmane.org>
Message-ID: <51DC6980.6070101@burner.com>

On 7/9/2013 6:01 AM, Carl Eugen Hoyos wrote:

> Should be fixed, thank you for the report!

Cool, thanks!

From lambda.clozure at gmail.com  Wed Jul 10 13:31:44 2013
From: lambda.clozure at gmail.com (Bieber Butzeman)
Date: Wed, 10 Jul 2013 13:31:44 +0200
Subject: [FFmpeg-user] Correct usage of the GNU Public licence and using
	mjpeg in mp4 container.
Message-ID: <CACB006cvLM7JStpySKMAxBH2ObUMq6swJMu21wyipbsfh2Md+g@mail.gmail.com>

Hi folks,
i developed a small tool for a german university in java for windows, mac
and linux that convert an animated gif into a movie.

Now i having some questing regarding to release it under the GNU Public
license:
1. As i am not sure if i use only Lesser GNU Public stuff is it ok if i
just go for GNU Public license, because it is the more "restricted" one and
don't have to worry about licensing concerns?

2. Is it sufficent to state in the License file that this Program
(Gif2Video) is using the GNU Public license (2+) plus a copy of that
license in a file nearby. And also mention that it uses ffmpeg (with a
link) that use different licenses, most of it is in the Lesser GNU Public
license (also place a copy of that License file). Or must i include the
hole source with a binary release (doubt it, links should be sufficend,
aren't they?)


The other thing is that i having problems of using motion jpeg in a mp4
container, at least when it comes down to watch them in Quicktime or
Windows Media Player. VLC works fine, maybe because both use libavcodec?
Here is an example code that would my programm use in windows for making a
mjpeg mp4 video out of jpeg images with i extract before form the gifs:
./lib/ffmpeg.exe -f image2 -r 15 -i C:\temp\CrocFAIL\CrocFAIL%d.jpeg -r 25
-vcodec mjpeg -pix_fmt yuv420p C:\temp\CrocFAIL.mp4
more readable ->
ffmpeg -f image2 -r 15 -i test%d.jpeg -r 25 -vcodec mjpeg -pix_fmt yuv420p
test.mp4

thanks for any help in advance and sorry for my broken english, trying to
improve it, but i am no native speaker.
Sven Gleitsmann

btw if anyone need that tool, just send me an mail and i send you a
download link. will release i most likely this month on sourceforge but for
now it should be fine so.

From cehoyos at ag.or.at  Wed Jul 10 13:40:54 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 10 Jul 2013 11:40:54 +0000 (UTC)
Subject: [FFmpeg-user] Correct usage of the GNU Public licence and using
	mjpeg in mp4 container.
References: <CACB006cvLM7JStpySKMAxBH2ObUMq6swJMu21wyipbsfh2Md+g@mail.gmail.com>
Message-ID: <loom.20130710T133559-568@post.gmane.org>

Bieber Butzeman <lambda.clozure <at> gmail.com> writes:

> Or must i include the hole source with a binary release

No, a link to the *exact* sources that you used to 
compile FFmpeg are sufficient. ("... accompany ...")
Please understand that usually, this link points to 
a website controlled by you and not to "ffmpeg.org".

> ffmpeg -f image2 -r 15 -i test%d.jpeg -r 25 
> -vcodec mjpeg -pix_fmt yuv420p test.mp4

I wanted to write:
"Why do you think isom support mjpeg?"
but it actually is supported.
Do you have a mjpeg-in-isom file that is played 
by QuickTime?

Carl Eugen

IANAL, please read the GPL and the LGPL to know the 
exact requirements for binary distribution.


From dflett at bigpond.net.au  Wed Jul 10 13:50:43 2013
From: dflett at bigpond.net.au (Dan Flett)
Date: Wed, 10 Jul 2013 21:50:43 +1000
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
In-Reply-To: <20120827155600.GC21473@arborea>
References: <20120827191454.GA23763@Xeslaro> <20120827155600.GC21473@arborea>
Message-ID: <000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Stefano Sabatini
> Sent: Tuesday, 28 August 2012 01:56
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb
> drivers
> 
> On date Monday 2012-08-27 19:14:58 +0000, Xeslaro encoded:

..snip

> >   however, when i wanna grab something using ffmpeg -f fbdev -i
> >   /dev/fb0, ffmpeg just complains that the pixel format couldn't be
> >   recognized.
> >
> >   i used the exactly the ffmpeg binary running on anther computer
> >   with intel 945gm integrated graphics card, and ffmpeg -f fbdev -i
> >   /dev/fb0 just worked fine.
> 
> >   are there some limitations in ffmpeg with framebuffer device?
> >   anyway to solve that problem? thanks.
> 
> I implemented only the formats which I could test. Can you debug the
device
> and test the information passed to get_pixfmt_from_fb_varinfo()? It should
> be rather easy to add an entry to the rgb_pixfmt_map array in fbdev.c.
> 

Hi ,
I'm trying to use the fbdev format on the Raspberry Pi  (Raspbian).  When I
try to to a screen grab or screencast I too get told "pixel format not
supported".

pi at raspberrypi ~/videos/test $ /usr/local/bin/ffmpeg -f fbdev -frames:v 1 -r
1 -i /dev/fb0 screenshot.jpeg
ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  9 2013 14:47:41 with gcc 4.6 (Debian 4.6.3-14+rpi1)
  configuration:
  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
[fbdev @ 0xeae430] Framebuffer pixel format not supported.
/dev/fb0: Invalid argument

But I've discovered that this does work:

pi at raspberrypi ~/videos/test $ /usr/local/bin/ffmpeg -f rawvideo -codec:v
rawvideo -s 1680x1050 -pix_fmt rgb565le -frames:v 1 -r 1 -i /dev/fb0
screenshot-ff01.bmp
ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  9 2013 14:47:41 with gcc 4.6 (Debian 4.6.3-14+rpi1)
  configuration:
  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
[rawvideo @ 0x2a835d0] Estimating duration from bitrate, this may be
inaccurate
Input #0, rawvideo, from '/dev/fb0':
  Duration: N/A, start: 0.000000, bitrate: 28224 kb/s
    Stream #0:0: Video: rawvideo (RGB[16] / 0x10424752), rgb565le,
1680x1050, 28224 kb/s, 1 tbr, 1 tbn, 1 tbc
Output #0, image2, to 'screenshot-ff01.bmp':
  Metadata:
    encoder         : Lavf54.63.104
    Stream #0:0: Video: bmp, rgb565le, 1680x1050, q=2-31, 200 kb/s, 90k tbn,
1 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> bmp)
Press [q] to stop, [?] for help
[rawvideo @ 0x2a8c3a0] Invalid buffer size, packet size 20160 < expected
length 3528000
Error while decoding stream #0:0: Invalid argument
frame=    1 fps=0.4 q=0.0 Lsize=N/A time=00:00:01.00 bitrate=N/A
video:3445kB audio:0kB subtitle:0 global headers:0kB muxing overhead
-100.000624%

So it looks like the Raspberry Pi uses pixel format rgb565le.  But I can't
use this to create a video stream (I want to pipe it into gstreamer and use
the omxh264enc encoder).

Can I help fix this?  If so, you'll have to take me through the debug steps
as I'm not much of  a c coder at all. :)

Cheers,

Dan


From cehoyos at ag.or.at  Wed Jul 10 14:06:13 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 10 Jul 2013 12:06:13 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
References: <20120827191454.GA23763@Xeslaro> <20120827155600.GC21473@arborea>
	<000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>
Message-ID: <loom.20130710T140109-897@post.gmane.org>

Dan Flett <dflett <at> bigpond.net.au> writes:

> But I've discovered that this does work:
> 
> $ /usr/local/bin/ffmpeg -f rawvideo -codec:v rawvideo 
> -s 1680x1050 -pix_fmt rgb565le -frames:v 1 -r 1 
> -i /dev/fb0 screenshot-ff01.bmp

> [rawvideo  <at>  0x2a8c3a0] Invalid buffer size, 
> packet size 20160 < expected length 3528000

This line indicates that it does not work.
Please add some printf's to get_pixfmt_from_fb_varinfo() 
in libavdevice/fbdev.c to show the contents of varinfo.
(Although I fear a compressed format is used.)

Carl Eugen


From lucielemonnier at hotmail.fr  Wed Jul 10 15:43:57 2013
From: lucielemonnier at hotmail.fr (lucie Lemonnier)
Date: Wed, 10 Jul 2013 15:43:57 +0200
Subject: [FFmpeg-user] try to change complex filters on runtime
Message-ID: <DUB111-W71942C66995E0A11E9D04BB97A0@phx.gbl>

Hello, 
I want to change 
complex filters on runtime by changing the code of ffmpeg.c to change 
between different overlays or to change the value of x in crop 
parameters.
Before, i tried to change the overlays and crop with the 
use of simple filter.To try this, I call a fuction "filter_change" in 
the function transcode_step : 

void filter_change(char* newfilter){
               ost->avfilter = av_strdup(newfilter);
		if (configure_filtergraph(ost->filter->graph)) {
			av_log(NULL, AV_LOG_FATAL, "Error opening filters!\n");
			exit(1);
		}

}

I call for example : filter_change("movie = objets.png [mv1];[in][mv1] overlay=0:0 [out];[out] crop=1920:1080:0:0");
or 

filter_change("movie = transp.mov [mv1];[in][mv1] overlay=0:0 [out];[out] crop=1920:1080:0:0");
               

I
 use rtp to output. But when I add a video overlay for example, the 
output video (rtp) freeze and the output with overlay reappears after 5 
to 10 seconds. It's too long. I need a real-time change or almost.
That's why I test with complex filters.
I create a function reinit_filters_complex that I call in transcode_step :

void reinit_filters_complex(const char *arg){
	int i,j;
    for (i = 0; i < nb_filtergraphs; i++) {
        avfilter_graph_free(&filtergraphs[i]->graph);
        for (j = 0; j < filtergraphs[i]->nb_inputs; j++) {
            av_freep(&filtergraphs[i]->inputs[j]->name);
            av_freep(&filtergraphs[i]->inputs[j]);
        }
        av_freep(&filtergraphs[i]->inputs);
        for (j = 0; j < filtergraphs[i]->nb_outputs; j++) {
            av_freep(&filtergraphs[i]->outputs[j]->name);
            av_freep(&filtergraphs[i]->outputs[j]);
        }
        av_freep(&filtergraphs[i]->outputs);
        av_freep(&filtergraphs[i]);
    }
    av_freep(&filtergraphs);
	nb_filtergraphs = 0;
	GROW_ARRAY(filtergraphs, nb_filtergraphs);
	 if (!(filtergraphs[nb_filtergraphs - 1] = av_mallocz(sizeof(*filtergraphs[0]))))
        return AVERROR(ENOMEM);
    filtergraphs[nb_filtergraphs - 1]->index       = nb_filtergraphs - 1;
    filtergraphs[nb_filtergraphs - 1]->graph_desc = arg;

    configure_complex_filters();

}

For
 example , I call reinit_filters_complex("[0:0] crop=960:1080:0:0 
[tin];[2:0] crop=960:1080:0:0 [tmv]);[tin][tmv]overlay=0:0").
I 
passed in parameters of ffmpeg -i videoSource.mp4 -i transp.mov -i 
objets.png -filter_complex "[0:0] crop=960:1080:0:0 [tin];[1:0] 
crop=960:1080:0:0 [tmv]".
But It crash on this change.
I think it's because I free all in filtergraphs.
But I don't know what is missing for it to work.
Can you help me, please?
Thanks.

			

		

		
			 
	

		 		 	   		  

From jam at tigger.ws  Wed Jul 10 15:59:20 2013
From: jam at tigger.ws (James Linder)
Date: Wed, 10 Jul 2013 21:59:20 +0800
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <loom.20130709T135126-473@post.gmane.org>
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
	<loom.20130709T094054-726@post.gmane.org>
	<70EF77B2-4254-417D-8D2B-4849B793A5D9@tigger.ws>
	<loom.20130709T135126-473@post.gmane.org>
Message-ID: <3424D80A-6E67-4AC2-9359-0DEE7F47E4F4@tigger.ws>


On 09/07/2013, at 8:00 PM, Carl Eugen Hoyos wrote:

> James Linder <jam <at> tigger.ws> writes:
> 
>> ffmpeg version 1.0.6
> 
> This is old, please update to current git head.
> 
>> Input #0, pulse, from 'default':
>>  Duration: N/A, start: 0.636342, bitrate: 1536 kb/s
> 
>> Input #1, video4linux2,v4l2, from '/dev/video0':
>>  Duration: N/A, start: 1373359581.412447, bitrate: 23040 kb/s
> 
>> [avi at 0x6ae480] st:0 PTS: 352982 DTS: 
>> 352982 < 353004 invalid, clipping
> 
> These are the culprits (note the different start 
> times apart from the warning).
> 
> Please try -use_wallclock_as_timestamps (it is 
> possible that this only works with alsa).
> 
> [...]
> 
>>> is the problem also reproducible with ffmpeg 
>>> alone (without the pipe to ffplay)?
>> 
>> Hmmm without the ffplay display all looks good.
> 
> Maybe somebody who knows more about pipes can comment.
> 
>> I watch cpu usage at 50% odd.
> 
> What happens if you significantly reduce the load by 
> using a much smaller resolution?

If any kind soul can tell me why I'm being an idiot I would appreciate it so:

Just to recap:
I want to get audio and video synced
I want to display what is being recorded
Pulse lets them set the audio level visually
I want to NOT start jerk-display after an hour or so

I built git-head
I followed 
http://ffmpeg.org/trac/ffmpeg/wiki/Creating%20multiple%20outputs
and
http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option

[dvr] /home/dvr [132]% ./ffmpeg -f pulse -i default -f v4l2 -i /dev/video0 -vtag divx -s 384x288 -r pal -q:v 5 -f tee  -map 0:a -map 1:v Wed.avi | cat >Wed1.avi
ffmpeg version N-54580-g694550f Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 10 2013 11:43:58 with gcc 4.7 (SUSE Linux)
  configuration: --enable-libv4l2 --enable-libpulse
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 11.101 / 55. 11.101
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 79.100 /  3. 79.100
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, pulse, from 'default':
  Duration: N/A, start: 0.005521, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Input #1, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 1373462222.886516, bitrate: 92160 kb/s
    Stream #1:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480, 92160 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
Output #0, tee, to 'Wed.avi':
    Stream #0:0: Audio: none, 0 channels, 128 kb/s
    Stream #0:1: Video: none (divx / 0x78766964), 384x288, q=2-31, 128 kb/s, 90k tbn
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> ?)
  Stream #1:0 -> #0:1 (rawvideo -> ?)
Encoder (codec none) not found for output stream #0:0

Step #1 is to get 2 streams that are the same (I assume | cat will not cause performance issues)

Thanks
James


From cehoyos at ag.or.at  Wed Jul 10 16:06:16 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 10 Jul 2013 14:06:16 +0000 (UTC)
Subject: [FFmpeg-user] Comments about audio sync invited
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
	<loom.20130709T094054-726@post.gmane.org>
	<70EF77B2-4254-417D-8D2B-4849B793A5D9@tigger.ws>
	<loom.20130709T135126-473@post.gmane.org>
	<3424D80A-6E67-4AC2-9359-0DEE7F47E4F4@tigger.ws>
Message-ID: <loom.20130710T160535-257@post.gmane.org>

James Linder <jam <at> tigger.ws> writes:

> Output #0, tee, to 'Wed.avi':
>     Stream #0:0: Audio: none, 0 channels, 128 kb/s
>     Stream #0:1: Video: none (divx / 0x78766964), 384x288

The tee muxer has no default codecs, so you have to 
specify them with -acodec and -vcodec.

Carl Eugen


From nicolas.george at normalesup.org  Wed Jul 10 16:19:45 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 10 Jul 2013 16:19:45 +0200
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <3424D80A-6E67-4AC2-9359-0DEE7F47E4F4@tigger.ws>
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
	<loom.20130709T094054-726@post.gmane.org>
	<70EF77B2-4254-417D-8D2B-4849B793A5D9@tigger.ws>
	<loom.20130709T135126-473@post.gmane.org>
	<3424D80A-6E67-4AC2-9359-0DEE7F47E4F4@tigger.ws>
Message-ID: <20130710141944.GA8186@phare.normalesup.org>

Le duodi 22 messidor, an CCXXI, James Linder a ?crit?:
> [dvr] /home/dvr [132]% ./ffmpeg -f pulse -i default -f v4l2 -i /dev/video0 -vtag divx -s 384x288 -r pal -q:v 5 -f tee  -map 0:a -map 1:v Wed.avi | cat >Wed1.avi

The tee muxer in this command line does not make any sense. The cat is
completely useless too.

Regards,

-- 
  Nicolas George
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From solarmobiletrailers at gmail.com  Wed Jul 10 14:33:39 2013
From: solarmobiletrailers at gmail.com (Kesten Broughton)
Date: Wed, 10 Jul 2013 07:33:39 -0500
Subject: [FFmpeg-user] Fwd: .configure can't locate x264.h
In-Reply-To: <CAO2fFsWUeo1oR=P+SdRuf1yhdO4=3Nxh0KG-CiY49CMR7qu0mw@mail.gmail.com>
References: <CAO2fFsWUeo1oR=P+SdRuf1yhdO4=3Nxh0KG-CiY49CMR7qu0mw@mail.gmail.com>
Message-ID: <CAO2fFsVgf3=exKq3eDDDmiO9C+=OcM6+ch5+fHSH3RBwsrh=wg@mail.gmail.com>

Hi,
Thanks for the quick response.

I was following the instructions in the error message from running
./configure which asked to attach config.log and send it to this email.
The config.log was around 108k.
Is that an unusually large config.log or perhaps the message should be
modified?

Here is the message again without the log.

I'm running ubuntu 12.04.

kesten

---------- Forwarded message ----------
From: Kesten Broughton <solarmobiletrailers at gmail.com>
Date: Tue, Jul 9, 2013 at 8:42 PM
Subject: .configure can't locate x264.h
To: ffmpeg-user at mplayerhq.hu


I just pulled fresh from trunk.
I'm trying to build with all options to enable gource dependencies.

last lines from config.log:

gcc -D_ISOC99_SOURCE -D_POSIX_C_SOURCE=200112 -D_FILE_OFFSET_BITS=64
-D_LARGEFILE_SOURCE -std=c99 -fomit-frame-pointer -pthread -E -o
/tmp/ffconf.2RxuVyAx.o /tmp/ffconf.gaVjnbSu.c
/tmp/ffconf.gaVjnbSu.c:1:18: fatal error: x264.h: No such file or directory
compilation terminated.
ERROR: libx264 not found

However, i did get and build it, and it's in a standard place.
kesten at kesten-K42Jr:/opt/lib/gource-0.40/ffmpeg$ locate x264.h
/usr/include/x264.h

how can i find where configure is looking for it?

thanks,

kesten


-- 

Kesten Broughton
President and Technology Director,
Solar Mobile Trailers
kesten at solarmobiletrailers.com
www.sunfarmkitchens.ca <http://www.sunfarmkitchens.ca>
512 701 4209



--

From compuvision.sr at gmail.com  Wed Jul 10 20:56:55 2013
From: compuvision.sr at gmail.com (Albert Scholtalbers)
Date: Wed, 10 Jul 2013 15:56:55 -0300
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
Message-ID: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>

Hi there,

I'm trying to produce MPEG2 files with hardcoded subs using ffmpeg under
Windows.
I'm following the instructions here:
http://trac.ffmpeg.org/wiki/How%20to%20burn%20subtitles%20into%20the%20video
I get perfect MPEG2 files with the default template in winff.
ffmpeg.exe -y -i "E:\src.mkv" -f dvd -target ntsc-dvd -r 29.97 -vf
scale=720:480 -aspect 16:9 -b:v 8000k -g 12 -mbd rd -trellis 1 -flags +mv0
-cmp 0 -subcmp 2 "E:\dest.mpg"

When I enter the option -vf subtitles=src.srt, like this

ffmpeg.exe -y -i "E:\src.mkv" -vf subtitles=src.srt -f dvd -target ntsc-dvd
-r 29.97 -vf scale=720:480 -aspect 16:9 -b:v 8000k -g 12 -mbd rd -trellis 1
-flags +mv0 -cmp 0 -subcmp 2 "E:\dest.mpg"

I get still the same mpeg file, without any subs....

When I type ffmpeg.exe I see it's compiled with libass
ffmpeg version N-51433-g551f683 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Mar 28 2013 21:58:26 with gcc 4.8.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfi
g --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray
--enable-libcaca --enable-libfreetype --enable-libg
sm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenjpeg --ena
ble-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-lib
twolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libxa
vs --enable-libxvid --enable-zlib
  libavutil      52. 22.101 / 52. 22.101
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  0.100 / 55.  0.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 48.105 /  3. 48.105
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100

What is the problem, what am I doing wrong?

The output is this:
ffmpeg version N-51433-g551f683 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Mar 28 2013 21:58:26 with gcc 4.8.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfi
g --enable-frei0r --enable-gnutls --enable-libass --enable-libbluray
--enable-libcaca --enable-libfreetype --enable-libg
sm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenjpeg --ena
ble-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-lib
twolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libxa
vs --enable-libxvid --enable-zlib
  libavutil      52. 22.101 / 52. 22.101
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  0.100 / 55.  0.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 48.105 /  3. 48.105
  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 src.mkv':
  Metadata:
    creation_time   : 2013-04-11 22:04:12
  Duration: 01:26:25.18, start: 0.000000, bitrate: 6955 kb/s
    Stream #0:0(eng): Video: h264 (High), yuv420p, 1280x688 [SAR 1:1 DAR
80:43], 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)
Output #0, dvd, to 'E:\testmpg.mpg':
  Metadata:
    encoder         : Lavf55.0.100
    Stream #0:0(eng): Video: mpeg2video (hq), yuv420p, 720x480 [SAR 32:27
DAR 16:9], q=2-31, 8000 kb/s, 90k tbn, 29.97 t
bc (default)
    Stream #0:1: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s (default)
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg2video)
  Stream #0:1 -> #0:1 (dca -> ac3)
Press [q] to stop, [?] for help

From cehoyos at ag.or.at  Wed Jul 10 22:02:20 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 10 Jul 2013 20:02:20 +0000 (UTC)
Subject: [FFmpeg-user] Fwd: .configure can't locate x264.h
References: <CAO2fFsWUeo1oR=P+SdRuf1yhdO4=3Nxh0KG-CiY49CMR7qu0mw@mail.gmail.com>
	<CAO2fFsVgf3=exKq3eDDDmiO9C+=OcM6+ch5+fHSH3RBwsrh=wg@mail.gmail.com>
Message-ID: <loom.20130710T220129-746@post.gmane.org>

Kesten Broughton <solarmobiletrailers <at> gmail.com> writes:

> last lines from config.log:

(The last lines is actually what is needed.)

> gcc -D_ISOC99_SOURCE -D_POSIX_C_SOURCE=200112 -D_FILE_OFFSET_BITS=64
> -D_LARGEFILE_SOURCE -std=c99 -fomit-frame-pointer -pthread -E -o
> /tmp/ffconf.2RxuVyAx.o /tmp/ffconf.gaVjnbSu.c
> /tmp/ffconf.gaVjnbSu.c:1:18: fatal error: x264.h: No such file or directory
> compilation terminated.
> ERROR: libx264 not found
> 
> However, i did get and build it, and it's in a standard place.
> kesten <at> kesten-K42Jr:/opt/lib/gource-0.40/ffmpeg$ locate x264.h
> /usr/include/x264.h

What is your configure line?

Does gcc -v show anything unusual?

Carl Eugen


From cehoyos at ag.or.at  Wed Jul 10 22:03:40 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 10 Jul 2013 20:03:40 +0000 (UTC)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
Message-ID: <loom.20130710T220231-979@post.gmane.org>

Albert Scholtalbers <compuvision.sr <at> gmail.com> writes:

> ffmpeg.exe -y -i "E:\src.mkv" -vf subtitles=src.srt 
> -f dvd -target ntsc-dvd -r 29.97 -vf scale=720:480 
> -aspect 16:9 -b:v 8000k -g 12 -mbd rd -trellis 1
> -flags +mv0 -cmp 0 -subcmp 2 "E:\dest.mpg"

I didn't test but I suspect specifying two 
"-vf" options for one output file is either 
undefined behaviour or forces only the 
second option.

Carl Eugen


From compuvision.sr at gmail.com  Wed Jul 10 22:21:26 2013
From: compuvision.sr at gmail.com (Albert Scholtalbers)
Date: Wed, 10 Jul 2013 17:21:26 -0300
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <loom.20130710T220231-979@post.gmane.org>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
Message-ID: <CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>

>I didn't test but I suspect specifying two"-vf" options for one output
file is either
> undefined behaviour or forces only the
> second option.

>Carl Eugen

I found a working command under Linux:
/usr/local/bin/ffmpeg -y -i src.mkv  -f dvd -target ntsc-dvd -r 29.97 -vf
scale=720:480 -vf subtitles=filename=src.srt -aspect 16:9 -b:v 8000k -g 12
-mbd rd -trellis 1 -flags +mv0 -cmp 0 -subcmp 2 dest.mpg

That "seems"to be easier. But I haven't found any option to set
font/size/color in document or Google yet.

From bostjan.strojan at gmail.com  Wed Jul 10 22:37:39 2013
From: bostjan.strojan at gmail.com (=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=)
Date: Wed, 10 Jul 2013 22:37:39 +0200
Subject: [FFmpeg-user] Fwd: .configure can't locate x264.h
In-Reply-To: <CAO2fFsVgf3=exKq3eDDDmiO9C+=OcM6+ch5+fHSH3RBwsrh=wg@mail.gmail.com>
References: <CAO2fFsWUeo1oR=P+SdRuf1yhdO4=3Nxh0KG-CiY49CMR7qu0mw@mail.gmail.com>
	<CAO2fFsVgf3=exKq3eDDDmiO9C+=OcM6+ch5+fHSH3RBwsrh=wg@mail.gmail.com>
Message-ID: <CAMCbGjznxNcSdwj4U1p6aEoFwJmTXY=_qAyvfjBrFgicpp20Wg@mail.gmail.com>

On Wed, Jul 10, 2013 at 2:33 PM, Kesten Broughton
<solarmobiletrailers at gmail.com> wrote:

> ERROR: libx264 not found

you may need to add
--extra-ldflags="-ldl"
to the ffmpeg ./configure part
(This is due to the new way x264 is dynamically loading opencl stuff.)

b.

From cehoyos at ag.or.at  Wed Jul 10 22:42:09 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 10 Jul 2013 20:42:09 +0000 (UTC)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
Message-ID: <loom.20130710T224032-641@post.gmane.org>

Albert Scholtalbers <compuvision.sr <at> gmail.com> writes:

> I found a working command under Linux:
> -vf subtitles=filename=src.srt

> But I haven't found any option to set
> font/size/color in document or Google yet.

I am really no srt expert, but I would have 
expected that the srt file contains all 
definitions for font/size/color.

Carl Eugen


From compuvision.sr at gmail.com  Wed Jul 10 23:08:47 2013
From: compuvision.sr at gmail.com (Albert Scholtalbers)
Date: Wed, 10 Jul 2013 18:08:47 -0300
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <loom.20130710T224032-641@post.gmane.org>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
Message-ID: <CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>

Thx for your honesty.

It seems besides libass, you also need to enable fontconfig
so --enable-libass --enable-fontconfig

I think otherwise you only have one basic font.

.srt files only contain txt and starttime code and stoptime code
so very easy and straight forward.

>> I found a working command under Linux:
>> -vf subtitles=filename=src.srt

>> But I haven't found any option to set
>> font/size/color in document or Google yet.

>I am really no srt expert, but I would have
>expected that the srt file contains all
>definitions for font/size/color.

From cehoyos at ag.or.at  Wed Jul 10 23:13:48 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 10 Jul 2013 21:13:48 +0000 (UTC)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
Message-ID: <loom.20130710T231313-408@post.gmane.org>

Albert Scholtalbers <compuvision.sr <at> gmail.com> writes:

> .srt files only contain txt and starttime code 
> and stoptime code

This is at least not true for the one 
srt file that FFmpeg uses for 
regression tests.

Please avoid top-posting here, Carl Eugen


From compuvision.sr at gmail.com  Wed Jul 10 23:35:04 2013
From: compuvision.sr at gmail.com (Albert Scholtalbers)
Date: Wed, 10 Jul 2013 18:35:04 -0300
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <loom.20130710T231313-408@post.gmane.org>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
Message-ID: <CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>

> Please avoid top-posting here, Carl Eugen

I don't know what you mean by that.
Why I discovered is that there is a gui for the mailinglists now.
So it will be easier to reply and format than in Gmail :)

From barsnick at gmx.net  Wed Jul 10 23:52:35 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Wed, 10 Jul 2013 23:52:35 +0200
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
Message-ID: <20130710215235.GA20341@sunshine.barsnick.net>

On Wed, Jul 10, 2013 at 18:35:04 -0300, Albert Scholtalbers wrote:
> > Please avoid top-posting here, Carl Eugen
> I don't know what you mean by that.

A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
A: Top-posting.
Q: What is the most annoying thing in e-mail?

M.

From compuvision.sr at gmail.com  Wed Jul 10 23:38:22 2013
From: compuvision.sr at gmail.com (gepebril)
Date: Wed, 10 Jul 2013 14:38:22 -0700 (PDT)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
Message-ID: <1373492302767-4659911.post@n4.nabble.com>

Aha,

Via the Gui now :)
Is there a libass .srt specialist withing the FFMpeg community
It seems there is hardly any info on this matter on the Linux platform



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/make-mpeg2-file-with-hardcoded-subs-from-srt-file-tp4659902p4659911.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From compuvision.sr at gmail.com  Wed Jul 10 23:59:10 2013
From: compuvision.sr at gmail.com (gepebril)
Date: Wed, 10 Jul 2013 14:59:10 -0700 (PDT)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <20130710215235.GA20341@sunshine.barsnick.net>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
	<20130710215235.GA20341@sunshine.barsnick.net>
Message-ID: <1373493550748-4659913.post@n4.nabble.com>

@Moritz

Thx for the reply.
If somebody writes that they don't know what somebody means.
You are not helping them by explaining why it's bad or good. Just an example
what top posting textually is..... 



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/make-mpeg2-file-with-hardcoded-subs-from-srt-file-tp4659902p4659913.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From dflett at bigpond.net.au  Thu Jul 11 01:39:28 2013
From: dflett at bigpond.net.au (Dan Flett)
Date: Thu, 11 Jul 2013 09:39:28 +1000
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
In-Reply-To: <loom.20130710T140109-897@post.gmane.org>
References: <20120827191454.GA23763@Xeslaro>
	<20120827155600.GC21473@arborea>	<000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>
	<loom.20130710T140109-897@post.gmane.org>
Message-ID: <000a01ce7dc6$b82d5ba0$288812e0$@bigpond.net.au>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos
> Sent: Wednesday, 10 July 2013 22:06


> > But I've discovered that this does work:
> >
> > $ /usr/local/bin/ffmpeg -f rawvideo -codec:v rawvideo -s 1680x1050
> > -pix_fmt rgb565le -frames:v 1 -r 1 -i /dev/fb0 screenshot-ff01.bmp
> 
> > [rawvideo  <at>  0x2a8c3a0] Invalid buffer size, packet size 20160 <
> > expected length 3528000
> 
> This line indicates that it does not work.
> Please add some printf's to get_pixfmt_from_fb_varinfo() in
> libavdevice/fbdev.c to show the contents of varinfo.
> (Although I fear a compressed format is used.)

Yes obviously there's some issue there too, but despite the error message,
ffmpeg does work in that I do get an output (a BMP screenshot in this case).

I will do as you suggest and report back.

Dan


From jam at tigger.ws  Thu Jul 11 03:29:01 2013
From: jam at tigger.ws (James Linder)
Date: Thu, 11 Jul 2013 09:29:01 +0800
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <20130710141944.GA8186@phare.normalesup.org>
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
	<loom.20130709T094054-726@post.gmane.org>
	<70EF77B2-4254-417D-8D2B-4849B793A5D9@tigger.ws>
	<loom.20130709T135126-473@post.gmane.org>
	<3424D80A-6E67-4AC2-9359-0DEE7F47E4F4@tigger.ws>
	<20130710141944.GA8186@phare.normalesup.org>
Message-ID: <502E99C1-2AC6-4ED9-B290-261A4BFCA3DE@tigger.ws>


On 10/07/2013, at 10:19 PM, Nicolas George wrote:

> Le duodi 22 messidor, an CCXXI, James Linder a ?crit :
>> [dvr] /home/dvr [132]% ./ffmpeg -f pulse -i default -f v4l2 -i /dev/video0 -vtag divx -s 384x288 -r pal -q:v 5 -f tee  -map 0:a -map 1:v Wed.avi | cat >Wed1.avi
> 
> The tee muxer in this command line does not make any sense. The cat is
> completely useless too. 

Hi Nicolas

Please explain a little more.

I want to output the pipe to ffplay or mplayer but after a while my display and my recorded file show pauses every 5 seconds or so of 100mS or so. The cat command would isolate the display process from the mix. vmstat, top, iostat and gnome-system-monitor do not give any clues to the reason for jerky display - they look the same when steady and jerky recording is taking place.

Cat also gives two files that I may inspect to confirm they are identical before proceeding with record-and-display.
Why is my cat completely useless?

snip from 1st post: (well the whole post :-) -----------------------------------------------------------------------------
Some years ago I setup a record-v4l-camera-from-interface with audio on line in using transcode. Perfect, but at kernel 2.6.32

I modernized the recording system only to find transcode still uses VIDIOC_S_PARM ioctls that were removed from kernel cica 2.6.33

I use ffmpeg now. All good except the audio is not synced. I apply -itsoffset and all is good for an hour or two then the recording hesitates every 5 or so seconds for say 100mS. I've looked at top, vmstat, iostat even gnome-system-monitor and can see nothing that would be causing it.

ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 -i /dev/video0 -vtag divx -s svga -r pal -q:v 5 file.avi -s svga -f rawvideo - | ffplay -f rawvideo -i - -s svga

There is not any swap or obvious unusual disk activity.

Any ideas welcome
James

PS
I tried avconv instead of ffmpeg and the stuttering ceased (joy, rapture) but I cannot sync audio (sad)
avplay stutters where ffplay does not
--------------------------------------------------------------------------------------------------------------------------------------

James
 

From soho123.2012 at gmail.com  Thu Jul 11 05:12:26 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Thu, 11 Jul 2013 11:12:26 +0800
Subject: [FFmpeg-user] how to use presets files
Message-ID: <CAFd2cZUGdnvRgwoPKrW3Zjx8V3Sj1P=zKF0kkqRYbBpZN4Tddw@mail.gmail.com>

Hi All,

I see the files about "presets" in  the folder "/ffmpeg/presets".
for example:
libvpx-1080p.ffpreset

it seems some parameter settings.
Does someone have idea about the purpose of presets?
How to use the presets for reduce latency when the video codec is libx264?

Thanks!

From zoleexfz at tanoda.suli.hu  Thu Jul 11 07:15:28 2013
From: zoleexfz at tanoda.suli.hu (=?ISO-8859-2?Q?F=FCves_Zolt=E1n?=)
Date: Thu, 11 Jul 2013 07:15:28 +0200
Subject: [FFmpeg-user] strange preset file parse error on gentoo from on
 10.0.7 and above in june
Message-ID: <51DE3F70.3050302@tanoda.suli.hu>

Hi all,

  I use ffmpeg (and fffserver) for streaming the tv to my laptop since 
a few months without any error, but a few days ago I got a parse error
when I used preset file some flags2 option didn't work anymore
(+fastpskip+mbtree) and with some other sets too

with this error:
Undefined constant or missing '(' in 'fastpskip'
Unable to parse option value "fastpskip+mbtree"
.ffmpeg/libx264-lossless_ultrafast.ffpreset: Invalid option or argument:
'flags2=+fastpskip+mbtree', parsed as 'flags2' = '+fastpskip+mbtree'

:O

I checked (double checked) gentoo forums but I didn't have any relevant
comment about this, so at this point I don't know what to do ( and I
used stable branch of the potage mostly so nothing magic stuff I prefer 
'it's just work operation' )

When I use these option (-fast-pskip 1 -mbtree 1 )  with ffmpeg command
it looks good and working but why and why not from the preset

Thanks for your time and help

keep the good work

Z.

From cehoyos at ag.or.at  Thu Jul 11 08:42:26 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 11 Jul 2013 06:42:26 +0000 (UTC)
Subject: [FFmpeg-user] strange preset file parse error on gentoo from on
	10.0.7 and above in june
References: <51DE3F70.3050302@tanoda.suli.hu>
Message-ID: <loom.20130711T084209-106@post.gmane.org>

F?ves Zolt?n <zoleexfz <at> tanoda.suli.hu> writes:

> with this error:
> Undefined constant or missing '(' in 'fastpskip'

Command line and complete, uncut console output 
missing.

Carl Eugen


From oliver at fromme.com  Thu Jul 11 08:49:41 2013
From: oliver at fromme.com (Oliver Fromme)
Date: Thu, 11 Jul 2013 08:49:41 +0200 (CEST)
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <502E99C1-2AC6-4ED9-B290-261A4BFCA3DE@tigger.ws>
Message-ID: <201307110649.r6B6nflk025866@haluter.fromme.com>

James Linder wrote:
 > 
 > On 10/07/2013, at 10:19 PM, Nicolas George wrote:
 > 
 > > Le duodi 22 messidor, an CCXXI, James Linder a ?crit :
 > > > [dvr] /home/dvr [132]% ./ffmpeg -f pulse -i default -f v4l2 -i /dev/video0 -vtag divx -s 384x288 -r pal -q:v 5 -f tee  -map 0:a -map 1:v Wed.avi | cat >Wed1.avi
 > > 
 > > The tee muxer in this command line does not make any sense. The cat is
 > > completely useless too. 
 > [...]
 > Why is my cat completely useless?

Because the following two commands produce the same result:

$ ffmpeg ... | cat > foo.avi
$ ffmpeg ... > foo.avi

The difference is that the first has an additional overhead
of one execve() (probably insignificant in this case) and
of one pipe and one process that have to handle the output
data.  The latter might be significant because it usually
involves at least two in-memory copy operations, depending
on the implementations of pipes and cat in your operating
system.

The above is a typical "useless use of cat".  It should be
avoided.

If everybody checked his/her shell scripts for useless uses
of cat and removed them, we could probably save a few
megawatts of electrical power per day that would otherwise
be wasted.  ;-)

Best regards
   Oliver


-- 
``We are all but compressed light'' (Albert Einstein)

From zoleexfz at tanoda.suli.hu  Thu Jul 11 09:24:24 2013
From: zoleexfz at tanoda.suli.hu (=?UTF-8?B?RsO8dmVzIFpvbHTDoW4=?=)
Date: Thu, 11 Jul 2013 09:24:24 +0200
Subject: [FFmpeg-user] strange preset file parse error on gentoo from on
 10.0.7 and above in june
In-Reply-To: <loom.20130711T084209-106@post.gmane.org>
References: <51DE3F70.3050302@tanoda.suli.hu>
	<loom.20130711T084209-106@post.gmane.org>
Message-ID: <51DE5DA8.7060207@tanoda.suli.hu>

Hi,

 thanks for your reply and than here is the command line and the console
output

but what i said before these setting worked for me many times until that
strange and sunny  summer day :)

ffmpeg  -f video4linux2 -i /dev/video0 -f alsa -i pulse -async 1
http://*.*.*.1:8090/feed1.ffm 

the preset file :

cat  .ffmpeg/libx264-lossless_ultrafast.ffpreset
coder=0
flags=+loop
cmp=+chroma
partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
me_method=dia
subq=0
me_range=16
g=250
keyint_min=25
sc_threshold=40
i_qfactor=0.71
b_strategy=1
qcomp=0.6
qmin=10
qmax=51
qdiff=4
directpred=1
flags2=+fastpskip+mbtree
cqp=0

ffserver stream  sets:

<Stream live>       # Output stream URL definition
   Feed feed1.ffm              # Feed from which to receive video
   Format flv

   # Audio settings
    AudioCodec aac
    Strict -2
    AudioBitRate 64
    AudioChannels 1
    AudioSampleRate 44100

   # Video settings
   VideoCodec libx264
   VideoSize 720x576
   VideoFrameRate 25 
   AVOptionVideo flags +global_header
    AVPresetVideo lossless_ultrafast
   AVOptionAudio flags +global_header

    StartSendOnKey
    VideoBitRate 700            # Video bitrate
</Stream>

and the ffserver output:

 ffserver -f  /*notpublic/FfserverTV
ffserver version 1.0.7 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 11 2013 09:15:01 with gcc 4.5.4 (Gentoo 4.5.4 p1.0,
pie-0.4.7)
  configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib
--mandir=/usr/share/man --enable-shared --cc=i686-pc-linux-gnu-gcc
--cxx=i686-pc-linux-gnu-g++ --ar=i686-pc-linux-gnu-ar
--optflags='-march=native -Os -pipe' --extra-cflags='-march=native -Os
-pipe' --extra-cxxflags='-march=native -Os -pipe' --disable-static
--enable-gpl --enable-postproc --enable-avfilter --enable-avresample
--disable-stripping --disable-debug --disable-doc --disable-vdpau
--enable-libmp3lame --enable-libx264 --enable-libxvid --enable-libfaac
--enable-nonfree --disable-indev=oss --disable-indev=jack
--enable-x11grab --enable-libpulse --disable-outdev=oss
--enable-libfreetype --enable-libvorbis --enable-libvpx
--enable-libopenjpeg --disable-amd3dnow --disable-amd3dnowext
--disable-altivec --disable-avx --disable-mmxext --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
[NULL @ 0xbf89c58c] [Eval @ 0xbf89aa98] Undefined constant or missing
'(' in 'fastpskip'
[NULL @ 0xbf89c58c] Unable to parse option value "fastpskip+mbtree"
/*notpublic/.ffmpeg/libx264-lossless_ultrafast.ffpreset: Invalid option
or argument: 'flags2=+fastpskip+mbtree', parsed as 'flags2' =
'+fastpskip+mbtree'
/*notpublic/FfserverTV:52: AVPreset error: lossless_ultrafast
Incorrect config file - exiting.

same results from the git repo:

ffserver version N-54624-g2e6338b Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 10 2013 16:51:26 with gcc 4.5.4 (Gentoo 4.5.4 p1.0,
pie-0.4.7)
  configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib
--mandir=/usr/share/man --enable-shared --cc=i686-pc-linux-gnu-gcc
--cxx=i686-pc-linux-gnu-g++ --ar=i686-pc-linux-gnu-ar
--optflags='-march=native -Os -pipe' --extra-cflags='-march=native -Os
-pipe' --extra-cxxflags='-march=native -Os -pipe' --disable-static
--enable-gpl --enable-postproc --enable-avfilter --enable-avresample
--disable-stripping --enable-nonfree --disable-indev=oss
--disable-indev=jack --disable-outdev=oss --enable-bzlib
--enable-runtime-cpudetect --disable-debug --disable-doc
--disable-gnutls --enable-hardcoded-tables --enable-iconv
--enable-network --disable-openssl --enable-ffplay --enable-vaapi
--disable-vdpau --enable-zlib --disable-libvo-aacenc
--disable-libvo-amrwbenc --enable-libmp3lame --disable-libfdk-aac
--disable-libaacplus --enable-libfaac --disable-libtheora
--disable-libtwolame --disable-libwavpack --enable-libx264
--enable-libxvid --disable-libcdio --disable-libiec61883
--disable-libdc1394 --disable-libcaca   libavutil      52. 39.100 / 52.
39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.100 / 55. 12.100
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 80.100 /  3. 80.100
  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
[NULL @ 0xbfa66cf4] [Eval @ 0xbfa65208] Undefined constant or missing
'(' in 'fastpskip'
[NULL @ 0xbfa66cf4] Unable to parse option value "fastpskip+mbtree"
/*notpublic/.ffmpeg/libx264-lossless_ultrafast.ffpreset: Invalid option
or argument: 'flags2=+fastpskip+mbtree', parsed as 'flags2' =
'+fastpskip+mbtree'
/*notpublic/FfserverTV:52: AVPreset error: lossless_ultrafast
Incorrect config file - exiting.

Thanks again for your time and help

Have a nice day

Z.


2013.07.11. 8:42 keltez?ssel, Carl Eugen Hoyos ?rta:
> F?ves Zolt?n <zoleexfz <at> tanoda.suli.hu> writes:
>
>> with this error:
>> Undefined constant or missing '(' in 'fastpskip'
> Command line and complete, uncut console output 
> missing.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From andrey.aleksandrovich at googlemail.com  Thu Jul 11 11:08:07 2013
From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich)
Date: Thu, 11 Jul 2013 12:08:07 +0300
Subject: [FFmpeg-user] how to use presets files
In-Reply-To: <CAFd2cZUGdnvRgwoPKrW3Zjx8V3Sj1P=zKF0kkqRYbBpZN4Tddw@mail.gmail.com>
References: <CAFd2cZUGdnvRgwoPKrW3Zjx8V3Sj1P=zKF0kkqRYbBpZN4Tddw@mail.gmail.com>
Message-ID: <CAPGkfYHTLJriy69c_HbjbL7yiSObwa--ShFtz4C9qCcPj=98hQ@mail.gmail.com>

http://ffmpeg.org/ffmpeg.html#Preset-files-1

On 7/11/13, Soho Soho123 <soho123.2012 at gmail.com> wrote:
> Hi All,
>
> I see the files about "presets" in  the folder "/ffmpeg/presets".
> for example:
> libvpx-1080p.ffpreset
>
> it seems some parameter settings.
> Does someone have idea about the purpose of presets?
> How to use the presets for reduce latency when the video codec is libx264?
>
> Thanks!
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From krueger at lesspain.de  Thu Jul 11 11:15:45 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 11 Jul 2013 11:15:45 +0200
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <1373493550748-4659913.post@n4.nabble.com>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
	<20130710215235.GA20341@sunshine.barsnick.net>
	<1373493550748-4659913.post@n4.nabble.com>
Message-ID: <CAEnqZEUWLhANnNYHRfC3ZoBzgdab-Amtb4Z+ZZPLTTJfiNNAKA@mail.gmail.com>

On Wed, Jul 10, 2013 at 11:59 PM, gepebril <compuvision.sr at gmail.com> wrote:
> @Moritz
>
> Thx for the reply.
> If somebody writes that they don't know what somebody means.
> You are not helping them by explaining why it's bad or good. Just an example

He actually did. Please read the mail again and if you don't
understand the implicit explanation, try googling "top-posting".

From phil_rhodes at rocketmail.com  Thu Jul 11 13:01:49 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Thu, 11 Jul 2013 04:01:49 -0700 (PDT)
Subject: [FFmpeg-user] how to use presets files
In-Reply-To: <CAPGkfYHTLJriy69c_HbjbL7yiSObwa--ShFtz4C9qCcPj=98hQ@mail.gmail.com>
References: <CAFd2cZUGdnvRgwoPKrW3Zjx8V3Sj1P=zKF0kkqRYbBpZN4Tddw@mail.gmail.com>
	<CAPGkfYHTLJriy69c_HbjbL7yiSObwa--ShFtz4C9qCcPj=98hQ@mail.gmail.com>
Message-ID: <1373540509.88508.YahooMailNeo@web121103.mail.ne1.yahoo.com>

I'm sure I was told at one point that presets are no longer files, is this still true?

P


________________________________
 From: Andrey Aleksandrovich <andrey.aleksandrovich at googlemail.com>
To: FFmpeg user questions <ffmpeg-user at ffmpeg.org> 
Sent: Thursday, 11 July 2013, 10:08
Subject: Re: [FFmpeg-user] how to use presets files
 

http://ffmpeg.org/ffmpeg.html#Preset-files-1

On 7/11/13, Soho Soho123 <soho123.2012 at gmail.com> wrote:
> Hi All,
>
> I see the files about "presets" in? the folder "/ffmpeg/presets".
> for example:
> libvpx-1080p.ffpreset
>
> it seems some parameter settings.
> Does someone have idea about the purpose of presets?
> How to use the presets for reduce latency when the video codec is libx264?
>
> Thanks!
> _______________________________________________
> 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 doenerbudenchef at yahoo.de  Thu Jul 11 13:33:53 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Thu, 11 Jul 2013 04:33:53 -0700 (PDT)
Subject: [FFmpeg-user] Increase volume while using amerge
Message-ID: <1373542433407-4659924.post@n4.nabble.com>

Hi all, 

I've tried to increase the volume of my output file. But when I use the
amerge parameter to merge my mono audio streams the volume filter does not
work.

Output:

C:\>ffmpeg -i different.mxf -s 640x360  -aspect 16:9 -y -vb 1500k -bt 1500k
-vcodec libx264 -r 25 -vf yadif -pix_fmt yuv420p -f flv -filter_complex
amerge -af volume=15.8dB -strict experimental -acodec aac -ab 128k -ar 44100
-ss 0 -pass 1 -fastfirstpass 1 different.flv && ffmpeg -i different.mxf -s
640x360  -aspect 16:9 -y -vb 1500k -bt 1500k -vcodec libx264 -r 25 -vf yadif
-pix_fmt yuv420p -f flv -filter_complex amerge -af volume=15.8dB -strict
experimental -acodec aac -ab 128k -ar 44100 -ss 0 -pass 2 -fastfirstpass 0
different.flv

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
Guessed Channel Layout for  Input Stream #0.1 : mono
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
Guessed Channel Layout for  Input Stream #0.6 : mono
Guessed Channel Layout for  Input Stream #0.7 : mono
Guessed Channel Layout for  Input Stream #0.8 : mono
Input #0, mxf, from 'different.mxf':
  Metadata:
    uid             : 90798c31-2fe4-11e2-b193-00e081d26969
    generation_uid  : 90798c32-2fe4-11e2-9f65-00e081d26969
    company_name    : Adobe Systems Incorporated
    product_name    : Adobe Premiere Pro
    product_version : 5.0.0
    application_platform: win32
    product_uid     : 10ab07a9-e89e-7510-a923-ea9220524153
    modification_date: 2012-11-16 11:55:49
    timecode        : 00:00:00:00
  Duration: 00:05:00.48, start: 0.000000, bitrate: 59264 kb/s
    Stream #0:0: Video: mpeg2video (4:2:2), yuv422p, 1920x1080 [SAR 1:1 DAR
16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc
    Stream #0:1: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:2: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:3: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:4: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:5: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:6: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:7: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:8: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
[Parsed_amerge_0 @ 00000000024ad500] No channel layout for input 1
[Parsed_amerge_0 @ 00000000024ad500] Input channel layouts overlap: output
layout will be determined by the number of distinct input channels
[libx264 @ 0000000002523900] using SAR=1/1
[libx264 @ 0000000002523900] using cpu capabilities: MMX2 SSE2Fast SSSE3
SSE4.2 AVX
[libx264 @ 0000000002523900] profile Main, level 3.0
[libx264 @ 0000000002523900] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4 AVC
codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options:
cabac=1 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=6 lookahead_threads=2
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=abr mbtree=1 bitrate=1500 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69
qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, flv, to 'different.flv':
  Metadata:
    uid             : 90798c31-2fe4-11e2-b193-00e081d26969
    generation_uid  : 90798c32-2fe4-11e2-9f65-00e081d26969
    company_name    : Adobe Systems Incorporated
    product_name    : Adobe Premiere Pro
    product_version : 5.0.0
    application_platform: win32
    product_uid     : 10ab07a9-e89e-7510-a923-ea9220524153
    modification_date: 2012-11-16 11:55:49
    timecode        : 00:00:00:00
    encoder         : Lavf55.10.100
    Stream #0:0: Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, stereo,
fltp, 128 kb/s
    Stream #0:1: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p,
640x360 [SAR 1:1 DAR 16:9], q=-1--1, pass 1, 1500 kb/s, 1k tbn, 25 tbc
Stream mapping:
  Stream #0:1 (pcm_s24le) -> amerge:in0 (graph 0)
  Stream #0:2 (pcm_s24le) -> amerge:in1 (graph 0)
  amerge (graph 0) -> Stream #0:0 (aac)
  Stream #0:0 -> #0:1 (mpeg2video -> libx264)
Press [q] to stop, [?] for help
[mpeg2video @ 00000000024ac8a0] warning: first frame is no keyframe
    Last message repeated 1 times
frame= 7512 fps= 67 q=-1.0 Lsize=   59746kB time=00:05:00.49
bitrate=1628.8kbits/s
video:54679kB audio:4704kB subtitle:0 global headers:0kB muxing overhead
0.610088%
[libx264 @ 0000000002523900] frame I:117   Avg QP:11.66  size: 30327
[libx264 @ 0000000002523900] frame P:4401  Avg QP:15.46  size:  9736
[libx264 @ 0000000002523900] frame B:2994  Avg QP:17.18  size:  3205
[libx264 @ 0000000002523900] consecutive B-frames: 32.3% 38.9% 14.4% 14.4%
[libx264 @ 0000000002523900] mb I  I16..4: 36.9%  0.0% 63.1%
[libx264 @ 0000000002523900] mb P  I16..4: 28.9%  0.0%  0.0%  P16..4: 60.4% 
0.0%  0.0%  0.0%  0.0%    skip:10.7%
[libx264 @ 0000000002523900] mb B  I16..4:  3.6%  0.0%  0.0%  B16..8: 36.8% 
0.0%  0.0%  direct:22.1%  skip:37.5%  L0:27.8% L1:38.1% BI:34.2%
[libx264 @ 0000000002523900] final ratefactor: 14.89
[libx264 @ 0000000002523900] coded y,uvDC,uvAC intra: 52.5% 38.9% 22.6%
inter: 34.7% 19.5% 2.3%
[libx264 @ 0000000002523900] i16 v,h,dc,p: 48% 15% 27% 10%
[libx264 @ 0000000002523900] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 24% 17% 13%  7% 
8%  8%  7%  8%  7%
[libx264 @ 0000000002523900] i8c dc,h,v,p: 67% 12% 17%  4%
[libx264 @ 0000000002523900] Weighted P-Frames: Y:9.5% UV:1.6%
[libx264 @ 0000000002523900] kb/s:1490.71
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
Guessed Channel Layout for  Input Stream #0.1 : mono
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
Guessed Channel Layout for  Input Stream #0.6 : mono
Guessed Channel Layout for  Input Stream #0.7 : mono
Guessed Channel Layout for  Input Stream #0.8 : mono
Input #0, mxf, from 'different.mxf':
  Metadata:
    uid             : 90798c31-2fe4-11e2-b193-00e081d26969
    generation_uid  : 90798c32-2fe4-11e2-9f65-00e081d26969
    company_name    : Adobe Systems Incorporated
    product_name    : Adobe Premiere Pro
    product_version : 5.0.0
    application_platform: win32
    product_uid     : 10ab07a9-e89e-7510-a923-ea9220524153
    modification_date: 2012-11-16 11:55:49
    timecode        : 00:00:00:00
  Duration: 00:05:00.48, start: 0.000000, bitrate: 59264 kb/s
    Stream #0:0: Video: mpeg2video (4:2:2), yuv422p, 1920x1080 [SAR 1:1 DAR
16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc
    Stream #0:1: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:2: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:3: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:4: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:5: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:6: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:7: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
    Stream #0:8: Audio: pcm_s24le, 48000 Hz, mono, s32, 1152 kb/s
[Parsed_amerge_0 @ 00000000024d4aa0] No channel layout for input 1
[Parsed_amerge_0 @ 00000000024d4aa0] Input channel layouts overlap: output
layout will be determined by the number of distinct input channels
[libx264 @ 00000000047753a0] using SAR=1/1
[libx264 @ 00000000047753a0] using cpu capabilities: MMX2 SSE2Fast SSSE3
SSE4.2 AVX
[libx264 @ 00000000047753a0] profile High, level 3.0
[libx264 @ 00000000047753a0] 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 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 dead
zone=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_lookahe
ad=40 rc=2pass mbtree=1 bitrate=1500 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
Output #0, flv, to 'different.flv':
  Metadata:
    uid             : 90798c31-2fe4-11e2-b193-00e081d26969
    generation_uid  : 90798c32-2fe4-11e2-9f65-00e081d26969
    company_name    : Adobe Systems Incorporated
    product_name    : Adobe Premiere Pro
    product_version : 5.0.0
    application_platform: win32
    product_uid     : 10ab07a9-e89e-7510-a923-ea9220524153
    modification_date: 2012-11-16 11:55:49
    timecode        : 00:00:00:00
    encoder         : Lavf55.10.100
    Stream #0:0: Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, stereo,
fltp, 128 kb/s
    Stream #0:1: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p,
640x360 [SAR 1:1 DAR 16:9], q=-1--1, pass 2, 1500 kb/s, 1k tbn, 25 tbc
Stream mapping:
  Stream #0:1 (pcm_s24le) -> amerge:in0 (graph 0)
  Stream #0:2 (pcm_s24le) -> amerge:in1 (graph 0)
  amerge (graph 0) -> Stream #0:0 (aac)
  Stream #0:0 -> #0:1 (mpeg2video -> libx264)
Press [q] to stop, [?] for help
[mpeg2video @ 00000000024abaa0] warning: first frame is no keyframe
    Last message repeated 1 times
frame= 7512 fps= 60 q=-1.0 Lsize=   60088kB time=00:05:00.49
bitrate=1638.1kbits/s
video:55022kB audio:4704kB subtitle:0 global headers:0kB muxing overhead
0.606587%
[libx264 @ 00000000047753a0] frame I:117   Avg QP:13.20  size: 27204
[libx264 @ 00000000047753a0] frame P:4401  Avg QP:15.57  size:  9706
[libx264 @ 00000000047753a0] frame B:2994  Avg QP:17.51  size:  3488
[libx264 @ 00000000047753a0] consecutive B-frames: 32.3% 38.9% 14.4% 14.4%
[libx264 @ 00000000047753a0] mb I  I16..4: 13.5% 69.2% 17.3%
[libx264 @ 00000000047753a0] mb P  I16..4:  1.0%  7.0%  1.2%  P16..4: 37.3%
30.2% 15.1%  0.0%  0.0%    skip: 8.2%
[libx264 @ 00000000047753a0] mb B  I16..4:  0.0%  0.7%  0.1%  B16..8: 45.6%
12.2%  2.9%  direct: 4.1%  skip:34.4%  L0:40.2% L1:45.2% BI:14.6%
[libx264 @ 00000000047753a0] 8x8 transform intra:75.0% inter:66.6%
[libx264 @ 00000000047753a0] coded y,uvDC,uvAC intra: 81.7% 64.3% 45.8%
inter: 38.6% 23.9% 3.7%
[libx264 @ 00000000047753a0] i16 v,h,dc,p: 47% 18%  6% 29%
[libx264 @ 00000000047753a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 10% 27%  5% 
7%  9%  6%  8%  6%
[libx264 @ 00000000047753a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 14% 14%  6% 
9% 10%  8%  7%  5%
[libx264 @ 00000000047753a0] i8c dc,h,v,p: 58% 14% 19%  8%
[libx264 @ 00000000047753a0] Weighted P-Frames: Y:9.5% UV:1.6%
[libx264 @ 00000000047753a0] ref P L0: 60.0% 11.9% 18.7%  9.0%  0.5%
[libx264 @ 00000000047753a0] ref B L0: 78.2% 19.4%  2.5%
[libx264 @ 00000000047753a0] ref B L1: 94.8%  5.2%
[libx264 @ 00000000047753a0] kb/s:1500.05

Thanks,
Alex





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From cehoyos at ag.or.at  Thu Jul 11 14:09:16 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 11 Jul 2013 12:09:16 +0000 (UTC)
Subject: [FFmpeg-user] Increase volume while using amerge
References: <1373542433407-4659924.post@n4.nabble.com>
Message-ID: <loom.20130711T140730-496@post.gmane.org>

Alex <doenerbudenchef <at> yahoo.de> writes:

> C:\>ffmpeg -i different.mxf -s 640x360  -aspect 16:9 
> -y -vb 1500k -bt 1500k -vcodec libx264 -r 25 -vf yadif 
> -pix_fmt yuv420p -f flv -filter_complex amerge 
> -af volume=15.8dB -strict experimental -acodec aac 
> -ab 128k -ar 44100 -ss 0 -pass 1 -fastfirstpass 1 
> different.flv

-filter_complex amerge,volume=15.8dB (untested)

Generally:
Please try to shorten your command line as long 
as you can still reproduce your issue, it makes 
helping you significantly easier.

Carl Eugen


From doenerbudenchef at yahoo.de  Thu Jul 11 14:31:11 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Thu, 11 Jul 2013 05:31:11 -0700 (PDT)
Subject: [FFmpeg-user] Increase volume while using amerge
In-Reply-To: <loom.20130711T140730-496@post.gmane.org>
References: <1373542433407-4659924.post@n4.nabble.com>
	<loom.20130711T140730-496@post.gmane.org>
Message-ID: <1373545871656-4659926.post@n4.nabble.com>

Hi Carl Eugen,

thanks, it works. But I don't want to shorten the command, because we are
using a special script which include some parameters dynamically. 

Thanks in advance
Alex



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From dflett at bigpond.net.au  Thu Jul 11 15:13:14 2013
From: dflett at bigpond.net.au (Dan Flett)
Date: Thu, 11 Jul 2013 23:13:14 +1000
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
In-Reply-To: <000a01ce7dc6$b82d5ba0$288812e0$@bigpond.net.au>
References: <20120827191454.GA23763@Xeslaro>	<20120827155600.GC21473@arborea>	<000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>	<loom.20130710T140109-897@post.gmane.org>
	<000a01ce7dc6$b82d5ba0$288812e0$@bigpond.net.au>
Message-ID: <000e01ce7e38$682b9c80$3882d580$@bigpond.net.au>

> -----Original Message-----
> Sent: Thursday, 11 July 2013 09:39
> To: 'FFmpeg user questions'
> Subject: Re: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb
> drivers

> > Please add some printf's to get_pixfmt_from_fb_varinfo() in
> > libavdevice/fbdev.c to show the contents of varinfo.
> > (Although I fear a compressed format is used.)
> 
> Yes obviously there's some issue there too, but despite the error message,
> ffmpeg does work in that I do get an output (a BMP screenshot in this
case).
> 
> I will do as you suggest and report back.

OK, sorry, doing the printf's was a bit over my head I'm afraid.

What I did do is look at the output of "fbset" on my raspberry pi:

pi at raspberrypi ~ $ fbset -i

mode "1680x1050"
    geometry 1680 1050 1680 1050 16
    timings 0 0 0 0 0 0 0
    rgba 5/11,6/5,5/0,0/16
endmode

Frame buffer device information:
    Name        : BCM2708 FB
    Address     : 0x50006000
    Size        : 3528000
    Type        : PACKED PIXELS
    Visual      : TRUECOLOR
    XPanStep    : 0
    YPanStep    : 0
    YWrapStep   : 0
    LineLength  : 3360
    Accelerator : No

>From this I worked out the RGBA offsets and produced this patch for fbdev.c:

pi at raspberrypi ~/ffmpeg-dmo-1.2.1/libavdevice $ diff -u fbdev_orig.c fbdev.c
--- fbdev_orig.c        2013-07-11 12:28:28.554753775 +0000
+++ fbdev.c     2013-07-11 12:27:01.559807736 +0000
@@ -59,6 +59,7 @@
     {  32,       3,           2,           8,            0,
AV_PIX_FMT_ABGR  },
     {  24,       0,           8,          16,            0,
AV_PIX_FMT_RGB24 },
     {  24,      16,           8,           0,            0,
AV_PIX_FMT_BGR24 },
+    {  16,       11,          5,           0,            16,
AV_PIX_FMT_RGB565 }, // raspi
 };

 static enum AVPixelFormat get_pixfmt_from_fb_varinfo(struct
fb_var_screeninfo *varinfo)

This seems to work without error now:

pi at raspberrypi ~ $ /usr/local/bin/ffmpeg -y -f fbdev -r 30 -i /dev/fb0
-codec:v copy -frames 1000 -f rawvideo /dev/null
ffmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  9 2013 14:47:41 with gcc 4.6 (Debian 4.6.3-14+rpi1)
  configuration:
  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
[fbdev @ 0x20a9440] w:1024 h:576 bpp:16 pixfmt:rgb565le fps:30/1
bit_rate:283115520
[fbdev @ 0x20a9440] Estimating duration from bitrate, this may be inaccurate
Input #0, fbdev, from '/dev/fb0':
  Duration: N/A, start: 1373547949.464476, bitrate: 283115 kb/s
    Stream #0:0: Video: rawvideo (RGB[16] / 0x10424752), rgb565le, 1024x576,
283115 kb/s, 30 tbr, 1000k tbn, 30 tbc
Output #0, rawvideo, to '/dev/null':
  Metadata:
    encoder         : Lavf54.63.104
    Stream #0:0: Video: rawvideo (RGB[16] / 0x10424752), rgb565le, 1024x576,
q=2-31, 283115 kb/s, 90k tbn, 30 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
frame= 1000 fps= 30 q=-1.0 Lsize= 1152000kB time=00:00:33.34
bitrate=283045.8kbits/s
video:1152000kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.000000%

With the framebuffer at 1680x1050 I was getting a capture rate of about
4.6fps to /dev/null on the Raspberry Pi.  Lowering the resolution to
1024x576 and capturing at 30fps works in real-time, with ffmpeg getting
about 85% CPU usage on the Raspberry Pi.  I'm hoping to pipe this into
gstreamer with their omxh264enc hardware encoding module, but at the moment
I don't think gstreamer handles rgb565le raw video very well.

Cheers,

Dan


From cehoyos at ag.or.at  Thu Jul 11 15:20:11 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 11 Jul 2013 13:20:11 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
References: <20120827191454.GA23763@Xeslaro>	<20120827155600.GC21473@arborea>	<000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>	<loom.20130710T140109-897@post.gmane.org>
	<000a01ce7dc6$b82d5ba0$288812e0$@bigpond.net.au>
	<000e01ce7e38$682b9c80$3882d580$@bigpond.net.au>
Message-ID: <loom.20130711T151749-928@post.gmane.org>

Dan Flett <dflett <at> bigpond.net.au> writes:

> What I did do is look at the output of "fbset" on my raspberry pi:
> 
> pi <at> raspberrypi ~ $ fbset -i
> 
> mode "1680x1050"
>     geometry 1680 1050 1680 1050 16
>     timings 0 0 0 0 0 0 0
>     rgba 5/11,6/5,5/0,0/16
> endmode

Thank you for this hint!
I wonder what the output here is trying to tell me:
rgba 8/0,8/0,8/0,0/0

> +    {  16,       11,          5,           0,            16,
> AV_PIX_FMT_RGB565 }, // raspi

Could you confirm that "AV_PIX_FMT_RGB565LE" works 
but "AV_PIX_FMT_RGB565BE" shows incorrect colours?

Thank you, Carl Eugen


From cehoyos at ag.or.at  Thu Jul 11 15:22:20 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 11 Jul 2013 13:22:20 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
References: <20120827191454.GA23763@Xeslaro>	<20120827155600.GC21473@arborea>	<000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>	<loom.20130710T140109-897@post.gmane.org>
	<000a01ce7dc6$b82d5ba0$288812e0$@bigpond.net.au>
	<000e01ce7e38$682b9c80$3882d580$@bigpond.net.au>
	<loom.20130711T151749-928@post.gmane.org>
Message-ID: <loom.20130711T152144-670@post.gmane.org>

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

> > +    {  16,       11,          5,           0,            16,
> > AV_PIX_FMT_RGB565 }, // raspi
> 
> Could you confirm that "AV_PIX_FMT_RGB565LE" works 
> but "AV_PIX_FMT_RGB565BE" shows incorrect colours?

Never mind, this is not the necessary test.

I will commit to my repo and ask Michael to merge.

Thank you!

Carl Eugen


From rob.ludwig at lyricallabs.com  Thu Jul 11 17:57:02 2013
From: rob.ludwig at lyricallabs.com (Robert Ludwig)
Date: Thu, 11 Jul 2013 10:57:02 -0500
Subject: [FFmpeg-user] MSVC or Intel Compiler Versions
Message-ID: <51DED5CE.7080806@lyricallabs.com>

Hi-
I'm trying to compile a ffmpeg for Windows, and I see in the FATE 
results listed at fate.ffmpeg.org that there are test results for a 
win32 binary compiled with the Intel C++ Compiler and the Microsoft 
Compiler. Is there any way to get the source code that was used for 
those compilations specifically? Neither of those compilers will compile 
ffmpeg directly from the git head, because of C99 compatibility issues, 
among other things, so if someone has a working version that would help 
a lot.

Thanks,
-Rob

From andrey.aleksandrovich at googlemail.com  Thu Jul 11 18:19:21 2013
From: andrey.aleksandrovich at googlemail.com (Andrey Aleksandrovich)
Date: Thu, 11 Jul 2013 19:19:21 +0300
Subject: [FFmpeg-user] how to use presets files
In-Reply-To: <1373540509.88508.YahooMailNeo@web121103.mail.ne1.yahoo.com>
References: <CAFd2cZUGdnvRgwoPKrW3Zjx8V3Sj1P=zKF0kkqRYbBpZN4Tddw@mail.gmail.com>
	<CAPGkfYHTLJriy69c_HbjbL7yiSObwa--ShFtz4C9qCcPj=98hQ@mail.gmail.com>
	<1373540509.88508.YahooMailNeo@web121103.mail.ne1.yahoo.com>
Message-ID: <CAPGkfYGnagcDus_8GrMV1rnHksUySCoHbhv86purc=Uyg3N62w@mail.gmail.com>

AFAIK, the presets are internal part of codecs now i.e. any codec has
(or not) his own preset, which is like specific flags. To learn how to
use them you have to look at codec manuals, not at ffmpeg. In your
case - 'libvpx' manual.

On 7/11/13, Phil Rhodes <phil_rhodes at rocketmail.com> wrote:
> I'm sure I was told at one point that presets are no longer files, is this
> still true?
>
> P
>
>
> ________________________________
>  From: Andrey Aleksandrovich <andrey.aleksandrovich at googlemail.com>
> To: FFmpeg user questions <ffmpeg-user at ffmpeg.org>
> Sent: Thursday, 11 July 2013, 10:08
> Subject: Re: [FFmpeg-user] how to use presets files
>
>
> http://ffmpeg.org/ffmpeg.html#Preset-files-1
>
> On 7/11/13, Soho Soho123 <soho123.2012 at gmail.com> wrote:
>> Hi All,
>>
>> I see the files about "presets" in? the folder "/ffmpeg/presets".
>> for example:
>> libvpx-1080p.ffpreset
>>
>> it seems some parameter settings.
>> Does someone have idea about the purpose of presets?
>> How to use the presets for reduce latency when the video codec is
>> libx264?
>>
>> Thanks!
>> _______________________________________________
>> 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 tevans.uk at googlemail.com  Thu Jul 11 18:41:08 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Thu, 11 Jul 2013 17:41:08 +0100
Subject: [FFmpeg-user] how to use presets files
In-Reply-To: <1373540509.88508.YahooMailNeo@web121103.mail.ne1.yahoo.com>
References: <CAFd2cZUGdnvRgwoPKrW3Zjx8V3Sj1P=zKF0kkqRYbBpZN4Tddw@mail.gmail.com>
	<CAPGkfYHTLJriy69c_HbjbL7yiSObwa--ShFtz4C9qCcPj=98hQ@mail.gmail.com>
	<1373540509.88508.YahooMailNeo@web121103.mail.ne1.yahoo.com>
Message-ID: <CAFHbX1Kb28DtfXN4xFPKvi7b0769-ENFRJR4wA87NLeS47cezg@mail.gmail.com>

On Thu, Jul 11, 2013 at 12:01 PM, Phil Rhodes
<phil_rhodes at rocketmail.com> wrote:
> I'm sure I was told at one point that presets are no longer files, is this still true?
>
> P

ISTR that preset files still exist, but they only contain ffmpeg
specific options.

Encoding profiles used to be specified for things like h264, and you
would do that with -pre or -vpre. This was then changed to just pass
these values through to the encoding library itself, so with x264 for
instance you can pass through a profile, a preset and a tuning, with
-profile:v -preset:v and -tune:v.

Ie, you can still use -pre, but if you want to specify x264, profile
"main", preset "fast", then it is not "-vcodec x264 -vpre
x264-main-fast", it is "-c:v x264 -profile:v main -preset:v fast". You
can get the arguments for profile/preset/tune from the x264 binary
with --fullhelp.

Cheers

Tom

From cehoyos at ag.or.at  Thu Jul 11 18:42:59 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 11 Jul 2013 16:42:59 +0000 (UTC)
Subject: [FFmpeg-user] MSVC or Intel Compiler Versions
References: <51DED5CE.7080806@lyricallabs.com>
Message-ID: <loom.20130711T183939-544@post.gmane.org>

Robert Ludwig <rob.ludwig <at> lyricallabs.com> writes:

> I'm trying to compile a ffmpeg for Windows, and I 
> see in the FATE results listed at fate.ffmpeg.org 
> that there are test results for a win32 binary 
> compiled with the Intel C++ Compiler and the 
> Microsoft Compiler. Is there any way to get the 
> source code that was used for those compilations 
> specifically?

Definitely, please see http://ffmpeg.org/download.html

I will spare you the link to our platform 
documentation that explains how to compile 
FFmpeg with MSVC (and icl) because I am 
sure you read it before posting here, but ...

> Neither of those compilers will compile 
> ffmpeg directly from the git head

... what is unfortunately missing is the 
configure line you used, and the complete, 
uncut output of "make V=1" after you ran 
"make" several times and saw identical 
errors.
(Please do never post the complete, uncut 
output of the first failing make call.)

Carl Eugen

PS:
http://ffmpeg.org/platform.html#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows


From compuvision.sr at gmail.com  Thu Jul 11 19:04:18 2013
From: compuvision.sr at gmail.com (gepebril)
Date: Thu, 11 Jul 2013 10:04:18 -0700 (PDT)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <CAEnqZEUWLhANnNYHRfC3ZoBzgdab-Amtb4Z+ZZPLTTJfiNNAKA@mail.gmail.com>
References: <CAECKL3qni1x-yqchqRQ3Vr0xK=W3L_xc1tpTAUXurQAvDi-wQg@mail.gmail.com>
	<loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
	<20130710215235.GA20341@sunshine.barsnick.net>
	<1373493550748-4659913.post@n4.nabble.com>
	<CAEnqZEUWLhANnNYHRfC3ZoBzgdab-Amtb4Z+ZZPLTTJfiNNAKA@mail.gmail.com>
Message-ID: <1373562258940-4659934.post@n4.nabble.com>

Ok,

Still not clear though. I see that the gui removes all previous text from
the former mail. So that is probably not-top posting.

Anyway, after some hours of testing and wandering around the Net. I managed
to get it working on Windows with ffmpeg. Although it seems possible to
generate all kinds of different fonts using .srt, there is no actual working
sample off it on the Net, which I found.
That left me to a working solution, thanks to this  this
<http://forum.serviio.org/viewtopic.php?f=3&t=8255>   webpage. Basically the
trick is:
1) convert your .srt to .ass file (with ffmpeg)
2) Use Windows 
3) Follow the instruction on the website 
4) Render.

In commands it looks like this (create .ass file from .srt):
ffmpeg.exe -i subtitle_file.srt -y -scodec ass subtitle_file.ass


Make sure you have the local variables set right
set FC_CONFIG_DIR=E:\temp\Subtitles
set FONTCONFIG_FILE=fonts.xml
set FONTCONFIG_PATH=E:\temp\Subtitles

And use the related example of fonts.xml
<?xml version="1.0"?>
<fontconfig>

<dir>C:\WINDOWS\Fonts</dir>

<match target="pattern">
   <test qual="any" name="family"><string>mono</string></test>
   <edit name="family" mode="assign"><string>monospace</string></edit>
</match>

<match target="pattern">
   <test qual="all" name="family"
mode="not_eq"><string>sans-serif</string></test>
   <test qual="all" name="family"
mode="not_eq"><string>serif</string></test>
   <test qual="all" name="family"
mode="not_eq"><string>monospace</string></test>
   <edit name="family" mode="append_last"><string>sans-serif</string></edit>
</match>

<alias>
   <family>Times</family>
   <prefer><family>Times New Roman</family></prefer>
   <default><family>serif</family></default>
</alias>
<alias>
   <family>Helvetica</family>
   <prefer><family>Arial</family></prefer>
   <default><family>sans</family></default>
</alias>
<alias>
   <family>Courier</family>
   <prefer><family>Courier New</family></prefer>
   <default><family>monospace</family></default>
</alias>
<alias>
   <family>serif</family>
   <prefer><family>Times New Roman</family></prefer>
</alias>
<alias>
   <family>sans</family>
   <prefer><family>Arial</family></prefer>
</alias>
<alias>
   <family>monospace</family>
   <prefer><family>Andale Mono</family></prefer>
</alias>
<match target="pattern">
   <test name="family" mode="eq">
      <string>Courier New</string>
   </test>
   <edit name="family" mode="prepend">
      <string>monospace</string>
   </edit>
</match>
<match target="pattern">
   <test name="family" mode="eq">
      <string>Courier</string>
   </test>
   <edit name="family" mode="prepend">
      <string>monospace</string>
   </edit>
</match>

</fontconfig>

Create your video file with burned in subs
ffmpeg.exe -y -i "src.mkv"  -f dvd -target ntsc-dvd -r 29.97 -vf
scale=720:480 -vf "ass=subtitle_file.ass" -aspect 16:9 -b:v 8000k -g 12 -mbd
rd -trellis 1 -flags +mv0 -cmp 0 -subcmp 2 e:\dst.mpg


Note: make sure that the .ass file is in the same path as where are, as I
haven't managed to get a path reference working with ass 



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/make-mpeg2-file-with-hardcoded-subs-from-srt-file-tp4659902p4659934.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From thilo.borgmann at googlemail.com  Thu Jul 11 21:05:56 2013
From: thilo.borgmann at googlemail.com (Thilo Borgmann)
Date: Thu, 11 Jul 2013 21:05:56 +0200
Subject: [FFmpeg-user] Reading EXIF from mjpeg
Message-ID: <51DF0214.3010502@googlemail.com>

Hi,

can we read EXIF metadata from motion jpeg streams?

Any hint is welcome, using another lib is almost not ;)

Cheers,
Thilo

From onemda at gmail.com  Thu Jul 11 21:10:35 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 11 Jul 2013 19:10:35 +0000
Subject: [FFmpeg-user] Reading EXIF from mjpeg
In-Reply-To: <51DF0214.3010502@googlemail.com>
References: <51DF0214.3010502@googlemail.com>
Message-ID: <CAPYw7P5e6Li9FTZgk4x9c_EDSMSsOMcj4sMYvdXDtEbRPSMOyg@mail.gmail.com>

On 7/11/13, Thilo Borgmann <thilo.borgmann at googlemail.com> wrote:
> Hi,
>
> can we read EXIF metadata from motion jpeg streams?
>
> Any hint is welcome, using another lib is almost not ;)

Not at all, There is open bug report.

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

From cehoyos at ag.or.at  Thu Jul 11 22:13:47 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 11 Jul 2013 20:13:47 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
References: <20120827191454.GA23763@Xeslaro>	<20120827155600.GC21473@arborea>	<000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>	<loom.20130710T140109-897@post.gmane.org>
	<000a01ce7dc6$b82d5ba0$288812e0$@bigpond.net.au>
	<000e01ce7e38$682b9c80$3882d580$@bigpond.net.au>
Message-ID: <loom.20130711T221310-741@post.gmane.org>

Dan Flett <dflett <at> bigpond.net.au> writes:

> +    {  16,       11,          5,           0,            16,
> AV_PIX_FMT_RGB565 }, // raspi

Your patch was merged, thank you for 
working on this!

Carl Eugen


From johncrossman at berkeley.edu  Fri Jul 12 00:07:30 2013
From: johncrossman at berkeley.edu (John Crossman)
Date: Thu, 11 Jul 2013 15:07:30 -0700
Subject: [FFmpeg-user] Stream specifier ':v' in filtergraph ... matches no
 streams. (Seeking help on parsing output)
Message-ID: <CANdKVX4gDec-K=aLNRq5S0zHoACBmcD1yGoKkxzHG4V0=6RbLQ@mail.gmail.com>

Can you help me understand the problem?  (Thanks)

*The command:*

/usr/bin/ffmpeg -i audio.mp2 -i screen.mp4 -filter_complex
[1:v]pad=iw*2:ih[bg];[bg][0:v]overlay=W/2+((W/2-w)/2):(H-h)/2 -qscale:v 2
-strict -2 output.mp4

*The output:*

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
[mp3 @ 0x141dbe0] max_analyze_duration 5000000 reached at 5015510
microseconds
[mp3 @ 0x141dbe0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from '/export/mh_data/opencast/shared/workspace/mediapackage/
3d4024ec-6f38-4a46-9505-7c11dffdff8a/aeda9898-4f20-
4eec-a654-a8b18baf106a/audio.mp2':
  Duration: 00:04:59.46, start: 0.000000, bitrate: 192 kb/s
    Stream #0:0: Audio: mp2, 44100 Hz, stereo, s16p, 192 kb/s
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from '/export/mh_data/opencast/
shared/workspace/mediapackage/3d4024ec-6f38-4a46-9505-
7c11dffdff8a/8638cca9-f829-4edf-8020-4443f07c6e72/screen.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.63.104
  Duration: 00:04:59.00, start: 0.000000, bitrate: 168 kb/s
    Stream #1:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1024x768 [SAR 1:1 DAR 4:3], 28 kb/s, 100 fps, 100 tbr, 12800 tbn, 200 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #1:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
fltp, 123 kb/s
    Metadata:
      handler_name    : SoundHandler
Stream specifier ':v' in filtergraph description
[1:v]pad=iw*2:ih[bg];[bg][0:v]overlay=W/2+((W/2-w)/2):(H-h)/2 matches no
streams.

From rob.ludwig at lyricallabs.com  Fri Jul 12 00:36:10 2013
From: rob.ludwig at lyricallabs.com (Robert Ludwig)
Date: Thu, 11 Jul 2013 17:36:10 -0500
Subject: [FFmpeg-user] MSVC or Intel Compiler Versions
In-Reply-To: <loom.20130711T183939-544@post.gmane.org>
References: <51DED5CE.7080806@lyricallabs.com>
	<loom.20130711T183939-544@post.gmane.org>
Message-ID: <51DF335A.3040903@lyricallabs.com>

On 7/11/2013 11:42 AM, Carl Eugen Hoyos wrote:
> Robert Ludwig <rob.ludwig <at> lyricallabs.com> writes:
>
>> I'm trying to compile a ffmpeg for Windows, and I
>> see in the FATE results listed at fate.ffmpeg.org
>> that there are test results for a win32 binary
>> compiled with the Intel C++ Compiler and the
>> Microsoft Compiler. Is there any way to get the
>> source code that was used for those compilations
>> specifically?
> Definitely, please see http://ffmpeg.org/download.html
>
> I will spare you the link to our platform
> documentation that explains how to compile
> FFmpeg with MSVC (and icl) because I am
> sure you read it before posting here, but ...
>
>> Neither of those compilers will compile
>> ffmpeg directly from the git head
> ... what is unfortunately missing is the
> configure line you used, and the complete,
> uncut output of "make V=1" after you ran
> "make" several times and saw identical
> errors.
> (Please do never post the complete, uncut
> output of the first failing make call.)
>
> Carl Eugen
>
> PS:
> http://ffmpeg.org/platform.html#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
Thanks for your help! I had seen the instructions for the C99to89 builds 
that also required MSYS, etc. to execute. I was hoping that the FATE 
tests were referring to a native Visual Studio version, e.g. one that 
could be executed within Visual Studio using cmd.exe and would allow 
access to Visual Studio debugging tools. If such is not the case, I am 
probably out of luck.

Thanks again,
-Rob

From lou at lrcd.com  Fri Jul 12 01:10:38 2013
From: lou at lrcd.com (Lou)
Date: Thu, 11 Jul 2013 15:10:38 -0800
Subject: [FFmpeg-user] Stream specifier ':v' in filtergraph ... matches
 no streams. (Seeking help on parsing output)
In-Reply-To: <CANdKVX4gDec-K=aLNRq5S0zHoACBmcD1yGoKkxzHG4V0=6RbLQ@mail.gmail.com>
References: <CANdKVX4gDec-K=aLNRq5S0zHoACBmcD1yGoKkxzHG4V0=6RbLQ@mail.gmail.com>
Message-ID: <20130711151038.3e19223f@lrcd.com>

On Thu, 11 Jul 2013 15:07:30 -0700
John Crossman <johncrossman at berkeley.edu> wrote:

> Can you help me understand the problem?  (Thanks)
> 
> *The command:*
> 
> /usr/bin/ffmpeg -i audio.mp2 -i screen.mp4 -filter_complex
> [1:v]pad=iw*2:ih[bg];[bg][0:v]overlay=W/2+((W/2-w)/2):(H-h)/2 -qscale:v 2
> -strict -2 output.mp4

[0:v] does not refer to any stream. Your first input is audio only. I
assume you probably meant to use [1:v].

Note that, in your case, libx264 will be used for mp4 output, but this
encoder ignores "-qscale:v", so the default of "-crf 23" will be used
instead:

FFmpeg and x264 Encoding Guide
https://trac.ffmpeg.org/wiki/x264EncodingGuide

From dflett at bigpond.net.au  Fri Jul 12 05:38:11 2013
From: dflett at bigpond.net.au (Dan Flett)
Date: Fri, 12 Jul 2013 13:38:11 +1000
Subject: [FFmpeg-user] ffmpeg -f fbdev doesn't work with some fb drivers
In-Reply-To: <loom.20130711T221310-741@post.gmane.org>
References: <20120827191454.GA23763@Xeslaro>	<20120827155600.GC21473@arborea>	<000301ce7d63$b57633e0$20629ba0$@bigpond.net.au>	<loom.20130710T140109-897@post.gmane.org>	<000a01ce7dc6$b82d5ba0$288812e0$@bigpond.net.au>	<000e01ce7e38$682b9c80$3882d580$@bigpond.net.au>
	<loom.20130711T221310-741@post.gmane.org>
Message-ID: <001201ce7eb1$3d8b17c0$b8a14740$@bigpond.net.au>

> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Carl Eugen Hoyos
> Sent: Friday, 12 July 2013 06:14
> Dan Flett <dflett <at> bigpond.net.au> writes:
> 
> > +    {  16,       11,          5,           0,            16,
> > AV_PIX_FMT_RGB565 }, // raspi
> 
> Your patch was merged, thank you for
> working on this!

My pleasure!


From lamia at jeack.com.au  Fri Jul 12 06:49:44 2013
From: lamia at jeack.com.au (Bazza)
Date: Fri, 12 Jul 2013 14:49:44 +1000
Subject: [FFmpeg-user] This strea is something else ?
Message-ID: <jo1vt8tg2t7ovt26o51mi2edd79vrqb72v@4ax.com>

I'm seeing this quite a bit. Especially if I use a -target command.

Of itself, I don't mind that there's a 'dvd_nav_packet' stream but
 it does seem to intefere with other subsequent conversions. 
The example output text below  is  from a vob file generated by
 the Sony Vegas editor. FFMPEG will also generate the *0x1bf' as
I mention above but does it really need to 'classify' the nav_packet
as being stream #0 ? If it does, where can I find the reasoning 
behind this? (Using Zaranoe Windows version)


C:\Users\Admin>ffprobe -i z:\video_ts\vts_01_1.vob
ffprobe version N-53033-g56ba331 Copyright (c) 2007-2013 the FFmpeg
developers
  built on May 13 2013 22:23:31 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-font
config --enable-frei0r --enable-gnutls --enable-iconv --enable-libass
--enable-libbluray --enable-libcaca --enable-
libfreetype --enable-libgsm --enable-libilbc --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-am
rwb --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 --enabl
e-libvpx --enable-libx264 --enable-libxavs --enable-libxvid
--enable-zlib
  libavutil      52. 31.100 / 52. 31.100
  libavcodec     55.  9.100 / 55.  9.100
  libavformat    55.  7.100 / 55.  7.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 65.100 /  3. 65.100
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[mpeg @ 000000000033e760] max_analyze_duration 5000000 reached at
5000000 microseconds
Input #0, mpeg, from 'z:\video_ts\vts_01_1.vob':
  Duration: 00:09:10.19, start: 0.228544, bitrate: 5743 kb/s
    Stream #0:0[0x1bf]: Data: dvd_nav_packet     <*****************
    Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576
[SAR 16:15 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50
 tbc
    Stream #0:2[0xa0]: Audio: pcm_s16be, 48000 Hz, 2 channels, s16,
1536 kb/s
Unsupported codec with id 1145979222 for input stream 0  <***********



From cehoyos at ag.or.at  Fri Jul 12 08:25:53 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 12 Jul 2013 06:25:53 +0000 (UTC)
Subject: [FFmpeg-user] MSVC or Intel Compiler Versions
References: <51DED5CE.7080806@lyricallabs.com>
	<loom.20130711T183939-544@post.gmane.org>
	<51DF335A.3040903@lyricallabs.com>
Message-ID: <loom.20130712T082228-62@post.gmane.org>

Robert Ludwig <rob.ludwig <at> lyricallabs.com> writes:

> I was hoping that the FATE tests were referring to a 
> native Visual Studio version, e.g. one that could be 
> executed within Visual Studio using cmd.exe and 
> would allow access to Visual Studio debugging tools.

Of course builds made with msvc or icl are native 
Visual Studio versions that allow debugging with 
Visual Studio.

Carl Eugen


From cehoyos at ag.or.at  Fri Jul 12 09:04:46 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 12 Jul 2013 07:04:46 +0000 (UTC)
Subject: [FFmpeg-user] This strea is something else ?
References: <jo1vt8tg2t7ovt26o51mi2edd79vrqb72v@4ax.com>
Message-ID: <loom.20130712T090407-164@post.gmane.org>

Bazza <lamia <at> jeack.com.au> writes:

> Of itself, I don't mind that there's a 'dvd_nav_packet' 
> stream but it does seem to intefere with other 
> subsequent conversions.

Please post an example for such an "interference".

Carl Eugen


From jam at tigger.ws  Fri Jul 12 09:42:14 2013
From: jam at tigger.ws (James Linder)
Date: Fri, 12 Jul 2013 15:42:14 +0800
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <201307110649.r6B6nflk025866@haluter.fromme.com>
References: <201307110649.r6B6nflk025866@haluter.fromme.com>
Message-ID: <A5A2C9C7-0D67-4696-A4BA-02C930527CD7@tigger.ws>


On 11/07/2013, at 2:49 PM, Oliver Fromme wrote:

> James Linder wrote:
>> 
>> On 10/07/2013, at 10:19 PM, Nicolas George wrote:
>> 
>>> Le duodi 22 messidor, an CCXXI, James Linder a ?crit :
>>>> [dvr] /home/dvr [132]% ./ffmpeg -f pulse -i default -f v4l2 -i /dev/video0 -vtag divx -s 384x288 -r pal -q:v 5 -f tee  -map 0:a -map 1:v Wed.avi | cat >Wed1.avi
>>> 
>>> The tee muxer in this command line does not make any sense. The cat is
>>> completely useless too. 
>> [...]
>> Why is my cat completely useless?
> 
> Because the following two commands produce the same result:
> 
> $ ffmpeg ... | cat > foo.avi
> $ ffmpeg ... > foo.avi
> 
> The difference is that the first has an additional overhead
> of one execve() (probably insignificant in this case) and
> of one pipe and one process that have to handle the output
> data.  The latter might be significant because it usually
> involves at least two in-memory copy operations, depending
> on the implementations of pipes and cat in your operating
> system.
> 
> The above is a typical "useless use of cat".  It should be
> avoided.

Methinks that you do not understand what cat was being used for (to determine if system load was an issue using a |pipe|)

What is clear after building git head

Stuttering is no longer an issue
Audio sync is manageable (I will look at hacking the code to force same-sort timestamps) but this is really a case of using the latest.
Many thanks to Carl: the questions and comments were very helpful.
James

From stefasab at gmail.com  Fri Jul 12 09:53:49 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Fri, 12 Jul 2013 09:53:49 +0200
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <1373562258940-4659934.post@n4.nabble.com>
References: <loom.20130710T220231-979@post.gmane.org>
	<CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
	<20130710215235.GA20341@sunshine.barsnick.net>
	<1373493550748-4659913.post@n4.nabble.com>
	<CAEnqZEUWLhANnNYHRfC3ZoBzgdab-Amtb4Z+ZZPLTTJfiNNAKA@mail.gmail.com>
	<1373562258940-4659934.post@n4.nabble.com>
Message-ID: <20130712075349.GN4978@barisone>

On date Thursday 2013-07-11 10:04:18 -0700, gepebril wrote:
> Ok,
[...] 
> Create your video file with burned in subs
> ffmpeg.exe -y -i "src.mkv"  -f dvd -target ntsc-dvd -r 29.97 -vf
> scale=720:480 -vf "ass=subtitle_file.ass" -aspect 16:9 -b:v 8000k -g 12 -mbd
> rd -trellis 1 -flags +mv0 -cmp 0 -subcmp 2 e:\dst.mpg

This is still wrong, -vf foo -vf bar has the same effects as -vf bar,
what you want instead:
-vf scale=720:480,ass=subtitle_file.ass

Also why do you need to convert to ass?
 
> Note: make sure that the .ass file is in the same path as where are, as I
> haven't managed to get a path reference working with ass 

You need to specify an absolute path (and be aware of escaping problems).

[...]

From soho123.2012 at gmail.com  Fri Jul 12 10:23:34 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Fri, 12 Jul 2013 16:23:34 +0800
Subject: [FFmpeg-user] how to mux h264 video and pcm_s16le by ffserver
Message-ID: <CAFd2cZVzUNVuWomHJ0ATgr+D-ThMwjfnXfhjkpzyx6RyHysEoA@mail.gmail.com>

hi all,

I try to output a rtp stream that include both video and audio.
the config I use:

<Stream test1-rtsp.mpg>
Feed feed1.ffm
Format rtp
AVOptionVideo flags +global_header
VideoSize 1280x720
VideoFrameRate 30
VideoCodec libx264
AVOptionAudio flags +global_header
AudioCodec pcm_s16le
AudioChannels 2
AudioSampleRate 48000
</Stream>


I can NOT play by ffplay and VLC.
Does anyone have idea how to mux h264 video and pcm_s16le audio?

From soho123.2012 at gmail.com  Fri Jul 12 11:19:20 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Fri, 12 Jul 2013 17:19:20 +0800
Subject: [FFmpeg-user] output asf stream fail since asf enc fail
Message-ID: <CAFd2cZXcz+2dYUcx0Gi8f6xaq9PpT9LinaKR4B5iBDHQz6VTTQ@mail.gmail.com>

Hi all,

I try to output asf stream that include video and audio
config file of ffserver :
<Stream test.asf>
Feed feed1.ffm
Format asf

AVOptionVideo flags +global_header
VideoSize 1280x720
VideoFrameRate 30
VideoCodec libx264

AVOptionAudio flags +global_header
AudioCodec pcm_s16le
AudioChannels 2
AudioSampleRate 48000
</Stream>

I got the error :
ffserver: libavformat/asfenc.c: 642: flush_packet: Assertion
`asf->packet_timestamp_end >= asf->packet_tim
estamp_start' failed.



Does anyone have ideas?

From thilo.borgmann at googlemail.com  Fri Jul 12 13:39:34 2013
From: thilo.borgmann at googlemail.com (Thilo Borgmann)
Date: Fri, 12 Jul 2013 13:39:34 +0200
Subject: [FFmpeg-user] Reading EXIF from mjpeg
In-Reply-To: <CAPYw7P5e6Li9FTZgk4x9c_EDSMSsOMcj4sMYvdXDtEbRPSMOyg@mail.gmail.com>
References: <51DF0214.3010502@googlemail.com>
	<CAPYw7P5e6Li9FTZgk4x9c_EDSMSsOMcj4sMYvdXDtEbRPSMOyg@mail.gmail.com>
Message-ID: <51DFEAF6.6020308@googlemail.com>

Am 11.07.13 21:10, schrieb Paul B Mahol:
> On 7/11/13, Thilo Borgmann <thilo.borgmann at googlemail.com> wrote:
>> Hi,
>>
>> can we read EXIF metadata from motion jpeg streams?
>>
>> Any hint is welcome, using another lib is almost not ;)
> 
> Not at all, There is open bug report.

I can see Ticket #1926 if that's what you're referring to, thanks.

I don't know about such things but is this the official specification?
http://www.exif.org/Exif2-2.PDF

Is there any newer specification?

Thanks,
Thilo

From macdroid53 at gmail.com  Fri Jul 12 13:48:13 2013
From: macdroid53 at gmail.com (Mac)
Date: Fri, 12 Jul 2013 07:48:13 -0400
Subject: [FFmpeg-user] Reading EXIF from mjpeg
In-Reply-To: <51DFEAF6.6020308@googlemail.com>
References: <51DF0214.3010502@googlemail.com>
	<CAPYw7P5e6Li9FTZgk4x9c_EDSMSsOMcj4sMYvdXDtEbRPSMOyg@mail.gmail.com>
	<51DFEAF6.6020308@googlemail.com>
Message-ID: <CAHEqMkG9HMWnJtwJ7K=OBS8WLcT5rQmzr6H_bDsKqyDRDrmQeg@mail.gmail.com>

I use this command to turn a directory of images into a movie.
ffmpeg -f image2 -r 1/5 -i img%03d.png -c:v libx264 -r 30 out.mp4

What do need to add to add the file name (well actually all I need is
the number of the file) as a subtitle?


Regards,
Mac

From onemda at gmail.com  Fri Jul 12 13:50:49 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 12 Jul 2013 11:50:49 +0000
Subject: [FFmpeg-user] Reading EXIF from mjpeg
In-Reply-To: <CAHEqMkG9HMWnJtwJ7K=OBS8WLcT5rQmzr6H_bDsKqyDRDrmQeg@mail.gmail.com>
References: <51DF0214.3010502@googlemail.com>
	<CAPYw7P5e6Li9FTZgk4x9c_EDSMSsOMcj4sMYvdXDtEbRPSMOyg@mail.gmail.com>
	<51DFEAF6.6020308@googlemail.com>
	<CAHEqMkG9HMWnJtwJ7K=OBS8WLcT5rQmzr6H_bDsKqyDRDrmQeg@mail.gmail.com>
Message-ID: <CAPYw7P7D6=2gdjyryoYfgmCpDWd2_EBFJ+KwhpRAufY2mMndmg@mail.gmail.com>

On 7/12/13, Mac <macdroid53 at gmail.com> wrote:
> I use this command to turn a directory of images into a movie.
> ffmpeg -f image2 -r 1/5 -i img%03d.png -c:v libx264 -r 30 out.mp4
>
> What do need to add to add the file name (well actually all I need is
> the number of the file) as a subtitle?

Why are you hijacking threads?

From compuvision.sr at gmail.com  Fri Jul 12 13:50:52 2013
From: compuvision.sr at gmail.com (gepebril)
Date: Fri, 12 Jul 2013 04:50:52 -0700 (PDT)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <20130712075349.GN4978@barisone>
References: <CAECKL3rJzyJfNsVLFVkZTEBHgTk4epXY=H8p6k4=6tZ_kc-oGA@mail.gmail.com>
	<loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
	<20130710215235.GA20341@sunshine.barsnick.net>
	<1373493550748-4659913.post@n4.nabble.com>
	<CAEnqZEUWLhANnNYHRfC3ZoBzgdab-Amtb4Z+ZZPLTTJfiNNAKA@mail.gmail.com>
	<1373562258940-4659934.post@n4.nabble.com>
	<20130712075349.GN4978@barisone>
Message-ID: <1373629852904-4659951.post@n4.nabble.com>

@ Stefano,

>This is still wrong, -vf foo -vf bar has the same effects as -vf bar, 
Maybe in theory. I will try your suggestion today.

I didn't say that you have to convert to .ass to get burned in subtitles.
It's just that FFMpeg takes the layout of subs from the .ass file, it's not
present in .srt files.
Maybe there is a way to do it with .srt files (change fonts etc), but from
the let's say 50 examples of how to place hardcoded subs with ffmpeg, 48 are
done with .ass and of all .srt files examples none of the discusses
lay-out/positioning/fonts/size. So I assume it's not possible/not yet
implemented.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/make-mpeg2-file-with-hardcoded-subs-from-srt-file-tp4659902p4659951.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From stefasab at gmail.com  Fri Jul 12 15:16:32 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Fri, 12 Jul 2013 15:16:32 +0200
Subject: [FFmpeg-user] try to change complex filters on runtime
In-Reply-To: <DUB111-W71942C66995E0A11E9D04BB97A0@phx.gbl>
References: <DUB111-W71942C66995E0A11E9D04BB97A0@phx.gbl>
Message-ID: <20130712131632.GO4978@barisone>

On date Wednesday 2013-07-10 15:43:57 +0200, lucie Lemonnier wrote:
[...]
> For
>  example , I call reinit_filters_complex("[0:0] crop=960:1080:0:0 
> [tin];[2:0] crop=960:1080:0:0 [tmv]);[tin][tmv]overlay=0:0").
> I 
> passed in parameters of ffmpeg -i videoSource.mp4 -i transp.mov -i 
> objets.png -filter_complex "[0:0] crop=960:1080:0:0 [tin];[1:0] 
> crop=960:1080:0:0 [tmv]".
> But It crash on this change.

> I think it's because I free all in filtergraphs.
> But I don't know what is missing for it to work.
> Can you help me, please?
> Thanks.

You could achieve the same by extending libavfilter and using commands
to send interactive commands to filters. For example check my patch on
ffmpeg-devel to change crop parameters, as for what regards changing
overlay you could connect two movie sources and disable/enable the
active overlay (check timeline), or maybe implement a switcher
controlled by a command. Dynamically changing movie input is not
currently supported.

From compuvision.sr at gmail.com  Fri Jul 12 15:17:30 2013
From: compuvision.sr at gmail.com (gepebril)
Date: Fri, 12 Jul 2013 06:17:30 -0700 (PDT)
Subject: [FFmpeg-user] make mpeg2 file with hardcoded subs from .srt file
In-Reply-To: <1373629852904-4659951.post@n4.nabble.com>
References: <loom.20130710T224032-641@post.gmane.org>
	<CAECKL3pXDbJu=v3ecRA1fR8wTR7A1HkEtspW_y-qxmvZU5HMow@mail.gmail.com>
	<loom.20130710T231313-408@post.gmane.org>
	<CAECKL3qefcSpYe0WGb424gxScE5eZG2DK34tS6WuMLJhUm1gxg@mail.gmail.com>
	<20130710215235.GA20341@sunshine.barsnick.net>
	<1373493550748-4659913.post@n4.nabble.com>
	<CAEnqZEUWLhANnNYHRfC3ZoBzgdab-Amtb4Z+ZZPLTTJfiNNAKA@mail.gmail.com>
	<1373562258940-4659934.post@n4.nabble.com>
	<20130712075349.GN4978@barisone>
	<1373629852904-4659951.post@n4.nabble.com>
Message-ID: <1373635050495-4659953.post@n4.nabble.com>

@ Stefano,
I tested your command, and it works. That command makes more sense, so -vf
<option 1>,<option 2>,<option 3> Thanks for that part.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/make-mpeg2-file-with-hardcoded-subs-from-srt-file-tp4659902p4659953.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jshupert at pps-inc.com  Fri Jul 12 17:20:15 2013
From: jshupert at pps-inc.com (Jim Shupert)
Date: Fri, 12 Jul 2013 11:20:15 -0400
Subject: [FFmpeg-user] capture via decklink
Message-ID: <51E01EAF.6070406@pps-inc.com>

Friends,
Does anyone know if it is possible to capture by a Blackmagic card ?
a DeckLink Mini Recorder or DeckLink SDI?
I would like to capture SDI and then turn that into various files 
including a live stream maybe being served by ffserver.
I have been googling and trying things I see a bmdcapture that wishes 
one use the libav fork .
So - I wondering if 'true ffmpeg' can do the same or something simular.

i have a recent ffmpeg built on a ubuntu 12.04 64bit machine with 
DeckLink SDI and the app media express is all working well.
But media express only captures as uncompressed . Then i use ffmpeg to 
make my desired finished files.
I am exploring a better means to pipe and tee and do what i need.

have a great day!

jS

From expandafter at yahoo.com  Fri Jul 12 13:31:11 2013
From: expandafter at yahoo.com (S. Fisher)
Date: Fri, 12 Jul 2013 04:31:11 -0700 (PDT)
Subject: [FFmpeg-user] Convert PAL to NTSC
Message-ID: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>

Usually films that are shot at a rate of 24fps are converted to
PAL's 25fps by simply speeding them up; no frames are added.
This results in a shorter playtime and an increase in the pitch
of the sound.

So to convert to NTSC's 24fps (24000/1001), all we have to do
is slow everything down; we don't have to change the number of
frames.

Here's a method that works well for me.  On ffmpeg's command line
the following appears twice---once before the input file and once
before the output file:

-r ntsc-film

To slow down the audio, add this to the audio filters:

atempo=0.959040959040959


From francesco.damato7 at gmail.com  Fri Jul 12 18:27:21 2013
From: francesco.damato7 at gmail.com (Francesco Damato)
Date: Fri, 12 Jul 2013 18:27:21 +0200
Subject: [FFmpeg-user] compile error: ffmpeg
	av_get_default_channel_layout
In-Reply-To: <004f01cd85d9$b47de780$1d79b680$@com>
References: <004f01cd85d9$b47de780$1d79b680$@com>
Message-ID: <51E02E69.401@gmail.com>

Hi Sven,

i have the same problem. How do you resolve it???

Thanks!!

rancesco

From cehoyos at ag.or.at  Fri Jul 12 19:58:26 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 12 Jul 2013 17:58:26 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?compile_error=3A_ffmpeg_av=5Fget=5Fdefaul?=
	=?utf-8?q?t=5Fchannel=5Flayout?=
References: <004f01cd85d9$b47de780$1d79b680$@com> <51E02E69.401@gmail.com>
Message-ID: <loom.20130712T195653-29@post.gmane.org>

Francesco Damato <francesco.damato7 <at> gmail.com> writes:

> i have the same problem.

If you tell us how to reproduce your problem 
we may be able to help you.

If you want me to guess:
You have different FFmpeg version installed 
(that may not be a problem so far) and you 
don't tell your linker which version to use.

Carl Eugen


From cehoyos at ag.or.at  Fri Jul 12 20:00:37 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 12 Jul 2013 18:00:37 +0000 (UTC)
Subject: [FFmpeg-user] Convert PAL to NTSC
References: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>
Message-ID: <loom.20130712T195836-399@post.gmane.org>

S. Fisher <expandafter <at> yahoo.com> writes:

> On ffmpeg's command line the following appears 
> twice---once before the input file and once
> before the output file:
> 
> -r ntsc-film

I am not convinced that this is documented (and 
guaranteed to work), the intended way iirc is 
to use the setpts filter (as you do for audio) 
and use -r for the output file.

Note that in theory, movies can be telecined 
(24:1) to produce PAL output, in this case 
both variants cannot work.

Carl Eugen


From phil_rhodes at rocketmail.com  Sat Jul 13 02:34:40 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Fri, 12 Jul 2013 17:34:40 -0700 (PDT)
Subject: [FFmpeg-user] Convert PAL to NTSC
In-Reply-To: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>
References: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>
Message-ID: <1373675680.58156.YahooMailNeo@web121106.mail.ne1.yahoo.com>

>?This results in a shorter playtime and an increase in the pitch
>?of the sound.

The sound is invariably corrected these days; I'm not sure if ffmpeg has a suitable audio retiming engine.

P

From dave at dericed.com  Sat Jul 13 03:46:25 2013
From: dave at dericed.com (Dave Rice)
Date: Fri, 12 Jul 2013 21:46:25 -0400
Subject: [FFmpeg-user] capture via decklink
In-Reply-To: <51E01EAF.6070406@pps-inc.com>
References: <51E01EAF.6070406@pps-inc.com>
Message-ID: <1CAB7801-BBEF-404E-8803-3C598D31A98F@dericed.com>

Hi Jim,

On Jul 12, 2013, at 11:20 AM, Jim Shupert <jshupert at pps-inc.com> wrote:

> Friends,
> Does anyone know if it is possible to capture by a Blackmagic card ?
> a DeckLink Mini Recorder or DeckLink SDI?
> I would like to capture SDI and then turn that into various files including a live stream maybe being served by ffserver.
> I have been googling and trying things I see a bmdcapture that wishes one use the libav fork .
> So - I wondering if 'true ffmpeg' can do the same or something simular.

Officially bmdcapture only support libav (and possibly requires it for installation), but there's a recent ffmpeg-user thread that includes working bmdcapture -> ffmpeg examples. http://comments.gmane.org/gmane.comp.video.ffmpeg.user/46502

> i have a recent ffmpeg built on a ubuntu 12.04 64bit machine with DeckLink SDI and the app media express is all working well.
> But media express only captures as uncompressed . Then i use ffmpeg to make my desired finished files.
> I am exploring a better means to pipe and tee and do what i need.

I'm interested in this as well, if you're willing to share your work.
Dave Rice

From lamia at jeack.com.au  Sat Jul 13 03:54:25 2013
From: lamia at jeack.com.au (Bazza)
Date: Sat, 13 Jul 2013 11:54:25 +1000
Subject: [FFmpeg-user] This strea is something else ?
References: <jo1vt8tg2t7ovt26o51mi2edd79vrqb72v@4ax.com>
	<loom.20130712T090407-164@post.gmane.org>
Message-ID: <a4c1u812kb8fneq2a46nn86qls2qeqk3h5@4ax.com>

On Fri, 12 Jul 2013 07:04:46 +0000 (UTC), Carl Eugen Hoyos
<cehoyos at ag.or.at> wrote:

>Please post an example for such an "interference".
>Carl Eugen

I wish I could Carl. I just tried to systematically repeat and
document the steps but things  worked properly this time.
A lot of other things didn't but therein lies a different mystery.
Perhaps, just to keep things simple for the moment,
someone could fill me in on FFMPEG's last line message that I posted
and why it should have appeared    . . . .

> 
>    Stream #0:0[0x1bf]: Data: dvd_nav_packet     <*****************
>    Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576
>   [SAR 16:15 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50  tbc
>   Stream #0:2[0xa0]: Audio: pcm_s16be, 48000 Hz, 2 channels, s16,
>   1536 kb/s

> Unsupported codec with id 1145979222 for input stream 0  <***********

PS: Is it valid to post FFPROBE output here?


From cehoyos at ag.or.at  Sat Jul 13 04:44:28 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 13 Jul 2013 02:44:28 +0000 (UTC)
Subject: [FFmpeg-user] This strea is something else ?
References: <jo1vt8tg2t7ovt26o51mi2edd79vrqb72v@4ax.com>
	<loom.20130712T090407-164@post.gmane.org>
	<a4c1u812kb8fneq2a46nn86qls2qeqk3h5@4ax.com>
Message-ID: <loom.20130713T044014-224@post.gmane.org>

Bazza <lamia <at> jeack.com.au> writes:

> >    Stream #0:0[0x1bf]: Data: dvd_nav_packet

(Command line and complete, uncut console output missing.)
I am not sure I understand:
Do you think your stream does not contain any dvd_nav 
packets? Or would you just prefer if ffmpeg shows the 
streams that you like it to show but does not show the 
streams that you would prefer not to see?

Seriously:
Third party applications use these packets (or can 
use these packets if your prefer), ffmpeg does not 
use them. Does this answer your question?
(Note that ffmpeg reports if a dca stream is 
dts-hd-master but it cannot decode dts-hd-master - 
just to name another example.)

> PS: Is it valid to post FFPROBE output here?

If you want to report a problem with ffprobe that 
is not reproducible with ffmpeg or if you have a 
question regarding ffprobe that cannot be solved 
with using ffmpeg, you should probably post 
ffprobe output, yes.

Carl Eugen


From bbutscheidt at yahoo.de  Sat Jul 13 10:27:41 2013
From: bbutscheidt at yahoo.de (Bernd Butscheidt)
Date: Sat, 13 Jul 2013 09:27:41 +0100 (BST)
Subject: [FFmpeg-user] Convert PAL to NTSC
In-Reply-To: <1373675680.58156.YahooMailNeo@web121106.mail.ne1.yahoo.com>
References: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>
	<1373675680.58156.YahooMailNeo@web121106.mail.ne1.yahoo.com>
Message-ID: <1373704061.80604.YahooMailNeo@web133201.mail.ir2.yahoo.com>



----- Original Message -----
> From: Phil Rhodes <phil_rhodes at rocketmail.com>

> Sent: Saturday, July 13, 2013 2:34 AM
> Subject: Re: [FFmpeg-user] Convert PAL to NTSC

>? I'm not sure if ffmpeg has a suitable audio retiming engine.

In addition I happen to have TV recordings where the audio and video of the reencoding is still in sync but looses sync if one of the suggested methods (setpts or using the -r options twice) is applied.
So sometimes this works like charm, but as Carl Eugen already suggested, it is not guaranteed to work. 

But anyway it is an interesting suggestion to solve the task and there are cases where it succeeds. 

From cehoyos at ag.or.at  Sat Jul 13 11:29:03 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 13 Jul 2013 09:29:03 +0000 (UTC)
Subject: [FFmpeg-user] Convert PAL to NTSC
References: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>
	<1373675680.58156.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<1373704061.80604.YahooMailNeo@web133201.mail.ir2.yahoo.com>
Message-ID: <loom.20130713T112521-172@post.gmane.org>

Bernd Butscheidt <bbutscheidt <at> yahoo.de> writes:

> In addition I happen to have TV recordings where the 
> audio and video of the reencoding is still in sync 
> but looses sync if one of the suggested methods 
> (setpts or using the -r options twice) is applied.

My comment that something may no work only applied to 
using -r twice , -vf setpts is - to the best of my 
knowledge - supposed to work fine.

*Please* report such issues (sample + command line + 
complete, uncut console output)!
I don't know how obvious this is but issues that get 
not reported are unlikely to get fixed while we have 
a (very) long history of fixed reported issues.
(There is of course no guarantee that issues get 
fixed, see ticket #5.)

Carl Eugen


From nicolas.george at normalesup.org  Sat Jul 13 13:38:10 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sat, 13 Jul 2013 13:38:10 +0200
Subject: [FFmpeg-user] Comments about audio sync invited
In-Reply-To: <502E99C1-2AC6-4ED9-B290-261A4BFCA3DE@tigger.ws>
References: <51DB2C45.8040004@burner.com>
	<loom.20130708T232242-289@post.gmane.org>
	<DBE7C145-52A7-4DC2-9CCA-5A001C80A5D9@tigger.ws>
	<loom.20130709T094054-726@post.gmane.org>
	<70EF77B2-4254-417D-8D2B-4849B793A5D9@tigger.ws>
	<loom.20130709T135126-473@post.gmane.org>
	<3424D80A-6E67-4AC2-9359-0DEE7F47E4F4@tigger.ws>
	<20130710141944.GA8186@phare.normalesup.org>
	<502E99C1-2AC6-4ED9-B290-261A4BFCA3DE@tigger.ws>
Message-ID: <20130713113810.GC30102@phare.normalesup.org>

Le tridi 23 messidor, an CCXXI, James Linder a ?crit?:
> > The tee muxer in this command line does not make any sense. The cat is
> > completely useless too. 
> Please explain a little more.

http://ffmpeg.org/ffmpeg-formats.html#tee

Outputs of the tee muxer are separated by | in the output file name; your
command line has no | in the output file name.

> I want to output the pipe to ffplay or mplayer but after a while my
> display and my recorded file show pauses every 5 seconds or so of 100mS or
> so.

Looks like a sound device over- or underflow.

>     The cat command would isolate the display process from the mix.

There is nothing to isolate. With the command you posted, cat will only sit
in the way doing nothing.

Regards,

-- 
  Nicolas George
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From krueger at lesspain.de  Sat Jul 13 14:20:59 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Sat, 13 Jul 2013 14:20:59 +0200
Subject: [FFmpeg-user] Convert PAL to NTSC
In-Reply-To: <1373675680.58156.YahooMailNeo@web121106.mail.ne1.yahoo.com>
References: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>
	<1373675680.58156.YahooMailNeo@web121106.mail.ne1.yahoo.com>
Message-ID: <CAEnqZEUAP+j_Doory5KE4njsH86rh2Puqw1Rc93mXugsoyY+OA@mail.gmail.com>

On Sat, Jul 13, 2013 at 2:34 AM, Phil Rhodes <phil_rhodes at rocketmail.com> wrote:
>> This results in a shorter playtime and an increase in the pitch
>> of the sound.
>
> The sound is invariably corrected these days; I'm not sure if ffmpeg has a suitable audio retiming engine.

the filter atempo does that and AFAICT it does a decent job from
listening to the output, although I have not tested this thoroughly in
comparison with accepted professional speed correction software.

From bbutscheidt at yahoo.de  Sat Jul 13 15:32:40 2013
From: bbutscheidt at yahoo.de (Bernd Butscheidt)
Date: Sat, 13 Jul 2013 14:32:40 +0100 (BST)
Subject: [FFmpeg-user] Convert PAL to NTSC
In-Reply-To: <loom.20130713T112521-172@post.gmane.org>
References: <1373628671.78104.YahooMailBasic@web163802.mail.gq1.yahoo.com>
	<1373675680.58156.YahooMailNeo@web121106.mail.ne1.yahoo.com>
	<1373704061.80604.YahooMailNeo@web133201.mail.ir2.yahoo.com>
	<loom.20130713T112521-172@post.gmane.org>
Message-ID: <1373722360.85386.YahooMailNeo@web133205.mail.ir2.yahoo.com>

----- Original Message -----

> From: Carl Eugen Hoyos <cehoyos at ag.or.at>
> Sent: Saturday, July 13, 2013 11:29 AM

> My comment that something may no work only applied to 
> using -r twice , -vf setpts is - to the best of my 
> knowledge - supposed to work fine.

Thanks for the clarification and sorry if I referenced to your statement wrongly.


So I may try to sum up:

To convert Pal, e.g. 25fps, progressive source material to ntsc film, the initial post suggested this way:

ffmpeg -r 24000/1001 -i PAL.input.video -vcodec libx264 -preset ultrafast -crf 14 -acodec libmp3lame -ab 224k -filter:a atempo=0.959040959040959 -r 24000/1001 NTSC.output.video

The suggested documented alternative would be to use the setpts filter instead of the initial r-option:

ffmpeg -i PAL.input.video -filter:v setpts=25025/24000*PTS -vcodec libx264 -preset ultrafast -crf 14 -acodec libmp3lame -ab 224k -filter:a atempo=0.959040959040959 -f matroska -r 24000/1001 NTSC.output.video

Of course, codec options and input and output file names are only meant to be examples.

With two of my test files the first proposal works on one of them. The use of setpts in combination with atempo looks and sounds promising on both files.

So please don't get me wrong even if I may not find a proper way to express myself correctly here. What I was referring to in addition was the problem of TV-Sat recordings which, for reasons which seems to be still mysterious, turn out to be faulty input for ffmpeg from time to time. With files like this, I still ran into sync problems.
?

> *Please* report such issues (sample + command line + 
> complete, uncut console output)!


Promised, I will try - as I tried already:
https://lists.ffmpeg.org/pipermail/ffmpeg-user/2013-February/013219.html

But this may be an example too of how difficult it is to narrow down TV-recordings of GB-size to a small sample of a few MB which gets to a point.


From o_lucian at yahoo.com  Sat Jul 13 09:41:12 2013
From: o_lucian at yahoo.com (lucian orasanu)
Date: Sat, 13 Jul 2013 00:41:12 -0700 (PDT)
Subject: [FFmpeg-user] Artefacts when recording an rtsp stream
Message-ID: <1373701272.60990.YahooMailNeo@web160801.mail.bf1.yahoo.com>

Good day,

I'am trayng to record an stream from an hikvision DS=2CD2032-I, with command "ffmpeg -i rtsp://37.43.41.213/h264/ch1/main/av_stream test.avi"
but i see allot of errors like"


[h264 @ 0x163a380] error while decoding MB 47 39, bytestream (-5)
[h264 @ 0x163a380] Cannot use next picture in error concealment
[h264 @ 0x163a380] concealing 482 DC, 482 AC, 482 MV errors in P frame
RTP: missed 10 packets31.0 size= ? ?6055kB time=00:03:31.36 bitrate= 234.7kbits/s dup=0 drop=5 ? ?
[h264 @ 0x15aa9a0] RTP: missed 12 packets

and

[h264 @ 0x163a380] concealing 2136 DC, 2136 AC, 2136 MV errors in I frame
[h264 @ 0x15aa9a0] RTP: missed 4 packets
[h264 @ 0x163a380] error while decoding MB 63 17, bytestream (-11)


this is the start of recording:

ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
? built on Jul 12 2013 21:57:15 with gcc 4.6.3 (Gentoo 4.6.3 p1.11, 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='-march=core2 -O2 -pipe' --extra-cflags='-march=core2 -O2 -pipe' --extra-cxxflags='-march=core2 -O2 -pipe' --disable-static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --enable-version3 --enable-nonfree --disable-indev=alsa --disable-indev=oss --disable-indev=jack --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-version3 --enable-bzlib --enable-runtime-cpudetect --disable-debug --disable-doc --disable-gnutls --enable-hardcoded-tables --enable-iconv --enable-network --enable-openssl --disable-ffplay --disable-vaapi --disable-vdpau --enable-zlib --enable-libvo-aacenc --disable-libvo-amrwbenc --enable-libmp3lame --disable-libfdk-aac --disable-libaacplus
 --enable-libfaac --enable-libtheora --disable-libtwolame --disable- ?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
[h264 @ 0x200e9a0] RTP: missed 4 packets
[h264 @ 0x200e9a0] RTP: missed 9 packets
[h264 @ 0x200e9a0] RTP: missed 4 packets
[h264 @ 0x200e9a0] RTP: missed 2 packets
[h264 @ 0x200e9a0] RTP: missed 3 packets
? ? Last message repeated 1 times
[h264 @ 0x200e9a0] RTP: missed 5 packets
? ? Last message repeated 1 times
[h264 @ 0x200e9a0] RTP: missed 9 packets
[h264 @ 0x200e9a0] RTP: missed 8 packets
? ? Last message repeated 1 times
[h264 @ 0x200e9a0] RTP: missed 3 packets
[h264 @ 0x200e9a0] left block unavailable for requested intra4x4 mode -1 at 0 18
[h264 @ 0x200e9a0] error while decoding MB 0 18, bytestream (15159)
[h264 @ 0x200e9a0] concealing 2209 DC, 2209 AC, 2209 MV errors in I frame
[h264 @ 0x200e9a0] RTP: missed 6 packets
[h264 @ 0x200e9a0] RTP: missed 2 packets
[h264 @ 0x200e9a0] RTP: missed 10 packets
[h264 @ 0x200e9a0] RTP: missed 8 packets
[h264 @ 0x200e9a0] RTP: missed 9 packets
[h264 @ 0x200e9a0] RTP: missed 11 packets
[h264 @ 0x200e9a0] RTP: missed 2 packets
? ? Last message repeated 1 times
[h264 @ 0x200e9a0] RTP: missed 1 packets
[h264 @ 0x200e9a0] RTP: missed 9 packets
[h264 @ 0x200e9a0] RTP: missed 7 packets
[h264 @ 0x200e9a0] RTP: missed 6 packets
[h264 @ 0x200e9a0] concealing 2142 DC, 2142 AC, 2142 MV errors in I frame
[h264 @ 0x200e9a0] RTP: missed 1 packets
[h264 @ 0x200e9a0] RTP: missed 6 packets
[h264 @ 0x200e9a0] RTP: missed 40 packets
[h264 @ 0x200e9a0] RTP: missed 42 packets
[h264 @ 0x200e9a0] RTP: missed 79 packets
[h264 @ 0x200e9a0] RTP: missed 9 packets
[h264 @ 0x200e9a0] RTP: missed 8 packets
[h264 @ 0x200e9a0] RTP: missed 9 packets
[h264 @ 0x200e9a0] RTP: missed 6 packets
[h264 @ 0x200e9a0] RTP: missed 16 packets
[h264 @ 0x200e9a0] RTP: missed 39 packets
[h264 @ 0x200e9a0] RTP: missed 23 packets
[h264 @ 0x200e9a0] RTP: missed 5 packets
[h264 @ 0x200e9a0] RTP: missed 1 packets
Input #0, rtsp, from 'rtsp://37.43.41.213/h264/ch1/main/av_stream':
? Metadata:
? ? title ? ? ? ? ? : Media Presentation
? Duration: N/A, start: 0.520133, bitrate: N/A
? ? Stream #0:0: Video: h264 (Main), yuv420p, 1280x720, 25 fps, 25 tbr, 90k tbn, 50 tbc



And the image is very distorsioned, allot of bad mackrobloks. here is an sample
http://zm.realfield.ro/test.avi


My observation is that with ffmpeg 0.7 is working beter than 2.0

can this be fixed??
camera is online and who wants can make an test!

From francesco.damato7 at gmail.com  Sat Jul 13 18:42:50 2013
From: francesco.damato7 at gmail.com (Francesco)
Date: Sat, 13 Jul 2013 09:42:50 -0700 (PDT)
Subject: [FFmpeg-user] compile error: ffmpeg
	av_get_default_channel_layout
In-Reply-To: <loom.20130712T195653-29@post.gmane.org>
References: <004901cd85cb$7ed07b60$7c717220$@com>
	<004f01cd85d9$b47de780$1d79b680$@com> <51E02E69.401@gmail.com>
	<loom.20130712T195653-29@post.gmane.org>
Message-ID: <1373733770814-4659969.post@n4.nabble.com>

HI  Carl Eugen Hoyos,

i get this error: relocation error: /usr/local/lib/libavcodec.so.54: symbol
av_get_default_channel_layout, version LIBAVUTIL_51 not defined in file
libavutil.so.51 with link time reference.

Thanks for your help!!! I am a new user of Linux and ffmpeg library.





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From cehoyos at ag.or.at  Sat Jul 13 21:41:55 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 13 Jul 2013 19:41:55 +0000 (UTC)
Subject: [FFmpeg-user] Artefacts when recording an rtsp stream
References: <1373701272.60990.YahooMailNeo@web160801.mail.bf1.yahoo.com>
Message-ID: <loom.20130713T211854-682@post.gmane.org>

lucian orasanu <o_lucian <at> yahoo.com> writes:

> I'am trayng to record an stream from an hikvision 
> DS=2CD2032-I, with command 
> ffmpeg -i rtsp://37.43.41.213/h264/ch1/main/av_stream 

I tested the stream several times, either I get many 
distortions and many lines of "RTP: missed x packets" 
on the console, or no messages and perfect reception.

I am not sure if there is something that can be fixed 
in FFmpeg.

Carl Eugen


From cehoyos at ag.or.at  Sat Jul 13 21:43:46 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 13 Jul 2013 19:43:46 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?compile_error=3A_ffmpeg_av=5Fget=5Fdefaul?=
	=?utf-8?q?t=5Fchannel=5Flayout?=
References: <004901cd85cb$7ed07b60$7c717220$@com>
	<004f01cd85d9$b47de780$1d79b680$@com> <51E02E69.401@gmail.com>
	<loom.20130712T195653-29@post.gmane.org>
	<1373733770814-4659969.post@n4.nabble.com>
Message-ID: <loom.20130713T214215-887@post.gmane.org>

Francesco <francesco.damato7 <at> gmail.com> writes:

> i get this error: relocation error: 
> /usr/local/lib/libavcodec.so.54: symbol 
> av_get_default_channel_layout, version LIBAVUTIL_51 
> not defined in file libavutil.so.51 with link time 
> reference.

This is unfortunately still missing nearly all 
information that is needed to reproduce the problem.

As said, you have different versions of FFmpeg installed.
This is not necessarily a problem, but if you are a new 
user, I would recommend to use static compilation 
(remove --enable-shared from your configure line).

Carl Eugen


From mail at robertoragusa.it  Sat Jul 13 22:33:16 2013
From: mail at robertoragusa.it (Roberto Ragusa)
Date: Sat, 13 Jul 2013 22:33:16 +0200
Subject: [FFmpeg-user] How to skip parts during transcoding? (aka edit
	decision list)
Message-ID: <51E1B98C.5080800@robertoragusa.it>

Hi,

which is the best way to transcode a video and skip some intervals of it?
(identified by timestamps or frame number)

Mencoder has some kind of edl (edit decision list) support, how can the
same result be obtained with ffmpeg?

Thanks.

-- 
   Roberto Ragusa    mail at robertoragusa.it

From horsley1953 at gmail.com  Sun Jul 14 00:07:30 2013
From: horsley1953 at gmail.com (Tom Horsley)
Date: Sat, 13 Jul 2013 18:07:30 -0400
Subject: [FFmpeg-user] How to skip parts during transcoding? (aka edit
 decision list)
In-Reply-To: <51E1B98C.5080800@robertoragusa.it>
References: <51E1B98C.5080800@robertoragusa.it>
Message-ID: <20130713180730.21643ce3@zooty>

On Sat, 13 Jul 2013 22:33:16 +0200
Roberto Ragusa wrote:

> Mencoder has some kind of edl (edit decision list) support, how can the
> same result be obtained with ffmpeg?

I've been using the -ss and -t option *following* (that's the important
bit) the -i input option.

That gets me frame accurate seeks and encodes one bit of a video I
want to keep (kind of slow, but you can't have everything). Basically
I do this for all the segments of a recorded TV program that are
not commercials :-).

I then use the concat filter to join together the segments.

Slightly old versions of all my script can be found here:

http://home.comcast.net/~tomhorsley/game/ffmpeg.html

From mohanakannan.bts at gmail.com  Sat Jul 13 21:25:24 2013
From: mohanakannan.bts at gmail.com (mohan kannan)
Date: Sun, 14 Jul 2013 00:55:24 +0530
Subject: [FFmpeg-user] Error On installing FFMPEG-MASTER for opencv
Message-ID: <CAEin2Na7NB5G4YAosHNPc6idNqcFweLZmPGHpG1LHCCvwpvzrg@mail.gmail.com>

Dear Developer,
                         On configuring FFMPEG-MASTER,it is telling that
,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.
                           I've tried different method,even installed
newest FFMPEG zip....It is still showing this error..What to do.Please
solve this bug and please send me a solution.

From lou at lrcd.com  Sun Jul 14 00:22:34 2013
From: lou at lrcd.com (Lou)
Date: Sat, 13 Jul 2013 14:22:34 -0800
Subject: [FFmpeg-user] Error On installing FFMPEG-MASTER for opencv
In-Reply-To: <CAEin2Na7NB5G4YAosHNPc6idNqcFweLZmPGHpG1LHCCvwpvzrg@mail.gmail.com>
References: <CAEin2Na7NB5G4YAosHNPc6idNqcFweLZmPGHpG1LHCCvwpvzrg@mail.gmail.com>
Message-ID: <20130713142234.73090cc4@lrcd.com>

On Sun, 14 Jul 2013 00:55:24 +0530
mohan kannan <mohanakannan.bts at gmail.com> wrote:

> Dear Developer,
>                          On configuring FFMPEG-MASTER,it is telling that
> ,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.
>                            I've tried different method,even installed
> newest FFMPEG zip....It is still showing this error..What to do.Please
> solve this bug and please send me a solution.

You did not provide enough information to determine why ffmpeg is not
finding libx264. If you compiled x264 you should show the configure
options you used, the configure options you used for ffmpeg, and the
tail of config.log. The name of the OS or distro you are using would be
useful too.

With the limited information provided I can only guess that you need to
add "--extra-libs=-ldl" to your ffmpeg configure.

From lamia at jeack.com.au  Sun Jul 14 05:14:27 2013
From: lamia at jeack.com.au (Bazza)
Date: Sun, 14 Jul 2013 13:14:27 +1000
Subject: [FFmpeg-user] This strea is something else ?
References: <jo1vt8tg2t7ovt26o51mi2edd79vrqb72v@4ax.com>
	<loom.20130712T090407-164@post.gmane.org>
	<a4c1u812kb8fneq2a46nn86qls2qeqk3h5@4ax.com>
	<loom.20130713T044014-224@post.gmane.org>
Message-ID: <2v34u89ivf3bt80fkjdcd3c440cg76cuf5@4ax.com>

On Sat, 13 Jul 2013 02:44:28 +0000 (UTC), Carl Eugen Hoyos
<cehoyos at ag.or.at> wrote:

>Bazza <lamia <at> jeack.com.au> writes:
>
>> >    Stream #0:0[0x1bf]: Data: dvd_nav_packet
>
>I am not sure I understand:
>Do you think your stream does not contain any dvd_nav 
>packets? Or would you just prefer if ffmpeg shows the 
>streams that you like it to show but does not show the 
>streams that you would prefer not to see?

No, I'm happy enough about the recognition and display of the stream.
I'm questioning the "why" of FFMPEG classifying the (0x1bf) stream as
being stream 0 which is exactly what it was doing in my initial post.
I had always thought the term #0:0 -always- meant the video stream.
Would it be the case that FFMPEG concentrates its  attention on the
[0x1e0] ... type of description (that sounds muddled I know).

e.g. 

Here's the O/P from a dvd vob file using current FFMPEG (Win)
Note that #0:0 _IS_ a video etc and that the nav_packet is placed last
All well and good to my expectations and way of thinking

Input #0, mpeg, from 'e:\saturdays\kw.vob':
  Duration: 00:10:59.62, start: 3660.168000, bitrate: 6568 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[0x81]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s
    Stream #0:2[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
    Stream #0:3[0x1bf]: Data: dvd_nav_packet
---------------------------------------------------------------------------
But in contrast, my initial post shows a stream as below

Input #0, mpeg, from 'z:\video_ts\vts_01_1.vob':
  Duration: 00:09:10.19, start: 0.228544, bitrate: 5743 kb/s
    Stream #0:0[0x1bf]: Data: dvd_nav_packet     <<<< This bit here -)
    Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576
            [SAR 16:15 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0:2[0xa0]: Audio: pcm_s16be, 48000 Hz, 2 channels, s16,
1536 kb/s

The streams are what ... misnamed ...? This confuses me
especially if I'm a habit of re-mapping particular expected 
streams - only to find out they've been called something else.
So. I'm just seeking enlightenment.


>> PS: Is it valid to post FFPROBE output here?
>ffprobe output, yes.

Ok. I'm comfortable with that -) Just wanted to know.



From cehoyos at ag.or.at  Sun Jul 14 10:35:38 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 14 Jul 2013 08:35:38 +0000 (UTC)
Subject: [FFmpeg-user] This strea is something else ?
References: <jo1vt8tg2t7ovt26o51mi2edd79vrqb72v@4ax.com>
	<loom.20130712T090407-164@post.gmane.org>
	<a4c1u812kb8fneq2a46nn86qls2qeqk3h5@4ax.com>
	<loom.20130713T044014-224@post.gmane.org>
	<2v34u89ivf3bt80fkjdcd3c440cg76cuf5@4ax.com>
Message-ID: <loom.20130714T103503-717@post.gmane.org>

Bazza <lamia <at> jeack.com.au> writes:

> I had always thought the term #0:0 -always- meant 
> the video stream.

This is not correct, and to the best of my 
knowledge, it has never been.

Carl Eugen


From stefasab at gmail.com  Sun Jul 14 12:58:18 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Sun, 14 Jul 2013 12:58:18 +0200
Subject: [FFmpeg-user] Set start time offset in ts video segment output
 from mp4 movie clip input
In-Reply-To: <CAL22iBhJ8pRMGrrMzqKFdwLM83SyL5718C9pw9mC6zpJy6Bpsg@mail.gmail.com>
References: <CAL22iBhJ8pRMGrrMzqKFdwLM83SyL5718C9pw9mC6zpJy6Bpsg@mail.gmail.com>
Message-ID: <20130714105818.GA4978@barisone>

On date Friday 2013-07-05 19:47:53 -0400, Peter Chen wrote:
> I'm converting an mp4 movie clip into a ts video segment using the
> following command:
> 
>    > ffmpeg -i input-segment-4.mp4 -vcodec copy -acodec copy -qscale 0
> -vbsf h264_mp4toannexb -y -r 24 -vf transpose=1 -f
> mpegts output-segment-4.ts
> 
> This ts video segment will be part of an HLS (HTTP Live Streaming) video
> stream.
> 
> The ffmpeg conversion works great.
> 
> I might not be asking the following question using the proper terminology,
> but here's what I'd like to do. I'd like the output file, called
> output-segment-4.ts in the ffmpeg command above, to have a start time
> offset of, say, 40 seconds. So, in an HLS video stream that contains these
> output ts segments, output-segment-0.ts would have a start time of 0
> seconds, output-segment-1.ts would have a start time of 10 seconds, etc...
> 
> Currently, if I'm reading the output of ffprobe correctly, every ts file
> shows a start time offset of about 1.4 seconds. Below is the output of
> "ffprobe output-segment-4.ts". I'm basically trying to
> manipulate output-segment-4.ts so "start: 1.421678" becomes "start: 40.0"
> in the line "Duration: 00:00:09.95, start: 1.421678, bitrate: 1718
> kb/s".
> 
> In summary, how do I set that start field?

You might want to look at the segment muxer and the options
reset_timestamps and initial_offset, which was added very recently.
-- 
FFmpeg = Fabulous and Faithful Mere Puritan Elastic Game

From rohit.tvm.86 at gmail.com  Mon Jul 15 08:44:04 2013
From: rohit.tvm.86 at gmail.com (rohit a)
Date: Mon, 15 Jul 2013 12:14:04 +0530
Subject: [FFmpeg-user] Sampling frequency conversion
Message-ID: <CAPJ_JDxaqnEefhy72rvMPmRXkkSog8uKZVUpnHezWtccEwSDcA@mail.gmail.com>

Hi,

I tried to perform sampling frequency conversion in ffmpeg version 1.2
using -ar command.  But the output file had the same sampling frequency as
the input (32000 Hz).   The command given is as below
./ffmpeg -i  input.mpeg  output.avi -ar 24000

Please inform me the correct procedure for sampling frequency conversion.

Regards,
Rohit

From soho123.2012 at gmail.com  Mon Jul 15 11:29:23 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 17:29:23 +0800
Subject: [FFmpeg-user] mux flv problem
Message-ID: <CAFd2cZU=Ym0=mkY-sWyEaQpzEhfuFLfd2EVxhEkJWd_V25axag@mail.gmail.com>

Hi ,

When I try to mux video +audio to flv format with ffserver,
it says :FLV does not support sample rate 48000,choose from (44100, 22050,
11025)

why flv can not use 48000 sample for audio?

Does someone have ideas?

From doenerbudenchef at yahoo.de  Mon Jul 15 11:27:12 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Mon, 15 Jul 2013 02:27:12 -0700 (PDT)
Subject: [FFmpeg-user] Increase volume while using amerge
In-Reply-To: <1373545871656-4659926.post@n4.nabble.com>
References: <1373542433407-4659924.post@n4.nabble.com>
	<loom.20130711T140730-496@post.gmane.org>
	<1373545871656-4659926.post@n4.nabble.com>
Message-ID: <1373880432667-4659981.post@n4.nabble.com>

Hi all,

is there a way to unshorten the command? 

Thanks,
Alex



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Sent from the FFmpeg-users mailing list archive at Nabble.com.

From onemda at gmail.com  Mon Jul 15 12:24:33 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 15 Jul 2013 10:24:33 +0000
Subject: [FFmpeg-user] Increase volume while using amerge
In-Reply-To: <1373880432667-4659981.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>
Message-ID: <CAPYw7P70EQGdMVcoBswSYABTin=VAnepLJF33fTnvDuNMaF7=A@mail.gmail.com>

On 7/15/13, Alex <doenerbudenchef at yahoo.de> wrote:
> Hi all,
>
> is there a way to unshorten the command?

Unshorten what exactly?

>
> Thanks,
> Alex
>
>
>
> --
> View this message in context:
> http://ffmpeg-users.933282.n4.nabble.com/Increase-volume-while-using-amerge-tp4659924p4659981.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 doenerbudenchef at yahoo.de  Mon Jul 15 12:31:42 2013
From: doenerbudenchef at yahoo.de (Alex)
Date: Mon, 15 Jul 2013 03:31:42 -0700 (PDT)
Subject: [FFmpeg-user] Increase volume while using amerge
In-Reply-To: <CAPYw7P70EQGdMVcoBswSYABTin=VAnepLJF33fTnvDuNMaF7=A@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>
Message-ID: <1373884302257-4659984.post@n4.nabble.com>

I want to have these part...

-filter_complex amerge,volume=15.8dB 

...splitted in two seperate parameters as I wrote in my first post:

 -filter_complex amerge -af volume=15.8dB

But this doesn't work! 



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From satyagowtham.k at gmail.com  Mon Jul 15 12:58:10 2013
From: satyagowtham.k at gmail.com (satya gowtham kudupudi)
Date: Mon, 15 Jul 2013 16:28:10 +0530
Subject: [FFmpeg-user] Cortex-A8 neon gpu optimiazation
Message-ID: <CADY_UEdt+VUozw7dM+mYQC5Moe0ojR89WXHuSk3TUAh=rJDFUQ@mail.gmail.com>

on beaglebone black ubuntu i'm doing

ffmpeg -f video4linux2 -vcodec mjpeg -r 15 -s 320x240 -i test.flv

its taking 60% of cpu. can i take support of gpu and optimize? I tried
doing some thing like

ffmpeg -cpuflags ARM neon -f video4linux2 -vcodec mjpeg -r 15 -s 320x240 -i
test.flv

i donno exact syntax, i haven't understood -cpuflags options in ffmpeg man
page.
please suggest me some optimization

Thanks!
*Gowtham*

From satyagowtham.k at gmail.com  Mon Jul 15 13:08:04 2013
From: satyagowtham.k at gmail.com (satya gowtham kudupudi)
Date: Mon, 15 Jul 2013 16:38:04 +0530
Subject: [FFmpeg-user] Cortex-A8 neon gpu optimiazation
In-Reply-To: <CADY_UEdt+VUozw7dM+mYQC5Moe0ojR89WXHuSk3TUAh=rJDFUQ@mail.gmail.com>
References: <CADY_UEdt+VUozw7dM+mYQC5Moe0ojR89WXHuSk3TUAh=rJDFUQ@mail.gmail.com>
Message-ID: <CADY_UEemut4BVdV4Jv3h9z+=5FxvctAMPTSHZFKmJGNjm5q9Cg@mail.gmail.com>

ffmpeg -f video4linux2 -vcodec mjpeg -r 15 -s 320x240 -i /dev/video0
test.flv

correction: i forgot */dev/video0* input option


On Mon, Jul 15, 2013 at 4:28 PM, satya gowtham kudupudi <
satyagowtham.k at gmail.com> wrote:

> on beaglebone black ubuntu i'm doing
>
> ffmpeg -f video4linux2 -vcodec mjpeg -r 15 -s 320x240 -i test.flv
>
> its taking 60% of cpu. can i take support of gpu and optimize? I tried
> doing some thing like
>
> ffmpeg -cpuflags ARM neon -f video4linux2 -vcodec mjpeg -r 15 -s 320x240
> -i test.flv
>
> i donno exact syntax, i haven't understood -cpuflags options in ffmpeg man
> page.
> please suggest me some optimization
>
> Thanks!
> *Gowtham*
>



-- 
*- Gowtham*

From onemda at gmail.com  Mon Jul 15 13:18:57 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 15 Jul 2013 11:18:57 +0000
Subject: [FFmpeg-user] Increase volume while using amerge
In-Reply-To: <1373884302257-4659984.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>
Message-ID: <CAPYw7P7Z9GLwCOkNuGkqegLavAkv3MmsEGc4Vs+T+pM7OA0usQ@mail.gmail.com>

On 7/15/13, Alex <doenerbudenchef at yahoo.de> wrote:
> I want to have these part...
>
> -filter_complex amerge,volume=15.8dB
>
> ...splitted in two seperate parameters as I wrote in my first post:
>
>  -filter_complex amerge -af volume=15.8dB
>
> But this doesn't work!

For good reason. You can't use both of them at same time.

From cehoyos at ag.or.at  Mon Jul 15 13:21:07 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 11:21:07 +0000 (UTC)
Subject: [FFmpeg-user] mux flv problem
References: <CAFd2cZU=Ym0=mkY-sWyEaQpzEhfuFLfd2EVxhEkJWd_V25axag@mail.gmail.com>
Message-ID: <loom.20130715T132033-890@post.gmane.org>

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

> When I try to mux video +audio to flv format with ffserver,

(How is this ffserver-related?)

> it says :FLV does not support sample rate 48000,choose 
> from (44100, 22050, 11025)

> why flv can not use 48000 sample for audio?

Wrong mailing list?

Carl Eugen


From cehoyos at ag.or.at  Mon Jul 15 13:22:59 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 11:22:59 +0000 (UTC)
Subject: [FFmpeg-user] Cortex-A8 neon gpu optimiazation
References: <CADY_UEdt+VUozw7dM+mYQC5Moe0ojR89WXHuSk3TUAh=rJDFUQ@mail.gmail.com>
Message-ID: <loom.20130715T132123-604@post.gmane.org>

satya gowtham kudupudi <satyagowtham.k <at> gmail.com> writes:

> on beaglebone black ubuntu i'm doing
> 
> ffmpeg -f video4linux2 -vcodec mjpeg -r 15 -s 320x240 -i test.flv
> 
> its taking 60% of cpu. can i take support of gpu and optimize?

Patch welcome.

> I tried doing some thing like
> 
> ffmpeg -cpuflags ARM neon

cpuflags is a debugging option, I cannot imagine a use-case atm 
that helps users.

Carl Eugen


From cehoyos at ag.or.at  Mon Jul 15 13:27:29 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 11:27:29 +0000 (UTC)
Subject: [FFmpeg-user] Sampling frequency conversion
References: <CAPJ_JDxaqnEefhy72rvMPmRXkkSog8uKZVUpnHezWtccEwSDcA@mail.gmail.com>
Message-ID: <loom.20130715T132316-690@post.gmane.org>

rohit a <rohit.tvm.86 <at> gmail.com> writes:

> ./ffmpeg -i  input.mpeg  output.avi -ar 24000

(Complete, uncut console output missing, note that 
ffmpeg prints a warning here explaining that 
something is wrong with the command line.)

The general ffmpeg syntax is:
$ ffmpeg [input file 1 options] -i input_file-1 
[[input file x options] -i input_file_x]
[output file 1 options] output_file_1
[[output file x options] output_file_x]

Your command line asks ffmeg to change the sample 
rate for the second output file to 24kHz but you 
did not define a second output file.

Carl Eugen


From satyagowtham.k at gmail.com  Mon Jul 15 13:39:52 2013
From: satyagowtham.k at gmail.com (satya gowtham kudupudi)
Date: Mon, 15 Jul 2013 17:09:52 +0530
Subject: [FFmpeg-user] Cortex-A8 neon gpu optimiazation
In-Reply-To: <loom.20130715T132123-604@post.gmane.org>
References: <CADY_UEdt+VUozw7dM+mYQC5Moe0ojR89WXHuSk3TUAh=rJDFUQ@mail.gmail.com>
	<loom.20130715T132123-604@post.gmane.org>
Message-ID: <CADY_UEeAiCm_w2x4BPRKadeA3to4jbe6TiMAt+1JZR_YHenQ0Q@mail.gmail.com>

so cant i optimize anything further?


On Mon, Jul 15, 2013 at 4:52 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> satya gowtham kudupudi <satyagowtham.k <at> gmail.com> writes:
>
> > on beaglebone black ubuntu i'm doing
> >
> > ffmpeg -f video4linux2 -vcodec mjpeg -r 15 -s 320x240 -i test.flv
> >
> > its taking 60% of cpu. can i take support of gpu and optimize?
>
> Patch welcome.
>
> > I tried doing some thing like
> >
> > ffmpeg -cpuflags ARM neon
>
> cpuflags is a debugging option, I cannot imagine a use-case atm
> that helps users.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>



-- 
*- Gowtham*

From soho123.2012 at gmail.com  Mon Jul 15 13:51:27 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 19:51:27 +0800
Subject: [FFmpeg-user] mux flv problem
In-Reply-To: <loom.20130715T132033-890@post.gmane.org>
References: <CAFd2cZU=Ym0=mkY-sWyEaQpzEhfuFLfd2EVxhEkJWd_V25axag@mail.gmail.com>
	<loom.20130715T132033-890@post.gmane.org>
Message-ID: <CAFd2cZXTmpwq3=0wxKQJmaHdwF0YH85nF0jFTq_J5EXW9baKow@mail.gmail.com>

Hi All,

I use ffmpeg + ffserver to do a stream server.
the stream includes video + audio.
the config for ffserver :
<Stream test.flv>
Feed feed1.ffm
Format flv

AVOptionVideo flags +global_header
VideoSize 1280x720
VideoFrameRate 12
VideoCodec libx264
AVOptionAudio flags +global_header
AudioCodec pcm_s16le
AudioChannels 2
AudioSampleRate 48000
</Stream>
the command line of ffmpeg:
ffmpeg -loglevel debug -f alsa -ar 48000 -ac 2 -i hw:0 -f video4linux2 -r
30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy -acodec copy
http://localhost:8090/feed1.ffm


then I got the error message:
FLV does not support sample rate 48000,choose from (44100, 22050, 11025)
it is output from :libavformat/flvenc.c, that used when ffserver do mux.


Hence, my question is ffserver related.
Does someone have idea about why flv can not support 48000 sample rate in
audio

From rohit.tvm.86 at gmail.com  Mon Jul 15 14:01:00 2013
From: rohit.tvm.86 at gmail.com (rohit a)
Date: Mon, 15 Jul 2013 17:31:00 +0530
Subject: [FFmpeg-user] Sampling frequency conversion
In-Reply-To: <loom.20130715T132316-690@post.gmane.org>
References: <CAPJ_JDxaqnEefhy72rvMPmRXkkSog8uKZVUpnHezWtccEwSDcA@mail.gmail.com>
	<loom.20130715T132316-690@post.gmane.org>
Message-ID: <CAPJ_JDw-UQ0Dz2fM0vwxrr4ZK+cmb-dSEz08rtQqOjfDBnEQ8A@mail.gmail.com>

Hi Carl,

I used an input of 32000 Hz. I wanted to convert the output to 24000 Hz.

So I used the below command.
./ffmpeg  -i   Input.mpeg  -ar  24000  Output.avi

But still I am getting the output sampling frequency as 32000 Hz.  So can
you  please provide me the correct command line for converting 32000 Hz
to  24000 Hz.

Regards,
Rohit
On 15 Jul 2013 17:00, "Carl Eugen Hoyos" <cehoyos at ag.or.at> wrote:

> rohit a <rohit.tvm.86 <at> gmail.com> writes:
>
> > ./ffmpeg -i  input.mpeg  output.avi -ar 24000
>
> (Complete, uncut console output missing, note that
> ffmpeg prints a warning here explaining that
> something is wrong with the command line.)
>
> The general ffmpeg syntax is:
> $ ffmpeg [input file 1 options] -i input_file-1
> [[input file x options] -i input_file_x]
> [output file 1 options] output_file_1
> [[output file x options] output_file_x]
>
> Your command line asks ffmeg to change the sample
> rate for the second output file to 24kHz but you
> did not define a second output file.
>
> 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 Jul 15 14:27:40 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 20:27:40 +0800
Subject: [FFmpeg-user] ffplay can NOT play 48000 sample rate flv stream
Message-ID: <CAFd2cZVQLPkqFboamoVSP3G6DFhRiuK6X9-AWD9oVgaG_uKjuQ@mail.gmail.com>

Hi All,

when I try to play the video + audio  stream by ffplay,
even I modify the sample rate to 48000 in the src , ffplay just can play
flv in 44100 sample rate.
why ffplay can NOT play 48000 sample rate audio stream?

From rohit.tvm.86 at gmail.com  Mon Jul 15 14:35:39 2013
From: rohit.tvm.86 at gmail.com (rohit a)
Date: Mon, 15 Jul 2013 18:05:39 +0530
Subject: [FFmpeg-user] mux flv problem
In-Reply-To: <CAFd2cZU=Ym0=mkY-sWyEaQpzEhfuFLfd2EVxhEkJWd_V25axag@mail.gmail.com>
References: <CAFd2cZU=Ym0=mkY-sWyEaQpzEhfuFLfd2EVxhEkJWd_V25axag@mail.gmail.com>
Message-ID: <CAPJ_JDwgJSsP7z0tsfMBVmkX-GcgcReaQBDFguE486S_x+DNhA@mail.gmail.com>

Hi,

As per my understanding FLV format supports sampling frequency of 11, 22
and 44 KHz only.   All other sampling frequencies are considered as
invalid.

Regards,
Rohit
On 15 Jul 2013 15:00, "Soho Soho123" <soho123.2012 at gmail.com> wrote:

> Hi ,
>
> When I try to mux video +audio to flv format with ffserver,
> it says :FLV does not support sample rate 48000,choose from (44100, 22050,
> 11025)
>
> why flv can not use 48000 sample for audio?
>
> Does someone have ideas?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From jam at tigger.ws  Mon Jul 15 14:44:48 2013
From: jam at tigger.ws (James Linder)
Date: Mon, 15 Jul 2013 20:44:48 +0800
Subject: [FFmpeg-user] Help Please
Message-ID: <A5C0165C-F3C2-4F90-9045-9AD34D7326AC@tigger.ws>

If any kind soul can show me the error of my ways I'd appreciate it lots:

I want to record audio and video to a file
I want to display the video as it is being recorded.

This has worked with different v4l hardware :-(

I record:
/usr/bin/ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 -i /dev/video0 -vtag divx -s 720x576 -r pal -q:v 5 mon.avi -s 720x576 -f rawvideo - |/usr/bin/ffplay -f rawvideo -i - -s 720x576

The rawvideo size has no effect.

I get:
ffplay version N-54580-g694550f Copyright (c) 2003-2013 the FFmpeg developers
  built on Jul 10 2013 11:43:58 with gcc 4.7 (SUSE Linux)
  configuration: --enable-libv4l2 --enable-libpulse
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 11.101 / 55. 11.101
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 79.100 /  3. 79.100
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
Option -s is deprecated, use -video_size.
ffmpeg version N-54580-g694550f Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 10 2013 11:43:58 with gcc 4.7 (SUSE Linux)
  configuration: --enable-libv4l2 --enable-libpulse
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 11.101 / 55. 11.101
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 79.100 /  3. 79.100
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, pulse, from 'default':
  Duration: N/A, start: 0.010790, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Input #1, video4linux2,v4l2, from '/dev/video0':B sq=    0B f=0/0   
  Duration: N/A, start: 1373874168.744534, bitrate: 165888 kb/s
    Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x576, 165888 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
File 'mon.avi' already exists. Overwrite ? [y/N]     nan    :  0.000 fd=   0 aq=    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0    
Output #0, avi, to 'mon.avi':
  Metadata:
    ISFT            : Lavf55.11.101
    Stream #0:0: Video: mpeg4 (divx / 0x78766964), yuv420p, 720x576, q=2-31, 200 kb/s, 25 tbn, 25 tbc
    Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp, 192 kb/s
Output #1, rawvideo, to 'pipe:':
  Metadata:
    encoder         : Lavf55.11.101
    Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 800x600, q=2-31, 200 kb/s, 90k tbn, 25 tbc
Stream mapping:
  Stream #1:0 -> #0:0 (rawvideo -> mpeg4)
  Stream #0:0 -> #0:1 (pcm_s16le -> ac3)
  Stream #1:0 -> #1:0 (rawvideo -> rawvideo)
Press [q] to stop, [?] for help
Input #0, rawvideo, from 'pipe:':
  Duration: N/A, start: 0.000000, bitrate: 144000 kb/s
    Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 800x600, 144000 kb/s, 25 tbr, 25 tbn, 25 tbc
frame=   16 fps=0.0 q=5.0 q=0.0 size=     155kB time=00:00:00.64 bitrate=1981.5k  10.36 M-V: -0.002 fd=   0 aq=    0KB vq= 4219KB sq=    0B f=0/0

My live picture is garbled/all one colour/lines
The recorded file is perfect

I get even more confused, I try to use tee

/usr/bin/ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 -i /dev/video0 -vtag divx -s 720x576 -r pal -q:v 5 -f tee -map a:0 -map v:0 -c copy mon.avi |mplayer -

and get:
ffmpeg version N-54580-g694550f Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 10 2013 11:43:58 with gcc 4.7 (SUSE Linux)
  configuration: --enable-libv4l2 --enable-libpulse
  libavutil      52. 38.100 / 52. 38.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 11.101 / 55. 11.101
  libavdevice    55.  2.100 / 55.  2.100
  libavfilter     3. 79.100 /  3. 79.100
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102
MPlayer dev-SVN-r35127-4.7-openSUSE Linux 12.3 (x86_64)-Packman (C) 2000-2012 MPlayer Team
Can't open joystick device /dev/input/js0: No such file or directory
Can't init input joystick
mplayer: could not connect to socket
mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.

Playing -.
Reading from stdin...
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, pulse, from 'default':
  Duration: N/A, start: 0.009793, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Input #1, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 1373877364.072402, bitrate: 165888 kb/s
    Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x576, 165888 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
Stream map 'v:0' matches no streams.
libavformat version 54.25.104 (internal)

I don't understand why tee does not do what I expect
I don't understand why rawvideo from one v4l device works fine, another does not

Thanks
James


From cehoyos at ag.or.at  Mon Jul 15 15:15:05 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 13:15:05 +0000 (UTC)
Subject: [FFmpeg-user] Help Please
References: <A5C0165C-F3C2-4F90-9045-9AD34D7326AC@tigger.ws>
Message-ID: <loom.20130715T151256-915@post.gmane.org>

James Linder <jam <at> tigger.ws> writes:

> Output #1, rawvideo, to 'pipe:':
>   Metadata:
>     encoder         : Lavf55.11.101
>     Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 
> 800x600, q=2-31, 200 kb/s, 90k tbn, 25 tbc
> Stream mapping:
>   Stream #1:0 -> #0:0 (rawvideo -> mpeg4)
>   Stream #0:0 -> #0:1 (pcm_s16le -> ac3)
>   Stream #1:0 -> #1:0 (rawvideo -> rawvideo)
> Press [q] to stop, [?] for help
> Input #0, rawvideo, from 'pipe:':
>   Duration: N/A, start: 0.000000, bitrate: 144000 kb/s
>     Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p

(Sorry if you disagree or if I miss something but don't 
you agree that above is nearly unreadable and the 
problem can be reproduced using a file instead of a 
pipe?)

You don't tell ffplay that your "input" is yuyv422 and the 
default for rawvideo is yuv420p, use -pix_fmt yuyv422

Carl Eugen


From soho123.2012 at gmail.com  Mon Jul 15 15:15:57 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 21:15:57 +0800
Subject: [FFmpeg-user] Help please !!!!! asf stream abort by ffserver
Message-ID: <CAFd2cZWfpYHyjBiV8AUpT+nzf-y1in3a-X9XaLP0DpydsfSd6Q@mail.gmail.com>

hi all,


When I try to do mux h264 video + pcm_s16le audio, ffserver abort since the
error:
Mon Jul 15 21:07:40 2013 Assertion asf->packet_timestamp_end >=
asf->packet_timestamp_start failed at l
ibavformat/asfenc.c:643

Does anyone have ideas?

==========================================
ffserver.conf:
<Stream test.asf>
Feed feed1.ffm
Format asf

AVOptionVideo flags +global_header
VideoSize 640x480
VideoFrameRate 30
VideoCodec libx264

AVOptionAudio flags +global_header
AudioCodec pcm_s16le
AudioChannels 2
AudioSampleRate 44100
</Stream>

ffmpeg command :
ffmpeg -f alsa -ar 44100 -ac 2 -i hw:0 -f video4linux2 -r 30 -s 640x480
-input_format h264 -i /dev/video1 -vcodec copy -acodec copy
http://localhost:8090/feed1.ffm
========================================================================

when I try to play by ffplay,
the log from ffplay :
Input #0, asf, from 'mmsh://192.168.1.254:8090/test.asf':0B f=0/0
  Metadata:
    encoder         : Lavf55.12.100
  Duration: N/A, start: 3827889.346000, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2
channels,
 s16, 1411 kb/s
    Stream #0:1: Video: h264 (Constrained Baseline) (H264 / 0x34363248),
yuv420p
, 640x480, 100 tbr, 1k tbn, 2k tbc
Read data packet header failed!aq=    0KB vq=   21KB sq=    0B f=0/0
Recv other type packet -5
Read data packet header failed!
Recv other type packet -5
Read data packet header failed!
Recv other type packet -5
Read data packet header failed!
Recv other type packet -5
Read data packet header failed!
Recv other type packet -5
3827898.56 A-V:  0.068 fd=  44 aq=    0KB vq=    0KB sq=    0B f=0/0

Does any one have ides!!

From cehoyos at ag.or.at  Mon Jul 15 15:19:16 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 13:19:16 +0000 (UTC)
Subject: [FFmpeg-user] mux flv problem
References: <CAFd2cZU=Ym0=mkY-sWyEaQpzEhfuFLfd2EVxhEkJWd_V25axag@mail.gmail.com>
	<loom.20130715T132033-890@post.gmane.org>
	<CAFd2cZXTmpwq3=0wxKQJmaHdwF0YH85nF0jFTq_J5EXW9baKow@mail.gmail.com>
Message-ID: <loom.20130715T151533-655@post.gmane.org>

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

> FLV does not support sample rate 48000,choose from 
> (44100, 22050, 11025) it is output from: 
> libavformat/flvenc.c, that used when ffserver do mux.
> 
> Hence, my question is ffserver related.

So you cannot reproduce the problem with ffmpeg?

You may have already noticed that ffserver is not 
very well maintained so reporting problems against 
ffserver that are reproducible with ffmpeg is 
often not a good idea.

(This is of course completely irrelevant for this 
specific question: I don't think anybody on this 
mailing list knows why Adobe - or Macromedia? - 
chose to support only a limited set of bitrates 
in flv.)

Carl Eugen


From cehoyos at ag.or.at  Mon Jul 15 15:21:32 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 13:21:32 +0000 (UTC)
Subject: [FFmpeg-user] Help please !!!!! asf stream abort by ffserver
References: <CAFd2cZWfpYHyjBiV8AUpT+nzf-y1in3a-X9XaLP0DpydsfSd6Q@mail.gmail.com>
Message-ID: <loom.20130715T151945-716@post.gmane.org>

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

> When I try to do mux h264 video + pcm_s16le audio, 

(Are you sure pcm_s16le is supported in asf?)

> ffserver abort since the error:
> Mon Jul 15 21:07:40 2013 Assertion 
> asf->packet_timestamp_end >= asf->packet_timestamp_start 
> failed at libavformat/asfenc.c:643

Complete, uncut console output missing.

Carl Eugen


From soho123.2012 at gmail.com  Mon Jul 15 15:28:24 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 21:28:24 +0800
Subject: [FFmpeg-user] rtp problem Help please! please!
Message-ID: <CAFd2cZW5q3qqF-4Cspx3uAUx5-UXWx9aeO3=vKvr4pPhvVp1jg@mail.gmail.com>

Hi All,

When I try to send out the stream (video + audio) by rtp. then using ffplay
to play the stream.
why ffplay can play s16be only? Why ffplay can not play s16le audio?
I can see the log from ffplay ,
it says:a=rtpmap:96 L16/44100/2, and [rtsp @ 02a64560] audio codec set to:
pcm_s16be,
why?

=========================================================================
and in advance,
my stream server :ffserver+ffmpeg send out the rtp data to the port that
rtsp message described.
but ffplay can not display the video data and output the audio data ?

Does any one have ideas?

From cehoyos at ag.or.at  Mon Jul 15 15:22:33 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 13:22:33 +0000 (UTC)
Subject: [FFmpeg-user] Sampling frequency conversion
References: <CAPJ_JDxaqnEefhy72rvMPmRXkkSog8uKZVUpnHezWtccEwSDcA@mail.gmail.com>
	<loom.20130715T132316-690@post.gmane.org>
	<CAPJ_JDw-UQ0Dz2fM0vwxrr4ZK+cmb-dSEz08rtQqOjfDBnEQ8A@mail.gmail.com>
Message-ID: <loom.20130715T152155-772@post.gmane.org>

rohit a <rohit.tvm.86 <at> gmail.com> writes:

> I used an input of 32000 Hz. I wanted to convert 
> the output to 24000 Hz.
> 
> So I used the below command.
> ./ffmpeg  -i   Input.mpeg  -ar  24000  Output.avi

Please provide (the command line together with) the 
complete, uncut console output if you request help 
on this mailing list.

Carl Eugen


From soho123.2012 at gmail.com  Mon Jul 15 15:31:18 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 21:31:18 +0800
Subject: [FFmpeg-user] how to debug when running ffplay
Message-ID: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>

Hi All,
I got the big trouble about using ffplay to play stream that send from
ffserver.
since ffplay can not display the video data and output the sound from
ffserver.
I would like to see what happen in ffplay,
can someone help me about how to debugging in ffplay side ?

From jshupert at pps-inc.com  Mon Jul 15 15:44:19 2013
From: jshupert at pps-inc.com (Jim Shupert)
Date: Mon, 15 Jul 2013 09:44:19 -0400
Subject: [FFmpeg-user] capture via decklink
In-Reply-To: <1CAB7801-BBEF-404E-8803-3C598D31A98F@dericed.com>
References: <51E01EAF.6070406@pps-inc.com>
	<1CAB7801-BBEF-404E-8803-3C598D31A98F@dericed.com>
Message-ID: <51E3FCB3.7080408@pps-inc.com>


On 7/12/2013 9:46 PM, Dave Rice wrote:
> Hi Jim,
>
> On Jul 12, 2013, at 11:20 AM, Jim Shupert <jshupert at pps-inc.com> wrote:
>
>> Friends,
>> Does anyone know if it is possible to capture by a Blackmagic card ?
>> a DeckLink Mini Recorder or DeckLink SDI?
>> I would like to capture SDI and then turn that into various files including a live stream maybe being served by ffserver.
>> I have been googling and trying things I see a bmdcapture that wishes one use the libav fork .
>> So - I wondering if 'true ffmpeg' can do the same or something simular.
> Officially bmdcapture only support libav (and possibly requires it for installation), but there's a recent ffmpeg-user thread that includes working bmdcapture -> ffmpeg examples. http://comments.gmane.org/gmane.comp.video.ffmpeg.user/46502
>
>> i have a recent ffmpeg built on a ubuntu 12.04 64bit machine with DeckLink SDI and the app media express is all working well.
>> But media express only captures as uncompressed . Then i use ffmpeg to make my desired finished files.
>> I am exploring a better means to pipe and tee and do what i need.
> I'm interested in this as well, if you're willing to share your work.
> Dave Rice
> ____________
I am willing to share - and will ----  I am currently going the avconv way.


From soho123.2012 at gmail.com  Mon Jul 15 15:50:07 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 21:50:07 +0800
Subject: [FFmpeg-user] Help please !!!!! asf stream abort by ffserver
In-Reply-To: <loom.20130715T151945-716@post.gmane.org>
References: <CAFd2cZWfpYHyjBiV8AUpT+nzf-y1in3a-X9XaLP0DpydsfSd6Q@mail.gmail.com>
	<loom.20130715T151945-716@post.gmane.org>
Message-ID: <CAFd2cZWwzsHq4CnBZpnhsiQae38F+9AJW5--R=wROKqG5U5sTw@mail.gmail.com>

(Are you sure pcm_s16le is supported in asf?)
>
which audio format is supported as your opinion?

> ffserver abort since the error:
> > Mon Jul 15 21:07:40 2013 Assertion
> > asf->packet_timestamp_end >= asf->packet_timestamp_start
> > failed at libavformat/asfenc.c:643
>
> Complete, uncut console output missing.
>

the log:


# ffserver version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
   built on Jul 15 2013 14:29:59 with gcc 4.4.5-1.5.5p4 (GCC)
   configuration: --enable-cross-compile --cross-prefix=sdk-linux-
--arch=mips --target-os=linux --disa
 ble-doc --disable-htmlpages --disable-manpages --disable-podpages
--disable-txtpages --disable-mips32r2
  --disable-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mips
 dspr1 --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,copy,format,fps,framestep,resample'
--disa  libav
 util      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
 Mon Jul 15 21:36:48 2013 FFserver started.
 ffmpeg -loglevel debug -f alsa -ar 44100 -ac 2 -i hw:0 -f video4linux2 -r
30 -
 s 640x480 -input_format h264 -i /dev/video1 -vcodec copy -acodec copy
http://lo
 calhost:8090/feed1.ffm
 ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
   built on Jul 15 2013 14:29:59 with gcc 4.4.5-1.5.5p4 (GCC)
   configuration: --enable-cross-compile --cross-prefix=sdk-linux-
--arch=mips --target-os=linux --disa
 ble-doc --disable-htmlpages --disable-manpages --disable-podpages
--disable-txtpages --disable-mips32r2
  --disable-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mips
 dspr1 --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,copy,format,fps,framestep,resample'
--disa  libav
 util      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
'alsa'.
 Reading option '-ar' ... matched as option 'ar' (set audio sampling rate
(in Hz)) with argument '44100'
 .
 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 '-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)) wit
 h argument '30'.
 Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '640
 x480'.
 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 '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) wit
 h argument 'copy'.
 Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) wit
 h 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 hw:0.
 Applying option f (force format) with argument alsa.
 Applying option ar (set audio sampling rate (in Hz)) with argument 44100.
 Applying option ac (set number of audio channels) with argument 2.
 Successfully parsed a group of options.
 Opening an input file: hw:0.
 [alsa @ 0x6bdd20] audio_device=hw:0
 [alsa @ 0x6bdd20] codec_id ==NONE
 [alsa @ 0x6bdd20] Do RECORD,1
 [alsa @ 0x6bdd20] audio fmt=2
 Hardware PCM card 0 'USB Camera' device 0 subdevice 0
 Its setup is:
   stream       : CAPTURE
   access       : RW_INTERLEAVED
   format       : S16_LE
   subformat    : STD
   channels     : 2
   rate         : 44100
   exact rate   : 44100 (44100/1)
   msbits       : 16
   buffer_size  : 65536
   period_size  : 64
   period_time  : 1451
   tstamp_mode  : NONE
   period_step  : 1
   avail_min    : 64
   period_event : 0
   start_threshold  : 1
   stop_threshold   : 65536
   silence_threshold: 0
   silence_size : 0
   boundary     : 1073741824
   appl_ptr     : 0
   hw_ptr       : 0
 [alsa @ 0x6bdd20] All info found
 Guessed Channel Layout for  Input Stream #0.0 : stereo
 Input #0, alsa, from 'hw:0':
   Duration: N/A, start: 1373924221.998844, bitrate: 1411 kb/s
     Stream #0:0, 1, 1/1000000: Audio: pcm_s16le, 44100 Hz, stereo, s16,
1411 kb/s
 Successfully opened the file.
 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 @ 0x6c8cf0] fd:3 capabilities:4000001
 [video4linux2,v4l2 @ 0x6c8cf0] Current input_channel: 0, input_name:
Camera 1
 [video4linux2,v4l2 @ 0x6c8cf0] Setting time per frame to 1/30
 [h264 @ 0x6ca030] Frame num gap 8 6
 [video4linux2,v4l2 @ 0x6c8cf0] All info found
 Input #1, video4linux2,v4l2, from '/dev/video1':
   Duration: N/A, start: 54.020000, bitrate: N/A
     Stream #1:0, 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: 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.
 Successfully parsed a group of options.
 Opening an output file: http://localhost:8090/feed1.ffm.
 [ffm @ 0x794b40] Format ffm probed with size=2048 and score=101
 [AVIOContext @ 0x7be580] Statistics: 4096 bytes read, 0 seeks
 Mon Jul 15 21:37:03 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_s16le, 44100 Hz, stereo, 1411
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 #0:0 -> #0:0 (copy)
   Stream #1:0 -> #0:1 (copy)
 Press [q] to stop, [?] for help
 [alsa @ 0x6bdd20] ALSA buffer xrun.
 frame=   25 fps=0.0 q=-1.0 size=     124kB time=00:00:02.39 bitrate=
423.7kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 2494875, current:
2494829; changing to 2494876. This
  may result in incorrect timestamps in the output file.
 frame=   40 fps= 40 q=-1.0 size=     220kB time=00:00:02.89 bitrate=
622.1kbits/s
     frame=   55 fps= 36 q=-1.0 size=     436kB time=00:00:03.40
bitrate=1049.7kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 3522565, current:
3522541; changing to 3522566. This
  may result in incorrect timestamps in the output file.
 frame=   70 fps= 35 q=-1.0 size=     648kB time=00:00:03.90
bitrate=1358.9kbits/s
     frame=   85 fps= 34 q=-1.0 size=     844kB time=00:00:04.40
bitrate=1569.7kbits/s
     frame=  100 fps= 33 q=-1.0 size=    1040kB time=00:00:04.90
bitrate=1736.4kbits/s
     frame=  115 fps= 33 q=-1.0 size=    1244kB time=00:00:05.40
bitrate=1885.0kbits/s
     frame=  130 fps= 32 q=-1.0 size=    1428kB time=00:00:05.90
bitrate=1981.1kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 6222550, current:
6222549; changing to 6222551. This
  may result in incorrect timestamps in the output file.
 frame=  145 fps= 32 q=-1.0 size=    1628kB time=00:00:06.40
bitrate=2082.0kbits/s
     frame=  160 fps= 32 q=-1.0 size=    1824kB time=00:00:06.90
bitrate=2164.1kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 7122529, current:
7122506; changing to 7122530. This
  may result in incorrect timestamps in the output file.
 frame=  175 fps= 32 q=-1.0 size=    2020kB time=00:00:07.40
bitrate=2234.5kbits/s
     frame=  190 fps= 32 q=-1.0 size=    2212kB time=00:00:07.90
bitrate=2291.9kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 8022437, current:
8022415; changing to 8022438. This
  may result in incorrect timestamps in the output file.
 frame=  205 fps= 31 q=-1.0 size=    2412kB time=00:00:08.40
bitrate=2350.7kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 8722440, current:
8722440; changing to 8722441. This
  may result in incorrect timestamps in the output file.
 frame=  220 fps= 31 q=-1.0 size=    2608kB time=00:00:08.90
bitrate=2399.0kbits/s
     frame=  235 fps= 31 q=-1.0 size=    2808kB time=00:00:09.40
bitrate=2445.7kbits/s
     frame=  250 fps= 31 q=-1.0 size=    3004kB time=00:00:09.90
bitrate=2484.1kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 10222280, current:
10222262; changing to 10222281. T
 his may result in incorrect timestamps in the output file.
 [ffm @ 0x794630] Non-monotonous DTS in output stream 0:0; previous:
10322332, current: 10322313; changi
 ng to 10322333. This may result in incorrect timestamps in the output file.
 frame=  265 fps= 31 q=-1.0 size=    3196kB time=00:00:10.40
bitrate=2516.2kbits/s
     frame=  280 fps= 31 q=-1.0 size=    3392kB time=00:00:10.90
bitrate=2548.0kbits/s
     frame=  295 fps= 31 q=-1.0 size=    3580kB time=00:00:11.40
bitrate=2571.3kbits/s
     frame=  310 fps= 31 q=-1.0 size=    3772kB time=00:00:11.90
bitrate=2595.3kbits/s
     Non-monotonous DTS in output stream 0:0; previous: 12022369, current:
12022349; changing to 12022370. T
 his may result in incorrect timestamps in the output file.
 frame=  325 fps= 31 q=-1.0 size=    3972kB time=00:00:12.27
bitrate=2651.1kbits/s
     frame=  340 fps= 31 q=-1.0 size=    4168kB time=00:00:12.56
bitrate=2717.6kbits/s
     frame=  355 fps= 31 q=-1.0 size=    4368kB time=00:00:12.85
bitrate=2783.7kbits/s
     frame=  370 fps= 31 q=-1.0 size=    4564kB time=00:00:13.14
bitrate=2844.4kbits/s
     frame=  385 fps= 31 q=-1.0 size=    4752kB time=00:00:13.43
bitrate=2897.6kbits/s
     frame=  400 fps= 31 q=-1.0 size=    4940kB time=00:00:13.77
bitrate=2938.2kbits/s
     frame=  414 fps= 31 q=-1.0 size=    5048kB time=00:00:14.24
bitrate=2903.3kbits/s
     ALSA buffer xrun.
 [ffm @ 0x794630] Non-monotonous DTS in output stream 0:0; previous:
15681195, current: 15680422; changi
 ng to 15681196. This may result in incorrect timestamps in the output file.
 [ffm @ 0x794630] Non-monotonous DTS in output stream 0:0; previous:
15681196, current: 15681053; changi
 ng to 15681197. This may result in incorrect timestamps in the output file.
 frame=  430 fps= 31 q=-1.0 size=    5220kB time=00:00:15.79
bitrate=2707.9kbits/s
     frame=  445 fps= 31 q=-1.0 size=    5328kB time=00:00:16.08
bitrate=2713.9kbits/s
     frame=  460 fps= 31 q=-1.0 size=    5440kB time=00:00:16.37
bitrate=2721.8kbits/s
     frame=  475 fps= 31 q=-1.0 size=    5640kB time=00:00:16.66
bitrate=2772.9kbits/s
     frame=  490 fps= 31 q=-1.0 size=    5836kB time=00:00:16.95
bitrate=2820.1kbits/s
     Mon Jul 15 21:37:20 2013 192.168.1.3 - - [GET] "/test.asf HTTP/1.1"
200 1021
 frame=  505 fps= 31 q=-1.0 size=    6032kB time=00:00:17.27
bitrate=2860.7kbits/s
     frame=  512 fps= 30 q=-1.0 size=    6092kB time=00:00:17.50
bitrate=2851.2kbits/s
     ALSA buffer xrun.
 frame=  537 fps= 31 q=-1.0 size=    6284kB time=00:00:19.27
bitrate=2671.2kbits/s
     frame=  552 fps= 31 q=-1.0 size=    6376kB time=00:00:19.56
bitrate=2670.1kbits/s
     Mon Jul 15 21:37:22 2013 Assertion asf->packet_timestamp_end >=
asf->packet_timestamp_start failed at l
 ibavformat/asfenc.c:643
 av_interleaved_write_frame(): Connection reset by peer
 [AVIOContext @ 0x7be5a0] Statistics: 0 seeks, 1624 writeouts
 [1] + Aborted                    ffserver -f /etc/ffserver.conf
 #

From iskilian at hotmail.com  Mon Jul 15 15:51:06 2013
From: iskilian at hotmail.com (S Kilian)
Date: Mon, 15 Jul 2013 09:51:06 -0400
Subject: [FFmpeg-user] capture via decklink
In-Reply-To: <51E3FCB3.7080408@pps-inc.com>
References: <51E01EAF.6070406@pps-inc.com>,
	<1CAB7801-BBEF-404E-8803-3C598D31A98F@dericed.com>,
	<51E3FCB3.7080408@pps-inc.com>
Message-ID: <BAY178-W11176DA717FFDEE2DE44DADD670@phx.gbl>

 

Try bmdtools (bmdcapture). I haven't tried it. Also see Blackmagic Design's forum. I saw some posting that might help.

 

https://github.com/lu-zero/bmdtools/

 

 

 - Steve K
 		 	   		  

From soho123.2012 at gmail.com  Mon Jul 15 16:01:05 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 22:01:05 +0800
Subject: [FFmpeg-user] HELP :ffplay CANOT play h264 video + pcm_S16LE
	audio rtp stream
In-Reply-To: <CAFd2cZWqcgBhVPhO_0F2FegSY-ZcGCf-odpb2Xc_3DdW2MuQqQ@mail.gmail.com>
References: <CAFd2cZWh0EHepSghXrvKZ-OJ=cGz3BOJpNbs62vGyjqrZ9s-Zw@mail.gmail.com>
	<CAFd2cZWqcgBhVPhO_0F2FegSY-ZcGCf-odpb2Xc_3DdW2MuQqQ@mail.gmail.com>
Message-ID: <CAFd2cZU14i_tQDyb=PzjOuZ9QyX60Dabk+Fa2oROUB9YJTDiBw@mail.gmail.com>

Hi All,


Help Please !!!!!
Help Please !!!!!

2013/7/9 Soho Soho123 <soho123.2012 at gmail.com>

> Hi All,
>
> the command line of ffmpeg I used:
> ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format
> h264 -i /dev/video1 -f alsa -i hw:0 -ar 48000 -ac 2 -acodec copy -vcodec
> copy  http://localhost:8090/feed1.ffm
>
> It is very appreciated for any input!!
>
> 2013/7/9 Soho Soho123 <soho123.2012 at gmail.com>
>
>> Hi all,
>>
>> I got the big trouble about :
>> ffplay can not play play h264 video + pcm_S16LE audio rtp stream.
>> originally, ffplay can play video stream if no audio.
>> after add audio part , then ffplay can not play
>> Does anyone have ideas?
>> the configuration of ffserver.conf:
>> ======================================================
>> Port 8090
>> RTSPPort 5554
>> <Feed feed1.ffm>
>> File /tmp/feed1.ffm FileMaxSize 5M
>> ACL allow 127.0.0.1
>> </Feed>
>> <Stream test1-rtsp.mpg>
>> Feed feed1.ffm
>> Format rtp
>> AVOptionVideo flags +global_header
>> VideoSize 1280x720
>> VideoFrameRate 30
>> VideoCodec libx264
>> AVOptionAudio flags +global_header
>> AudioCodec pcm_s16le
>> AudioChannels 2
>> AudioSampleRate 48000
>> </Stream>
>> ======================================================
>> the detail log :
>> ======================================================
>> ffserver version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
>>   built on Jul  9 2013 15:51:03 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 --dis
>> able-ffplay --disable-postproc --disable-runtime-cpudetect
>> --disable-swscale-alpha --disable-mipsdspr1 --d
>> isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
>> --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
>> --enable-libx2
>> 64 --extra-cflags=-I/ShareLib_Install/bin/../include
>> --extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
>> --enable-filter='aformat,aresample,copy,format,fps
>> ,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
>> 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
>> Tue Jul  9 20:57:52 2013 FFserver started.
>> ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
>>   built on Jul  9 2013 15:51:03 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 --dis
>> able-ffplay --disable-postproc --disable-runtime-cpudetect
>> --disable-swscale-alpha --disable-mipsdspr1 --d
>> isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
>> --bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
>> --enable-libx2
>> 64 --extra-cflags=-I/ShareLib_Install/bin/../include
>> --extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
>> --enable-filter='aformat,aresample,copy,format,fps
>> ,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
>> 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
>> Splitting the commandline.
>> Reading option '-loglevel' ... matched as option 'loglevel' (set libav*
>> logging level) with argument 'debu
>> g'.
>> 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 a
>> rgument '30'.
>> Reading option '-s' ... matched as option 's' (set frame size (WxH or
>> abbreviation)) with argument '1280x7
>> 20'.
>> 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 '-i' ... matched as input file with argument 'hw:0'.
>> 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 '-acodec' ... matched as option 'acodec' (force audio
>> codec ('copy' to copy stream)) with a
>> rgument 'copy'.
>> Reading option '-vcodec' ... matched as option 'vcodec' (force video
>> codec ('copy' to copy stream)) with a
>> rgument '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 libav* 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
>> 1280x720.
>> Successfully parsed a group of options.
>> Opening an input file: /dev/video1.
>> [video4linux2,v4l2 @ 0x69e240] fd:3 capabilities:4000001
>> [video4linux2,v4l2 @ 0x69e240] Selecting input_channel: 0
>> [video4linux2,v4l2 @ 0x69e240] input_channel: 0, input_name: Camera 1
>> [video4linux2,v4l2 @ 0x69e240] Setting time per frame to 1/30
>> [h264 @ 0x69ede0] Using externally provided dimensions
>> [h264 @ 0x69ede0] Frame num gap 18 16
>> [h264 @ 0x69ede0] Frame num gap 21 19
>> [h264 @ 0x69ede0] Frame num gap 27 25
>> [video4linux2,v4l2 @ 0x69e240] All info found
>> [video4linux2,v4l2 @ 0x69e240] Estimating duration from bitrate, this may
>> be inaccurate
>> Input #0, video4linux2,v4l2, from '/dev/video1':
>>   Duration: N/A, start: 24.690000, bitrate: N/A
>>     Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 1280x720, 1/2000000,
>> -5 kb/s, 30 fps, 30 tbr, 1000k t
>> bn, 2000k tbc
>> Successfully opened the file.
>> Parsing a group of options: input file hw:0.
>> Applying option f (force format) with argument alsa.
>> Successfully parsed a group of options.
>> Opening an input file: hw:0.
>> [alsa @ 0x837ba0] codec_id ==NONE
>> [alsa @ 0x837ba0] All info found
>> [alsa @ 0x837ba0] Estimating duration from bitrate, this may be inaccurate
>> Guessed Channel Layout for  Input Stream #1.0 : stereo
>> Input #1, alsa, from 'hw:0':
>>   Duration: N/A, start: 1373403477.887958, 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://localhost:8090/feed1.ffm.
>> Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
>> Applying option ac (set number of audio channels) with argument 2.
>> Applying option acodec (force audio codec ('copy' to copy stream)) with
>> argument copy.
>> Applying option vcodec (force video codec ('copy' to copy stream)) with
>> argument copy.
>> Successfully parsed a group of options.
>> Opening an output file: http://localhost:8090/feed1.ffm.
>> [ffm @ 0x842e40] Format ffm probed with size=2048 and score=101
>> [AVIOContext @ 0x842700] Statistics: 4096 bytes read, 0 seeks
>> Tue Jul  9 20:57:57 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         : Lavf54.63.104
>>     Stream #0:0, 0, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, 1536
>> kb/s
>>     Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 1280x720, 1/1000000,
>> q=2-31, -5 kb/s, 30 fps, 1000k t
>> bn, 1000k tbc
>> Stream mapping:
>>   Stream #1:0 -> #0:0 (copy)
>>   Stream #0:0 -> #0:1 (copy)
>> Press [q] to stop, [?] for help
>> frame=   23 fps=0.0 q=-1.0 size=     136kB time=00:00:02.37 bitrate=
>> 470.1kbits/s
>>     frame=   38 fps= 37 q=-1.0 size=     272kB time=00:00:02.87 bitrate=
>> 776.4kbits/s
>>     frame=   53 fps= 35 q=-1.0 size=     388kB time=00:00:03.37 bitrate=
>> 943.2kbits/s
>>     frame=   68 fps= 34 q=-1.0 size=     532kB time=00:00:03.87
>> bitrate=1126.1kbits/s
>>     frame=   83 fps= 33 q=-1.0 size=     740kB time=00:00:04.37
>> bitrate=1387.2kbits/s
>>     frame=   98 fps= 32 q=-1.0 size=     972kB time=00:00:04.87
>> bitrate=1635.0kbits/s
>>     frame=  113 fps= 32 q=-1.0 size=    1192kB time=00:00:05.37
>> bitrate=1818.4kbits/s
>>     frame=  128 fps= 32 q=-1.0 size=    1416kB time=00:00:05.87
>> bitrate=1976.1kbits/s
>>     frame=  143 fps= 32 q=-1.0 size=    1632kB time=00:00:06.37
>> bitrate=2098.8kbits/s
>>     frame=  158 fps= 31 q=-1.0 size=    1864kB time=00:00:06.87
>> bitrate=2222.7kbits/s
>>     frame=  173 fps= 31 q=-1.0 size=    2068kB time=00:00:07.37
>> bitrate=2298.7kbits/s
>>     frame=  188 fps= 31 q=-1.0 size=    2276kB time=00:00:07.87
>> bitrate=2369.1kbits/s
>>     frame=  203 fps= 31 q=-1.0 size=    2484kB time=00:00:08.37
>> bitrate=2431.2kbits/s
>>     frame=  218 fps= 31 q=-1.0 size=    2692kB time=00:00:08.87
>> bitrate=2486.2kbits/s
>>     frame=  233 fps= 31 q=-1.0 size=    2904kB time=00:00:09.37
>> bitrate=2538.9kbits/s
>>     frame=  248 fps= 31 q=-1.0 size=    3116kB time=00:00:09.87
>> bitrate=2586.2kbits/s
>>     frame=  263 fps= 31 q=-1.0 size=    3328kB time=00:00:10.37
>> bitrate=2629.0kbits/s
>>     frame=  278 fps= 31 q=-1.0 size=    3544kB time=00:00:10.87
>> bitrate=2670.9kbits/s
>>     frame=  293 fps= 31 q=-1.0 size=    3756kB time=00:00:11.37
>> bitrate=2706.2kbits/s
>>     frame=  308 fps= 31 q=-1.0 size=    3968kB time=00:00:11.87
>> bitrate=2738.5kbits/s
>>     frame=  323 fps= 31 q=-1.0 size=    4180kB time=00:00:12.37
>> bitrate=2768.2kbits/s
>>     Tue Jul  9 20:58:08 2013 192.168.1.3:34550 - - "PLAY
>> test1-rtsp.mpg/streamid=0 RTP/UDP"
>> Tue Jul  9 20:58:08 2013 192.168.1.3:34552 - - "PLAY
>> test1-rtsp.mpg/streamid=1 RTP/UDP"
>> frame=  338 fps= 31 q=-1.0 size=    4308kB time=00:00:12.87
>> bitrate=2742.1kbits/s
>>     frame=  353 fps= 31 q=-1.0 size=    4424kB time=00:00:13.37
>> bitrate=2710.7kbits/s
>>     frame=  368 fps= 31 q=-1.0 size=    4536kB time=00:00:13.87
>> bitrate=2679.1kbits/s
>>     frame=  383 fps= 31 q=-1.0 size=    4652kB time=00:00:14.37
>> bitrate=2652.0kbits/s
>>     frame=  398 fps= 31 q=-1.0 size=    4764kB time=00:00:14.87
>> bitrate=2624.5kbits/s
>>     frame=  413 fps= 31 q=-1.0 size=    4876kB time=00:00:15.37
>> bitrate=2598.8kbits/s
>>     ALSA buffer xrun.
>> Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
>> Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]cannot find FFM syncword
>> Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
>> frame=  428 fps= 31 q=-1.0 size=    5280kB time=00:00:15.87
>> bitrate=2725.5kbits/s
>>     frame=  443 fps= 30 q=-1.0 size=    5396kB time=00:00:16.37
>> bitrate=2700.3kbits/s
>>     frame=  458 fps= 30 q=-1.0 size=    5512kB time=00:00:16.87
>> bitrate=2676.6kbits/s
>>     frame=  473 fps= 30 q=-1.0 size=    5624kB time=00:00:17.37
>> bitrate=2652.4kbits/s
>>     find Key
>> frame=  488 fps= 30 q=-1.0 size=    5748kB time=00:00:17.87
>> bitrate=2635.0kbits/s
>>     ALSA buffer xrun.
>> frame=  503 fps= 30 q=-1.0 size=    6196kB time=00:00:18.37
>> bitrate=2763.1kbits/s
>>     ALSA buffer xrun.
>> frame=  509 fps= 26 q=-1.0 size=    6224kB time=00:00:18.57
>> bitrate=2745.7kbits/s
>>     frame=  520 fps= 23 q=-1.0 size=    6776kB time=00:00:21.37
>> bitrate=2597.5kbits/s
>>     ALSA buffer xrun.
>> frame=  537 fps= 22 q=-1.0 size=    6856kB time=00:00:24.47
>> bitrate=2295.2kbits/s
>>     ALSA buffer xrun.
>> frame=  561 fps= 22 q=-1.0 size=    7040kB time=00:00:27.30
>> bitrate=2112.5kbits/s
>>     frame=  576 fps= 22 q=-1.0 size=    7092kB time=00:00:27.80
>> bitrate=2089.8kbits/s
>>     frame=  591 fps= 22 q=-1.0 size=    7144kB time=00:00:28.30
>> bitrate=2068.0kbits/s
>>     frame=  606 fps= 22 q=-1.0 size=    7200kB time=00:00:28.80
>> bitrate=2048.0kbits/s
>>     frame=  621 fps= 23 q=-1.0 size=    7252kB time=00:00:29.30
>> bitrate=2027.6kbits/s
>>     ALSA buffer xrun.
>> frame=  636 fps= 23 q=-1.0 size=    7572kB time=00:00:29.80
>> bitrate=2081.5kbits/s
>>     frame=  652 fps= 23 q=-1.0 size=    7808kB time=00:00:30.34
>> bitrate=2108.2kbits/s
>>     frame=  667 fps= 23 q=-1.0 size=    8020kB time=00:00:30.84
>> bitrate=2130.3kbits/s
>>     frame=  682 fps= 23 q=-1.0 size=    8132kB time=00:00:31.34
>> bitrate=2125.6kbits/s
>>     frame=  697 fps= 23 q=-1.0 size=    8244kB time=00:00:31.84
>> bitrate=2121.1kbits/s
>>     frame=  712 fps= 23 q=-1.0 size=    8360kB time=00:00:32.34
>> bitrate=2117.7kbits/s
>>     frame=  726 fps= 23 q=-1.0 size=    8472kB time=00:00:32.80
>> bitrate=2115.9kbits/s
>>     frame=  741 fps= 24 q=-1.0 size=    8584kB time=00:00:33.30
>> bitrate=2111.7kbits/s
>>     ALSA buffer xrun.
>> frame=  756 fps= 24 q=-1.0 size=    8996kB time=00:00:33.80
>> bitrate=2180.3kbits/s
>>     frame=  771 fps= 24 q=-1.0 size=    9108kB time=00:00:34.30
>> bitrate=2175.3kbits/s
>>     frame=  786 fps= 24 q=-1.0 size=    9220kB time=00:00:34.80
>> bitrate=2170.4kbits/s
>>     frame=  802 fps= 24 q=-1.0 size=    9352kB time=00:00:35.34
>> bitrate=2167.8kbits/s
>>     frame=  816 fps= 24 q=-1.0 size=    9460kB time=00:00:35.80
>> bitrate=2164.7kbits/s
>>     frame=  831 fps= 24 q=-1.0 size=    9576kB time=00:00:36.30
>> bitrate=2161.1kbits/s
>>     ALSA buffer xrun.
>> frame=  846 fps= 24 q=-1.0 size=   10008kB time=00:00:36.80
>> bitrate=2227.9kbits/s
>>     frame=  861 fps= 24 q=-1.0 size=   10120kB time=00:00:37.30
>> bitrate=2222.6kbits/s
>>     frame=  876 fps= 24 q=-1.0 size=   10232kB time=00:00:37.80
>> bitrate=2217.5kbits/s
>>     frame=  891 fps= 24 q=-1.0 size=   10348kB time=00:00:38.30
>> bitrate=2213.3kbits/s
>>     Tue Jul  9 20:58:34 2013 192.168.1.3 - - [PLAY] "rtsp://
>> 192.168.1.254:5554/test1-rtsp.mpg/ RTSP/1.0" 200 1
>> 030
>> frame=  907 fps= 25 q=-1.0 size=   10460kB time=00:00:38.84
>> bitrate=2206.2kbits/s
>>     frame=  922 fps= 25 q=-1.0 size=   10572kB time=00:00:39.34
>> bitrate=2201.5kbits/s
>>     ALSA buffer xrun.
>> [ffm @ 0x842940] st:0 PTS: 37891681 DTS: 37891681 < 37892211 invalid,
>> clipping
>> frame=  937 fps= 25 q=-1.0 size=   10984kB time=00:00:39.84
>> bitrate=2258.6kbits/s
>>     frame=  952 fps= 25 q=-1.0 size=   11096kB time=00:00:40.34
>> bitrate=2253.3kbits/s
>>     qframe=  967 fps= 25 q=-1.0 size=   11208kB time=00:00:40.84
>> bitrate=2248.2kbits/s
>>     frame=  982 fps= 25 q=-1.0 size=   11324kB time=00:00:41.34
>> bitrate=2244.0kbits/s
>>     frame=  997 fps= 25 q=-1.0 size=   11444kB time=00:00:41.84
>> bitrate=2240.7kbits/s
>>     frame= 1012 fps= 25 q=-1.0 size=   11560kB time=00:00:42.34
>> bitrate=2236.6kbits/s
>>     ALSA buffer xrun.
>> frame= 1027 fps= 25 q=-1.0 size=   11988kB time=00:00:42.84
>> bitrate=2292.4kbits/s
>>     ALSA buffer xrun.
>> frame= 1029 fps= 23 q=-1.0 size=   12004kB time=00:00:42.90
>> bitrate=2292.2kbits/s
>>     frame= 1052 fps= 23 q=-1.0 size=   12572kB time=00:00:46.90
>> bitrate=2195.9kbits/s
>>     frame= 1067 fps= 23 q=-1.0 size=   12624kB time=00:00:47.40
>> bitrate=2181.8kbits/s
>>
>> =============================================================================
>> ffplay log:
>> ffplay -loglevel debug rtsp://192.168.1.254:5554/test1-rtsp.mpg
>> ffplay version N-54362-ge0be3cb Copyright (c) 2003-2013 the FFmpeg
>> developers
>>   built on Jul  2 2013 22:15:59 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. 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
>> SDP:nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
>> v=0
>> o=- 0 0 IN IP4 127.0.0.1
>> s=No Title
>> c=IN IP4 0.0.0.0
>> t=0 0
>> a=tool:libavformat 54.63.104
>> m=audio 0 RTP/AVP 96
>> b=AS:1536
>> a=control:streamid=0
>> m=video 0 RTP/AVP 97
>> b=AS:-5
>> a=rtpmap:97 H264/90000
>> a=fmtp:97 packetization-mode=1;
>> sprop-parameter-sets=Z0JAH5ZUAoAtyA==,aM44gA==;
>> profile-level-id=42401F
>> a=control:streamid=1
>>
>> [rtsp @ 02a64500] video codec set to: h264
>> [NULL @ 02a679e0] RTP Packetization Mode: 1
>> [NULL @ 02a679e0] Extradata set to 02a67ea0 (size: 22)!
>> [NULL @ 02a679e0] RTP Profile IDC: 42 Profile IOP: 40 Level: 1f
>> hello state=0 0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
>>     nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
>>
>>
>>
>>
>>
>>
>>
>

From jam at tigger.ws  Mon Jul 15 16:05:09 2013
From: jam at tigger.ws (James Linder)
Date: Mon, 15 Jul 2013 22:05:09 +0800
Subject: [FFmpeg-user] Help Please
In-Reply-To: <loom.20130715T151256-915@post.gmane.org>
References: <A5C0165C-F3C2-4F90-9045-9AD34D7326AC@tigger.ws>
	<loom.20130715T151256-915@post.gmane.org>
Message-ID: <F9D7A571-5F45-4950-AC32-DAC4CC84BB46@tigger.ws>


On 15/07/2013, at 9:15 PM, Carl Eugen Hoyos wrote:

> James Linder <jam <at> tigger.ws> writes:
> 
>> Output #1, rawvideo, to 'pipe:':
>>  Metadata:
>>    encoder         : Lavf55.11.101
>>    Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 
>> 800x600, q=2-31, 200 kb/s, 90k tbn, 25 tbc
>> Stream mapping:
>>  Stream #1:0 -> #0:0 (rawvideo -> mpeg4)
>>  Stream #0:0 -> #0:1 (pcm_s16le -> ac3)
>>  Stream #1:0 -> #1:0 (rawvideo -> rawvideo)
>> Press [q] to stop, [?] for help
>> Input #0, rawvideo, from 'pipe:':
>>  Duration: N/A, start: 0.000000, bitrate: 144000 kb/s
>>    Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p
> 
> (Sorry if you disagree or if I miss something but don't 
> you agree that above is nearly unreadable and the 
> problem can be reproduced using a file instead of a 
> pipe?)
> 
> You don't tell ffplay that your "input" is yuyv422 and the 
> default for rawvideo is yuv420p, use -pix_fmt yuyv422

Carl thanks, that fixed the picture immediately.

This records perfectly
/usr/bin/ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 -i /dev/video0 -vtag divx -s 720x576 -r pal -q:v 5 mon.avi

As soon as I add 
-s 720x576 -f rawvideo - |/usr/bin/ffplay -f rawvideo -i - -pix_fmy yuyv422 -s 720x576
or indeed
-s 720x576 -f rawvideo - -pix_fmt yuv420p  |/usr/bin/ffplay -f rawvideo -i - -s 720x576

BOTH the display and the recorded file are [repeat_frame/ 1/2 frame new stuff every few seconds /old frame repeated]

where would I start to look for the cause. Display Size does not make any difference. Sys load is 10% (90% idle) Thanks
James

From mkfrank at gmx.net  Mon Jul 15 16:22:43 2013
From: mkfrank at gmx.net (Michael Frank)
Date: Mon, 15 Jul 2013 16:22:43 +0200
Subject: [FFmpeg-user] Strange? output from configure
Message-ID: <51E405B3.3050104@gmx.net>

Hello,
when configuring ffmpeg (1.2 or 2.0) I pass --arch=core2 to configure 
and in the output I get "3DNow! enabled   yes" and "3DNow! extended 
enabled   yes" as well. This is confusing as the Intel Core2 is not 
capable of 3DNow! (verified by /proc/cpuinfo).

Can someone please point me in the right direction.

Thanks
Michael

From cehoyos at ag.or.at  Mon Jul 15 17:10:22 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 15:10:22 +0000 (UTC)
Subject: [FFmpeg-user] Strange? output from configure
References: <51E405B3.3050104@gmx.net>
Message-ID: <loom.20130715T170421-44@post.gmane.org>

Michael Frank <mkfrank <at> gmx.net> writes:

> when configuring ffmpeg (1.2 or 2.0) I pass --arch=core2 

(I suspect --arch mostly / only makes sense for cross-
compilation, I don't thin it activates or deactivates 
any SIMD capabilities on any system.)

> to configure and in the output I get "3DNow! enabled   yes" 
> and "3DNow! extended enabled   yes" as well.

If you don't like it, pass --disable-amd3dnow to configure.

I don't think this has many advantages though.

Carl Eugen


From soho123.2012 at gmail.com  Mon Jul 15 17:18:34 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Mon, 15 Jul 2013 23:18:34 +0800
Subject: [FFmpeg-user] stream server solution
Message-ID: <CAFd2cZUNAz8JjoU1bbHa8-Zu+-i8Yxm_Tk7wZxSZNna0Jg_F+w@mail.gmail.com>

Hi All,

I use ffserver + ffmpeg for the stream server application.
but it seems the solution does not popular.
Does anyone have idea about stream server that can run in linux OS?

As my understanding, ffmeg can send stream to the specific client host.
but it can not provide the service for multiple client with unicast packet.
Any ideas?

Thanks

From jpboard2 at yahoo.com  Mon Jul 15 18:56:53 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 15 Jul 2013 09:56:53 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
Message-ID: <1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>

I'd like to write a C or C++ program which uses libavcodec to retrieve
individual frames of an AVI file.? I can't find much documentation on the WWW.
Does anyone have documentation on how to do this, or some current examples?

I think this question is appropriate for this mailing list, since the ffmpeg group 
developed libavcodec. If not, please let me know.? Thanks.

From o_lucian at yahoo.com  Mon Jul 15 09:16:08 2013
From: o_lucian at yahoo.com (lucian orasanu)
Date: Mon, 15 Jul 2013 00:16:08 -0700 (PDT)
Subject: [FFmpeg-user] Artefacts when recording an rtsp stream
Message-ID: <1373872568.86623.YahooMailNeo@web160803.mail.bf1.yahoo.com>

>>I'am trayng to record an stream from an hikvision >>DS=2CD2032-I, with command >>ffmpeg -i rtsp://37.43.41.213/h264/ch1/main/av_stream  > I tested the stream several times, either I get many 
> distortions and many lines of "RTP: missed x packets" 
> on the console, or no messages and perfect reception. > I am not sure if there is something that can be fixed 
> in FFmpeg. > Carl Eugen
But this is happening even if camera is connected directly to the pc with an ethernet cable!!
speed is too high?
i have an stream recorded with camera! maybe if i upload the file to my server you can analyze the file and see if ffmpeg should handle that format?

From o_lucian at yahoo.com  Mon Jul 15 09:33:52 2013
From: o_lucian at yahoo.com (lucian orasanu)
Date: Mon, 15 Jul 2013 00:33:52 -0700 (PDT)
Subject: [FFmpeg-user] Artefacts when recording an rtsp stream
Message-ID: <1373873632.21905.YahooMailNeo@web160802.mail.bf1.yahoo.com>

back with an link with recorded file from camera:

http://zm.realfield.ro/57609.avi

another thing i observed, is if I reduce the quality of the camera there are less of thees?"RTP: missed x packets", and also there are less artefacts.
If you what to look at the camera settings user is admin, and password 12345. default

From cehoyos at ag.or.at  Mon Jul 15 20:51:32 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 18:51:32 +0000 (UTC)
Subject: [FFmpeg-user] Strange? output from configure
References: <51E405B3.3050104@gmx.net>
Message-ID: <loom.20130715T204841-175@post.gmane.org>

Michael Frank <mkfrank <at> gmx.net> writes:

> when configuring ffmpeg (1.2 or 2.0) I pass 
> --arch=core2 to configure 

I should have added that you probably want to use:
$ configure --cpu=core2
This does not disable 3dnow but enables core2-
optimizations.

Carl Eugen


From cehoyos at ag.or.at  Mon Jul 15 20:54:22 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 18:54:22 +0000 (UTC)
Subject: [FFmpeg-user] libavcodec documentation and usage
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
Message-ID: <loom.20130715T205304-102@post.gmane.org>

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

> 
> I'd like to write a C or C++ program which uses libavcodec to retrieve
> individual frames of an AVI file.? I can't find much documentation on the WWW.
> Does anyone have documentation on how to do this, or some current examples?

Did you look into the doc/examples directory?

Please do not high jack threads, Carl Eugen





From bouke at videotoolshed.com  Mon Jul 15 20:55:00 2013
From: bouke at videotoolshed.com (Bouke (VideoToolShed))
Date: Mon, 15 Jul 2013 20:55:00 +0200
Subject: [FFmpeg-user] OT, OHM 2013
Message-ID: <31F878335868481B9B25950BF3AF9AF9@HPKANTOOR>

Anyone going?
Reason, as my contribution to the open source community, i'm volunteering 
bringing myself and my video hardware to do (live) streaming.
Now of course i count myself for at least 5 mortals, but even then the 
broadcast team is highly understaffed...
So, if you do want to come and help us out a little, that would be great.
More info here:
https://ohm2013.org/site/
And contacting me direct is also possible of course.
Small disclaimer, i'm just a volunteer and my words are just mine, i have no 
relation with the organisation whatsoever.

Bouke

VideoToolShed
van Oldenbarneveltstraat 33
6512 AS  NIJMEGEN, the Netherlands
+31 24 3553311 


From jpboard2 at yahoo.com  Mon Jul 15 21:07:38 2013
From: jpboard2 at yahoo.com (James Board)
Date: Mon, 15 Jul 2013 12:07:38 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <loom.20130715T205304-102@post.gmane.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
Message-ID: <1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>

>> I'd like to write a C or C++ program which uses libavcodec to retrieve
>> individual frames of an AVI file.? I can't find much documentation on the WWW.
>> Does anyone have documentation on how to do this, or some current examples?
>
>Did you look into the doc/examples directory?
>
>Please do not high jack threads, Carl Eugen


No.? I only looked on the web, not in the doc/examples directory.


What thread did I hijack?

Thank you for the doc/examples pointer.

From tanjimhossainsifat at gmail.com  Mon Jul 15 22:12:22 2013
From: tanjimhossainsifat at gmail.com (Tanjim Hossain Sifat)
Date: Tue, 16 Jul 2013 02:12:22 +0600
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
Message-ID: <CAFQBXYB_Umszv3dO8ahBdSjcBnTopcRaQq=p4nMV=GHG_6zQaA@mail.gmail.com>

Dear James,
If you have some question on libavcodec then subscribe to the libav-user
mailing list (link given below):
https://lists.ffmpeg.org/mailman/listinfo/libav-user/

I have gathered some amateur level knowledge on libavcodec, which I can
share with you, but we should do it on the libavcodec mail thread. This
thread is on the command-line usage of ffmpeg.


On Tue, Jul 16, 2013 at 1:07 AM, James Board <jpboard2 at yahoo.com> wrote:

> >> I'd like to write a C or C++ program which uses libavcodec to retrieve
> >> individual frames of an AVI file.  I can't find much documentation on
> the WWW.
> >> Does anyone have documentation on how to do this, or some current
> examples?
> >
> >Did you look into the doc/examples directory?
> >
> >Please do not high jack threads, Carl Eugen
>
>
> No.  I only looked on the web, not in the doc/examples directory.
>
>
> What thread did I hijack?
>
> Thank you for the doc/examples pointer.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>



-- 
MD. Tanjim Hossain Sifat
Bangladesh University of Engineering and Technology,
Dhaka

From cehoyos at ag.or.at  Tue Jul 16 01:10:08 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 23:10:08 +0000 (UTC)
Subject: [FFmpeg-user] libavcodec documentation and usage
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
Message-ID: <loom.20130716T010918-419@post.gmane.org>

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

> >Please do not high jack threads, Carl Eugen

> What thread did I hijack?

http://thread.gmane.org/gmane.comp.video.ffmpeg.user/46634
http://www.google.com/search?q=thread+hijacking

Carl Eugen


From cehoyos at ag.or.at  Tue Jul 16 01:15:08 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 23:15:08 +0000 (UTC)
Subject: [FFmpeg-user] Help please !!!!! asf stream abort by ffserver
References: <CAFd2cZWfpYHyjBiV8AUpT+nzf-y1in3a-X9XaLP0DpydsfSd6Q@mail.gmail.com>
	<loom.20130715T151945-716@post.gmane.org>
	<CAFd2cZWwzsHq4CnBZpnhsiQae38F+9AJW5--R=wROKqG5U5sTw@mail.gmail.com>
Message-ID: <loom.20130716T011354-843@post.gmane.org>

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

> > (Are you sure pcm_s16le is supported in asf?)
>
> which audio format is supported as your opinion?

Sorry, apparently asf does support formats 
that are supported in wav.

[...]

> Assertion asf->packet_timestamp_end >=
> asf->packet_timestamp_start failed at libavformat/asfenc.c:643

Does this assertion only trigger with ffserver or 
also with ffmpeg?

Carl Eugen


From cehoyos at ag.or.at  Tue Jul 16 01:20:56 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 15 Jul 2013 23:20:56 +0000 (UTC)
Subject: [FFmpeg-user] Help Please
References: <A5C0165C-F3C2-4F90-9045-9AD34D7326AC@tigger.ws>
	<loom.20130715T151256-915@post.gmane.org>
	<F9D7A571-5F45-4950-AC32-DAC4CC84BB46@tigger.ws>
Message-ID: <loom.20130716T011911-849@post.gmane.org>

James Linder <jam <at> tigger.ws> writes:

> As soon as I add 
> -s 720x576 -f rawvideo - |/usr/bin/ffplay -f rawvideo 
> -i - -pix_fmy yuyv422 -s 720x576
> or indeed
> -s 720x576 -f rawvideo - -pix_fmt yuv420p  |
> /usr/bin/ffplay -f rawvideo -i - -s 720x576
> 
> BOTH the display and the recorded file are 
> [repeat_frame/ 1/2 frame new stuff every few 
> seconds /old frame repeated]

Sorry, I don't understand this sentence.
Consider starting with command line together 
with complete, uncut console output after 
testing if the same problem happens without 
the pipe.

Carl Eugen


From phil_rhodes at rocketmail.com  Tue Jul 16 01:41:14 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Mon, 15 Jul 2013 16:41:14 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <loom.20130716T010918-419@post.gmane.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
Message-ID: <1373931674.51294.YahooMailNeo@web121104.mail.ne1.yahoo.com>

> What thread did I hijack?


Some mailing lists use special message headers which allow clients which are aware of them to group the messages into threads. Personally I think if you want that you should use a web forum, but that's besides the point. This is such a list.

No, there is no way you could have been expected to know that. Yes, Carl's response is deliberately incomplete and unhelpful. Welcome to open source.?

P

From expandafter at yahoo.com  Tue Jul 16 02:16:15 2013
From: expandafter at yahoo.com (S. Fisher)
Date: Mon, 15 Jul 2013 17:16:15 -0700 (PDT)
Subject: [FFmpeg-user] Forcing ffmpeg to detect duration reliably
Message-ID: <1373933775.67420.YahooMailBasic@web163806.mail.gq1.yahoo.com>

tracey jaquith <tracey <at> archive.org> wrote:

> I use this for our archives when I get an estimated duration
> that seems wrong.  Much faster than the "ffmpeg -i input -f null
> -", though I'm sure there may be some pros/cons.
> 
> It uses "ffprobe" to "count video packets" and their durations,
> finally printing a total #seconds.
> 
> 
> ffprobe -select_streams v  -show_packets -print_format compact  *mpg  2>/dev/null | egrep -o 'duration_time=[^ ]+'|cut -f2 -d= | perl -ne 'chop; $n+=$_; print (int($n*10)/10)."\n" if (eof());'

Instead of using egrep, cut, and perl, it seems somewhat simpler
to use the small language awk.

ffprobe -select_streams v -show_packets -print_format compact foo.mp4 | mawk -F "duration_time=" "NF>1 {t+=$2} END{printf '%.1f', t}"


Tested in a DOS shell under windoze.


From soho123.2012 at gmail.com  Tue Jul 16 03:47:18 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 16 Jul 2013 09:47:18 +0800
Subject: [FFmpeg-user] Help please !!!!! asf stream abort by ffserver
In-Reply-To: <loom.20130716T011354-843@post.gmane.org>
References: <CAFd2cZWfpYHyjBiV8AUpT+nzf-y1in3a-X9XaLP0DpydsfSd6Q@mail.gmail.com>
	<loom.20130715T151945-716@post.gmane.org>
	<CAFd2cZWwzsHq4CnBZpnhsiQae38F+9AJW5--R=wROKqG5U5sTw@mail.gmail.com>
	<loom.20130716T011354-843@post.gmane.org>
Message-ID: <CAFd2cZUE1geW97XoO5oPsG7CWBUwwJ6matD+dUzRgV8ZvWVJrw@mail.gmail.com>

> > (Are you sure pcm_s16le is supported in asf?)
> >
> > which audio format is supported as your opinion?
>
> Sorry, apparently asf does support formats
> that are supported in wav.
>

So, the audio format like umlaw, a-law, does not support, too.Right?
And,
which encode is  recommend?
AAC? or other

>
> [...]
>
> > Assertion asf->packet_timestamp_end >=
> > asf->packet_timestamp_start failed at libavformat/asfenc.c:643
>
> Does this assertion only trigger with ffserver or
> also with ffmpeg?
>
> Currently , the test is trigger with ffserver only,
I do not have idea how to use ffmpeg to mux asf stream.
Could you kindly give me the hint to use ffmpeg to mux asf stream?

From soho123.2012 at gmail.com  Tue Jul 16 03:51:05 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 16 Jul 2013 09:51:05 +0800
Subject: [FFmpeg-user] HELP!HELP!Please!!!ffplay can not receive the rtp
	stream from udp socket
Message-ID: <CAFd2cZUSfugOQq1ikZP_kQeQ6JgSAk8OELrnhQ1hEBMK0n65pQ@mail.gmail.com>

Hi all,

I got the big trouble about :
ffplay can not play play h264 video + pcm_S16LE audio rtp stream.
originally, ffplay can play video stream if no audio.
after add audio part , then ffplay can not play.
It seems ffplay is wait rtp data arrived, but ffserver is sending rtp data
continuous to the udp port that rtsp negotiate
Does anyone have ideas?
the configuration of ffserver.conf:
==============================
========================
Port 8090
RTSPPort 5554
<Feed feed1.ffm>
File /tmp/feed1.ffm FileMaxSize 5M
ACL allow 127.0.0.1
</Feed>
<Stream test1-rtsp.mpg>
Feed feed1.ffm
Format rtp
AVOptionVideo flags +global_header
VideoSize 1280x720
VideoFrameRate 30
VideoCodec libx264
AVOptionAudio flags +global_header
AudioCodec pcm_s16le
AudioChannels 2
AudioSampleRate 48000
</Stream>
======================================================
the detail log :
======================================================
ffserver version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  9 2013 15:51:03 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 --dis
able-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --d
isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
--bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
--enable-libx2
64 --extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,copy,format,fps
,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
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
Tue Jul  9 20:57:52 2013 FFserver started.
ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul  9 2013 15:51:03 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 --dis
able-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --d
isable-mipsdspr2 --disable-mipsfpu --enable-small --prefix=/ffmpeg/romfs
--bindir=/ffmpeg/romfs --disable-bsfs --disable-filters --enable-gpl
--enable-libx2
64 --extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264'
--enable-filter='aformat,aresample,copy,format,fps
,framestep,resample' --disable-encoders --enable-encoder='mjpeg,h264,
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
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set libav*
logging level) with argument 'debu
g'.
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 a
rgument '30'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '1280x7
20'.
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 '-i' ... matched as input file with argument 'hw:0'.
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 '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with a
rgument 'copy'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with a
rgument '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 libav* 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
1280x720.
Successfully parsed a group of options.
Opening an input file: /dev/video1.
[video4linux2,v4l2 @ 0x69e240] fd:3 capabilities:4000001
[video4linux2,v4l2 @ 0x69e240] Selecting input_channel: 0
[video4linux2,v4l2 @ 0x69e240] input_channel: 0, input_name: Camera 1
[video4linux2,v4l2 @ 0x69e240] Setting time per frame to 1/30
[h264 @ 0x69ede0] Using externally provided dimensions
[h264 @ 0x69ede0] Frame num gap 18 16
[h264 @ 0x69ede0] Frame num gap 21 19
[h264 @ 0x69ede0] Frame num gap 27 25
[video4linux2,v4l2 @ 0x69e240] All info found
[video4linux2,v4l2 @ 0x69e240] Estimating duration from bitrate, this may
be inaccurate
Input #0, video4linux2,v4l2, from '/dev/video1':
  Duration: N/A, start: 24.690000, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 1280x720, 1/2000000,
-5 kb/s, 30 fps, 30 tbr, 1000k t
bn, 2000k tbc
Successfully opened the file.
Parsing a group of options: input file hw:0.
Applying option f (force format) with argument alsa.
Successfully parsed a group of options.
Opening an input file: hw:0.
[alsa @ 0x837ba0] codec_id ==NONE
[alsa @ 0x837ba0] All info found
[alsa @ 0x837ba0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #1.0 : stereo
Input #1, alsa, from 'hw:0':
  Duration: N/A, start: 1373403477.887958, 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://localhost:8090/feed1.ffm.
Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
Applying option ac (set number of audio channels) with argument 2.
Applying option acodec (force audio codec ('copy' to copy stream)) with
argument copy.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument copy.
Successfully parsed a group of options.
Opening an output file: http://localhost:8090/feed1.ffm.
[ffm @ 0x842e40] Format ffm probed with size=2048 and score=101
[AVIOContext @ 0x842700] Statistics: 4096 bytes read, 0 seeks
Tue Jul  9 20:57:57 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         : Lavf54.63.104
    Stream #0:0, 0, 1/1000000: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s
    Stream #0:1, 0, 1/1000000: Video: h264, yuv420p, 1280x720, 1/1000000,
q=2-31, -5 kb/s, 30 fps, 1000k t
bn, 1000k tbc
Stream mapping:
  Stream #1:0 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=   23 fps=0.0 q=-1.0 size=     136kB time=00:00:02.37 bitrate=
470.1kbits/s
    frame=   38 fps= 37 q=-1.0 size=     272kB time=00:00:02.87 bitrate=
776.4kbits/s
    frame=   53 fps= 35 q=-1.0 size=     388kB time=00:00:03.37 bitrate=
943.2kbits/s
    frame=   68 fps= 34 q=-1.0 size=     532kB time=00:00:03.87
bitrate=1126.1kbits/s
    frame=   83 fps= 33 q=-1.0 size=     740kB time=00:00:04.37
bitrate=1387.2kbits/s
    frame=   98 fps= 32 q=-1.0 size=     972kB time=00:00:04.87
bitrate=1635.0kbits/s
    frame=  113 fps= 32 q=-1.0 size=    1192kB time=00:00:05.37
bitrate=1818.4kbits/s
    frame=  128 fps= 32 q=-1.0 size=    1416kB time=00:00:05.87
bitrate=1976.1kbits/s
    frame=  143 fps= 32 q=-1.0 size=    1632kB time=00:00:06.37
bitrate=2098.8kbits/s
    frame=  158 fps= 31 q=-1.0 size=    1864kB time=00:00:06.87
bitrate=2222.7kbits/s
    frame=  173 fps= 31 q=-1.0 size=    2068kB time=00:00:07.37
bitrate=2298.7kbits/s
    frame=  188 fps= 31 q=-1.0 size=    2276kB time=00:00:07.87
bitrate=2369.1kbits/s
    frame=  203 fps= 31 q=-1.0 size=    2484kB time=00:00:08.37
bitrate=2431.2kbits/s
    frame=  218 fps= 31 q=-1.0 size=    2692kB time=00:00:08.87
bitrate=2486.2kbits/s
    frame=  233 fps= 31 q=-1.0 size=    2904kB time=00:00:09.37
bitrate=2538.9kbits/s
    frame=  248 fps= 31 q=-1.0 size=    3116kB time=00:00:09.87
bitrate=2586.2kbits/s
    frame=  263 fps= 31 q=-1.0 size=    3328kB time=00:00:10.37
bitrate=2629.0kbits/s
    frame=  278 fps= 31 q=-1.0 size=    3544kB time=00:00:10.87
bitrate=2670.9kbits/s
    frame=  293 fps= 31 q=-1.0 size=    3756kB time=00:00:11.37
bitrate=2706.2kbits/s
    frame=  308 fps= 31 q=-1.0 size=    3968kB time=00:00:11.87
bitrate=2738.5kbits/s
    frame=  323 fps= 31 q=-1.0 size=    4180kB time=00:00:12.37
bitrate=2768.2kbits/s
    Tue Jul  9 20:58:08 2013 192.168.1.3:34550 - - "PLAY
test1-rtsp.mpg/streamid=0 RTP/UDP"
Tue Jul  9 20:58:08 2013 192.168.1.3:34552 - - "PLAY
test1-rtsp.mpg/streamid=1 RTP/UDP"
frame=  338 fps= 31 q=-1.0 size=    4308kB time=00:00:12.87
bitrate=2742.1kbits/s
    frame=  353 fps= 31 q=-1.0 size=    4424kB time=00:00:13.37
bitrate=2710.7kbits/s
    frame=  368 fps= 31 q=-1.0 size=    4536kB time=00:00:13.87
bitrate=2679.1kbits/s
    frame=  383 fps= 31 q=-1.0 size=    4652kB time=00:00:14.37
bitrate=2652.0kbits/s
    frame=  398 fps= 31 q=-1.0 size=    4764kB time=00:00:14.87
bitrate=2624.5kbits/s
    frame=  413 fps= 31 q=-1.0 size=    4876kB time=00:00:15.37
bitrate=2598.8kbits/s
    ALSA buffer xrun.
Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]cannot find FFM syncword
Tue Jul  9 20:58:11 2013 [ffm @ 0x679a80]resyncing
frame=  428 fps= 31 q=-1.0 size=    5280kB time=00:00:15.87
bitrate=2725.5kbits/s
    frame=  443 fps= 30 q=-1.0 size=    5396kB time=00:00:16.37
bitrate=2700.3kbits/s
    frame=  458 fps= 30 q=-1.0 size=    5512kB time=00:00:16.87
bitrate=2676.6kbits/s
    frame=  473 fps= 30 q=-1.0 size=    5624kB time=00:00:17.37
bitrate=2652.4kbits/s
    find Key
frame=  488 fps= 30 q=-1.0 size=    5748kB time=00:00:17.87
bitrate=2635.0kbits/s
    ALSA buffer xrun.
frame=  503 fps= 30 q=-1.0 size=    6196kB time=00:00:18.37
bitrate=2763.1kbits/s
    ALSA buffer xrun.
frame=  509 fps= 26 q=-1.0 size=    6224kB time=00:00:18.57
bitrate=2745.7kbits/s
    frame=  520 fps= 23 q=-1.0 size=    6776kB time=00:00:21.37
bitrate=2597.5kbits/s
    ALSA buffer xrun.
frame=  537 fps= 22 q=-1.0 size=    6856kB time=00:00:24.47
bitrate=2295.2kbits/s
    ALSA buffer xrun.
frame=  561 fps= 22 q=-1.0 size=    7040kB time=00:00:27.30
bitrate=2112.5kbits/s
    frame=  576 fps= 22 q=-1.0 size=    7092kB time=00:00:27.80
bitrate=2089.8kbits/s
    frame=  591 fps= 22 q=-1.0 size=    7144kB time=00:00:28.30
bitrate=2068.0kbits/s
    frame=  606 fps= 22 q=-1.0 size=    7200kB time=00:00:28.80
bitrate=2048.0kbits/s
    frame=  621 fps= 23 q=-1.0 size=    7252kB time=00:00:29.30
bitrate=2027.6kbits/s
    ALSA buffer xrun.
frame=  636 fps= 23 q=-1.0 size=    7572kB time=00:00:29.80
bitrate=2081.5kbits/s
    frame=  652 fps= 23 q=-1.0 size=    7808kB time=00:00:30.34
bitrate=2108.2kbits/s
    frame=  667 fps= 23 q=-1.0 size=    8020kB time=00:00:30.84
bitrate=2130.3kbits/s
    frame=  682 fps= 23 q=-1.0 size=    8132kB time=00:00:31.34
bitrate=2125.6kbits/s
    frame=  697 fps= 23 q=-1.0 size=    8244kB time=00:00:31.84
bitrate=2121.1kbits/s
    frame=  712 fps= 23 q=-1.0 size=    8360kB time=00:00:32.34
bitrate=2117.7kbits/s
    frame=  726 fps= 23 q=-1.0 size=    8472kB time=00:00:32.80
bitrate=2115.9kbits/s
    frame=  741 fps= 24 q=-1.0 size=    8584kB time=00:00:33.30
bitrate=2111.7kbits/s
    ALSA buffer xrun.
frame=  756 fps= 24 q=-1.0 size=    8996kB time=00:00:33.80
bitrate=2180.3kbits/s
    frame=  771 fps= 24 q=-1.0 size=    9108kB time=00:00:34.30
bitrate=2175.3kbits/s
    frame=  786 fps= 24 q=-1.0 size=    9220kB time=00:00:34.80
bitrate=2170.4kbits/s
    frame=  802 fps= 24 q=-1.0 size=    9352kB time=00:00:35.34
bitrate=2167.8kbits/s
    frame=  816 fps= 24 q=-1.0 size=    9460kB time=00:00:35.80
bitrate=2164.7kbits/s
    frame=  831 fps= 24 q=-1.0 size=    9576kB time=00:00:36.30
bitrate=2161.1kbits/s
    ALSA buffer xrun.
frame=  846 fps= 24 q=-1.0 size=   10008kB time=00:00:36.80
bitrate=2227.9kbits/s
    frame=  861 fps= 24 q=-1.0 size=   10120kB time=00:00:37.30
bitrate=2222.6kbits/s
    frame=  876 fps= 24 q=-1.0 size=   10232kB time=00:00:37.80
bitrate=2217.5kbits/s
    frame=  891 fps= 24 q=-1.0 size=   10348kB time=00:00:38.30
bitrate=2213.3kbits/s
    Tue Jul  9 20:58:34 2013 192.168.1.3 - - [PLAY] "rtsp://
192.168.1.254:5554/test1-rtsp.mpg/ RTSP/1.0" 200 1
030
frame=  907 fps= 25 q=-1.0 size=   10460kB time=00:00:38.84
bitrate=2206.2kbits/s
    frame=  922 fps= 25 q=-1.0 size=   10572kB time=00:00:39.34
bitrate=2201.5kbits/s
    ALSA buffer xrun.
[ffm @ 0x842940] st:0 PTS: 37891681 DTS: 37891681 < 37892211 invalid,
clipping
frame=  937 fps= 25 q=-1.0 size=   10984kB time=00:00:39.84
bitrate=2258.6kbits/s
    frame=  952 fps= 25 q=-1.0 size=   11096kB time=00:00:40.34
bitrate=2253.3kbits/s
    qframe=  967 fps= 25 q=-1.0 size=   11208kB time=00:00:40.84
bitrate=2248.2kbits/s
    frame=  982 fps= 25 q=-1.0 size=   11324kB time=00:00:41.34
bitrate=2244.0kbits/s
    frame=  997 fps= 25 q=-1.0 size=   11444kB time=00:00:41.84
bitrate=2240.7kbits/s
    frame= 1012 fps= 25 q=-1.0 size=   11560kB time=00:00:42.34
bitrate=2236.6kbits/s
    ALSA buffer xrun.
frame= 1027 fps= 25 q=-1.0 size=   11988kB time=00:00:42.84
bitrate=2292.4kbits/s
    ALSA buffer xrun.
frame= 1029 fps= 23 q=-1.0 size=   12004kB time=00:00:42.90
bitrate=2292.2kbits/s
    frame= 1052 fps= 23 q=-1.0 size=   12572kB time=00:00:46.90
bitrate=2195.9kbits/s
    frame= 1067 fps= 23 q=-1.0 size=   12624kB time=00:00:47.40
bitrate=2181.8kbits/s

=============================================================================
ffplay log:
ffplay -loglevel debug rtsp://192.168.1.254:5554/test1-rtsp.mpg
ffplay version N-54362-ge0be3cb Copyright (c) 2003-2013 the FFmpeg
developers
  built on Jul  2 2013 22:15:59 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. 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
SDP:nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Title
c=IN IP4 0.0.0.0
t=0 0
a=tool:libavformat 54.63.104
m=audio 0 RTP/AVP 96
b=AS:1536
a=control:streamid=0
m=video 0 RTP/AVP 97
b=AS:-5
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=1;
sprop-parameter-sets=Z0JAH5ZUAoAtyA==,aM44gA==;
profile-level-id=42401F
a=control:streamid=1

[rtsp @ 02a64500] video codec set to: h264
[NULL @ 02a679e0] RTP Packetization Mode: 1
[NULL @ 02a679e0] Extradata set to 02a67ea0 (size: 22)!
[NULL @ 02a679e0] RTP Profile IDC: 42 Profile IOP: 40 Level: 1f
hello state=0 0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

From soho123.2012 at gmail.com  Tue Jul 16 04:37:30 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 16 Jul 2013 10:37:30 +0800
Subject: [FFmpeg-user] Help please !!!!! asf stream abort by ffserver
In-Reply-To: <loom.20130716T011354-843@post.gmane.org>
References: <CAFd2cZWfpYHyjBiV8AUpT+nzf-y1in3a-X9XaLP0DpydsfSd6Q@mail.gmail.com>
	<loom.20130715T151945-716@post.gmane.org>
	<CAFd2cZWwzsHq4CnBZpnhsiQae38F+9AJW5--R=wROKqG5U5sTw@mail.gmail.com>
	<loom.20130716T011354-843@post.gmane.org>
Message-ID: <CAFd2cZVnyQncQ=HsHvBpN3RAD3GZezU6dHNC7ZN8adi6qw7GqQ@mail.gmail.com>

Sorry, apparently asf does support formats
> that are supported in wav.
>
> I try the encode "wmav2" for audio format.
Then try to play the asf stream.

I do not get the Assertion, but both vlc and ffplay can not output the
sound normally.
Because the sound can not be recognized.
Do you have any idea?

From soho123.2012 at gmail.com  Tue Jul 16 07:24:18 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 16 Jul 2013 13:24:18 +0800
Subject: [FFmpeg-user] ffmpeg how to set stream layout
Message-ID: <CAFd2cZUe8WJ13D+XtUY6Cgr_au1-v33Dmzm23=fLoX2LrmHZGA@mail.gmail.com>

Hi All,

I use ffmpeg command to generate audio and video stream.
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
-i /dev/video1 -f alsa -ar 44100 -ac 2 -i hw:0 -vcodec copy -acodec copy
http://localhost:8090/feed1.ffm

I got the stream layout :
 Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 44100 Hz, stereo, 1411 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

Can I modify the layout to
Stream 0 is video , and stream 1 is audio?

From jpboard2 at yahoo.com  Tue Jul 16 14:57:38 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 16 Jul 2013 05:57:38 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <loom.20130715T205304-102@post.gmane.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
Message-ID: <1373979458.33899.YahooMailNeo@web163901.mail.gq1.yahoo.com>





________________________________
 From: Carl Eugen Hoyos <cehoyos at ag.or.at>
To: ffmpeg-user at ffmpeg.org 
Sent: Monday, July 15, 2013 2:54 PM
Subject: Re: [FFmpeg-user] libavcodec documentation and usage
 

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

> 
> I'd like to write a C or C++ program which uses libavcodec to retrieve
> individual frames of an AVI file.? I can't find much documentation on the WWW.
> Does anyone have documentation on how to do this, or some current examples?

Did you look into the doc/examples directory?

Please do not high jack threads, Carl Eugen




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

Okay, I tried to compile the examples in the doc/examples directory.? Compilation
failed because there's no channel_layout.h.? Is there a workaround?

Jim

From jpboard2 at yahoo.com  Tue Jul 16 14:58:39 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 16 Jul 2013 05:58:39 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373931674.51294.YahooMailNeo@web121104.mail.ne1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373931674.51294.YahooMailNeo@web121104.mail.ne1.yahoo.com>
Message-ID: <1373979519.84391.YahooMailNeo@web163903.mail.gq1.yahoo.com>

>> What thread did I hijack?

>
>
>Some mailing lists use special message headers which allow clients which are aware of them to group the messages into threads. Personally I think if >you want that you should use a web forum, but that's besides the point. This is such a list.
>
>No, there is no way you could have been expected to know that. Yes, Carl's response is deliberately incomplete and unhelpful. Welcome to open >source.?
>

Okay, well, what headers should I use for what topics?

From jpboard2 at yahoo.com  Tue Jul 16 15:02:32 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 16 Jul 2013 06:02:32 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <loom.20130716T010918-419@post.gmane.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
Message-ID: <1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>

>> >Please do not high jack threads, Carl Eugen

>
>> What thread did I hijack?
>
>http://thread.gmane.org/gmane.comp.video.ffmpeg.user/46634
>http://www.google.com/search?q=thread+hijacking
>
>Carl Eugen

I still don't understand.? How did I 'hijack' that thread?? My email had a 
completely different subject line, and it was a completely new topic.

From philip at turmel.org  Tue Jul 16 15:46:53 2013
From: philip at turmel.org (Phil Turmel)
Date: Tue, 16 Jul 2013 09:46:53 -0400
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
Message-ID: <51E54ECD.7030003@turmel.org>

Hi James,

On 07/16/2013 09:02 AM, James Board wrote:
>>>> Please do not high jack threads, Carl Eugen
> 
>>> What thread did I hijack?
>>
>> http://thread.gmane.org/gmane.comp.video.ffmpeg.user/46634
>> http://www.google.com/search?q=thread+hijacking
>>
>> Carl Eugen
> 
> I still don't understand.  How did I 'hijack' that thread?  My email had a 
> completely different subject line, and it was a completely new topic.

When you used "reply" to start an email to this list, your email client
inserted codes for "In-Reply-To:" and "References:" that point to that
email.  Changing the subject doesn't removes those codes.  So your email
has codes that say that it belongs to the topic "how to debug when
running ffplay", but you were talking about something else.

Many people are ignorant of this activity because they are using an
email client that creates these codes (per standard) but do not display
messages in threads accordingly.  You can guess what email client is the
biggest offender.

If you want to be a good citizen on mailing lists (not just this one),
you need to compose a new message to the list, not use reply.  If you
just want to avoid typing the list address, many clients allow you to
right-click on the address and choose "Compose Message To" or something
similar.

People like Carl who provide a great deal of help on mailing lists like
this are especially sensitive to this issue as they tend to rely on
modern mail clients that use the standard codes to organize the
different simultaneous conversations where they are involved.  Such
people usually ignore conversations they don't have expertise in (I do
this on the linux lists where I can help).  Telling you not to hijack is
a mutual benefit--you don't want your message accidentally missed
because it's buried inside a different thread.

Hope this helps,

Phil

From phil_rhodes at rocketmail.com  Tue Jul 16 15:58:43 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Tue, 16 Jul 2013 06:58:43 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373979519.84391.YahooMailNeo@web163903.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373931674.51294.YahooMailNeo@web121104.mail.ne1.yahoo.com>
	<1373979519.84391.YahooMailNeo@web163903.mail.gq1.yahoo.com>
Message-ID: <1373983123.30278.YahooMailNeo@web121105.mail.ne1.yahoo.com>

>?Okay, well, what headers should I use for what topics?

I have no idea; it's effectively impossible to post here without offending someone, somehow.

To your original point, I'd also like to use the libraries (to retrieve the first second or two of audio from various a/v files), but I too have been stymied by the almost complete lack of documentation. The unique parts of the libavcodec documentation on the ffmpeg site are three lines long, which seems a trifle inadequate for something weighing in at tens of kilobytes of source.

P

From phil_rhodes at rocketmail.com  Tue Jul 16 16:01:02 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Tue, 16 Jul 2013 07:01:02 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
Message-ID: <1373983262.32223.YahooMailNeo@web121105.mail.ne1.yahoo.com>

>?I still don't understand.? How did I 'hijack' that thread?? My email had a?
>?completely different subject line, and it was a completely new topic.

There are additional headers in some mailing-list emails that your email program probably doesn't show you, which identify the thread. If you post to the list by replying to a message from another thread, these headers can sometimes be duplicated into the new email, meaning you appear - to people whose email programs support all this - to "alter the subject" of a thread partway down.

These special "threading" headers are not all that widely supported, especially on the webmail providers such as Yahoo that a lot of people use, so again, there's no reason you should be expected to know about them.

P

From nicolas.george at normalesup.org  Tue Jul 16 16:06:22 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Tue, 16 Jul 2013 16:06:22 +0200
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
Message-ID: <20130716140622.GA17997@phare.normalesup.org>

L'octidi 28 messidor, an CCXXI, James Board a ?crit?:
> >http://thread.gmane.org/gmane.comp.video.ffmpeg.user/46634

> I still don't understand.?? How did I 'hijack' that thread??? My email had a
> completely different subject line, and it was a completely new topic.

Did you look at this link in Carl Eugen's message? You can see that your
first mail is shown as a reply to "Soho Soho123"'s mail "how to debug
running ffplay". In this particular case it is not a big issue because this
mail has no real reply of its own, but if some come they will be mixed with
this discussion.

Regards,

-- 
  Nicolas George
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From nicolas.george at normalesup.org  Tue Jul 16 15:06:29 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Tue, 16 Jul 2013 15:06:29 +0200
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373979519.84391.YahooMailNeo@web163903.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373931674.51294.YahooMailNeo@web121104.mail.ne1.yahoo.com>
	<1373979519.84391.YahooMailNeo@web163903.mail.gq1.yahoo.com>
Message-ID: <20130716130629.GA30592@phare.normalesup.org>

L'octidi 28 messidor, an CCXXI, James Board a ?crit?:
> Okay, well, what headers should I use for what topics?

In-Reply-To (and References). All mail user agents do that automatically,
you do not have to do anything except be careful how you use your mail user
agent: if you want to start a new discussion, you use the command to write a
brand new mail, and if you want to reply to a discussion you use the command
to reply to a mail; not the other way around.

The message you were answering to just now is a useless troll, the best is
to just ignore it and all similar messages.

Regards,

-- 
  Nicolas George
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From lucielemonnier at hotmail.fr  Tue Jul 16 17:27:42 2013
From: lucielemonnier at hotmail.fr (lucie Lemonnier)
Date: Tue, 16 Jul 2013 17:27:42 +0200
Subject: [FFmpeg-user] try to change complex filters on runtime
In-Reply-To: <20130712131632.GO4978@barisone>
References: <DUB111-W71942C66995E0A11E9D04BB97A0@phx.gbl>,
	<20130712131632.GO4978@barisone>
Message-ID: <DUB111-W119EDB52F533A6B46FA327EB9600@phx.gbl>

I try to use your patch to change parameters of crop, it works fine except when the input file has reached its end because the function filter_frame in vf_crop.c is not called when the input file has reached its end.
I try to find where the AVFilterLink is destroy, I found some lines of code that make me think that the link is destroyed but when I try to modify these lines, it doesn't work.
Can you help me, please?
Thanks.

> Date: Fri, 12 Jul 2013 15:16:32 +0200
> From: stefasab at gmail.com
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] try to change complex filters on runtime
> 
> On date Wednesday 2013-07-10 15:43:57 +0200, lucie Lemonnier wrote:
> [...]
> > For
> >  example , I call reinit_filters_complex("[0:0] crop=960:1080:0:0 
> > [tin];[2:0] crop=960:1080:0:0 [tmv]);[tin][tmv]overlay=0:0").
> > I 
> > passed in parameters of ffmpeg -i videoSource.mp4 -i transp.mov -i 
> > objets.png -filter_complex "[0:0] crop=960:1080:0:0 [tin];[1:0] 
> > crop=960:1080:0:0 [tmv]".
> > But It crash on this change.
> 
> > I think it's because I free all in filtergraphs.
> > But I don't know what is missing for it to work.
> > Can you help me, please?
> > Thanks.
> 
> You could achieve the same by extending libavfilter and using commands
> to send interactive commands to filters. For example check my patch on
> ffmpeg-devel to change crop parameters, as for what regards changing
> overlay you could connect two movie sources and disable/enable the
> active overlay (check timeline), or maybe implement a switcher
> controlled by a command. Dynamically changing movie input is not
> currently supported.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
 		 	   		  

From jpboard2 at yahoo.com  Tue Jul 16 18:05:36 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 16 Jul 2013 09:05:36 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <51E54ECD.7030003@turmel.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
	<51E54ECD.7030003@turmel.org>
Message-ID: <1373990736.92719.YahooMailNeo@web163902.mail.gq1.yahoo.com>



>When you used "reply" to start an email to this list, your email client
>inserted codes for "In-Reply-To:" and "References:" that point to that
>email.? Changing the subject doesn't removes those codes.? So your email
>has codes that say that it belongs to the topic "how to debug when
>running ffplay", but you were talking about something else.

Okay.? Yes, I forgot what the mailing list email address was, so I
merely replied to an existing email and changed the subject line.? I had
always thought that was enough to start a new thread, but was wrong.? 
I'll fix this in future emails.

From jpboard2 at yahoo.com  Tue Jul 16 19:22:58 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 16 Jul 2013 10:22:58 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <loom.20130715T205304-102@post.gmane.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
Message-ID: <1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>

>> I'd like to write a C or C++ program which uses libavcodec to retrieve

>> individual frames of an AVI file.? I can't find much documentation on the WWW.
>> Does anyone have documentation on how to do this, or some current examples?
>
>Did you look into the doc/examples directory?

I tried to compile those examples, but failed because it could not find
the file?
channel_layout.hWhat is the cause of the problem?? Is channel_layout.h obsolete and the
examples have not been updated?? Or was my build of ffmpeg broken 
(I built from source on the latest git checkout) and I have to recofigure
and rebuild ffmpeg?? Or is the problem something else?? I only need
to do simple things with the libavcodecs, so I think if I had a working
example, I can figure out how to code my applications.

Thanks.

ps. Sorry for problems with improper formatting. I'm getting the impression
that the yahoo email client sucks and is the cause of many of the problems.

From phil_rhodes at rocketmail.com  Tue Jul 16 19:55:55 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Tue, 16 Jul 2013 10:55:55 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
Message-ID: <1373997355.43499.YahooMailNeo@web121104.mail.ne1.yahoo.com>




>Did you look into the doc/examples directory?

As far as I'm aware none of those has worked for years.

P

From philip at turmel.org  Tue Jul 16 20:16:24 2013
From: philip at turmel.org (Phil Turmel)
Date: Tue, 16 Jul 2013 14:16:24 -0400
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373983262.32223.YahooMailNeo@web121105.mail.ne1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
	<1373983262.32223.YahooMailNeo@web121105.mail.ne1.yahoo.com>
Message-ID: <51E58DF8.1030304@turmel.org>

I should know better, but:

On 07/16/2013 10:01 AM, Phil Rhodes wrote:
>> I still don't understand.  How did I 'hijack' that thread?  My
>> email had a completely different subject line, and it was a
>> completely new topic.
> 
> There are additional headers in some mailing-list emails that your
> email program probably doesn't show you, which identify the thread.
> If you post to the list by replying to a message from another thread,
> these headers can sometimes be duplicated into the new email, meaning
> you appear - to people whose email programs support all this - to
> "alter the subject" of a thread partway down.

The headers in question are *not* mailing list headers.  They appear in
all email replies from every desktop email client out there, and all the
major webmail clients, whether to a mailing list or not.  Both James'
mail client and *your* mail client are generating these headers.

> These special "threading" headers are not all that widely supported,
> especially on the webmail providers such as Yahoo that a lot of
> people use, so again, there's no reason you should be expected to
> know about them.

That's simply not true.  These headers are extremely widely supported,
and have been from the days of Eudora and Outlook 1.0.  The exceptions
are a handful of old webmail packages for self-hosted mail servers that
fail to generate these headers.  Such mails always appear to start a new
thread, never to hijack a thread.

Some long-standing contributors may be short when attempting to educate
newcomers, but the posting rules for these mailings lists are published
(with explanations) in the very first paragraph on ffmpeg's contact page.

You've been around long enough to know better, Phil.  You may disagree
with how the ffmpeg runs things, but it is not up to you or I.

Regards,

Phil

{I know, I shouldn't feed the troll.  But leaving that without a
rebuttal would be bad for newcomers.}

From nicolas.george at normalesup.org  Tue Jul 16 20:25:52 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Tue, 16 Jul 2013 20:25:52 +0200
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1373997355.43499.YahooMailNeo@web121104.mail.ne1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<1373997355.43499.YahooMailNeo@web121104.mail.ne1.yahoo.com>
Message-ID: <20130716182552.GA16270@phare.normalesup.org>

L'octidi 28 messidor, an CCXXI, Phil Rhodes a ?crit?:
> As far as I'm aware none of those has worked for years.

As always, this is completely untrue.

Regards,

-- 
  Nicolas George
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From cehoyos at ag.or.at  Tue Jul 16 23:01:42 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 16 Jul 2013 21:01:42 +0000 (UTC)
Subject: [FFmpeg-user] libavcodec documentation and usage
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
Message-ID: <loom.20130716T230016-117@post.gmane.org>

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

> >Did you look into the doc/examples directory?
> 
> I tried to compile those examples, but failed 
> because it could not find the file?
> channel_layout.h
> What is the cause of the problem?

The examples expect your FFmpeg version to be 
installed (just as any of your own programs 
that you compiled) as opposed to the ffmpeg 
and the ffplay executable that can be built 
without installed libraries (I never install).

Carl Eugen


From nicolas.george at normalesup.org  Tue Jul 16 23:24:00 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Tue, 16 Jul 2013 23:24:00 +0200
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <loom.20130716T230016-117@post.gmane.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130716T230016-117@post.gmane.org>
Message-ID: <20130716212400.GA3771@phare.normalesup.org>

L'octidi 28 messidor, an CCXXI, Carl Eugen Hoyos a ?crit?:
> The examples expect your FFmpeg version to be 
> installed (just as any of your own programs 
> that you compiled) as opposed to the ffmpeg 
> and the ffplay executable that can be built 
> without installed libraries (I never install).

You probably have found the right explanation. An additional note: the
pc-uninstalled directory contains pkg-config files that allow to build
without installing. Point the PKG_CONFIG_PATH environment variable to this
directory and it should work.

Regards,

-- 
  Nicolas George
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From phil_rhodes at rocketmail.com  Wed Jul 17 00:49:06 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Tue, 16 Jul 2013 15:49:06 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <20130716182552.GA16270@phare.normalesup.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<1373997355.43499.YahooMailNeo@web121104.mail.ne1.yahoo.com>
	<20130716182552.GA16270@phare.normalesup.org>
Message-ID: <1374014946.33649.YahooMailNeo@web121106.mail.ne1.yahoo.com>

> As always, this is completely untrue.

Well, that's not the case.?

Presumably the whole "The examples expect your FFmpeg version to be?
installed" thing that Carl Eugen tells us about is documented somewhere?

Oh no wait.

P

From phil_rhodes at rocketmail.com  Wed Jul 17 01:07:48 2013
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Tue, 16 Jul 2013 16:07:48 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <51E58DF8.1030304@turmel.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
	<1373983262.32223.YahooMailNeo@web121105.mail.ne1.yahoo.com>
	<51E58DF8.1030304@turmel.org>
Message-ID: <1374016068.1713.YahooMailNeo@web121103.mail.ne1.yahoo.com>

>?That's simply not true.? These headers are extremely widely supported,
>?and have been from the days of Eudora and Outlook 1.0.??

How do I make Yahoo Mail display threads, then?

Look; the reason I pick people up on this is precisely and absolutely because the prevailing attitude is bad for newcomers. People are deliberately and provocatively incomplete and give deliberately bad advice (break lines at 70 characters? What is this, 1976?), but the point is that all of this is a smokescreen for the real issue: this has become a mailing list thread about email etiquette, not libavcodec documentation.

Now, it can't really become a thread about libavcodec documentation because to all practical purposes there isn't any, and the people responsible are presumably embarrassed about that (I would be). So, they make it a thread about stomping on the new guy, asserting their technical knowledge, boasting, and busily fulfilling every available cliche about how unapproachable and difficult FOSS projects can be.

And I'm the troll? Drivel. The only person who actually offered the poor guy a human-readable explanation of why he'd pressed Carl Eugen's buttons was ME.

P

From jpboard2 at yahoo.com  Wed Jul 17 01:25:25 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 16 Jul 2013 16:25:25 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <loom.20130716T230016-117@post.gmane.org>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130716T230016-117@post.gmane.org>
Message-ID: <1374017125.3616.YahooMailNeo@web163902.mail.gq1.yahoo.com>

> The examples expect your FFmpeg version to be 

> installed (just as any of your own programs 
> that you compiled) as opposed to the ffmpeg 
> and the ffplay executable that can be built 
> without installed libraries (I never install).

Okay.? Thanks for your advice.

I did a 'make install' of ffmpeg, as Carl suggested, and after that, I tried
to build the libavcodec examples in doc/examples.? I received lots of
'undefined reference' errors after running 'make' for these subroutines:

??? av_get_channel_layout_nb_channels avcodec_alloc_context3
??? av_get_sample_fmt_name av_get_channel_layout_nb_channels avcodec_open2
??? av_samples_get_buffer_size avcodec_fill_audio_frame avcodec_encode_audio2
??? avcodec_encode_audio2 avcodec_free_frame avcodec_alloc_context3 avcodec_open2
??? avcodec_decode_audio4 av_samples_get_buffer_size avcodec_free_frame
??? avcodec_alloc_context3 av_opt_set avcodec_open2 av_image_alloc
??? avcodec_encode_video2 avcodec_encode_video2 avcodec_free_frame
??? avcodec_alloc_context3 avcodec_open2 avcodec_free_frame

I'm able to fix that.? The compiler didn't know where to look for the
libraries so I added -L/usr/local/lib to the compile command line.

However, after that, I received about 100 additional 'undefined reference'
error messages for these subroutines:
??? compress compress2 crc32 deflate deflateBound deflateEnd deflateInit2_
??? deflateInit_ deflateReset inflate inflateEnd inflateInit_ inflateReset
??? inflateSync pthread_create pthread_join uncompress

I don't know how to fix that.? What can I do to get the examples in
doc/examples in the ffmpeg source code to compile?

From jpboard2 at yahoo.com  Wed Jul 17 01:38:02 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 16 Jul 2013 16:38:02 -0700 (PDT)
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1374017125.3616.YahooMailNeo@web163902.mail.gq1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130716T230016-117@post.gmane.org>
	<1374017125.3616.YahooMailNeo@web163902.mail.gq1.yahoo.com>
Message-ID: <1374017882.21003.YahooMailNeo@web163903.mail.gq1.yahoo.com>

> What can I do to get the examples in

> doc/examples in the ffmpeg source code to compile?


Never mind.? I did the following and then the examples compiled:
??? setenv PKG_CONFIG_PATH /usr/local/lib/pkgconfig

From stefasab at gmail.com  Wed Jul 17 01:40:26 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Wed, 17 Jul 2013 01:40:26 +0200
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1374014946.33649.YahooMailNeo@web121106.mail.ne1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373995378.55618.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<1373997355.43499.YahooMailNeo@web121104.mail.ne1.yahoo.com>
	<20130716182552.GA16270@phare.normalesup.org>
	<1374014946.33649.YahooMailNeo@web121106.mail.ne1.yahoo.com>
Message-ID: <20130716234026.GD3431@barisone>

On date Tuesday 2013-07-16 15:49:06 -0700, Phil Rhodes wrote:
> > As always, this is completely untrue.
> 
> Well, that's not the case.?
> 

> Presumably the whole "The examples expect your FFmpeg version to be?
> installed" thing that Carl Eugen tells us about is documented somewhere?

Yes in the README file in the same directory.

In general the examples user is expected to have some familiarity with
the building process and the usual Linux/Unix conventions and with
some of its subtleties, which are not specific of FFmpeg. These days
people tend to have several installations of FFmpeg in the same system
and in different locations, and selecting the right one is usually
confusing to the unexperienced user.
-- 
FFmpeg = Furious Formidable Mean Peaceless Elitarian Gargoyle

From jam at tigger.ws  Wed Jul 17 01:56:56 2013
From: jam at tigger.ws (James Linder)
Date: Wed, 17 Jul 2013 07:56:56 +0800
Subject: [FFmpeg-user] libavcodec documentation and usage
In-Reply-To: <1374016068.1713.YahooMailNeo@web121103.mail.ne1.yahoo.com>
References: <CAFd2cZWJJKN+cAnHb3wmG8Ecz_UYYPkHXOy_KbvMbGdB79CGtA@mail.gmail.com>
	<1373907413.51076.YahooMailNeo@web163904.mail.gq1.yahoo.com>
	<loom.20130715T205304-102@post.gmane.org>
	<1373915258.45486.YahooMailNeo@web163905.mail.gq1.yahoo.com>
	<loom.20130716T010918-419@post.gmane.org>
	<1373979752.6624.YahooMailNeo@web163906.mail.gq1.yahoo.com>
	<1373983262.32223.YahooMailNeo@web121105.mail.ne1.yahoo.com>
	<51E58DF8.1030304@turmel.org>
	<1374016068.1713.YahooMailNeo@web121103.mail.ne1.yahoo.com>
Message-ID: <D02D8339-B03D-4D8E-9A56-02C5DBE9C752@tigger.ws>


On 17/07/2013, at 7:07 AM, Phil Rhodes wrote:

>>  That's simply not true.  These headers are extremely widely supported,
>>  and have been from the days of Eudora and Outlook 1.0.  
> 
> How do I make Yahoo Mail display threads, then?
> 
> Look; the reason I pick people up on this is precisely and absolutely because the prevailing attitude is bad for newcomers. People are deliberately and provocatively incomplete and give deliberately bad advice (break lines at 70 characters? What is this, 1976?), but the point is that all of this is a smokescreen for the real issue: this has become a mailing list thread about email etiquette, not libavcodec documentation.
> 
> Now, it can't really become a thread about libavcodec documentation because to all practical purposes there isn't any, and the people responsible are presumably embarrassed about that (I would be). So, they make it a thread about stomping on the new guy, asserting their technical knowledge, boasting, and busily fulfilling every available cliche about how unapproachable and difficult FOSS projects can be.
> 
> And I'm the troll? Drivel. The only person who actually offered the poor guy a human-readable explanation of why he'd pressed Carl Eugen's buttons was ME.

Carl needs to say the same thing 10 times a week he can be excused for being terce.
I thought that Phil's comments were kind and nice for someone who did not understand the nuances.
James

From ajsinclair at gmail.com  Wed Jul 17 05:09:17 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Wed, 17 Jul 2013 13:09:17 +1000
Subject: [FFmpeg-user] stream server solution
In-Reply-To: <CAFd2cZUNAz8JjoU1bbHa8-Zu+-i8Yxm_Tk7wZxSZNna0Jg_F+w@mail.gmail.com>
References: <CAFd2cZUNAz8JjoU1bbHa8-Zu+-i8Yxm_Tk7wZxSZNna0Jg_F+w@mail.gmail.com>
Message-ID: <CAOeAR14oi+2gL4RAk9pHA2ToZHEQ03kQt=8a+YHSwuJV51sE8A@mail.gmail.com>

On 16 July 2013 01:18, Soho Soho123 <soho123.2012 at gmail.com> wrote:

> Hi All,
>
> I use ffserver + ffmpeg for the stream server application.
> but it seems the solution does not popular.
> Does anyone have idea about stream server that can run in linux OS?
>
> As my understanding, ffmeg can send stream to the specific client host.
> but it can not provide the service for multiple client with unicast packet.
> Any ideas?
>
>  It depends on what kind of clients are consuming your video, some options
(there are many others)

Flash video: ffmpeg rtmp -> wowza media server
Multicast/iptv: ffmpeg udp -> network supporting multicast
iOS/Android: ffmpeg hls -> filesystem + Apache webserver

From digamma2function at gmail.com  Wed Jul 17 09:49:36 2013
From: digamma2function at gmail.com (Feng Duan)
Date: Wed, 17 Jul 2013 15:49:36 +0800
Subject: [FFmpeg-user] How to use mpegvideo_xvmc in ffmpeg?
Message-ID: <CACWOFwm3eRtBN-mRLg15uAv-9TfGOBQjir_Zh2RbHAQMVmYwnQ@mail.gmail.com>

Hi,

I run the mpegvideo_xvmc to test the GPU decoder.
I use the command as follow:
ffmpeg -vcodec mpegvideo_xvmc -i input.mpg -y output.m2ts

And error is in attachment.

Best Regards

Feng
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From cehoyos at ag.or.at  Wed Jul 17 10:31:30 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 17 Jul 2013 08:31:30 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?How_to_use_mpegvideo=5Fxvmc_in_ffmpeg=3F?=
References: <CACWOFwm3eRtBN-mRLg15uAv-9TfGOBQjir_Zh2RbHAQMVmYwnQ@mail.gmail.com>
Message-ID: <loom.20130717T102743-995@post.gmane.org>

Feng Duan <digamma2function <at> gmail.com> writes:

> ffmpeg -vcodec mpegvideo_xvmc -i input.mpg -y output.m2ts

I did not look at your attachment (and the error message 
you see) but the xvmc decoder cannot be used from ffmpeg 
(the application), it is used with the following MPlayer 
command line:
$ mplayer -vo xvmc -vc ffmpeg12mc
(and nowadays mplayer -vo xvmc -vc ffmpeg12)

I suspect it does not only work with MPlayer but that 
was the application it was originally written for.

Since it does not work with new (and middle-aged) 
Nvidia hardware and does only support MPEG1/2, it is 
often not (anymore) a recommended way of decoding.

Carl Eugen


From digamma2function at gmail.com  Wed Jul 17 13:16:38 2013
From: digamma2function at gmail.com (Feng Duan)
Date: Wed, 17 Jul 2013 19:16:38 +0800
Subject: [FFmpeg-user] How to use mpegvideo_xvmc in ffmpeg?
In-Reply-To: <loom.20130717T102743-995@post.gmane.org>
References: <CACWOFwm3eRtBN-mRLg15uAv-9TfGOBQjir_Zh2RbHAQMVmYwnQ@mail.gmail.com>
	<loom.20130717T102743-995@post.gmane.org>
Message-ID: <CACWOFwnDi9A1ShFYo4PN0UjE_XaE7RUo7thSvSUNJFR4gOE-iQ@mail.gmail.com>

Hi,

Thanks for your reply, I add the report at the attachment.
I use the application to do the work. As you said ,the xvmc decoder cannot
be used.

If I want do some work to support the xvmc decoder on XVMC_MPEG2_IDCT
format, How could I do to begin the work?

Best Regards

Feng


On Wed, Jul 17, 2013 at 4:31 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Feng Duan <digamma2function <at> gmail.com> writes:
>
> > ffmpeg -vcodec mpegvideo_xvmc -i input.mpg -y output.m2ts
>
> I did not look at your attachment (and the error message
> you see) but the xvmc decoder cannot be used from ffmpeg
> (the application), it is used with the following MPlayer
> command line:
> $ mplayer -vo xvmc -vc ffmpeg12mc
> (and nowadays mplayer -vo xvmc -vc ffmpeg12)
>
> I suspect it does not only work with MPlayer but that
> was the application it was originally written for.
>
> Since it does not work with new (and middle-aged)
> Nvidia hardware and does only support MPEG1/2, it is
> often not (anymore) a recommended way of decoding.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
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From jam at tigger.ws  Wed Jul 17 14:43:55 2013
From: jam at tigger.ws (James Linder)
Date: Wed, 17 Jul 2013 20:43:55 +0800
Subject: [FFmpeg-user] Help Please
In-Reply-To: <A5C0165C-F3C2-4F90-9045-9AD34D7326AC@tigger.ws>
References: <A5C0165C-F3C2-4F90-9045-9AD34D7326AC@tigger.ws>
Message-ID: <FBCDC352-6699-4AE3-A400-2406C06A6645@tigger.ws>

I discovered that this is all due to faulty hardware!
James

On 15/07/2013, at 8:44 PM, James Linder wrote:

> If any kind soul can show me the error of my ways I'd appreciate it lots:
> 
> I want to record audio and video to a file
> I want to display the video as it is being recorded.
> 
> This has worked with different v4l hardware :-(
> 
> I record:
> /usr/bin/ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 -i /dev/video0 -vtag divx -s 720x576 -r pal -q:v 5 mon.avi -s 720x576 -f rawvideo - |/usr/bin/ffplay -f rawvideo -i - -s 720x576
> 
> The rawvideo size has no effect.
> 
> I get:
> ffplay version N-54580-g694550f Copyright (c) 2003-2013 the FFmpeg developers
>  built on Jul 10 2013 11:43:58 with gcc 4.7 (SUSE Linux)
>  configuration: --enable-libv4l2 --enable-libpulse
>  libavutil      52. 38.100 / 52. 38.100
>  libavcodec     55. 18.102 / 55. 18.102
>  libavformat    55. 11.101 / 55. 11.101
>  libavdevice    55.  2.100 / 55.  2.100
>  libavfilter     3. 79.100 /  3. 79.100
>  libswscale      2.  3.100 /  2.  3.100
>  libswresample   0. 17.102 /  0. 17.102
> Option -s is deprecated, use -video_size.
> ffmpeg version N-54580-g694550f Copyright (c) 2000-2013 the FFmpeg developers
>  built on Jul 10 2013 11:43:58 with gcc 4.7 (SUSE Linux)
>  configuration: --enable-libv4l2 --enable-libpulse
>  libavutil      52. 38.100 / 52. 38.100
>  libavcodec     55. 18.102 / 55. 18.102
>  libavformat    55. 11.101 / 55. 11.101
>  libavdevice    55.  2.100 / 55.  2.100
>  libavfilter     3. 79.100 /  3. 79.100
>  libswscale      2.  3.100 /  2.  3.100
>  libswresample   0. 17.102 /  0. 17.102
> Guessed Channel Layout for  Input Stream #0.0 : stereo
> Input #0, pulse, from 'default':
>  Duration: N/A, start: 0.010790, bitrate: 1536 kb/s
>    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
> Input #1, video4linux2,v4l2, from '/dev/video0':B sq=    0B f=0/0   
>  Duration: N/A, start: 1373874168.744534, bitrate: 165888 kb/s
>    Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x576, 165888 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
> File 'mon.avi' already exists. Overwrite ? [y/N]     nan    :  0.000 fd=   0 aq=    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0    
> Output #0, avi, to 'mon.avi':
>  Metadata:
>    ISFT            : Lavf55.11.101
>    Stream #0:0: Video: mpeg4 (divx / 0x78766964), yuv420p, 720x576, q=2-31, 200 kb/s, 25 tbn, 25 tbc
>    Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp, 192 kb/s
> Output #1, rawvideo, to 'pipe:':
>  Metadata:
>    encoder         : Lavf55.11.101
>    Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 800x600, q=2-31, 200 kb/s, 90k tbn, 25 tbc
> Stream mapping:
>  Stream #1:0 -> #0:0 (rawvideo -> mpeg4)
>  Stream #0:0 -> #0:1 (pcm_s16le -> ac3)
>  Stream #1:0 -> #1:0 (rawvideo -> rawvideo)
> Press [q] to stop, [?] for help
> Input #0, rawvideo, from 'pipe:':
>  Duration: N/A, start: 0.000000, bitrate: 144000 kb/s
>    Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 800x600, 144000 kb/s, 25 tbr, 25 tbn, 25 tbc
> frame=   16 fps=0.0 q=5.0 q=0.0 size=     155kB time=00:00:00.64 bitrate=1981.5k  10.36 M-V: -0.002 fd=   0 aq=    0KB vq= 4219KB sq=    0B f=0/0
> 
> My live picture is garbled/all one colour/lines
> The recorded file is perfect
> 
> I get even more confused, I try to use tee
> 
> /usr/bin/ffmpeg -f pulse -itsoffset 0.2 -i default -f v4l2 -i /dev/video0 -vtag divx -s 720x576 -r pal -q:v 5 -f tee -map a:0 -map v:0 -c copy mon.avi |mplayer -
> 
> and get:
> ffmpeg version N-54580-g694550f Copyright (c) 2000-2013 the FFmpeg developers
>  built on Jul 10 2013 11:43:58 with gcc 4.7 (SUSE Linux)
>  configuration: --enable-libv4l2 --enable-libpulse
>  libavutil      52. 38.100 / 52. 38.100
>  libavcodec     55. 18.102 / 55. 18.102
>  libavformat    55. 11.101 / 55. 11.101
>  libavdevice    55.  2.100 / 55.  2.100
>  libavfilter     3. 79.100 /  3. 79.100
>  libswscale      2.  3.100 /  2.  3.100
>  libswresample   0. 17.102 /  0. 17.102
> MPlayer dev-SVN-r35127-4.7-openSUSE Linux 12.3 (x86_64)-Packman (C) 2000-2012 MPlayer Team
> Can't open joystick device /dev/input/js0: No such file or directory
> Can't init input joystick
> mplayer: could not connect to socket
> mplayer: No such file or directory
> Failed to open LIRC support. You will not be able to use your remote control.
> 
> Playing -.
> Reading from stdin...
> Guessed Channel Layout for  Input Stream #0.0 : stereo
> Input #0, pulse, from 'default':
>  Duration: N/A, start: 0.009793, bitrate: 1536 kb/s
>    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
> Input #1, video4linux2,v4l2, from '/dev/video0':
>  Duration: N/A, start: 1373877364.072402, bitrate: 165888 kb/s
>    Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x576, 165888 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
> Stream map 'v:0' matches no streams.
> libavformat version 54.25.104 (internal)
> 
> I don't understand why tee does not do what I expect
> I don't understand why rawvideo from one v4l device works fine, another does not
> 
> Thanks
> James
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From lucielemonnier at hotmail.fr  Wed Jul 17 18:38:32 2013
From: lucielemonnier at hotmail.fr (lucie Lemonnier)
Date: Wed, 17 Jul 2013 18:38:32 +0200
Subject: [FFmpeg-user] loop option on movie filter doesn't work
Message-ID: <DUB111-W98221409B00AD1668C9631B9610@phx.gbl>

Hello,
I try to use the loop option in the movie filter but it doesn't work :
ffmpeg -i source.mpg -vf "movie=transp.mov:loop=10 [mv];[in][mv] overlay=0:0".
I found this ticket :
https://ffmpeg.org/trac/ffmpeg/ticket/1799
Has been there any changes on this issue?
Does someone could help me?
Thanks.
Lucie
 		 	   		  

From lucian.orasanu at gmail.com  Wed Jul 17 11:05:34 2013
From: lucian.orasanu at gmail.com (Orasanu Lucian)
Date: Wed, 17 Jul 2013 09:05:34 +0000
Subject: [FFmpeg-user] Artefacts when recording an rtsp stream "Working
	perfect with openrtsp"
Message-ID: <1738812.iDgqYui39v@laptop>

Hello,

	I used live555's opertsp utility to capture from camera, here is a video 
file recorded with openrtsp from camera 
http://zm.realfield.ro/openrtsp_dump.avi.
with openrtsp video captured from camera is crystal clear without artefacts,
so camera has no network problems, I had to increase buffer size to work 
perfectly with openrtsp. So droping packets is comming from ffmpeg code,
maybe if I increase buffer size on ffmpeg will work ??? and how?

Thank you.

From aseto97 at yahoo.com  Wed Jul 17 22:10:03 2013
From: aseto97 at yahoo.com (Andrew Seto)
Date: Wed, 17 Jul 2013 13:10:03 -0700 (PDT)
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <1374087114.79043.YahooMailNeo@web121801.mail.ne1.yahoo.com>
References: <BEED9B6344438248AE283ABEC3662C4718900C705B@AICEXMBXCL02.LGE.NET>
	<1374087114.79043.YahooMailNeo@web121801.mail.ne1.yahoo.com>
Message-ID: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>



Hi,


I have the same ffmpeg problem that described in ticket 1734 http://trac.ffmpeg.org/ticket/1734
I downloaded the latest 2.0 release, but the problem is still there.
Can you tell me what I may be doing wrong? Below is my debug log.

Thanks
Andy


ffmpeg -v 9 -loglevel 99 -rtsp_transport tcp -i rtsp://root:root at 192.168.8.132:554/axis-media/media.amp?streamprofile=Quality?tcp -vcodec copy -acodec an -t 4 /tmp/media/axis.mp4? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
? built on Jul 17 2013 13:31:38 with gcc 4.3.2 (GCC)
? configuration: --arch=ppc --cpu=e500v2 --target-os=linux --samples=fate-suite/ --logfile=./config.log --disable-optimizations --disable-stripping --enable-cross-compile --cross-prefix=powerpc-none-linux-gnuspe- --sysroot=/opt/freescale/usr/local/gcc-4.3.74-eglibc-2.8.74-dp-2/powerpc-none-linux-gnuspe/powerpc-none-linux-gnuspe/libc --target-path=/opt/projects/pcs200/ltib/rootfs
? 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 '-v' ... matched as option 'v' (set logging level) with argument '9'.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'.
Reading option '-rtsp_transport' ... matched as AVOption 'rtsp_transport' with argument 'tcp'.
Reading option '-i' ... matched as input file with argument 'rtsp://root:root at 192.168.8.132:554/axis-media/media.amp?streamprofile=Quality?tcp'.
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 'an'.
Reading option '-t' ... matched as option 't' (record or transcode
 "duration" seconds of audio/video) with argument '4'.
Reading option '/tmp/media/axis.mp4' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option v (set logging level) with argument 9.
Successfully parsed a group of options.
Parsing a group of options: input file rtsp://root:root at 192.168.8.132:554/axis-media/media.amp?streamprofile=Quality?tcp.
Successfully parsed a group of options.
Opening an input file: rtsp://root:root at 192.168.8.132:554/axis-media/media.amp?streamprofile=Quality?tcp.
[rtsp @ 0x112b0d60] SDP:
v=0
o=- 1365518175096287 1365518175096287 IN IP4 192.168.8.132
s=Media Presentation
e=NONE
b=AS:50000
t=0
 0
a=control:rtsp://192.168.8.132:554/axis-media/media.amp?streamprofile=Quality?tcp
a=range:npt=0.000000-
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
b=AS:50000
a=framerate:30.0
a=transform:1,0,0;0,1,0;0,0,1
a=control:rtsp://192.168.8.132:554/axis-media/media.amp/trackID=1?streamprofile=Quality?tcp
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1; profile-level-id=4D4029; sprop-parameter-sets=Z01AKZpmAoAt2AtQEBAQXpw=,aO48gA==

[rtsp @ 0x112b0d60] video codec set to: h264
[NULL @ 0x112b34f0] RTP Packetization Mode: 1
[NULL @ 0x112b34f0] RTP Profile IDC: 4d Profile IOP: 40 Level: 29
[NULL @ 0x112b34f0] Extradata set to 0x112b3a60 (size: 29)!
[rtsp @ 0x112b0d60] hello state=0
[h264 @ 0x112b34f0] no picture 
[rtsp @ 0x112b0d60] All info found
rfps: 29.583333 0.015255
rfps: 29.666667 0.009949
rfps: 29.750000 0.005794
rfps: 29.833333 0.002789
rfps: 29.916667 0.000936
rfps: 30.000000
 0.000233
rfps: 30.083333 0.000682
? ? Last message repeated 1 times
rfps: 30.166667 0.002281
? ? Last message repeated 1 times
rfps: 30.250000 0.005032
? ? Last message repeated 1 times
rfps: 30.333333 0.008933
? ? Last message repeated 1 times
rfps: 30.416667 0.013985
? ? Last message repeated 1 times
rfps: 59.583333 0.016590
rfps: 59.666667 0.011157
rfps: 59.750000 0.006874
rfps: 59.833333 0.003743
rfps: 59.916667 0.001763
rfps: 60.000000 0.000933
rfps: 29.970030 0.000353
rfps: 59.940060 0.001414
Input #0, rtsp, from 'rtsp://root:root at 192.168.8.132:554/axis-media/media.amp?streamprofile=Quality?tcp':
? Metadata:
? ? title? ? ? ? ?  : Media Presentation
? Duration: N/A, start: 0.034389, bitrate: N/A
? ?
 Stream #0:0, 23, 1/90000: Video: h264 (Main), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 1/180000, 30 tbr, 90k tbn, 180k tbc
Successfully opened the file.
Parsing a group of options: output file /tmp/media/axis.mp4.
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 an.
Applying option t (record or transcode "duration" seconds of audio/video) with argument 4.
Successfully parsed a group of options.
Opening an output file: /tmp/media/axis.mp4.
File '/tmp/media/axis.mp4' already exists. Overwrite ? [y/N] y
Successfully opened the file.
Output #0, mp4, to '/tmp/media/axis.mp4':
? Metadata:
? ? title? ? ? ? ?  : Media Presentation
? ? encoder? ? ? ?  : Lavf55.12.100
? ? Stream #0:0, 0, 1/90000: Video: h264 ([33][0][0][0] /
 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 1/90000, q=2-31, 90k tbn, 90k tbc
Stream mapping:
? Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x112cdbc0] pts has no value
[mp4 @ 0x112cdbc0] pts (0) < dts (1) in stream 0
av_interleaved_write_frame(): Invalid argument
[AVIOContext @ 0x112cdac0] Statistics: 0 seeks, 2 writeouts

From iskilian at hotmail.com  Wed Jul 17 22:33:50 2013
From: iskilian at hotmail.com (S Kilian)
Date: Wed, 17 Jul 2013 16:33:50 -0400
Subject: [FFmpeg-user] Issue with Hyperdeck and Apple ProRes media created
	by FFMPEG
Message-ID: <BAY178-W85327E9198E50DEAB5F2ADD610@phx.gbl>

Hello,

 

I am having problems with generating Apple Pro Res content to play on Blackmagic's Hyperdeck Pro. The problem is that the video will appear choppy and left and right video will be swapped when I try to play content generated by FFMPEG's prores or prores_ks. The swapped video will be split vertically at the half screen or 2/3 screen point and then the left will be on the right and right will be left.

 

The Blackmagic Hyperdeck will only play files that are tagged as high quality (apch).

 

I think the problem could be solved if a working file was analyzed by an expert (and I'm far from being that expert). I have a 10 second sample (266 MB) that I can upload to somewhere.


Using:
ffmpeg version N-54653-g531f016 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 11 2013 09:48:15 with gcc 4.5.3 (GCC)
  configuration:
  libavutil      52. 39.100 / 52. 39.100
  libavcodec     55. 18.102 / 55. 18.102
  libavformat    55. 12.101 / 55. 12.101
  libavdevice    55.  3.100 / 55.  3.100
  libavfilter     3. 80.100 /  3. 80.100
  libswscale      2.  3.100 /  2.  3.100
  libswresample   0. 17.102 /  0. 17.102


Here are some of the things I have tried but all display the same issue.

00. General command: ffmpeg -i audio_in -i video_in -a:c pcm_s16le -c:v prores -profile:v 3 -pix_fmt yuv422p10le output
01. prores_ks and prores encoders
02. Modified code so that all profiles will tag as "apch" and tried all profiles
03. Set prores_ks' bits_per_mb from 128 to 7000. (128, 256, ... 768, 1000, 2000 ... 7000)
04. Set prores_ks' mbs_per_slice from 1 to 8 (1, 2, 4 and 8)
05. Set prores_ks' alpha_bits (0, 8 and 16)
06. Using prores_ks tried many qscale:v values (0, 1, 3, 5, 10, 15, 20, 25 and 30)
07. Tried -threads option at 8 and 16
08. Tried -timecode 00:00:00:00
09: Tried -filter_complex asetnsamples=n-16384:p=0
10: Tried pcm_s16le and pcm_s24le audio sample depth
11: Tried input jpeg video files
12: Tried different video rates (25, 50, 30000/1001, 60000/1001
13: Tried different video sizes (480, 720, 1080)
14: Tried creating separate audio pcm_s16le & video prores_ks files and then tried audio/video copy command to create final file
xx: Tried numerous other options

 

Notes:
 - Using video codec copy will play properly if video was generated from Final Cut Pro 7
 - No issues playing FFMPEG generated static video (color bars / no movement) with the Hyperdeck
 - The Blackmagic tech support was not helpful. He though it might be a "timing" or "pull down" issue.

 - Didn't see any differences other than data rate with ffprobe between Final Cut Pro file and FFMPEG file (which makes the pkt_pos and pkt_size different as well). 


Attachments:
hyperdeck-apple-pro-res-test.sh  (some of the stuff I tried)

 

Thanks in advance,

 - Steve K.

 
 		 	   		  
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From cehoyos at ag.or.at  Wed Jul 17 23:13:15 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 17 Jul 2013 21:13:15 +0000 (UTC)
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
References: <BEED9B6344438248AE283ABEC3662C4718900C705B@AICEXMBXCL02.LGE.NET>
	<1374087114.79043.YahooMailNeo@web121801.mail.ne1.yahoo.com>
	<1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
Message-ID: <loom.20130717T231205-344@post.gmane.org>

Andrew Seto <aseto97 <at> yahoo.com> writes:

> I have the same ffmpeg problem that described in 
> ticket 1734 http://trac.ffmpeg.org/ticket/1734

Please test 2e4ec89 to confirm that it is the same 
problem.

Carl Eugen


From aseto97 at yahoo.com  Wed Jul 17 23:27:33 2013
From: aseto97 at yahoo.com (Andy)
Date: Wed, 17 Jul 2013 14:27:33 -0700 (PDT)
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <loom.20130717T231205-344@post.gmane.org>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
Message-ID: <1374096453409-4660062.post@n4.nabble.com>

Hi Carl,

Is 2e4ec89 a release number?

Where can I download this version?

Thanks
Andy



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From aseto97 at yahoo.com  Thu Jul 18 00:32:21 2013
From: aseto97 at yahoo.com (Andy)
Date: Wed, 17 Jul 2013 15:32:21 -0700 (PDT)
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <1374096453409-4660062.post@n4.nabble.com>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
Message-ID: <1374100341199-4660063.post@n4.nabble.com>

Hi Carl, 

I did git checkout 2e4ec89 and my ffmpeg source is back to Sep.26,2012.
Did the build, tried to stream the camera again, and got the same error.

Do you have any idea why it is not working?

Thanks again.
Andy 


ffmpeg version N-44796-g2e4ec89 Copyright (c) 2000-2012 the FFmpeg
developers
  built on Jul 17 2013 17:06:17 with gcc 4.3.2 (GCC)
  configuration: --arch=ppc --cpu=e500v2 --target-os=linux
--samples=fate-suite/ --logfile=./config.log --disable-optimizations
--disable-stripping --enable-cross-compile
--cross-prefix=powerpc-none-linux-gnuspe-
--sysroot=/opt/freescale/usr/local/gcc-4.3.74-eglibc-2.8.74-dp-2/powerpc-none-linux-gnuspe/powerpc-none-linux-gnuspe/libc
--target-path=/opt/projects/pcs200/ltib/rootfs
  libavutil      51. 73.101 / 51. 73.101
  libavcodec     54. 59.100 / 54. 59.100
  libavformat    54. 29.101 / 54. 29.101
  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
[rtsp @ 0x110fa570] SDP:
v=0
o=- 1365531640853070 1365531640853070 IN IP4 192.168.8.132
s=Media Presentation
a=range:npt=0.000000-
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
b=AS:50000
a=framerate:30.0
a=transform:1,0,0;0,1,0;0,0,1
a=control:rtsp://192.168.8.132:554/axis-media/media.amp/trackID=1?streamprofile=Quality?tcp
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1; profile-level-id=4D4029;
sprop-parameter-sets=Z01AKZpmAoAt2AtQEBAQXpw=,aO48gA==

[rtsp @ 0x110fa570] video codec set to: h264
[NULL @ 0x11102760] RTP Packetization Mode: 1
[NULL @ 0x11102760] RTP Profile IDC: 4d Profile IOP: 40 Level: 29
[NULL @ 0x11102760] Extradata set to 0x11102be0 (size: 29)!
[rtsp @ 0x110fa570] hello state=0
[h264 @ 0x11102760] no picture 
[rtsp @ 0x110fa570] All info found
rfps: 29.583333 0.015360
rfps: 29.666667 0.009977
rfps: 29.750000 0.005769
rfps: 29.833333 0.002738
rfps: 29.916667 0.000883
rfps: 30.000000 0.000205
    Last message repeated 1 times
rfps: 30.083333 0.000702
    Last message repeated 1 times
rfps: 30.166667 0.002376
    Last message repeated 1 times
rfps: 30.250000 0.005225
    Last message repeated 1 times
rfps: 30.333333 0.009251
    Last message repeated 1 times
rfps: 30.416667 0.014453
    Last message repeated 1 times
rfps: 59.583333 0.016428
rfps: 59.666667 0.010954
rfps: 59.750000 0.006656
rfps: 59.833333 0.003534
rfps: 59.916667 0.001588
rfps: 60.000000 0.000819
    Last message repeated 1 times
rfps: 29.970030 0.000313
rfps: 59.940060 0.001253
[rtsp @ 0x110fa570] Estimating duration from bitrate, this may be inaccurate
Input #0, rtsp, from
'rtsp://root:root at 192.168.8.132:554/axis-media/media.amp?streamprofile=Quality?tcp':
  Metadata:
    title           : Media Presentation
  Duration: N/A, start: 0.034722, bitrate: N/A
    Stream #0:0, 23, 1/90000: Video: h264 (Main), yuv420p, 1280x720 [SAR 1:1
DAR 16:9], 1/180000, 30 tbr, 90k tbn, 180k tbc
File '/tmp/media/axis.mp4' already exists. Overwrite ? [y/N] y
Output #0, mp4, to '/tmp/media/axis.mp4':
  Metadata:
    title           : Media Presentation
    encoder         : Lavf54.29.101
    Stream #0:0, 0, 1/90000: Video: h264 ([33][0][0][0] / 0x0021), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], 1/90000, q=2-31, 90k tbn, 90k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x11103400] pts has no value
[mp4 @ 0x11103400] pts (0) < dts (1) in stream 0
av_interleaved_write_frame(): Invalid argument




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From pfischer at como.com  Thu Jul 18 09:44:40 2013
From: pfischer at como.com (=?utf-8?Q?Patrick_Fischer?=)
Date: Thu, 18 Jul 2013 09:44:40 +0200
Subject: [FFmpeg-user] how to handle missing pts values to keep audio sync
	(mpegts->mp4)
Message-ID: <zarafa.51e79ce8.7e11.1056c9af53416642@mail.como.com>

>

Hello!

 

?
 

This is my first mail on this list. So i hope i didn't got the wrong mailinglist.

 

?
 

Im writing an program which received a mpeg ts udp stream and convert it to mp4 on the fly.

 

Sometimes i have a problem with async audio.

 

?
 

[mpegts @ 0x35fd580] PES packet size mismatch
[mpeg2video @ 0x3604200] skipped MB in I frame at 25 32
[mpeg2video @ 0x3604200] skipped MB in I frame at 1 33
[mpeg2video @ 0x3604200] ac-tex damaged at 0 34
[mpeg2video @ 0x3604200] ac-tex damaged at 0 35
[mpeg2video @ 0x3604200] Warning MVs not available
[mpeg2video @ 0x3604200] concealing 176 DC, 176 AC, 176 MV errors in I frame
[mp2 @ 0x3649fa0] Header missing
[err]failed to decode audio packet

 

?
 

I use udp so it is possible to loose a packages. I have to handle it.

 

?
 

my video_push and audio_push are appended.

 

I dump out the PTS before i push it to the encoder.

 

So you can see that 67 PTS Values are missing.

 

With a 25fps video it is a delay of 2,68 sec. That's exactly what i see if i play the mp4 file via vlc.
 I have tried to skip some video frames, but din't have any effect.
 How to bring back sync audio?
 skip audio or video frames?
 recalc pts values?
 add dummy frames?
 all together?

 

?
 

push audio frame to 689152 (673)
push audio frame to 690176 (674)
[mpegts @ 0x2163580] PES packet size mismatch
push video frame to 392
[mpeg2video @ 0x216a200] skipped MB in I frame at 25 32
[mpeg2video @ 0x216a200] skipped MB in I frame at 1 33
[mpeg2video @ 0x216a200] ac-tex damaged at 0 34
[mpeg2video @ 0x216a200] ac-tex damaged at 0 35
[mpeg2video @ 0x216a200] Warning MVs not available
[mpeg2video @ 0x216a200] concealing 176 DC, 176 AC, 176 MV errors in I frame
push video frame to 393
push video frame to 394<----------------------------- 1 PTS Values is missing!!!!!!!
push video frame to 396
push video frame to 397
[mp2 @ 0x21affa0] Header missing
[err]failed to decode audio packet
push audio frame to 691200 (675)
push audio frame to 692224 (676)
push audio frame to 693248 (677)
push audio frame to 694272 (678)
push audio frame to 695296 (679)
push audio frame to 696320 (680)
push audio frame to 697344 (681)
push audio frame to 698368 (682)
push audio frame to 699392 (683)
push audio frame to 700416 (684)
push audio frame to 701440 (685)
push audio frame to 702464 (686)
push audio frame to 703488 (687)
push audio frame to 704512 (688)
push audio frame to 705536 (689)
push audio frame to 706560 (690)
push audio frame to 707584 (691)
push audio frame to 708608 (692)
push video frame to 398
push video frame to 465?????????????????????????? <----------------------------- 67 PTS Values are missing!!!!!!!
push video frame to 466
?
 

?
 

?
 

?
 

my video_push look something like that:

 

int enc_audio_frame (my_enc_t *enc,AVFrame *audio_frame){

? if ( enc->first_video_pts == AV_NOPTS_VALUE)
??????? return 0;

? if (enc->first_audio_pts == AV_NOPTS_VALUE) {
??????? enc->first_audio_pts = audio_frame->pts;
? }

? if (audio_frame->pts != AV_NOPTS_VALUE) {
????? audio_frame->pts = av_rescale_q(audio_frame->pts - enc->first_audio_pts, my_decoder_output.a_time_base, enc->conf.a_time_base);
? }
? AVPacket *enc_packet = (AVPacket *) malloc (sizeof (AVPacket));
? av_init_packet(enc_packet);
? enc_packet->size = 0;
? enc_packet->data=NULL;
? int packet_done = 0;
? if ( avcodec_encode_audio2 (enc->audio_codec_ctx,enc_packet,audio_frame,&packet_done) < 0) {
????? fprintf (stderr,"[err]%s failed to encode audio samples\n", __FUNCTION__);
????? return -1;
? }
? if ( packet_done ) {
??? enc_packet->stream_index= 1;
??? fprintf (stderr,"push audio frame to %"PRId64" (%"PRId64")\n", audio_frame->pts, audio_frame->pts/1024);
??? enc->push(enc->push_priv,enc_packet);
? }
? av_free_packet(enc_packet);
? av_free (enc_packet);

? return 0;

 

}

 

?
 

my audio_push look something like that:

 

int enc_audio_frame (my_enc_t *enc,AVFrame *audio_frame){

? if ( enc->first_video_pts == AV_NOPTS_VALUE)
??????? return 0;

? if (enc->first_audio_pts == AV_NOPTS_VALUE) {
??????? enc->first_audio_pts = audio_frame->pts;
? }

? if (audio_frame->pts != AV_NOPTS_VALUE) {
????? audio_frame->pts = av_rescale_q(audio_frame->pts - enc->first_audio_pts, my_decoder_output.a_time_base, enc->conf.a_time_base);
? }
? AVPacket *enc_packet = (AVPacket *) malloc (sizeof (AVPacket));
? av_init_packet(enc_packet);
? enc_packet->size = 0;
? enc_packet->data=NULL;
? int packet_done = 0;
? if ( avcodec_encode_audio2 (enc->audio_codec_ctx,enc_packet,audio_frame,&packet_done) < 0) {
????? fprintf (stderr,"[err]%s failed to encode audio samples\n", __FUNCTION__);
????? return -1;
? }
? if ( packet_done ) {
??? enc_packet->stream_index= 1;
??? fprintf (stderr,"push audio frame to %"PRId64" (%"PRId64")\n", audio_frame->pts, audio_frame->pts/1024);

 

??? enc->push(enc->push_priv,enc_packet);
? }
? av_free_packet(enc_packet);
? av_free (enc_packet);

? return 0;

 

?

From expandafter at yahoo.com  Thu Jul 18 09:56:02 2013
From: expandafter at yahoo.com (S. Fisher)
Date: Thu, 18 Jul 2013 00:56:02 -0700 (PDT)
Subject: [FFmpeg-user] How to skip parts during transcoding? (aka edit
	decision list)
Message-ID: <1374134162.20547.YahooMailBasic@web163801.mail.gq1.yahoo.com>

On Sat, 13 Jul 2013 22:33:16 +0200
Roberto Ragusa wrote:

> which is the best way to transcode a video and skip some intervals of it?
> (identified by timestamps or frame number)
> 
> Mencoder has some kind of edl (edit decision list) support, how can the
> same result be obtained with ffmpeg?


It can be done using the filters select, aselect, setpts, and asetpts.

Let's say you want to keep only two segments: the one starting at
5 seconds and ending at 20 seconds, and the one between 40 seconds
and 52 seconds.

ffmpeg -i "sample.avi" -vf select='-between(t,5,20)-between(t,40,52)',setpts=N/FRAME_RATE/TB
 -af aselect='-between(t,5,20)-between(t,40,52)',asetpts=N/SR/TB sample.mp4

You'll probably want to use a script to generate the command for you.


From cehoyos at ag.or.at  Thu Jul 18 09:55:49 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 18 Jul 2013 07:55:49 +0000 (UTC)
Subject: [FFmpeg-user] how to handle missing pts values to keep audio
	sync (mpegts->mp4)
References: <zarafa.51e79ce8.7e11.1056c9af53416642@mail.como.com>
Message-ID: <loom.20130718T095434-140@post.gmane.org>

Patrick Fischer <pfischer <at> como.com> writes:

> Im writing an program which received a mpeg ts 
> udp stream and convert it to mp4 on the fly.
> 
> Sometimes i have a problem with async audio.

Is the problem also reproducible with ffmpeg 
(the application)?
If yes, please add your command line and the 
complete, uncut console output.
If not, libav-user may be the better place to 
ask.

Carl Eugen


From cehoyos at ag.or.at  Thu Jul 18 10:00:27 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 18 Jul 2013 08:00:27 +0000 (UTC)
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
Message-ID: <loom.20130718T095657-197@post.gmane.org>

Andy <aseto97 <at> yahoo.com> writes:

> I did git checkout 2e4ec89 and my ffmpeg source is 
> back to Sep.26,2012. Did the build, tried to stream 
> the camera again, and got the same error.

Your problem is apparently not the same as ticket 1734.

> Do you have any idea why it is not working?

No, but consider opening a ticket adding the -fdebug ts 
output as requested by Michael in 1734.

> --disable-optimizations

Probably unrelated:
--disable-optimizations is a configure option that 
should only be used in special circumstances (debugging 
a crash that shows no useful backtrace with gdb or 
speeding up compilation when bisecting), I suspect 
you should remove it.

Carl Eugen


From krueger at lesspain.de  Thu Jul 18 12:05:57 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 18 Jul 2013 12:05:57 +0200
Subject: [FFmpeg-user] j2k encoder/decoder status
Message-ID: <CAEnqZEVCrda=i3AyMyvHOqpZCw4viu5aOhejq-R6gYD3oKfkEg@mail.gmail.com>

Hi,

there has been a lot of work on the j2k encoder/decoder recently.
Could someone post what the current status compared to libopenjpeg is
and what the short-term plan is or is this too early to ask because
this is still in progress?

Thanks in advance,

Robert

From cehoyos at ag.or.at  Thu Jul 18 12:24:47 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 18 Jul 2013 10:24:47 +0000 (UTC)
Subject: [FFmpeg-user] j2k encoder/decoder status
References: <CAEnqZEVCrda=i3AyMyvHOqpZCw4viu5aOhejq-R6gYD3oKfkEg@mail.gmail.com>
Message-ID: <loom.20130718T121538-342@post.gmane.org>

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

> there has been a lot of work on the j2k encoder/decoder 
> recently.

I don't think any work on the encoder was done recently.

> Could someone post what the current status compared to 
> libopenjpeg is

The native decoder is (still) not bit-exact, it does not 
support many (less common) pix-fmts.
(Or in other words: It is exactly as "experimental" as 
before.)

> and what the short-term plan is or is this too early 
> to ask because this is still in progress?

I suspect this is not the first time you ask such a 
question: Why do think there is a "plan"?
Do you know how open-source development works?
(And I don't mean kernel development that at least to 
some part is driven by hardware manufacturers.)

This is apparently also the reason why you misunderstood 
this mail (that was not meant nonconstructive at all):
http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45798/focus=45826

Carl Eugen


From krueger at lesspain.de  Thu Jul 18 13:42:05 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 18 Jul 2013 13:42:05 +0200
Subject: [FFmpeg-user] j2k encoder/decoder status
In-Reply-To: <loom.20130718T121538-342@post.gmane.org>
References: <CAEnqZEVCrda=i3AyMyvHOqpZCw4viu5aOhejq-R6gYD3oKfkEg@mail.gmail.com>
	<loom.20130718T121538-342@post.gmane.org>
Message-ID: <CAEnqZEXaF_2M8r2RN45N38Ugj2v=ppuHt2eSza+0JEL-_t9kaw@mail.gmail.com>

On Thu, Jul 18, 2013 at 12:24 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> there has been a lot of work on the j2k encoder/decoder
>> recently.
>
> I don't think any work on the encoder was done recently.

So I read the commit logs sloppily. Thanks for the clarification!

>
>> Could someone post what the current status compared to
>> libopenjpeg is
>
> The native decoder is (still) not bit-exact, it does not
> support many (less common) pix-fmts.
> (Or in other words: It is exactly as "experimental" as
> before.)

Thanks for that information!

>
>> and what the short-term plan is or is this too early
>> to ask because this is still in progress?
>
> I suspect this is not the first time you ask such a
> question: Why do think there is a "plan"?

There does not have to be one but sometimes a developer starts a unit
of work - paid or not - and thinks things like "I want to work on
module X now until it does Y and it will probably take me a week or
two". If that isn't the case here, the answer is "there is no plan"
but some people do work that way and in those cases might not mind
answering that but, of course, a possible outcome was that it isn't
the case here.

> Do you know how open-source development works?

That is not really meant as a question, I guess.

> (And I don't mean kernel development that at least to
> some part is driven by hardware manufacturers.)
>
> This is apparently also the reason why you misunderstood
> this mail (that was not meant nonconstructive at all):
> http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45798/focus=45826

Actually not, although this type of answer is probably perceived as a
provocation by some people because it might be interpreted to have a
"code something or piss off, you leach!" connotation because from past
interactions (bug-reports, patches etc.) you must of course know that
I do understand that sending patches to implement something a feature
I want to have, results in that feature being there, so it's really
hard to see what the constructive part in this is, because you
obviously are an intelligent person and your english is very good. If
that is not what you want to convey, a little work on that style would
help, no offense intended. To me the answer is a bit more than that. I
am asking these questions a) to maybe make decisions better for
planning the development of our software and b) to decide whether it
makes sense to offer funding for certain development tasks I am
interested in for whatever reason, maybe depending on how far
something is from being achieved. I do not expect anything and take
anything that comes back is a free bonus.

Peace,

Robert

From sven at nablet.com  Thu Jul 18 13:43:25 2013
From: sven at nablet.com (Sven Dueking)
Date: Thu, 18 Jul 2013 13:43:25 +0200
Subject: [FFmpeg-user] ffmpeg connection process to a ISS publishing point
Message-ID: <000001ce83ac$04051aa0$0c0f4fe0$@com>

Hi all,

 

Does anyone knows the details about the connection process to a ISS
publishing point ?

 

What happens for instance here :

 

ffmpeg -re <normal input/transcoding options> -movflags isml+frag_keyframe
-f ismv http://server/publishingpoint.isml/Streams(Encoder1

 

As far as I understand ffmpeg pushed the IMSL file to the ISS publishing
point, is this correct ?

How is this handled ? Open the HTTP Url and send data ?

 

Any kind of information/link would be great, I checked the souce code, but
didn?t managed to understand the details yet.

 

Best,

Sven

 


From cehoyos at ag.or.at  Thu Jul 18 13:50:16 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 18 Jul 2013 11:50:16 +0000 (UTC)
Subject: [FFmpeg-user] j2k encoder/decoder status
References: <CAEnqZEVCrda=i3AyMyvHOqpZCw4viu5aOhejq-R6gYD3oKfkEg@mail.gmail.com>
	<loom.20130718T121538-342@post.gmane.org>
	<CAEnqZEXaF_2M8r2RN45N38Ugj2v=ppuHt2eSza+0JEL-_t9kaw@mail.gmail.com>
Message-ID: <loom.20130718T134609-284@post.gmane.org>

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

> > Why do think there is a "plan"?
> 
> There does not have to be one but sometimes a developer 
> starts a unit of work - paid or not - and thinks things 
> like "I want to work on module X now until it does Y 
> and it will probably take me a week or two".

I don't remember that this was ever the case on 
ffmpeg-devel (if I understand you correctly).
Or actually, I do remember occurrences, but they 
never lead to a patch being sent;-(

[...]

> > This is apparently also the reason why you 
> > misunderstood this mail (that was not meant 
> > nonconstructive at all):
> > http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45798/focus=45826
> 
> Actually not, although this type of answer is probably 
> perceived as a provocation by some people because it 
> might be interpreted to have a "code something or piss 
> off, you leach!" connotation

Did you consider that the thread may also be interpreted 
exactly the other way 'round?
(Apparently it is neither how you nor I meant it.)

Carl Eugen


From stefasab at gmail.com  Thu Jul 18 13:56:43 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Thu, 18 Jul 2013 13:56:43 +0200
Subject: [FFmpeg-user] j2k encoder/decoder status
In-Reply-To: <loom.20130718T121538-342@post.gmane.org>
References: <CAEnqZEVCrda=i3AyMyvHOqpZCw4viu5aOhejq-R6gYD3oKfkEg@mail.gmail.com>
	<loom.20130718T121538-342@post.gmane.org>
Message-ID: <20130718115643.GD16930@barisone>

On date Thursday 2013-07-18 10:24:47 +0000, Carl Eugen Hoyos wrote:
[...]
> > and what the short-term plan is or is this too early 
> > to ask because this is still in progress?
> 
> I suspect this is not the first time you ask such a 
> question: Why do think there is a "plan"?
> Do you know how open-source development works?
> (And I don't mean kernel development that at least to 
> some part is driven by hardware manufacturers.)

There is not a formal organization entity which makes plans but
usually developers do, and thus it makes sense to ask the question and
is IMO the right thing to do in order to avoid work duplication.

OTOH ffmpeg-user is not likely the best place where to ask, since many
contributors/developers don't follow it.
-- 
FFmpeg = Fiendish & Faithful Mortal Problematic Entertaining God

From zoleexfz at tanoda.suli.hu  Thu Jul 18 13:59:42 2013
From: zoleexfz at tanoda.suli.hu (=?UTF-8?B?RsO8dmVzIFpvbHTDoW4=?=)
Date: Thu, 18 Jul 2013 13:59:42 +0200
Subject: [FFmpeg-user] strange preset file parse error on gentoo from on
 10.0.7 and above in june
In-Reply-To: <51DE5DA8.7060207@tanoda.suli.hu>
References: <51DE3F70.3050302@tanoda.suli.hu>
	<loom.20130711T084209-106@post.gmane.org>
	<51DE5DA8.7060207@tanoda.suli.hu>
Message-ID: <51E7D8AE.10901@tanoda.suli.hu>

Hi,

again unfortunately I cant discover any solution for my problem yet.

the parse error with the preset file is still valid in ffserver

If anybody could help me I really appreciate it

Thanks again for your time and help

Z.

2013-07-11 09:24 keltez?ssel, F?ves Zolt?n ?rta:
> Hi,
>
>  thanks for your reply and than here is the command line and the console
> output
>
> but what i said before these setting worked for me many times until that
> strange and sunny  summer day :)
>
> ffmpeg  -f video4linux2 -i /dev/video0 -f alsa -i pulse -async 1
> http://*.*.*.1:8090/feed1.ffm 
>
> the preset file :
>
> cat  .ffmpeg/libx264-lossless_ultrafast.ffpreset
> coder=0
> flags=+loop
> cmp=+chroma
> partitions=-parti8x8-parti4x4-partp8x8-partp4x4-partb8x8
> me_method=dia
> subq=0
> me_range=16
> g=250
> keyint_min=25
> sc_threshold=40
> i_qfactor=0.71
> b_strategy=1
> qcomp=0.6
> qmin=10
> qmax=51
> qdiff=4
> directpred=1
> flags2=+fastpskip+mbtree
> cqp=0
>
> ffserver stream  sets:
>
> <Stream live>       # Output stream URL definition
>    Feed feed1.ffm              # Feed from which to receive video
>    Format flv
>
>    # Audio settings
>     AudioCodec aac
>     Strict -2
>     AudioBitRate 64
>     AudioChannels 1
>     AudioSampleRate 44100
>
>    # Video settings
>    VideoCodec libx264
>    VideoSize 720x576
>    VideoFrameRate 25 
>    AVOptionVideo flags +global_header
>     AVPresetVideo lossless_ultrafast
>    AVOptionAudio flags +global_header
>
>     StartSendOnKey
>     VideoBitRate 700            # Video bitrate
> </Stream>
>
> and the ffserver output:
>
>  ffserver -f  /*notpublic/FfserverTV
> ffserver version 1.0.7 Copyright (c) 2000-2013 the FFmpeg developers
>   built on Jul 11 2013 09:15:01 with gcc 4.5.4 (Gentoo 4.5.4 p1.0,
> pie-0.4.7)
>   configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib
> --mandir=/usr/share/man --enable-shared --cc=i686-pc-linux-gnu-gcc
> --cxx=i686-pc-linux-gnu-g++ --ar=i686-pc-linux-gnu-ar
> --optflags='-march=native -Os -pipe' --extra-cflags='-march=native -Os
> -pipe' --extra-cxxflags='-march=native -Os -pipe' --disable-static
> --enable-gpl --enable-postproc --enable-avfilter --enable-avresample
> --disable-stripping --disable-debug --disable-doc --disable-vdpau
> --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-libfaac
> --enable-nonfree --disable-indev=oss --disable-indev=jack
> --enable-x11grab --enable-libpulse --disable-outdev=oss
> --enable-libfreetype --enable-libvorbis --enable-libvpx
> --enable-libopenjpeg --disable-amd3dnow --disable-amd3dnowext
> --disable-altivec --disable-avx --disable-mmxext --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
> [NULL @ 0xbf89c58c] [Eval @ 0xbf89aa98] Undefined constant or missing
> '(' in 'fastpskip'
> [NULL @ 0xbf89c58c] Unable to parse option value "fastpskip+mbtree"
> /*notpublic/.ffmpeg/libx264-lossless_ultrafast.ffpreset: Invalid option
> or argument: 'flags2=+fastpskip+mbtree', parsed as 'flags2' =
> '+fastpskip+mbtree'
> /*notpublic/FfserverTV:52: AVPreset error: lossless_ultrafast
> Incorrect config file - exiting.
>
> same results from the git repo:
>
> ffserver version N-54624-g2e6338b Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Jul 10 2013 16:51:26 with gcc 4.5.4 (Gentoo 4.5.4 p1.0,
> pie-0.4.7)
>   configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib
> --mandir=/usr/share/man --enable-shared --cc=i686-pc-linux-gnu-gcc
> --cxx=i686-pc-linux-gnu-g++ --ar=i686-pc-linux-gnu-ar
> --optflags='-march=native -Os -pipe' --extra-cflags='-march=native -Os
> -pipe' --extra-cxxflags='-march=native -Os -pipe' --disable-static
> --enable-gpl --enable-postproc --enable-avfilter --enable-avresample
> --disable-stripping --enable-nonfree --disable-indev=oss
> --disable-indev=jack --disable-outdev=oss --enable-bzlib
> --enable-runtime-cpudetect --disable-debug --disable-doc
> --disable-gnutls --enable-hardcoded-tables --enable-iconv
> --enable-network --disable-openssl --enable-ffplay --enable-vaapi
> --disable-vdpau --enable-zlib --disable-libvo-aacenc
> --disable-libvo-amrwbenc --enable-libmp3lame --disable-libfdk-aac
> --disable-libaacplus --enable-libfaac --disable-libtheora
> --disable-libtwolame --disable-libwavpack --enable-libx264
> --enable-libxvid --disable-libcdio --disable-libiec61883
> --disable-libdc1394 --disable-libcaca   libavutil      52. 39.100 / 52.
> 39.100
>   libavcodec     55. 18.102 / 55. 18.102
>   libavformat    55. 12.100 / 55. 12.100
>   libavdevice    55.  3.100 / 55.  3.100
>   libavfilter     3. 80.100 /  3. 80.100
>   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
> [NULL @ 0xbfa66cf4] [Eval @ 0xbfa65208] Undefined constant or missing
> '(' in 'fastpskip'
> [NULL @ 0xbfa66cf4] Unable to parse option value "fastpskip+mbtree"
> /*notpublic/.ffmpeg/libx264-lossless_ultrafast.ffpreset: Invalid option
> or argument: 'flags2=+fastpskip+mbtree', parsed as 'flags2' =
> '+fastpskip+mbtree'
> /*notpublic/FfserverTV:52: AVPreset error: lossless_ultrafast
> Incorrect config file - exiting.
>
> Thanks again for your time and help
>
> Have a nice day
>
> Z.
>
>
> 2013.07.11. 8:42 keltez?ssel, Carl Eugen Hoyos ?rta:
>> F?ves Zolt?n <zoleexfz <at> tanoda.suli.hu> writes:
>>
>>> with this error:
>>> Undefined constant or missing '(' in 'fastpskip'
>> Command line and complete, uncut console output 
>> missing.
>>
>> Carl Eugen
>>
>> _______________________________________________
>> 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 krueger at lesspain.de  Thu Jul 18 14:02:50 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 18 Jul 2013 14:02:50 +0200
Subject: [FFmpeg-user] j2k encoder/decoder status
In-Reply-To: <20130718115643.GD16930@barisone>
References: <CAEnqZEVCrda=i3AyMyvHOqpZCw4viu5aOhejq-R6gYD3oKfkEg@mail.gmail.com>
	<loom.20130718T121538-342@post.gmane.org>
	<20130718115643.GD16930@barisone>
Message-ID: <CAEnqZEVABWtseM-1ObH4Wsq+q7vwmzKeQHVXBWwXE+T=9birZw@mail.gmail.com>

On Thu, Jul 18, 2013 at 1:56 PM, Stefano Sabatini <stefasab at gmail.com> wrote:
> On date Thursday 2013-07-18 10:24:47 +0000, Carl Eugen Hoyos wrote:
> [...]
>> > and what the short-term plan is or is this too early
>> > to ask because this is still in progress?
>>
>> I suspect this is not the first time you ask such a
>> question: Why do think there is a "plan"?
>> Do you know how open-source development works?
>> (And I don't mean kernel development that at least to
>> some part is driven by hardware manufacturers.)
>
> There is not a formal organization entity which makes plans but
> usually developers do, and thus it makes sense to ask the question and
> is IMO the right thing to do in order to avoid work duplication.
>
> OTOH ffmpeg-user is not likely the best place where to ask, since many
> contributors/developers don't follow it.

My reason for that was that I thought other users who are interested
in j2k (there at least seem to be a few that are interested in using
ffmpeg to create DCP packages without the daunting slowness of the
openjpeg encoder, which I am too) would more likely see the answer
there and I assumed that the person most likely knowing the answer
(i.e. Michael) would follow ffmpeg-user but I can cross-post in these
cases in the future.

From krueger at lesspain.de  Thu Jul 18 14:17:14 2013
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Thu, 18 Jul 2013 14:17:14 +0200
Subject: [FFmpeg-user] j2k encoder/decoder status
In-Reply-To: <loom.20130718T134609-284@post.gmane.org>
References: <CAEnqZEVCrda=i3AyMyvHOqpZCw4viu5aOhejq-R6gYD3oKfkEg@mail.gmail.com>
	<loom.20130718T121538-342@post.gmane.org>
	<CAEnqZEXaF_2M8r2RN45N38Ugj2v=ppuHt2eSza+0JEL-_t9kaw@mail.gmail.com>
	<loom.20130718T134609-284@post.gmane.org>
Message-ID: <CAEnqZEXuvieiXGi3o=vEL9Pz5WV+ygGOc0kRwRu=cr6eh3_JaA@mail.gmail.com>

On Thu, Jul 18, 2013 at 1:50 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> > Why do think there is a "plan"?
>>
>> There does not have to be one but sometimes a developer
>> starts a unit of work - paid or not - and thinks things
>> like "I want to work on module X now until it does Y
>> and it will probably take me a week or two".
>
> I don't remember that this was ever the case on
> ffmpeg-devel (if I understand you correctly).
> Or actually, I do remember occurrences, but they
> never lead to a patch being sent;-(

No, that's why I wrote "thinks" not "posts". That is how I work on my
software projects and I had off-list communication with ffmpeg devs
who seem to do the same from time to time, no matter how well these
plans work out (I rarely keep the deadlines I set myself).

>
> [...]
>
>> > This is apparently also the reason why you
>> > misunderstood this mail (that was not meant
>> > nonconstructive at all):
>> > http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45798/focus=45826
>>
>> Actually not, although this type of answer is probably
>> perceived as a provocation by some people because it
>> might be interpreted to have a "code something or piss
>> off, you leach!" connotation
>
> Did you consider that the thread may also be interpreted
> exactly the other way 'round?
> (Apparently it is neither how you nor I meant it.)

Yes I did and do consider this a possibility. That's why I am not
saying "you meant it that way" but rather how the tone is perceived in
some cases. Sometimes this is a cultural thing (like me having been
perceived rude by americans I have done business with in the past for
being too direct) but especially in Austria my experience is rather
the opposite (like Christoph Waltz put it in the Conan interview
"first, they are very polite and second, they don't mean it" ;-)).

Cheers,

Robert

From smarrocco at ringsidecreative.com  Thu Jul 18 15:07:50 2013
From: smarrocco at ringsidecreative.com (Sam Marrocco)
Date: Thu, 18 Jul 2013 09:07:50 -0400
Subject: [FFmpeg-user] Is mixing of audio channels happening by default in
	ffmpeg?
Message-ID: <51E7E8A6.8070906@ringsidecreative.com>

I have an uncompressed Quicktime with 8 tracks of audio (in addition to it's one video and one Timecode track). I am transcoding it to Quicktime Prores with the following command line (and results):

"ffmpeg-20130507-git-399f6ef-win64-static\bin\ffmpeg.exe" -i "C:\Users\root\Desktop\MultiChannel.mov" -vcodec prores_ks -pix_fmt yuv444p10le -profile:v 3 -acodec copy -t 1 -f mov "D:\\MultiChannel\MultiChannel.mov"
ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg developers
  built on May  7 2013 01:14:14 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-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. 29.100 / 52. 29.100
  libavcodec     55.  7.100 / 55.  7.100
  libavformat    55.  4.101 / 55.  4.101
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 63.101 /  3. 63.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #1: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #2: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #3: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #4: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #5: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #6: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #7: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #8: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000239e760] Stream #9: not enough frames to estimate rate; consider increasing probesize
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\root\Desktop\MultiChannel.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2013-07-17 21:37:47
  Duration: 00:00:45.05, start: 0.000000, bitrate: 1199347 kb/s
    Stream #0:0(eng): Video: rawvideo (raw  / 0x20776172), rgb24, 1920x1080, 1193199 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 239760 tbn, 239760 tbc
    Metadata:
      creation_time   : 2013-07-17 21:37:47
      handler_name    : Apple Alias Data Handler
      timecode        : 00:59:45:00
    Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:27
      handler_name    : Apple Alias Data Handler
    Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:28
      handler_name    : Apple Alias Data Handler
    Stream #0:3(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:28
      handler_name    : Apple Alias Data Handler
    Stream #0:4(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:29
      handler_name    : Apple Alias Data Handler
    Stream #0:5(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:30
      handler_name    : Apple Alias Data Handler
    Stream #0:6(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:30
      handler_name    : Apple Alias Data Handler
    Stream #0:7(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:31
      handler_name    : Apple Alias Data Handler
    Stream #0:8(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:31
      handler_name    : Apple Alias Data Handler
    Stream #0:9(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      creation_time   : 2013-07-17 21:40:32
      handler_name    : Apple Alias Data Handler
      timecode        : 00:59:45:00
Output #0, mov, to 'D:\\MultiChannel\MultiChannel.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    encoder         : Lavf55.4.101
    Stream #0:0(eng): Video: prores (prores_ks) (apch / 0x68637061), yuv444p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 11988 tbn, 23.98 tbc
    Metadata:
      creation_time   : 2013-07-17 21:37:47
      handler_name    : Apple Alias Data Handler
      timecode        : 00:59:45:00
    Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:27
      handler_name    : Apple Alias Data Handler
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> prores_ks)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=   24 fps=2.5 q=0.0 Lsize=    8082kB time=00:00:01.00 bitrate=66139.3kbits/s dup=276 drop=0
video:7986kB audio:94kB subtitle:0 global headers:0kB muxing overhead 0.024802%



The resulting quicktime appears to contain a single audio track with the contents of original 8 tracks mixed together. I have not provided any audio filtering/mixing options. Wouldn't the default condition to be passing through of the audio tracks exactly as the source contains it? If not, is it possible to pass each track through as such or must it be explicitely told to do so? Ideally if provided with 8 tracks, the resulting file would have 8 tracks.


--

sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>

From cehoyos at ag.or.at  Thu Jul 18 15:16:45 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 18 Jul 2013 13:16:45 +0000 (UTC)
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
	in ffmpeg?
References: <51E7E8A6.8070906@ringsidecreative.com>
Message-ID: <loom.20130718T151116-7@post.gmane.org>

Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:

> "ffmpeg-20130507-git-399f6ef-win64-static\bin\ffmpeg.exe" 
> -i "C:\Users\root\Desktop\MultiChannel.mov" -vcodec prores_ks 
> -pix_fmt yuv444p10le -profile:v 3 -acodec copy -t 1 
> -f mov "D:\\MultiChannel\MultiChannel.mov"

> The resulting quicktime appears to contain a single audio 
> track with the contents of original 8 tracks mixed together.

This is very unlikely / no.
(The amerge filter allows to produce the result that you 
describe.)

The default action - since you did not specify a map 
parameter - is to output one audio track.
See the console output:
> Stream mapping:
>   Stream #0:0 -> #0:0 (rawvideo -> prores_ks)
>   Stream #0:1 -> #0:1 (copy)

Input audio track 0:1 is copied into the output file.

Note that if you specify -map 0 (to copy all streams), all 
output audio tracks except the first one will be silent 
(because that is useful for - imo - typical - usecases).
This is probably trivial to fix but I am still searching 
for somebody to help testing.
;-(

Carl Eugen


From tevans.uk at googlemail.com  Thu Jul 18 15:18:00 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Thu, 18 Jul 2013 14:18:00 +0100
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
	in ffmpeg?
In-Reply-To: <51E7E8A6.8070906@ringsidecreative.com>
References: <51E7E8A6.8070906@ringsidecreative.com>
Message-ID: <CAFHbX1JNdSeaNzAV9r5UdjNbCfy-GZWz9rSdHC6j2=WVtSRXyQ@mail.gmail.com>

On Thu, Jul 18, 2013 at 2:07 PM, Sam Marrocco
<smarrocco at ringsidecreative.com> wrote:
> I have an uncompressed Quicktime with 8 tracks of audio (in addition to it's
> one video and one Timecode track). I am transcoding it to Quicktime Prores
> with the following command line (and results):
>
> "ffmpeg-20130507-git-399f6ef-win64-static\bin\ffmpeg.exe" -i
> "C:\Users\root\Desktop\MultiChannel.mov" -vcodec prores_ks -pix_fmt
> yuv444p10le -profile:v 3 -acodec copy -t 1 -f mov
> "D:\\MultiChannel\MultiChannel.mov"

Add "-map 0" to your command line

See example 4 on this page:

http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option

Cheers

Tom

From smarrocco at ringsidecreative.com  Thu Jul 18 15:26:26 2013
From: smarrocco at ringsidecreative.com (Sam Marrocco)
Date: Thu, 18 Jul 2013 09:26:26 -0400
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <loom.20130718T151116-7@post.gmane.org>
References: <51E7E8A6.8070906@ringsidecreative.com>
	<loom.20130718T151116-7@post.gmane.org>
Message-ID: <51E7ED02.5080707@ringsidecreative.com>


On 7/18/2013 9:16 AM, Carl Eugen Hoyos wrote:


Input audio track 0:1 is copied into the output file.

Note that if you specify -map 0 (to copy all streams), all
output audio tracks except the first one will be silent
(because that is useful for - imo - typical - usecases).
This is probably trivial to fix but I am still searching
for somebody to help testing.
;-(





Is it possible to manually (through commandline args) tell it to copy all streams across without the (apparently bugged) -map 0 option?


sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>

From cehoyos at ag.or.at  Thu Jul 18 15:30:42 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 18 Jul 2013 13:30:42 +0000 (UTC)
Subject: [FFmpeg-user] 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>
Message-ID: <loom.20130718T153016-321@post.gmane.org>

Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:

> Is it possible to manually (through commandline args) 
> tell it to copy all streams across without the
> (apparently bugged) -map 0 option?

How is this option bugged?

Carl Eugen


From smarrocco at ringsidecreative.com  Thu Jul 18 15:36:14 2013
From: smarrocco at ringsidecreative.com (Sam Marrocco)
Date: Thu, 18 Jul 2013 09:36:14 -0400
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <loom.20130718T153016-321@post.gmane.org>
References: <51E7E8A6.8070906@ringsidecreative.com>
	<loom.20130718T151116-7@post.gmane.org>
	<51E7ED02.5080707@ringsidecreative.com>
	<loom.20130718T153016-321@post.gmane.org>
Message-ID: <51E7EF4E.8020100@ringsidecreative.com>


On 7/18/2013 9:30 AM, Carl Eugen Hoyos wrote:

Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:



Is it possible to manually (through commandline args)
tell it to copy all streams across without the
(apparently bugged) -map 0 option?



How is this option bugged?




Perhaps I am reading what you wrote incorrectly:

"Note that if you specify -map 0 (to copy all streams), all
output audio tracks except the first one will be silent....This is probably trivial to fix "

If when asked to copy all the streams with -map 0 (implying they will be copied as is) they are copied silent (without sound?) doesn't that imply that it does not work correctly? Shouldn't they be copied *with sound* and not silent? Or why would it need to be "fixed"? To me, this implies "bugged"? Am I reading this incorrectly?



sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>

From cehoyos at ag.or.at  Thu Jul 18 15:47:16 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 18 Jul 2013 13:47:16 +0000 (UTC)
Subject: [FFmpeg-user] 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>
Message-ID: <loom.20130718T154053-475@post.gmane.org>

Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:

> On 7/18/2013 9:30 AM, Carl Eugen Hoyos wrote:
> 
> Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:
> 
> > Is it possible to manually (through commandline args)
> > tell it to copy all streams across without the
> > (apparently bugged) -map 0 option?
> 
> How is this option bugged?
> 
> Perhaps I am reading what you wrote incorrectly:
> 
> "Note that if you specify -map 0 (to copy all streams), all
> output audio tracks except the first one will be silent....
> This is probably trivial to fix "
> 
> If when asked to copy all the streams with -map 0 
> (implying they will be copied as is) they are copied 
> silent (without sound?) doesn't that imply that it does 
> not work correctly? 

Sorry I am not a native speaker and probably did not 
explain well.

In the past, when a mov file was written with more than 
one audio stream, all audio streams were marked as 
"default" streams, meaning if you prepared a mov file 
for an iDevice containing a movie with several language 
tracks, all tracks were played simultaneously on playback.
(Making the feature of muxing several audio tracks into 
mov useless to some degree.)

A change was committed that marked the first of several 
audio tracks as "default" and all other tracks as "non-
default" (or as I badly named it: "silent").
This allows to select the language you like at playback 
and at the same time default playing (without selecting 
a language) works fine.

There is apparently a usecase (that I still don't 
understand completely) where you want all audio tracks to 
be marked as "default" (or non-silent).
I am searching for a volunteer to help me testing this 
use-case.

Note that even for the mentioned use-case that currently 
does not work out-of-the-box, it is still possible to 
activate the silent tracks in your application (iiuc).

Carl Eugen


From smarrocco at ringsidecreative.com  Thu Jul 18 16:01:49 2013
From: smarrocco at ringsidecreative.com (Sam Marrocco)
Date: Thu, 18 Jul 2013 10:01:49 -0400
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <loom.20130718T154053-475@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>
Message-ID: <51E7F54D.2050708@ringsidecreative.com>


On 7/18/2013 9:47 AM, Carl Eugen Hoyos wrote:


Sorry I am not a native speaker and probably did not
explain well.

In the past, when a mov file was written with more than
one audio stream, all audio streams were marked as
"default" streams, meaning if you prepared a mov file
for an iDevice containing a movie with several language
tracks, all tracks were played simultaneously on playback.
(Making the feature of muxing several audio tracks into
mov useless to some degree.)

A change was committed that marked the first of several
audio tracks as "default" and all other tracks as "non-
default" (or as I badly named it: "silent").
This allows to select the language you like at playback
and at the same time default playing (without selecting
a language) works fine.

There is apparently a usecase (that I still don't
understand completely) where you want all audio tracks to
be marked as "default" (or non-silent).
I am searching for a volunteer to help me testing this
use-case.

Note that even for the mentioned use-case that currently
does not work out-of-the-box, it is still possible to
activate the silent tracks in your application (iiuc).





Thank you for the explanation. I believe I now understand the issues better.
So using -map 0 should "activate" the other tracks and "pass them through". The resulting output is:


ffmpeg-20130507-git-399f6ef-win64-static\bin\ffmpeg.exe" -i "C:\Users\root\Desktop\MultiChannel.mov" -vcodec prores_ks -pix_fmt yuv444p10le -profile:v 3 -acodec copy -t 1 -map 0 -f mov "D:\\MultiChannel\MultiChannel.mov"
ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg developers
  built on May  7 2013 01:14:14 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-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. 29.100 / 52. 29.100
  libavcodec     55.  7.100 / 55.  7.100
  libavformat    55.  4.101 / 55.  4.101
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 63.101 /  3. 63.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  3.100 / 52.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #1: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #2: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #3: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #4: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #5: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #6: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #7: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #8: not enough frames to estimate rate; consider increasing probesize
[mov,mp4,m4a,3gp,3g2,mj2 @ 00000000003ce7e0] Stream #9: not enough frames to estimate rate; consider increasing probesize
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\root\Desktop\MultiChannel.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2013-07-17 21:37:47
  Duration: 00:00:45.05, start: 0.000000, bitrate: 1199347 kb/s
    Stream #0:0(eng): Video: rawvideo (raw  / 0x20776172), rgb24, 1920x1080, 1193199 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 239760 tbn, 239760 tbc
    Metadata:
      creation_time   : 2013-07-17 21:37:47
      handler_name    : Apple Alias Data Handler
      timecode        : 00:59:45:00
    Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:27
      handler_name    : Apple Alias Data Handler
    Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:28
      handler_name    : Apple Alias Data Handler
    Stream #0:3(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:28
      handler_name    : Apple Alias Data Handler
    Stream #0:4(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:29
      handler_name    : Apple Alias Data Handler
    Stream #0:5(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:30
      handler_name    : Apple Alias Data Handler
    Stream #0:6(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:30
      handler_name    : Apple Alias Data Handler
    Stream #0:7(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:31
      handler_name    : Apple Alias Data Handler
    Stream #0:8(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s
    Metadata:
      creation_time   : 2013-07-17 21:40:31
      handler_name    : Apple Alias Data Handler
    Stream #0:9(eng): Data: none (tmcd / 0x64636D74)
    Metadata:
      creation_time   : 2013-07-17 21:40:32
      handler_name    : Apple Alias Data Handler
      timecode        : 00:59:45:00
Data stream encoding not supported yet (only streamcopy)


No file is created--It looks as though the ninth track (apple timecode) is preventing completion using -map 0.



sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>

From tevans.uk at googlemail.com  Thu Jul 18 16:34:11 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Thu, 18 Jul 2013 15:34:11 +0100
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
	in ffmpeg?
In-Reply-To: <51E7F54D.2050708@ringsidecreative.com>
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>
	<51E7F54D.2050708@ringsidecreative.com>
Message-ID: <CAFHbX1KC3Z-dK3GG8U8_2=KL7uyMsdPTQzHbc6Tjk3N0Zqjwcw@mail.gmail.com>

On Thu, Jul 18, 2013 at 3:01 PM, Sam Marrocco
<smarrocco at ringsidecreative.com> wrote:
> Thank you for the explanation. I believe I now understand the issues better.
> So using -map 0 should "activate" the other tracks and "pass them through".
> The resulting output is:
>
>
> ffmpeg-20130507-git-399f6ef-win64-static\bin\ffmpeg.exe" -i
> "C:\Users\root\Desktop\MultiChannel.mov" -vcodec prores_ks -pix_fmt
> yuv444p10le -profile:v 3 -acodec copy -t 1 -map 0 -f mov
> "D:\\MultiChannel\MultiChannel.mov"
>
> ffmpeg version N-52837-g399f6ef Copyright (c) 2000-2013 the FFmpeg
> developers
> ?
> Data stream encoding not supported yet (only streamcopy)
>
>
> No file is created--It looks as though the ninth track (apple timecode) is
> preventing completion using -map 0.
>

-map 0 means add every input stream in to the output. You've said what
you want done with video, encode to prores_ks, and what to do with
audio, copy, you just need to either say don't add all streams (just
first video and all audio) or copy all input streams, including
metadata, and re-encode the video.

You can do both with -map, eg to not copy the metadata track:  "-map
0:v -map 0:a  -c:a copy -c:v prores_ks"
or to copy everything, but re-encode the video: "-map 0 -c copy -c:v prores_ks"

-c:v is an alias for -vcodec, -c:a for -acodec, in case you were wondering.

Cheers

Tom

From smarrocco at ringsidecreative.com  Thu Jul 18 16:43:37 2013
From: smarrocco at ringsidecreative.com (Sam Marrocco)
Date: Thu, 18 Jul 2013 10:43:37 -0400
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <51E7F54D.2050708@ringsidecreative.com>
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>
	<51E7F54D.2050708@ringsidecreative.com>
Message-ID: <51E7FF19.8040109@ringsidecreative.com>

It appears that the existence of a Timecode (data) track in the source
file causes -map 0 to fail with the error

"Data stream encoding not supported yet (only streamcopy)"

The Timecode track seems to be brought across by default when not using
the -map 0 option.
If ALL tracks are specified with -map 0, the encode fails because the
timecode track is a datatrack.
If -map specifies channels, it must specify the video and audio tracks,
but NOT the timecode track or the encode will fail. By *not* specifying
the Timecode track the encode will succeed, and the Timecode track is
brought across by default (somehow ignored by the -map option since it
was not specified, unlike audio and video tracks which must be specified).

Therefore, given a source of:

Track 1: Video
Track 2: Audio1
Track 3: Audio2
Track 4: TC

You cannot use the -map 0 options (because of the existence of the TC
track) and you must specify explicitly:

-map 0:0 -map 0:1 -map 0:2

to encode video, audio1, audio2 AND TC, even though the TC track is not
(and cannot) be specified.


sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>

From smarrocco at ringsidecreative.com  Thu Jul 18 17:08:27 2013
From: smarrocco at ringsidecreative.com (Sam Marrocco)
Date: Thu, 18 Jul 2013 11:08:27 -0400
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <CAFHbX1KC3Z-dK3GG8U8_2=KL7uyMsdPTQzHbc6Tjk3N0Zqjwcw@mail.gmail.com>
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>
	<51E7F54D.2050708@ringsidecreative.com>
	<CAFHbX1KC3Z-dK3GG8U8_2=KL7uyMsdPTQzHbc6Tjk3N0Zqjwcw@mail.gmail.com>
Message-ID: <51E804EB.4060805@ringsidecreative.com>


On 7/18/2013 10:34 AM, Tom Evans wrote:





-map 0 means add every input stream in to the output. You've said what
you want done with video, encode to prores_ks, and what to do with
audio, copy, you just need to either say don't add all streams (just
first video and all audio) or copy all input streams, including
metadata, and re-encode the video.

You can do both with -map, eg to not copy the metadata track:  "-map
0:v -map 0:a  -c:a copy -c:v prores_ks"
or to copy everything, but re-encode the video: "-map 0 -c copy -c:v prores_ks"

-c:v is an alias for -vcodec, -c:a for -acodec, in case you were wondering.





Thanks, Tom.
in order to get the video, 8 channels of audio and the ninth track TC (which can't be specified specifically) what you stated might work as:

-map 0:v -map 0:a

If I'm understanding this seems to indicate "copy all video tracks across, using my pre-specified video codec settings" and "copy across all audio tracks, using my pre-specified audio codec settings". The timecode seems to be brought across by default without specifying.

I'll test further, but this seems to do the job.

sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>

From hitesh.gupta at systematixindia.com  Thu Jul 18 16:43:22 2013
From: hitesh.gupta at systematixindia.com (HiteshG)
Date: Thu, 18 Jul 2013 07:43:22 -0700 (PDT)
Subject: [FFmpeg-user] How to create a video file during encoding from
	ffmpeg.
Message-ID: <1374158602016-4660086.post@n4.nabble.com>

Hi all,

I am doing encoding with camera to live Publish points. Now, what I need to
do is..
A video file should also create at the same time during encoding from same
camera.
The cammand what i am using for encoding is like this:

ffmpeg -f dshow -i video="HD Webcam C615":audio="Microphone (HD Webcam
C615)" -pix_fmt yuv420p -f ismv -threads 0 -c:v libx264 -profile:v baseline
-map 0:v -b:v:0  661k -s:v:0 456x256 -map 0:a -b:a:0 48k -bufsize 400k -r 24
http://localhost/PPS/publishpoint.isml/Streams(Encoder1)

and want to create a file on location "d:\my Videos\"
So, please suggest me what should I add into my command.

Thanks, 
Hitesh



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-to-create-a-video-file-during-encoding-from-ffmpeg-tp4660086.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From christian at maass.it  Thu Jul 18 23:45:36 2013
From: christian at maass.it (=?iso-8859-1?Q?Christian_Maa=DF?=)
Date: Thu, 18 Jul 2013 21:45:36 +0000
Subject: [FFmpeg-user] Converting Every Audio-Stream and copy all other
	Streams
Message-ID: <CE0E2EA1.80DD%christian@maass.it>

Hi Together,

I have the following Problem:

I want to use ffmpeg to convert my mkv files from DTS to AC3 by copying the video and subtitle streams.
I try the following command:

ffmpeg -i Video1.mkv -map 0 -vcodec copy -acodec ac3 -async 1000 -ab 640000 -scodec copy -y Video2.mkv

But when I try it I get the the following message:

Input #0, matroska,webm, from 'Video1.mkv':
  Duration: 01:33:42.04, start: 0.000000, bitrate: 3072 kb/s
    Stream #0.0(eng): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
    Stream #0.1(ger): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s (default)
    Stream #0.2(eng): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s
    Stream #0.3(ger): Subtitle: [0][0][0][0] / 0x0000 (default) (forced)
Number of stream maps must match number of output streams

If I try
ffmpeg -i Video1.mkv -vcodec copy -acodec ac3 -async 1000 -ab 640000 -scodec copy -y Video2.mkv
ffmpeg starts to convert the video, but with just the first audio-stream.

Is it possible to get a command which I can use for all my mkv files without checking the audio-streams for every file, so I can use it in a for-loop?

What do I do wrong?
Thank you for your help.
Kind Regards
Christian Maa?

From nicolas.george at normalesup.org  Fri Jul 19 00:23:15 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Fri, 19 Jul 2013 00:23:15 +0200
Subject: [FFmpeg-user] Converting Every Audio-Stream and copy all other
	Streams
In-Reply-To: <CE0E2EA1.80DD%christian@maass.it>
References: <CE0E2EA1.80DD%christian@maass.it>
Message-ID: <20130718222315.GA14918@phare.normalesup.org>

Le decadi 30 messidor, an CCXXI, Christian Maa? a ?crit?:
> But when I try it I get the the following message:
> 
> Input #0, matroska,webm, from 'Video1.mkv':
>   Duration: 01:33:42.04, start: 0.000000, bitrate: 3072 kb/s
>     Stream #0.0(eng): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
>     Stream #0.1(ger): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s (default)
>     Stream #0.2(eng): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s
>     Stream #0.3(ger): Subtitle: [0][0][0][0] / 0x0000 (default) (forced)
> Number of stream maps must match number of output streams

You trimmed your console output. You should not have done so. I suspect it
would have shown the version you are using is very old, because the error
message you quote was removed two years ago. The options you are trying to
use are recent.

Please post the complete output to confirm this hypothesis.

Regards,

-- 
  Nicolas George
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From bouke at videotoolshed.com  Fri Jul 19 02:27:26 2013
From: bouke at videotoolshed.com (Bouke (VideoToolShed))
Date: Fri, 19 Jul 2013 02:27:26 +0200
Subject: [FFmpeg-user] Offlist,
	Is mixing of audio channels happening by defaultin 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>
Message-ID: <0EF41E1BE75A4DBB80A1E74873E4DC91@HPKANTOOR>

Carl,
Since i'm working on audio mapping anyways, i can help testing if you like.
Since you're not a native english speaker (I'm neither), where are you from 
/ what's your native language?
(Concidering your name i would think Spanish, but one never knows...)

Bouke

VideoToolShed
van Oldenbarneveltstraat 33
6512 AS  NIJMEGEN, the Netherlands
+31 24 3553311
----- Original Message ----- 
From: "Carl Eugen Hoyos" <cehoyos at ag.or.at>
To: <ffmpeg-user at ffmpeg.org>
Sent: Thursday, July 18, 2013 3:47 PM
Subject: Re: [FFmpeg-user] Is mixing of audio channels happening by 
defaultin ffmpeg?


> Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:
>
>> On 7/18/2013 9:30 AM, Carl Eugen Hoyos wrote:
>>
>> Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:
>>
>> > Is it possible to manually (through commandline args)
>> > tell it to copy all streams across without the
>> > (apparently bugged) -map 0 option?
>>
>> How is this option bugged?
>>
>> Perhaps I am reading what you wrote incorrectly:
>>
>> "Note that if you specify -map 0 (to copy all streams), all
>> output audio tracks except the first one will be silent....
>> This is probably trivial to fix "
>>
>> If when asked to copy all the streams with -map 0
>> (implying they will be copied as is) they are copied
>> silent (without sound?) doesn't that imply that it does
>> not work correctly?
>
> Sorry I am not a native speaker and probably did not
> explain well.
>
> In the past, when a mov file was written with more than
> one audio stream, all audio streams were marked as
> "default" streams, meaning if you prepared a mov file
> for an iDevice containing a movie with several language
> tracks, all tracks were played simultaneously on playback.
> (Making the feature of muxing several audio tracks into
> mov useless to some degree.)
>
> A change was committed that marked the first of several
> audio tracks as "default" and all other tracks as "non-
> default" (or as I badly named it: "silent").
> This allows to select the language you like at playback
> and at the same time default playing (without selecting
> a language) works fine.
>
> There is apparently a usecase (that I still don't
> understand completely) where you want all audio tracks to
> be marked as "default" (or non-silent).
> I am searching for a volunteer to help me testing this
> use-case.
>
> Note that even for the mentioned use-case that currently
> does not work out-of-the-box, it is still possible to
> activate the silent tracks in your application (iiuc).
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user 


From bouke at videotoolshed.com  Fri Jul 19 02:30:51 2013
From: bouke at videotoolshed.com (Bouke (VideoToolShed))
Date: Fri, 19 Jul 2013 02:30:51 +0200
Subject: [FFmpeg-user] Offlist,
	Is mixing of audio channels happening by defaultin 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>
Message-ID: <BACA47B40A0E4FC5BF5FFCFE35FC18EB@HPKANTOOR>

hmm, all offlist messages are equal, but some...

Sorry!
(And for Carl, you'll pick it up anyways...)

Bouke

VideoToolShed
van Oldenbarneveltstraat 33
6512 AS  NIJMEGEN, the Netherlands
+31 24 3553311
----- Original Message ----- 
From: "Bouke (VideoToolShed)" <bouke at videotoolshed.com>
To: "FFmpeg user questions" <ffmpeg-user at ffmpeg.org>
Sent: Friday, July 19, 2013 2:27 AM
Subject: [FFmpeg-user] Offlist,Is mixing of audio channels happening by 
defaultin ffmpeg?


> Carl,
> Since i'm working on audio mapping anyways, i can help testing if you 
> like.
> Since you're not a native english speaker (I'm neither), where are you 
> from / what's your native language?
> (Concidering your name i would think Spanish, but one never knows...)
>
> Bouke
>
> VideoToolShed
> van Oldenbarneveltstraat 33
> 6512 AS  NIJMEGEN, the Netherlands
> +31 24 3553311
> ----- Original Message ----- 
> From: "Carl Eugen Hoyos" <cehoyos at ag.or.at>
> To: <ffmpeg-user at ffmpeg.org>
> Sent: Thursday, July 18, 2013 3:47 PM
> Subject: Re: [FFmpeg-user] Is mixing of audio channels happening by 
> defaultin ffmpeg?
>
>
>> Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:
>>
>>> On 7/18/2013 9:30 AM, Carl Eugen Hoyos wrote:
>>>
>>> Sam Marrocco <smarrocco <at> ringsidecreative.com> writes:
>>>
>>> > Is it possible to manually (through commandline args)
>>> > tell it to copy all streams across without the
>>> > (apparently bugged) -map 0 option?
>>>
>>> How is this option bugged?
>>>
>>> Perhaps I am reading what you wrote incorrectly:
>>>
>>> "Note that if you specify -map 0 (to copy all streams), all
>>> output audio tracks except the first one will be silent....
>>> This is probably trivial to fix "
>>>
>>> If when asked to copy all the streams with -map 0
>>> (implying they will be copied as is) they are copied
>>> silent (without sound?) doesn't that imply that it does
>>> not work correctly?
>>
>> Sorry I am not a native speaker and probably did not
>> explain well.
>>
>> In the past, when a mov file was written with more than
>> one audio stream, all audio streams were marked as
>> "default" streams, meaning if you prepared a mov file
>> for an iDevice containing a movie with several language
>> tracks, all tracks were played simultaneously on playback.
>> (Making the feature of muxing several audio tracks into
>> mov useless to some degree.)
>>
>> A change was committed that marked the first of several
>> audio tracks as "default" and all other tracks as "non-
>> default" (or as I badly named it: "silent").
>> This allows to select the language you like at playback
>> and at the same time default playing (without selecting
>> a language) works fine.
>>
>> There is apparently a usecase (that I still don't
>> understand completely) where you want all audio tracks to
>> be marked as "default" (or non-silent).
>> I am searching for a volunteer to help me testing this
>> use-case.
>>
>> Note that even for the mentioned use-case that currently
>> does not work out-of-the-box, it is still possible to
>> activate the silent tracks in your application (iiuc).
>>
>> Carl Eugen
>>
>> _______________________________________________
>> 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 christian at maass.it  Fri Jul 19 08:20:14 2013
From: christian at maass.it (=?iso-8859-1?Q?Christian_Maa=DF?=)
Date: Fri, 19 Jul 2013 06:20:14 +0000
Subject: [FFmpeg-user] Converting Every Audio-Stream and copy all
	other	Streams
In-Reply-To: <20130718222315.GA14918@phare.normalesup.org>
References: <CE0E2EA1.80DD%christian@maass.it>
	<20130718222315.GA14918@phare.normalesup.org>
Message-ID: <948585f94b44462395e1e3ccc2a344e3@DB3PR02MB060.eurprd02.prod.outlook.com>

Sorry, here is the complete console output:





# ffmpeg -i Video1.mkv -map 0 -vcodec copy -acodec ac3 -acodec ac3 -async 1000 -ab 640000 -scodec copy -y Video2.mkv

ffmpeg version 0.8.10, Copyright (c) 2000-2011 the FFmpeg developers

  built on Apr 26 2013 00:49:23 with gcc 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2)

  configuration: --prefix=/opt/cross-project/x86/sys-root/usr --disable-static --enable-shared --disable-yasm --enable-gpl --disable-encoder=snow --disable-decoder=snow

  libavutil    51.  9. 1 / 51.  9. 1

  libavcodec   53.  8. 0 / 53.  8. 0

  libavformat  53.  5. 0 / 53.  5. 0

  libavdevice  53.  1. 1 / 53.  1. 1

  libavfilter   2. 23. 0 /  2. 23. 0

  libswscale    2.  0. 0 /  2.  0. 0

  libpostproc  51.  2. 0 / 51.  2. 0

[matroska,webm @ 0x8068340] max_analyze_duration 5000000 reached at 5002000

[matroska,webm @ 0x8068340] Estimating duration from bitrate, this may be inaccurate

Input #0, matroska,webm, from 'Video1.mkv':

  Duration: 01:33:42.04, start: 0.000000, bitrate: 3072 kb/s

    Stream #0.0(eng): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)

    Stream #0.1(ger): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s (default)

    Stream #0.2(eng): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s

    Stream #0.3(ger): Subtitle: [0][0][0][0] / 0x0000 (default) (forced)

Number of stream maps must match number of output streams



It is the version which was perinstalled on my QNAP-NAS.



Kind regards

Christian Maa?



-----Urspr?ngliche Nachricht-----
Von: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] Im Auftrag von Nicolas George
Gesendet: Freitag, 19. Juli 2013 00:23
An: FFmpeg user questions
Betreff: Re: [FFmpeg-user] Converting Every Audio-Stream and copy all other Streams



Le decadi 30 messidor, an CCXXI, Christian Maa? a ?crit :

> But when I try it I get the the following message:

>

> Input #0, matroska,webm, from 'Video1.mkv':

>   Duration: 01:33:42.04, start: 0.000000, bitrate: 3072 kb/s

>     Stream #0.0(eng): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)

>     Stream #0.1(ger): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s (default)

>     Stream #0.2(eng): Audio: dca (DTS), 48000 Hz, 5.1, s16, 1536 kb/s

>     Stream #0.3(ger): Subtitle: [0][0][0][0] / 0x0000 (default)

> (forced) Number of stream maps must match number of output streams



You trimmed your console output. You should not have done so. I suspect it would have shown the version you are using is very old, because the error message you quote was removed two years ago. The options you are trying to use are recent.



Please post the complete output to confirm this hypothesis.



Regards,



--

  Nicolas George

From andrew at carrierdetect.com  Fri Jul 19 09:59:34 2013
From: andrew at carrierdetect.com (Andrew Back)
Date: Fri, 19 Jul 2013 08:59:34 +0100
Subject: [FFmpeg-user] Optimal hardware for use with ProRes.
Message-ID: <CADj1OPCVqDr1OmhaOaGG_3BcPx+JetVamk9O0TRL+AkfPLPgYQ@mail.gmail.com>

Hello,

I recently picked up an Atomos Ninja HDMI recorder for HDSLR video
use. This encodes to ProRes and I'd like to edit the video using
Cinelerra or Kdenlive on Linux.

My question is what would be the optimal hardware for this? Would
VDPAU hardware help at all? Or perhaps a decent GPU + OpenCL? Or is it
simply a case of as many CPU cores as possible?

Cheers,

Andrew

--
Andrew Back
http://carrierdetect.com

From cehoyos at ag.or.at  Fri Jul 19 10:10:07 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 19 Jul 2013 08:10:07 +0000 (UTC)
Subject: [FFmpeg-user] Optimal hardware for use with ProRes.
References: <CADj1OPCVqDr1OmhaOaGG_3BcPx+JetVamk9O0TRL+AkfPLPgYQ@mail.gmail.com>
Message-ID: <loom.20130719T100859-91@post.gmane.org>

Andrew Back <andrew <at> carrierdetect.com> writes:

> Or is it simply a case of as many CPU cores as possible?

(Assuming this question is FFmpeg-related)
Less faster CPU cores are much better, the overhead 
when doing multi-threading with FFmpeg is enormous.

Carl Eugen


From cehoyos at ag.or.at  Fri Jul 19 10:11:50 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 19 Jul 2013 08:11:50 +0000 (UTC)
Subject: [FFmpeg-user] Converting Every Audio-Stream and copy all other
	Streams
References: <CE0E2EA1.80DD%christian@maass.it>
	<20130718222315.GA14918@phare.normalesup.org>
	<948585f94b44462395e1e3ccc2a344e3@DB3PR02MB060.eurprd02.prod.outlook.com>
Message-ID: <loom.20130719T101104-262@post.gmane.org>

Christian Maa? <christian <at> maass.it> writes:

> ffmpeg version 0.8.10, Copyright (c) 2000-2011

Please update to current git head.

And please avoid top-posting here, it is 
considered rude.

Carl Eugen


From andrew at carrierdetect.com  Fri Jul 19 10:15:28 2013
From: andrew at carrierdetect.com (Andrew Back)
Date: Fri, 19 Jul 2013 09:15:28 +0100
Subject: [FFmpeg-user] Optimal hardware for use with ProRes.
In-Reply-To: <loom.20130719T100859-91@post.gmane.org>
References: <CADj1OPCVqDr1OmhaOaGG_3BcPx+JetVamk9O0TRL+AkfPLPgYQ@mail.gmail.com>
	<loom.20130719T100859-91@post.gmane.org>
Message-ID: <CADj1OPDt1wR1TufYYD21Fy8qaAZPzyZCQ9rLESeW7a35_kp0GQ@mail.gmail.com>

On 19 July 2013 09:10, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Andrew Back <andrew <at> carrierdetect.com> writes:
>
>> Or is it simply a case of as many CPU cores as possible?
>
> (Assuming this question is FFmpeg-related)
> Less faster CPU cores are much better, the overhead
> when doing multi-threading with FFmpeg is enormous.

It is, yes ? my understanding was that the aforementioned editing
software gets ProRes support via FFmpeg.

Good to know that fewer faster cores is better!

Cheers,

Andrew

--
Andrew Back
http://carrierdetect.com

From cehoyos at ag.or.at  Fri Jul 19 10:16:45 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 19 Jul 2013 08:16:45 +0000 (UTC)
Subject: [FFmpeg-user] 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>
	<51E7F54D.2050708@ringsidecreative.com>
	<CAFHbX1KC3Z-dK3GG8U8_2=KL7uyMsdPTQzHbc6Tjk3N0Zqjwcw@mail.gmail.com>
Message-ID: <loom.20130719T101553-921@post.gmane.org>

Tom Evans <tevans.uk <at> googlemail.com> writes:

> > No file is created--It looks as though the ninth 
> > track (apple timecode) is preventing completion 
> > using -map 0.

> You can do both with -map, eg to not copy the 
> metadata track: 
> "-map 0:v -map 0:a  -c:a copy -c:v prores_ks"

I didn't test but I expect -dn to also help.

Carl Eugen


From cehoyos at ag.or.at  Fri Jul 19 10:15:10 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 19 Jul 2013 08:15:10 +0000 (UTC)
Subject: [FFmpeg-user] Offlist,
	Is mixing of audio channels happening by defaultin 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>
Message-ID: <loom.20130719T101328-635@post.gmane.org>

Bouke (VideoToolShed <bouke <at> videotoolshed.com> writes:

> Since i'm working on audio mapping anyways, i can 
> help testing if you like.

I was hoping somebody could test if writing 0xf for 
all audio tracks as asked in this mail really helps:
http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45089/focus=45134

> Since you're not a native english speaker (I'm neither), 
> where are you from / what's your native language?
> (Concidering your name i would think Spanish, but one 
> never knows...)

No, Carl Eugen isn't really Spanish...

Carl Eugen


From stefasab at gmail.com  Fri Jul 19 10:33:30 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Fri, 19 Jul 2013 10:33:30 +0200
Subject: [FFmpeg-user] Optimal hardware for use with ProRes.
In-Reply-To: <CADj1OPCVqDr1OmhaOaGG_3BcPx+JetVamk9O0TRL+AkfPLPgYQ@mail.gmail.com>
References: <CADj1OPCVqDr1OmhaOaGG_3BcPx+JetVamk9O0TRL+AkfPLPgYQ@mail.gmail.com>
Message-ID: <20130719083330.GE16930@barisone>

On date Friday 2013-07-19 08:59:34 +0100, Andrew Back wrote:
> Hello,
> 
> I recently picked up an Atomos Ninja HDMI recorder for HDSLR video
> use. This encodes to ProRes and I'd like to edit the video using
> Cinelerra or Kdenlive on Linux.
> 

> My question is what would be the optimal hardware for this? Would
> VDPAU hardware help at all? Or perhaps a decent GPU + OpenCL? Or is it
> simply a case of as many CPU cores as possible?

OpenCL is used right now only for filters, and I can see specific
optimizations only for the X86 platform (but generic DSP routines have
optimizations for most platform), can't comment avoud VDPAU but I bet
it is unrelevant in this case.
-- 
FFmpeg = Fundamental and Forgiving Mysterious Practical Ecumenical Generator

From stefasab at gmail.com  Fri Jul 19 10:35:42 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Fri, 19 Jul 2013 10:35:42 +0200
Subject: [FFmpeg-user] How to create a video file during encoding from
 ffmpeg.
In-Reply-To: <1374158602016-4660086.post@n4.nabble.com>
References: <1374158602016-4660086.post@n4.nabble.com>
Message-ID: <20130719083542.GF16930@barisone>

On date Thursday 2013-07-18 07:43:22 -0700, HiteshG wrote:
> Hi all,
> 
> I am doing encoding with camera to live Publish points. Now, what I need to
> do is..
> A video file should also create at the same time during encoding from same
> camera.
> The cammand what i am using for encoding is like this:
> 
> ffmpeg -f dshow -i video="HD Webcam C615":audio="Microphone (HD Webcam
> C615)" -pix_fmt yuv420p -f ismv -threads 0 -c:v libx264 -profile:v baseline
> -map 0:v -b:v:0  661k -s:v:0 456x256 -map 0:a -b:a:0 48k -bufsize 400k -r 24
> http://localhost/PPS/publishpoint.isml/Streams(Encoder1)
> 
> and want to create a file on location "d:\my Videos\"
> So, please suggest me what should I add into my command.

See the tee muxer.
-- 
FFmpeg = Fancy & Freak Mere Prodigious Energized Gadget

From stefasab at gmail.com  Fri Jul 19 10:41:38 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Fri, 19 Jul 2013 10:41:38 +0200
Subject: [FFmpeg-user] ffplay can NOT play 48000 sample rate flv stream
In-Reply-To: <CAFd2cZVQLPkqFboamoVSP3G6DFhRiuK6X9-AWD9oVgaG_uKjuQ@mail.gmail.com>
References: <CAFd2cZVQLPkqFboamoVSP3G6DFhRiuK6X9-AWD9oVgaG_uKjuQ@mail.gmail.com>
Message-ID: <20130719084137.GG16930@barisone>

On date Monday 2013-07-15 20:27:40 +0800, Soho Soho123 wrote:
> Hi All,
> 
> when I try to play the video + audio  stream by ffplay,

> even I modify the sample rate to 48000 in the src , ffplay just can play
> flv in 44100 sample rate.

How can you modify the input sample rate? (AFAIK ffmpeg won't allow
it, since 48000 is not considered valid in FLV).

> why ffplay can NOT play 48000 sample rate audio stream?

See previous answer, consider also adding -strict unofficial to the
audio codec options.
-- 
FFmpeg = Fascinating and Frenzy Mere Plastic Egregious Gem

From bouke at videotoolshed.com  Fri Jul 19 11:24:45 2013
From: bouke at videotoolshed.com (Bouke (VideoToolShed))
Date: Fri, 19 Jul 2013 11:24:45 +0200
Subject: [FFmpeg-user] Offlist,
	Is mixing of audio channels happening by defaultin 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>
Message-ID: <EA3514503FCB4A868E0BB71A67D7A8A7@HPKANTOOR>

----- Original Message ----- 
From: "Carl Eugen Hoyos" <cehoyos at ag.or.at>
To: <ffmpeg-user at ffmpeg.org>
Sent: Friday, July 19, 2013 10:15 AM
Subject: Re: [FFmpeg-user] Offlist,Is mixing of audio channels happening by 
defaultin ffmpeg?


> Bouke (VideoToolShed <bouke <at> videotoolshed.com> writes:
>
>> Since i'm working on audio mapping anyways, i can
>> help testing if you like.
>
> I was hoping somebody could test if writing 0xf for
> all audio tracks as asked in this mail really helps:
> http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45089/focus=45134

Eehm Carl,
What do you want me to do? I'm an end user, not a C coder...

>> Since you're not a native english speaker (I'm neither),
>> where are you from / what's your native language?
>> (Concidering your name i would think Spanish, but one
>> never knows...)

> No, Carl Eugen isn't really Spanish...

But Hoyos is...


Bouke

VideoToolShed
van Oldenbarneveltstraat 33
6512 AS  NIJMEGEN, the Netherlands
+31 24 3553311

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


From 99191957 at qq.com  Fri Jul 19 11:29:23 2013
From: 99191957 at qq.com (99191957 at qq.com)
Date: Fri, 19 Jul 2013 17:29:23 +0800
Subject: [FFmpeg-user] Problem with openCL
Message-ID: <2013071917292215905417@qq.com>

I'm a new user of ffmpeg. English is not my mother tongue, please don't mind that I'm quite pool in English..
I configed FFmpeg2.0 with ?--enable-openCL ?. 
Can I use openCL to encode or decode a video---in oder to use GPU instead of CPU. If possible, how to do that. 
In user documetation, I found that: 
?-opencl_options options (global)?
   options can be:
?build_options?
?platform_idx?
?device_idx'
I cannot understand what does these options exact mean.
I'm wondering if anyone could help. 



99191957 at qq.com

From oliver at fromme.com  Fri Jul 19 11:47:18 2013
From: oliver at fromme.com (Oliver Fromme)
Date: Fri, 19 Jul 2013 11:47:18 +0200 (CEST)
Subject: [FFmpeg-user] Optimal hardware for use with ProRes.
In-Reply-To: <CADj1OPDt1wR1TufYYD21Fy8qaAZPzyZCQ9rLESeW7a35_kp0GQ@mail.gmail.com>
Message-ID: <201307190947.r6J9lIPq067638@haluter.fromme.com>

Andrew Back wrote:
 > On 19 July 2013 09:10, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
 > > Less faster CPU cores are much better, the overhead
 > > when doing multi-threading with FFmpeg is enormous.
 > [...]
 > Good to know that fewer faster cores is better!

I made an interesting observation recently:  When I bought
an intel Core i3-3225 (two cores) for a server machine, I
discovered that it encodes to h.264 with ffmpeg almost exactly
as fast as my workstation which has an AMD Phenom-II X6
(six cores).  And the latter draws twice as much electrical
power.

Best regards
   Oliver

-- 

From cehoyos at ag.or.at  Fri Jul 19 12:04:36 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 19 Jul 2013 10:04:36 +0000 (UTC)
Subject: [FFmpeg-user] Problem with openCL
References: <2013071917292215905417@qq.com>
Message-ID: <loom.20130719T120401-843@post.gmane.org>

99191957 <at> qq.com <99191957 <at> qq.com> writes:

> Can I use openCL to encode or decode a video

No, this is not implemented.
(Some people believe this would not have a 
positive performance effect.)

Carl Eugen


From cehoyos at ag.or.at  Fri Jul 19 12:06:35 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 19 Jul 2013 10:06:35 +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>
Message-ID: <loom.20130719T120527-55@post.gmane.org>

Bouke (VideoToolShed <bouke <at> videotoolshed.com> writes:

> > I was hoping somebody could test if writing 0xf for
> > all audio tracks as asked in this mail really helps:
> > http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45089/focus=45134
> 
> What do you want me to do? I'm an end user, not a C coder...

No C coder is needed, just somebody who can compile 
FFmpeg. Apparently, this is such an unimportant 
problem that I will probably not work on it.

Thanks, Carl Eugen


From info at patrick-fischer.de  Fri Jul 19 22:45:47 2013
From: info at patrick-fischer.de (Patrick Fischer)
Date: Fri, 19 Jul 2013 22:45:47 +0200
Subject: [FFmpeg-user] select stream depending on pmt changes
Message-ID: <51E9A57B.7060405@patrick-fischer.de>

Hello

I have a problem with dvb streams which have changes for the stream pid.
On some tv channels, two or more programs share the video pid for the 
most time of a day. But on some hours the program use a local window 
with local content.
This is known as pmt switch or pmt changes.

Ok take a look at some ffmpeg -i ... snapshots:

Now the Program 28726 has the vPID 0x191

Input #0, mpegts, from 'http://192.168.1.2:3000/TS/S19.2E-1-1051-28726.ts':
   Duration: 26:30:43.85, start: 34320.807400, bitrate: 0 kb/s
   Program 28726
     Metadata:
       service_name    : Test-R
       service_provider: ARD
     Stream #0.0[0x191]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 
64:45 DAR 16:9], 15000 kb/s, 37.50 fps, 25 tbr, 90k tbn, 50 tbc
     Stream #0.1[0x192](deu): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s
...
   Program 28724
     Metadata:
       service_name    : arte
       service_provider: ARD
   Program 28725
     Metadata:
       service_name    : PHOENIX
       service_provider: ARD


Now the Program 28726 has the vPID 0x1f5

Input #0, mpegts, from 'http://192.168.1.2:3000/TS/S19.2E-1-1051-28726.ts':
   Duration: N/A, start: 35760.215422, bitrate: 15192 kb/s
   Program 28726
     Metadata:
       service_name    : Test-R
       service_provider: ARD
     Stream #0.0[0x1f5]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 
64:45 DAR 16:9], 15000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
     Stream #0.1[0x1f6](deu): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s
...
   Program 28724
     Metadata:
       service_name    : arte
       service_provider: ARD
   Program 28725
     Metadata:
       service_name    : PHOENIX
       service_provider: ARD






And if ffmpeg got a good snapshout (pmt change was within the 
detectiontime )
i can see that there are both vPID's (0x1f5 and 0x191) are within the 
stream.

Input #0, mpegts, from 'xhx':
   Duration: 00:00:00.39, start: 34367.271400, bitrate: 4762 kb/s
...
   Program 28724
     Metadata:
       service_name    : arte
       service_provider: ARD
   Program 28725
     Metadata:
       service_name    : PHOENIX
       service_provider: ARD
   Program 28726
     Metadata:
       service_name    : Test-R
       service_provider: ARD
     Stream #0.0[0x1f5]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 
64:45 DAR 16:9], 15000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
     Stream #0.1[0x1f6](deu): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s
   No Program
     Stream #0.2[0x191]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 
64:45 DAR 16:9], 15000 kb/s, 50 fps, 25 tbr, 90k tbn, 50 tb
     Stream #0.3[0x192]: Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s


So now i hope i could describe the situation.
What did i want?
I want ffmpeg to detect the pmt switch and select the right Stream for 
output.
If this is not possible, a good beginning would be that i can put both 
PID's into the output ts file. (PMT is also needed)

The Channel Test-R is a test channel on Astra which switch each min from 
PHONEIX to Arte and back.
I would like to record this channel and store it including the right video.

Best Regards
Patrick

From lbrtchx at gmail.com  Sun Jul 21 00:10:58 2013
From: lbrtchx at gmail.com (Albretch Mueller)
Date: Sat, 20 Jul 2013 22:10:58 +0000
Subject: [FFmpeg-user] top left pixel coordinates in a screen capture ...
Message-ID: <CAFakBwhSMSD43RG6-u973UGajZKhwQ5usXVSJQqDZrK4_S7j6Q@mail.gmail.com>

 in order to do some screen capture using ffmpeg you would go:
~
 ffmpeg -f x11grab -r 25 -s 1024x768 -i :0.0+100,200 -f alsa -ac 2 -i
pulse output.flv
~
 where (100, 200) would be the top left position of the rectangular
area you would capture and output.flv will contain the audio also, but
how do you get the point  (100, 200)?
~
 lbrtchx

From lbrtchx at gmail.com  Sun Jul 21 00:10:58 2013
From: lbrtchx at gmail.com (Albretch Mueller)
Date: Sat, 20 Jul 2013 22:10:58 +0000
Subject: [FFmpeg-user] top left pixel coordinates in a screen capture ...
Message-ID: <CAFakBwhSMSD43RG6-u973UGajZKhwQ5usXVSJQqDZrK4_S7j6Q@mail.gmail.com>

 in order to do some screen capture using ffmpeg you would go:
~
 ffmpeg -f x11grab -r 25 -s 1024x768 -i :0.0+100,200 -f alsa -ac 2 -i
pulse output.flv
~
 where (100, 200) would be the top left position of the rectangular
area you would capture and output.flv will contain the audio also, but
how do you get the point  (100, 200)?
~
 lbrtchx

From jiachielee at live.com  Sun Jul 21 16:41:55 2013
From: jiachielee at live.com (thljcl)
Date: Sun, 21 Jul 2013 07:41:55 -0700 (PDT)
Subject: [FFmpeg-user] change frame rate
In-Reply-To: <EEF6503E-0646-4E13-BF52-F5570BAF183B@danielerossi.net>
References: <EEF6503E-0646-4E13-BF52-F5570BAF183B@danielerossi.net>
Message-ID: <1374417715380-4660111.post@n4.nabble.com>

The frame rate of your source may be constant or variable. If frame rate of
your source is constant,
You can just type the following:
ffmpeg -r "60" -i "input.avi" -vcodec "copy" "output.avi"
Here I assume that that the input video is with the file name of ?input.avi?
and output video is with the file name of ?output.avi?; this video does not
have an audio track.Basically, frame rate as an input option will make
ffmpeg ignores the original frame rate of the video and use the designated
frame rate without adding or dropping frames.If the frame rate of your
source is variable, the above method may work but the video may speed up or
slow down intermittently due to variable nature of the frame rate. In other
words, if you change the frame rate from one to another by adding/dropping
frames without changing the playback speed or the length of the video, the
video may be out of sync with the audio. One way of mitigating the effect of
variable frame rate is to re-encode the video to constant frame rate.Say the
average frame rate of the source is 30 FPS; the frame rate is variable.
Lossy encoding will inevitably result in ?generation loss?. To reduce the
quality loss, you can use lossless encoding in the process; then choose the
lossy encoding of appropriate level instead of choosing lossy encoding while
editing the videos.Basically, converting video to a series of image sequence
will force the frame rate to be constant.
mkdir "frames"
ffmpeg -i "input.avi" -r "30" "frames\f_%%1d.png"
ffmpeg -r "60" -i "frames\f_%%1d.png" -vcodec "libx264" -crf "0"
"output-crf0.mkv"
rmdir /s /q "frames"
?output-crf0.mkv? is of lossless encoding. You can choose the appropriate
compression level you see fit.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/change-frame-rate-tp939542p4660111.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From cehoyos at ag.or.at  Sun Jul 21 23:58:20 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 21 Jul 2013 21:58:20 +0000 (UTC)
Subject: [FFmpeg-user] change frame rate
References: <EEF6503E-0646-4E13-BF52-F5570BAF183B@danielerossi.net>
	<1374417715380-4660111.post@n4.nabble.com>
Message-ID: <loom.20130721T235713-441@post.gmane.org>

thljcl <jiachielee <at> live.com> writes:

> ffmpeg -r "60" -i "input.avi" -vcodec "copy" "output.avi"

As said, while this may work in some situations, 
the suggested way of changing speed of videos is 
using the setpts filter.

Unrelated: Please do not revive ancient threads.

Carl Eugen


From ajsinclair at gmail.com  Mon Jul 22 04:18:10 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Mon, 22 Jul 2013 12:18:10 +1000
Subject: [FFmpeg-user] Exit codes
Message-ID: <CAOeAR15n43RKrM0_aud-Gb52js4QD+ig_P3esX+fc9OxehmhkA@mail.gmail.com>

Hi,

Can anyone tell me if there is documented somewhere the different exit
codes used by ffmpeg and what they relate to?

I can see a whole stack of exit() calls in ffmpeg.c (mostly exit(1) as
expected for an error) but just wondering there is any convention on things
like exit(123).

Regards,
Andrew

From wimbonner at gmail.com  Mon Jul 22 05:47:45 2013
From: wimbonner at gmail.com (William C Bonner)
Date: Sun, 21 Jul 2013 20:47:45 -0700
Subject: [FFmpeg-user] FFMPeg reading from Logitech C920 generating H264
 source produces empty output file
Message-ID: <CALqHiHNzHNAgCELtYSBLCU01A0R=xEZFF8v3+NNa8j_rbdmvZg@mail.gmail.com>

I'm generally trying to do this in Linux, using the V4L2 interface, but I
decided to connect the webcam up to my windows desktop and test with dshow
and I got similar results, so I'm wondering if I'm doing something wrong,
or if this is a bug in FFMPEG? I find it interesting that FFMPEG seems to
acquire data when I'm asking it transcode and remux it, but it doesn't seem
to acquire it when all I'm asking it to do is to acquire and remux it.

In Windows I list devices, list options, and then try to stream video with
these commands:

ffmpeg -list_devices true -f dshow -i dummy
ffmpeg -f dshow -list_options true  -i video="Logitech HD Pro Webcam C920"
ffmpeg -f dshow -vcodec h264 -s 1280x720 -i video="Logitech HD Pro Webcam
C920" -vcodec copy -y test.mp4
ffmpeg version N-54691-g7f4a1fd Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 12 2013 16:31:48 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-libmp3
lame --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. 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
Input #0, dshow, from 'video=Logitech HD Pro Webcam C920':
  Duration: N/A, start: 930235.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Constrained Baseline), yuvj420p, 1280x720
[SAR 1:1 DAR 16:9], 31.25 tbr, 10000k tbn, 60 tbc
Output #0, mp4, to 'test.mp4':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuvj420p, 1280x720
[SAR 1:1 DAR 16:9], q=2-31, 10000k tbn, 10000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
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)

If I do everything the same, except omit the "-vcodec copy" I get a decent
quality output file, and it reports that it's transcoding from h264 to
libx264 and uses a lot of CPU in the process.

ffmpeg -f dshow -vcodec h264 -s 1280x720 -i video="Logitech HD Pro Webcam
C920" -y test.mp4
ffmpeg version N-54691-g7f4a1fd Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 12 2013 16:31:48 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-libmp3
lame --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. 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
Input #0, dshow, from 'video=Logitech HD Pro Webcam C920':
  Duration: N/A, start: 930119.199000, bitrate: N/A
    Stream #0:0: Video: h264 (Constrained Baseline), yuvj420p, 1280x720
[SAR 1:1 DAR 16:9], 31.25 tbr, 10000k tbn, 60 tbc
No pixel format specified, yuvj420p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0000000005fd1020] using SAR=1/1
[libx264 @ 0000000005fd1020] using cpu capabilities: MMX2 SSE2Fast SSSE3
Cache64 SlowShuffle
[libx264 @ 0000000005fd1020] profile High, level 3.2
[libx264 @ 0000000005fd1020] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4
AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html -
options: caba
c=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 'test.mp4':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuvj420p,
1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 16k tbn, 31.25 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
Press [q] to stop, [?] for help
frame=  911 fps= 30 q=-1.0 Lsize=    4253kB time=00:00:29.08
bitrate=1197.7kbits/s dup=226 drop=0
video:4243kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.229996%
[libx264 @ 0000000005fd1020] frame I:17    Avg QP:18.05  size: 21517
[libx264 @ 0000000005fd1020] frame P:423   Avg QP:21.14  size:  7715
[libx264 @ 0000000005fd1020] frame B:471   Avg QP:23.06  size:  1518
[libx264 @ 0000000005fd1020] consecutive B-frames: 28.9%  6.1%  1.3% 63.7%
[libx264 @ 0000000005fd1020] mb I  I16..4: 49.1% 45.3%  5.6%
[libx264 @ 0000000005fd1020] mb P  I16..4:  9.1%  6.2%  0.3%  P16..4: 37.8%
 3.6%  3.1%  0.0%  0.0%    skip:40.0%
[libx264 @ 0000000005fd1020] mb B  I16..4:  0.2%  0.2%  0.0%  B16..8: 29.3%
 0.4%  0.0%  direct: 0.9%  skip:69.0%  L0:47.3% L1:52.2% BI: 0.6%
[libx264 @ 0000000005fd1020] 8x8 transform intra:40.8% inter:78.3%
[libx264 @ 0000000005fd1020] coded y,uvDC,uvAC intra: 18.0% 49.1% 9.9%
inter: 4.1% 16.1% 0.5%
[libx264 @ 0000000005fd1020] i16 v,h,dc,p: 41% 22% 17% 21%
[libx264 @ 0000000005fd1020] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 24% 17% 41%  2%
 4%  4%  3%  2%  2%
[libx264 @ 0000000005fd1020] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 24% 18%  4%
 8%  7%  4%  2%  4%
[libx264 @ 0000000005fd1020] i8c dc,h,v,p: 56% 21% 21%  2%
[libx264 @ 0000000005fd1020] Weighted P-Frames: Y:3.1% UV:1.7%
[libx264 @ 0000000005fd1020] ref P L0: 69.7%  3.4% 19.0%  7.7%  0.1%
[libx264 @ 0000000005fd1020] ref B L0: 87.7%  9.4%  2.9%
[libx264 @ 0000000005fd1020] ref B L1: 95.4%  4.6%
[libx264 @ 0000000005fd1020] kb/s:1192.13

Here's the similar command in linux with a recent build of FFMPEG showing
similar results, and heavily using the CPU, which is what I'm trying to
avoid.

ffmpeg -f video4linux2 -list_formats all -i /dev/video0
ffmpeg -report -f v4l2 -s 1280x720 -vcodec h264 -i /dev/video0 -vcodec copy
-y test.mp4
ffmpeg started on 2000-01-01 at 00:34:54
Report written to "ffmpeg-20000101-003454.log"ffmpeg version
N-54692-gdd1382a Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 13 2013 05:15:21 with gcc 4.7.3 (Linaro GCC 4.7-2013.02-01)
20130205 (prerelease)
  configuration: --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
Input #0, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 135.904141, bitrate: N/A
    Stream #0:0: Video: h264 (Constrained Baseline), yuvj420p, 1280x720
[SAR 1:1 DAR 16:9], -5 kb/s, 30 fps, 30 tbr, 1000k tbn, 60 tbc
Output #0, mp4, to 'test.mp4':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: 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
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)

ffmpeg -report -f v4l2 -s 1280x720 -vcodec h264 -i /dev/video0 -y test.mp4

ffmpeg started on 2000-01-01 at 00:36:27
Report written to "ffmpeg-20000101-003627.log"
ffmpeg version N-54692-gdd1382a Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 13 2013 05:15:21 with gcc 4.7.3 (Linaro GCC 4.7-2013.02-01)
20130205 (prerelease)
  configuration: --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
Input #0, video4linux2,v4l2, from '/dev/video0':
  Duration: N/A, start: 229.309265, bitrate: N/A
    Stream #0:0: Video: h264 (Constrained Baseline), yuvj420p, 1280x720
[SAR 1:1 DAR 16:9], -5 kb/s, 30 fps, 30 tbr, 1000k tbn, 60 tbc
No pixel format specified, yuvj420p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x10ebf40] using SAR=1/1
[libx264 @ 0x10ebf40] using cpu capabilities: ARMv6 NEON
[libx264 @ 0x10ebf40] profile High, level 3.1
[libx264 @ 0x10ebf40] 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 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=1 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.mp4':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuvj420p,
1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 15360 tbn, 30 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
Press [q] to stop, [?] for help
frame=   86 fps=2.0 q=29.0 Lsize=     137kB time=00:00:02.80 bitrate=
402.1kbits/s dup=75 drop=0
video:136kB audio:0kB subtitle:0 global headers:0kB muxing overhead
1.240864%
[libx264 @ 0x10ebf40] frame I:1     Avg QP:19.67  size:  7304
[libx264 @ 0x10ebf40] frame P:33    Avg QP:17.52  size:  3784
[libx264 @ 0x10ebf40] frame B:52    Avg QP:27.16  size:   118
[libx264 @ 0x10ebf40] consecutive B-frames: 18.6%  2.3%  0.0% 79.1%
[libx264 @ 0x10ebf40] mb I  I16..4: 84.5% 11.4%  4.2%
[libx264 @ 0x10ebf40] mb P  I16..4:  6.7%  3.4%  0.3%  P16..4: 13.0%  1.3%
 1.0%  0.0%  0.0%    skip:74.3%
[libx264 @ 0x10ebf40] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  2.3%  0.0%
 0.0%  direct: 0.1%  skip:97.6%  L0:62.1% L1:37.4% BI: 0.5%
[libx264 @ 0x10ebf40] 8x8 transform intra:28.1% inter:67.4%
[libx264 @ 0x10ebf40] coded y,uvDC,uvAC intra: 13.0% 22.4% 5.0% inter: 1.3%
3.2% 0.3%
[libx264 @ 0x10ebf40] i16 v,h,dc,p: 53% 30% 15%  2%
[libx264 @ 0x10ebf40] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 18% 58%  1%  2%
 1%  1%  1%  2%
[libx264 @ 0x10ebf40] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 36% 20% 23%  4%  5%
 4%  3%  3%  2%
[libx264 @ 0x10ebf40] i8c dc,h,v,p: 67% 18% 14%  0%
[libx264 @ 0x10ebf40] Weighted P-Frames: Y:9.1% UV:9.1%
[libx264 @ 0x10ebf40] ref P L0: 80.6%  5.4%  9.5%  4.2%  0.4%
[libx264 @ 0x10ebf40] ref B L0: 77.6% 21.9%  0.5%
[libx264 @ 0x10ebf40] ref B L1: 96.4%  3.6%
[libx264 @ 0x10ebf40] kb/s:385.92

From oliver at fromme.com  Mon Jul 22 08:28:33 2013
From: oliver at fromme.com (Oliver Fromme)
Date: Mon, 22 Jul 2013 08:28:33 +0200 (CEST)
Subject: [FFmpeg-user] change frame rate
In-Reply-To: <loom.20130721T235713-441@post.gmane.org>
Message-ID: <201307220628.r6M6SXxm078276@haluter.fromme.com>

Carl Eugen Hoyos wrote:
 > Unrelated: Please do not revive ancient threads.

Why not?  I think it's OK to reply to an old thread if you
can contribute new information or insights in the context
of that thread that might be useful for others.

Best regards
   Oliver

-- 

From cehoyos at ag.or.at  Mon Jul 22 10:48:09 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 22 Jul 2013 08:48:09 +0000 (UTC)
Subject: [FFmpeg-user] Exit codes
References: <CAOeAR15n43RKrM0_aud-Gb52js4QD+ig_P3esX+fc9OxehmhkA@mail.gmail.com>
Message-ID: <loom.20130722T104631-850@post.gmane.org>

Andrew Sinclair <ajsinclair <at> gmail.com> writes:

> Can anyone tell me if there is documented somewhere 
> the different exit codes used by ffmpeg and what 
> they relate to?

No, generally, "1" means error, 123 means that several 
signals were catched and 254 has something to do with 
the error count.
I don't think you can use the exit value to interpret 
the problem (in case this was your question).

Carl Eugen


From cehoyos at ag.or.at  Mon Jul 22 10:51:59 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 22 Jul 2013 08:51:59 +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>
Message-ID: <loom.20130722T104914-806@post.gmane.org>

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


From jiachielee at live.com  Mon Jul 22 01:30:32 2013
From: jiachielee at live.com (thljcl)
Date: Sun, 21 Jul 2013 16:30:32 -0700 (PDT)
Subject: [FFmpeg-user] Frame rate and movies from images
In-Reply-To: <4D8EEF52-4CB2-40F8-BF10-5F02B9F288E6@uncommonprojects.com>
References: <4D8EEF52-4CB2-40F8-BF10-5F02B9F288E6@uncommonprojects.com>
Message-ID: <1374449432679-4660113.post@n4.nabble.com>

You made an error for not recognizing the difference between -r as an input
option or output option. If your source is a series of image sequence,
ffmpeg will assume the frame rate of 25 FPS if none is specified as the
input option; if your source is a video file, the frame rate of the source
will be regarded as input option if none is specified. Just type -r "6"
before -i should serve your purpose, if I understand you correctly. -r
before -i will force the frame rate on the source while ignoring the frame
rate of the source, which will result in slowing down or speeding up of the
playback of the video. -r after -i is the frame rate for output video, which
if the frame rate is high enough, will preserve the playback speed or the
length by adding or dropping frames. If the source is of variable frame
rate, specifying a different frame rate as the output option may cause the
output video out of sync with audio, mainly due to changing frame rate of
the source itself; even though the input video is perfectly in sync with
audio. Encoding a video into a series of image sequences and specifying a
frame rate as an output option can force the frame rate to be constant while
preserving the original length.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Frame-rate-and-movies-from-images-tp938477p4660113.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jpalmer at worldnow.com  Mon Jul 22 18:01:50 2013
From: jpalmer at worldnow.com (Jason Palmer)
Date: Mon, 22 Jul 2013 16:01:50 +0000
Subject: [FFmpeg-user] Frame number repeats and time resets when encoding
Message-ID: <F45E17C592E4A44F836D8B3718529BEE0A757F43@WNMAIL-MB01.worldnow.corp>

Hi

I am attempting to create a .mp4 file from an AJA Capture Source and I am seeing the following that I cannot explain:


1.       The audio is out of sync with the video.  I have tried the -async, -itsoffset, -ss flags and they did not work.

2.       The frame number starts at frame 167 (instead of 0) and it repeats itself towards the end of the recorded video.  In the output below, note how frame 871 is repeated multiple times.

3.       The time of the output of ffmpeg starts at 5.75 seconds (instead of 0 seconds) and starts over.  In the output below, note how once the output reaches 30 seconds (as specified by the -t flag), it starts over for about the same amount of time that ffmpeg takes to start the recording (~5.75 seconds).  It is also at this time when the frame number repeats itself.

Interestingly, the 5.75 seconds that ffmpeg takes to start recording is approximately how long the audio/video is out of sync.  Note that I am copying the video stream (which is in h264 format) and the audio stream (which is in aac format).  Any help/explanation would be greatly appreciated.

Here is the command I am using:

FFMPEG.EXE -y -rtbufsize 100M -f dshow -i video="Virtual Capture Source" -acodec aac -f dshow -i audio="Virtual Capture Source" -map 0:0 -map 1:0 -absf aac_adtstoasc -acodec copy -vcodec copy -vsync drop -r 29 -copyinkf -aspect 16:9 -t 00:00:30 out.mp4

Here is my output:

FFMPEG.EXE -y -rtbufsize 100M -f dshow -i video="Virtual Capture Sour
ce" -acodec aac -f dshow -i audio="Virtual Capture Source" -map 0:0 -map 1:0 -ab
sf aac_adtstoasc -acodec copy -vcodec copy -vsync drop -r 29 -copyinkf -aspect 1
6:9 -t 00:00:30 out.mp4
ffmpeg started on 2013-07-22 at 11:37:50
Report written to "log.txt"
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 15 2013 16:04:21 with gcc 4.8.0 (rubenvb-4.8.0)
  configuration: --enable-static --disable-shared --prefix=/mingw/i686-w64-mingw
32 --arch=x86 --target-os=mingw32 --enable-gpl --enable-nonfree --enable-runtime
-cpudetect --enable-openssl --enable-libfreetype --enable-fontconfig --enable-li
bass --enable-zlib --enable-librtmp --enable-w32threads --enable-libx264 --enabl
e-libfdk-aac --enable-debug --disable-optimizations --disable-stripping
  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
  libpostproc    52.  3.100 / 52.  3.100
[h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
[h264 @ 05e8e5e0] non-existing PPS 0 referenced
[h264 @ 05e8e5e0] decode_slice_header error
[h264 @ 05e8e5e0] no frame!
[h264 @ 05e8e5e0] non-existing SPS 0 referenced in buffering period
[h264 @ 05e8e5e0] non-existing PPS 0 referenced
[h264 @ 05e8e5e0] decode_slice_header error
[h264 @ 05e8e5e0] no frame!
[h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
[h264 @ 05e8e5e0] non-existing PPS 0 referenced
[h264 @ 05e8e5e0] decode_slice_header error
[h264 @ 05e8e5e0] no frame!
[h264 @ 05e8e5e0] non-existing SPS 0 referenced in buffering period
[h264 @ 05e8e5e0] non-existing PPS 0 referenced
[h264 @ 05e8e5e0] decode_slice_header error
[h264 @ 05e8e5e0] no frame!
[h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
[h264 @ 05e8e5e0] non-existing PPS 0 referenced
[h264 @ 05e8e5e0] decode_slice_header error
[h264 @ 05e8e5e0] no frame!
[h264 @ 05e8e5e0] non-existing SPS 0 referenced in buffering period
[h264 @ 05e8e5e0] non-existing PPS 0 referenced
[h264 @ 05e8e5e0] decode_slice_header error
[h264 @ 05e8e5e0] no frame!
[h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
[h264 @ 05e8e5e0] non-existing PPS 0 referenced
[h264 @ 05e8e5e0] decode_slice_header error
[h264 @ 05e8e5e0] no frame!
Input #0, dshow, from 'video=Virtual Capture Source':
  Duration: N/A, start: 2341547.916000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuv420p, 854x480 [SAR 1:1 DAR 427:240], 4 t
br, 10000k tbn, 58 tbc
Input #1, dshow, from 'audio=Virtual Capture Source':
  Duration: N/A, start: 2341552.949000, bitrate: N/A
    Stream #1:0: Audio: aac, 48000 Hz, stereo, fltp
Overriding aspect ratio with stream copy may produce invalid files
Output #0, mp4, to out.mp4':
  Metadata:
    encoder         : Lavf55.12.100
    Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 854x480 [SAR 128
0:1281 DAR 16:9], q=2-31, 14848 tbn, 29 tbc
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0767f040] Encoder did not produce proper pts, making some up.
frame=  167 fps=0.0 q=-1.0 size=     154kB time=00:00:05.75 bitrate= 219.6kbits/
frame=  182 fps=182 q=-1.0 size=     245kB time=00:00:06.27 bitrate= 319.9kbits/
frame=  196 fps=130 q=-1.0 size=     326kB time=00:00:06.75 bitrate= 395.6kbits/
frame=  211 fps=105 q=-1.0 size=     430kB time=00:00:07.27 bitrate= 483.7kbits/
frame=  225 fps= 90 q=-1.0 size=     559kB time=00:00:07.75 bitrate= 590.0kbits/
frame=  240 fps= 80 q=-1.0 size=     697kB time=00:00:08.27 bitrate= 689.8kbits/
frame=  254 fps= 72 q=-1.0 size=     776kB time=00:00:08.75 bitrate= 725.6kbits/
frame=  269 fps= 67 q=-1.0 size=     849kB time=00:00:09.27 bitrate= 750.0kbits/
frame=  283 fps= 63 q=-1.0 size=     910kB time=00:00:09.75 bitrate= 763.9kbits/
frame=  298 fps= 59 q=-1.0 size=     983kB time=00:00:10.27 bitrate= 784.0kbits/
frame=  312 fps= 57 q=-1.0 size=    1074kB time=00:00:10.75 bitrate= 818.0kbits/
frame=  327 fps= 54 q=-1.0 size=    1175kB time=00:00:11.27 bitrate= 854.0kbits/
frame=  341 fps= 52 q=-1.0 size=    1272kB time=00:00:11.75 bitrate= 886.3kbits/
frame=  356 fps= 51 q=-1.0 size=    1358kB time=00:00:12.27 bitrate= 905.9kbits/
frame=  371 fps= 49 q=-1.0 size=    1470kB time=00:00:12.79 bitrate= 941.4kbits/
frame=  385 fps= 48 q=-1.0 size=    1577kB time=00:00:13.27 bitrate= 973.1kbits/
frame=  400 fps= 47 q=-1.0 size=    1677kB time=00:00:13.79 bitrate= 996.0kbits/
frame=  414 fps= 46 q=-1.0 size=    1788kB time=00:00:14.27 bitrate=1026.2kbits/
frame=  429 fps= 45 q=-1.0 size=    1907kB time=00:00:14.79 bitrate=1055.8kbits/
frame=  443 fps= 44 q=-1.0 size=    2020kB time=00:00:15.27 bitrate=1083.3kbits/
frame=  458 fps= 43 q=-1.0 size=    2152kB time=00:00:15.79 bitrate=1116.2kbits/
frame=  472 fps= 43 q=-1.0 size=    2278kB time=00:00:16.27 bitrate=1146.6kbits/
frame=  487 fps= 42 q=-1.0 size=    2338kB time=00:00:16.79 bitrate=1140.3kbits/
frame=  501 fps= 42 q=-1.0 size=    2429kB time=00:00:17.27 bitrate=1152.0kbits/
frame=  516 fps= 41 q=-1.0 size=    2543kB time=00:00:17.79 bitrate=1171.0kbits/
frame=  530 fps= 41 q=-1.0 size=    2626kB time=00:00:18.27 bitrate=1177.2kbits/
frame=  545 fps= 40 q=-1.0 size=    2725kB time=00:00:18.79 bitrate=1187.9kbits/
frame=  559 fps= 40 q=-1.0 size=    2840kB time=00:00:19.27 bitrate=1207.1kbits/
frame=  574 fps= 39 q=-1.0 size=    2941kB time=00:00:19.79 bitrate=1217.1kbits/
frame=  588 fps= 39 q=-1.0 size=    3058kB time=00:00:20.27 bitrate=1235.4kbits/
frame=  603 fps= 39 q=-1.0 size=    3170kB time=00:00:20.79 bitrate=1249.1kbits/
frame=  617 fps= 38 q=-1.0 size=    3299kB time=00:00:21.27 bitrate=1270.2kbits/
frame=  632 fps= 38 q=-1.0 size=    3335kB time=00:00:21.79 bitrate=1253.6kbits/
frame=  646 fps= 38 q=-1.0 size=    3424kB time=00:00:22.27 bitrate=1259.1kbits/
frame=  661 fps= 38 q=-1.0 size=    3554kB time=00:00:22.79 bitrate=1277.4kbits/
frame=  675 fps= 37 q=-1.0 size=    3633kB time=00:00:23.27 bitrate=1278.6kbits/
frame=  690 fps= 37 q=-1.0 size=    3724kB time=00:00:23.79 bitrate=1282.1kbits/
frame=  704 fps= 37 q=-1.0 size=    3830kB time=00:00:24.27 bitrate=1292.4kbits/
frame=  719 fps= 37 q=-1.0 size=    3942kB time=00:00:24.79 bitrate=1302.4kbits/
frame=  733 fps= 37 q=-1.0 size=    4051kB time=00:00:25.27 bitrate=1312.9kbits/
frame=  748 fps= 36 q=-1.0 size=    4131kB time=00:00:25.79 bitrate=1312.0kbits/
frame=  762 fps= 36 q=-1.0 size=    4223kB time=00:00:26.27 bitrate=1316.6kbits/
frame=  777 fps= 36 q=-1.0 size=    4317kB time=00:00:26.79 bitrate=1320.1kbits/
frame=  791 fps= 36 q=-1.0 size=    4436kB time=00:00:27.27 bitrate=1332.4kbits/
frame=  806 fps= 36 q=-1.0 size=    4556kB time=00:00:27.79 bitrate=1342.8kbits/
frame=  820 fps= 36 q=-1.0 size=    4626kB time=00:00:28.27 bitrate=1340.2kbits/
frame=  835 fps= 35 q=-1.0 size=    4731kB time=00:00:28.79 bitrate=1346.1kbits/
frame=  849 fps= 35 q=-1.0 size=    4813kB time=00:00:29.27 bitrate=1346.8kbits/
frame=  864 fps= 35 q=-1.0 size=    4904kB time=00:00:29.79 bitrate=1348.4kbits/
frame=  871 fps= 35 q=-1.0 size=    4963kB time=00:00:25.28 bitrate=1608.3kbits/
frame=  871 fps= 34 q=-1.0 size=    5067kB time=00:00:25.79 bitrate=1609.5kbits/
frame=  871 fps= 33 q=-1.0 size=    5197kB time=00:00:26.28 bitrate=1619.9kbits/
frame=  871 fps= 33 q=-1.0 size=    5315kB time=00:00:26.77 bitrate=1626.1kbits/
frame=  871 fps= 32 q=-1.0 size=    5400kB time=00:00:27.28 bitrate=1621.2kbits/
frame=  871 fps= 32 q=-1.0 size=    5512kB time=00:00:27.77 bitrate=1625.7kbits/
frame=  871 fps= 31 q=-1.0 size=    5633kB time=00:00:28.28 bitrate=1631.4kbits/
frame=  871 fps= 30 q=-1.0 size=    5771kB time=00:00:28.77 bitrate=1642.7kbits/
frame=  871 fps= 30 q=-1.0 size=    5908kB time=00:00:29.29 bitrate=1652.3kbits/
frame=  871 fps= 29 q=-1.0 size=    6033kB time=00:00:29.78 bitrate=1659.5kbits/
frame=  871 fps= 29 q=-1.0 Lsize=    6094kB time=00:00:30.03 bitrate=1662.3kbits
/s
video:5493kB audio:586kB subtitle:0 global headers:0kB muxing overhead 0.249619%

From jpalmer at worldnow.com  Mon Jul 22 20:12:58 2013
From: jpalmer at worldnow.com (Jason Palmer)
Date: Mon, 22 Jul 2013 18:12:58 +0000
Subject: [FFmpeg-user] Frame rate and movies from images
In-Reply-To: <1374449432679-4660113.post@n4.nabble.com>
References: <4D8EEF52-4CB2-40F8-BF10-5F02B9F288E6@uncommonprojects.com>
	<1374449432679-4660113.post@n4.nabble.com>
Message-ID: <F45E17C592E4A44F836D8B3718529BEE0A757F84@WNMAIL-MB01.worldnow.corp>

Thanks for your reply.

If I attempt to use -r as an input option, ffmpeg gives me the error: Could not set video options: Input/output error.

I think this is because my input is not a series of image sequences nor is it a video file.  It is real time streaming from an AJA Capture Source.  As such, I am unable to set the frame rate of the input.

Any other ideas are welcome and appreciated.  Thanks.

-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of thljcl
Sent: Sunday, July 21, 2013 7:31 PM
To: ffmpeg-user at ffmpeg.org
Subject: Re: [FFmpeg-user] Frame rate and movies from images

You made an error for not recognizing the difference between -r as an input option or output option. If your source is a series of image sequence, ffmpeg will assume the frame rate of 25 FPS if none is specified as the input option; if your source is a video file, the frame rate of the source will be regarded as input option if none is specified. Just type -r "6"
before -i should serve your purpose, if I understand you correctly. -r before -i will force the frame rate on the source while ignoring the frame rate of the source, which will result in slowing down or speeding up of the playback of the video. -r after -i is the frame rate for output video, which if the frame rate is high enough, will preserve the playback speed or the length by adding or dropping frames. If the source is of variable frame rate, specifying a different frame rate as the output option may cause the output video out of sync with audio, mainly due to changing frame rate of the source itself; even though the input video is perfectly in sync with audio. Encoding a video into a series of image sequences and specifying a frame rate as an output option can force the frame rate to be constant while preserving the original length.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Frame-rate-and-movies-from-images-tp938477p4660113.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.
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From johncrossman at berkeley.edu  Mon Jul 22 23:26:29 2013
From: johncrossman at berkeley.edu (John Crossman)
Date: Mon, 22 Jul 2013 14:26:29 -0700
Subject: [FFmpeg-user] Better way to merge audio and video? (ffmpeg 1.2.1)
Message-ID: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com>

Is the following the best I can do? My current wait for process to finish
seems too long.

*ffmpeg -i audio.mp2 -i video.mp4 -qscale 0 -shortest -strict experimental
output.mp4
*

Also, should I replace  '-qscale 0'  with  '-q:v 0'?

From lou at lrcd.com  Mon Jul 22 23:38:40 2013
From: lou at lrcd.com (Lou)
Date: Mon, 22 Jul 2013 13:38:40 -0800
Subject: [FFmpeg-user] Better way to merge audio and video? (ffmpeg
 1.2.1)
In-Reply-To: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com>
References: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com>
Message-ID: <20130722133840.700b818d@lrcd.com>

On Mon, 22 Jul 2013 14:26:29 -0700
John Crossman <johncrossman at berkeley.edu> wrote:

> Is the following the best I can do? My current wait for process to finish
> seems too long.

What are you trying to do?

> *ffmpeg -i audio.mp2 -i video.mp4 -qscale 0 -shortest -strict experimental
> output.mp4
> *
> 
> Also, should I replace  '-qscale 0'  with  '-q:v 0'?

You did not include the complete console output. Note that the default
encoder used for mp4 output (libx264 or mpeg4) depends on your ffmpeg
configuration, and libx264 ignores "-qscale", but you did not provide
enough information for me to give you any suggestions.

From wimbonner at gmail.com  Mon Jul 22 23:47:02 2013
From: wimbonner at gmail.com (William C Bonner)
Date: Mon, 22 Jul 2013 14:47:02 -0700
Subject: [FFmpeg-user] FFMPeg reading from Logitech C920 generating H264
 source produces empty output file
In-Reply-To: <loom.20130722T104914-806@post.gmane.org>
References: <CALqHiHNzHNAgCELtYSBLCU01A0R=xEZFF8v3+NNa8j_rbdmvZg@mail.gmail.com>
	<loom.20130722T104914-806@post.gmane.org>
Message-ID: <CALqHiHO-4YmY2oJri-f3xg1nAT_bRbf03e3iRxXtHrZBFjbJZQ@mail.gmail.com>

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.

ffmpeg started on 2013-07-22 at 14:37:06
Report written to "ffmpeg-20130722-143706.log"
Command line:
ffmpeg -report -debug_ts -f dshow -vcodec h264 -s 1280x720 -i
"video=Logitech HD Pro Webcam C920" -vcodec copy -y test.mp4
ffmpeg version N-54691-g7f4a1fd Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 12 2013 16:31:48 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. 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
Splitting the commandline.
Reading option '-report' ... matched as option 'report' (generate a report)
with argument '1'.
Reading option '-debug_ts' ... matched as option 'debug_ts' (print
timestamp debugging info) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with argument
'dshow'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec
('copy' to copy stream)) with argument 'h264'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '1280x720'.
Reading option '-i' ... matched as input file with argument 'video=Logitech
HD Pro Webcam C920'.
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 '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 debug_ts (print timestamp debugging info) 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 video=Logitech HD Pro Webcam C920.
Applying option f (force format) with argument dshow.
Applying option vcodec (force video codec ('copy' to copy stream)) with
argument h264.
Applying option s (set frame size (WxH or abbreviation)) with argument
1280x720.
Successfully parsed a group of options.
Opening an input file: video=Logitech HD Pro Webcam C920.
[dshow @ 000000000262afa0] All info found
Input #0, dshow, from 'video=Logitech HD Pro Webcam C920':
  Duration: N/A, start: 996097.192000, bitrate: N/A
    Stream #0:0, 28, 1/10000000: Video: h264 (Constrained Baseline),
yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], 62.50 tbr, 10000k 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.
Successfully parsed a group of options.
Opening an output file: test.mp4.
Successfully opened the file.
Output #0, mp4, to 'test.mp4':
  Metadata:
    encoder         : Lavf55.12.102
    Stream #0:0, 0, 1/10000000: Video: h264 ([33][0][0][0] / 0x0021),
yuvj420p, 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 10000k tbn, 10000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
demuxer -> ist_index:0 type:video next_dts:NOPTS next_dts_time:NOPTS
next_pts:NOPTS next_pts_time:NOPTS pkt_pts:9960971920000
pkt_pts_time:996097 pkt_dts:9960971920000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:0 pkt_pts_time:0 pkt_dts:0
pkt_dts_time:0 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:16666 next_dts_time:0.016666
next_pts:16666 next_pts_time:0.016666 pkt_pts:9960972240000
pkt_pts_time:996097 pkt_dts:9960972240000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:320000 pkt_pts_time:0.032
pkt_dts:320000 pkt_dts_time:0.032 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:48666 next_dts_time:0.048666
next_pts:48666 next_pts_time:0.048666 pkt_pts:9960972560000
pkt_pts_time:996097 pkt_dts:9960972560000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:640000 pkt_pts_time:0.064
pkt_dts:640000 pkt_dts_time:0.064 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:80666 next_dts_time:0.080666
next_pts:80666 next_pts_time:0.080666 pkt_pts:9960972880000
pkt_pts_time:996097 pkt_dts:9960972880000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:960000 pkt_pts_time:0.096
pkt_dts:960000 pkt_dts_time:0.096 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:112666 next_dts_time:0.112666
next_pts:112666 next_pts_time:0.112666 pkt_pts:9960973200000
pkt_pts_time:996097 pkt_dts:9960973200000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:1280000 pkt_pts_time:0.128
pkt_dts:1280000 pkt_dts_time:0.128 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:144666 next_dts_time:0.144666
next_pts:144666 next_pts_time:0.144666 pkt_pts:9960973520000
pkt_pts_time:996097 pkt_dts:9960973520000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:1600000 pkt_pts_time:0.16
pkt_dts:1600000 pkt_dts_time:0.16 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:176666 next_dts_time:0.176666
next_pts:176666 next_pts_time:0.176666 pkt_pts:9960974000000
pkt_pts_time:996097 pkt_dts:9960974000000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:2080000 pkt_pts_time:0.208
pkt_dts:2080000 pkt_dts_time:0.208 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:224666 next_dts_time:0.224666
next_pts:224666 next_pts_time:0.224666 pkt_pts:9960974320000
pkt_pts_time:996097 pkt_dts:9960974320000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:2400000 pkt_pts_time:0.24
pkt_dts:2400000 pkt_dts_time:0.24 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:256666 next_dts_time:0.256666
next_pts:256666 next_pts_time:0.256666 pkt_pts:9960974640000
pkt_pts_time:996097 pkt_dts:9960974640000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:2720000 pkt_pts_time:0.272
pkt_dts:2720000 pkt_dts_time:0.272 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:288666 next_dts_time:0.288666
next_pts:288666 next_pts_time:0.288666 pkt_pts:9960974960000
pkt_pts_time:996097 pkt_dts:9960974960000 pkt_dts_time:996097
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:3040000 pkt_pts_time:0.304
pkt_dts:3040000 pkt_dts_time:0.304 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:320666 next_dts_time:0.320666
next_pts:320666 next_pts_time:0.320666 pkt_pts:9960975280000
pkt_pts_time:996098 pkt_dts:9960975280000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:3360000 pkt_pts_time:0.336
pkt_dts:3360000 pkt_dts_time:0.336 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:352666 next_dts_time:0.352666
next_pts:352666 next_pts_time:0.352666 pkt_pts:9960975600000
pkt_pts_time:996098 pkt_dts:9960975600000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:3680000 pkt_pts_time:0.368
pkt_dts:3680000 pkt_dts_time:0.368 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:384666 next_dts_time:0.384666
next_pts:384666 next_pts_time:0.384666 pkt_pts:9960975920000
pkt_pts_time:996098 pkt_dts:9960975920000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:4000000 pkt_pts_time:0.4
pkt_dts:4000000 pkt_dts_time:0.4 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:416666 next_dts_time:0.416666
next_pts:416666 next_pts_time:0.416666 pkt_pts:9960976240000
pkt_pts_time:996098 pkt_dts:9960976240000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:4320000 pkt_pts_time:0.432
pkt_dts:4320000 pkt_dts_time:0.432 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:448666 next_dts_time:0.448666
next_pts:448666 next_pts_time:0.448666 pkt_pts:9960976560000
pkt_pts_time:996098 pkt_dts:9960976560000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:4640000 pkt_pts_time:0.464
pkt_dts:4640000 pkt_dts_time:0.464 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:480666 next_dts_time:0.480666
next_pts:480666 next_pts_time:0.480666 pkt_pts:9960976880000
pkt_pts_time:996098 pkt_dts:9960976880000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:4960000 pkt_pts_time:0.496
pkt_dts:4960000 pkt_dts_time:0.496 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:512666 next_dts_time:0.512666
next_pts:512666 next_pts_time:0.512666 pkt_pts:9960977200000
pkt_pts_time:996098 pkt_dts:9960977200000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:5280000 pkt_pts_time:0.528
pkt_dts:5280000 pkt_dts_time:0.528 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:544666 next_dts_time:0.544666
next_pts:544666 next_pts_time:0.544666 pkt_pts:9960977680000
pkt_pts_time:996098 pkt_dts:9960977680000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:5760000 pkt_pts_time:0.576
pkt_dts:5760000 pkt_dts_time:0.576 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:592666 next_dts_time:0.592666
next_pts:592666 next_pts_time:0.592666 pkt_pts:9960978000000
pkt_pts_time:996098 pkt_dts:9960978000000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:6080000 pkt_pts_time:0.608
pkt_dts:6080000 pkt_dts_time:0.608 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:624666 next_dts_time:0.624666
next_pts:624666 next_pts_time:0.624666 pkt_pts:9960978320000
pkt_pts_time:996098 pkt_dts:9960978320000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:6400000 pkt_pts_time:0.64
pkt_dts:6400000 pkt_dts_time:0.64 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:656666 next_dts_time:0.656666
next_pts:656666 next_pts_time:0.656666 pkt_pts:9960978640000
pkt_pts_time:996098 pkt_dts:9960978640000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:6720000 pkt_pts_time:0.672
pkt_dts:6720000 pkt_dts_time:0.672 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:688666 next_dts_time:0.688666
next_pts:688666 next_pts_time:0.688666 pkt_pts:9960978960000
pkt_pts_time:996098 pkt_dts:9960978960000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:7040000 pkt_pts_time:0.704
pkt_dts:7040000 pkt_dts_time:0.704 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:720666 next_dts_time:0.720666
next_pts:720666 next_pts_time:0.720666 pkt_pts:9960979280000
pkt_pts_time:996098 pkt_dts:9960979280000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:7360000 pkt_pts_time:0.736
pkt_dts:7360000 pkt_dts_time:0.736 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:752666 next_dts_time:0.752666
next_pts:752666 next_pts_time:0.752666 pkt_pts:9960979600000
pkt_pts_time:996098 pkt_dts:9960979600000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:7680000 pkt_pts_time:0.768
pkt_dts:7680000 pkt_dts_time:0.768 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:784666 next_dts_time:0.784666
next_pts:784666 next_pts_time:0.784666 pkt_pts:9960979920000
pkt_pts_time:996098 pkt_dts:9960979920000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:8000000 pkt_pts_time:0.8
pkt_dts:8000000 pkt_dts_time:0.8 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:816666 next_dts_time:0.816666
next_pts:816666 next_pts_time:0.816666 pkt_pts:9960980240000
pkt_pts_time:996098 pkt_dts:9960980240000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:8320000 pkt_pts_time:0.832
pkt_dts:8320000 pkt_dts_time:0.832 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:848666 next_dts_time:0.848666
next_pts:848666 next_pts_time:0.848666 pkt_pts:9960980560000
pkt_pts_time:996098 pkt_dts:9960980560000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:8640000 pkt_pts_time:0.864
pkt_dts:8640000 pkt_dts_time:0.864 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:880666 next_dts_time:0.880666
next_pts:880666 next_pts_time:0.880666 pkt_pts:9960980880000
pkt_pts_time:996098 pkt_dts:9960980880000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:8960000 pkt_pts_time:0.896
pkt_dts:8960000 pkt_dts_time:0.896 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:912666 next_dts_time:0.912666
next_pts:912666 next_pts_time:0.912666 pkt_pts:9960981200000
pkt_pts_time:996098 pkt_dts:9960981200000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:9280000 pkt_pts_time:0.928
pkt_dts:9280000 pkt_dts_time:0.928 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:944000 next_dts_time:0.944
next_pts:944000 next_pts_time:0.944 pkt_pts:9960981520000
pkt_pts_time:996098 pkt_dts:9960981520000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:9600000 pkt_pts_time:0.96
pkt_dts:9600000 pkt_dts_time:0.96 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:976000 next_dts_time:0.976
next_pts:976000 next_pts_time:0.976 pkt_pts:9960982000000
pkt_pts_time:996098 pkt_dts:9960982000000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:10080000
pkt_pts_time:1.008 pkt_dts:10080000 pkt_dts_time:1.008 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1024000 next_dts_time:1.024
next_pts:1024000 next_pts_time:1.024 pkt_pts:9960982320000
pkt_pts_time:996098 pkt_dts:9960982320000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:10400000 pkt_pts_time:1.04
pkt_dts:10400000 pkt_dts_time:1.04 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1056000 next_dts_time:1.056
next_pts:1056000 next_pts_time:1.056 pkt_pts:9960982640000
pkt_pts_time:996098 pkt_dts:9960982640000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:10720000
pkt_pts_time:1.072 pkt_dts:10720000 pkt_dts_time:1.072 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1088000 next_dts_time:1.088
next_pts:1088000 next_pts_time:1.088 pkt_pts:9960982960000
pkt_pts_time:996098 pkt_dts:9960982960000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:11040000
pkt_pts_time:1.104 pkt_dts:11040000 pkt_dts_time:1.104 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1120000 next_dts_time:1.12
next_pts:1120000 next_pts_time:1.12 pkt_pts:9960983280000
pkt_pts_time:996098 pkt_dts:9960983280000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:11360000
pkt_pts_time:1.136 pkt_dts:11360000 pkt_dts_time:1.136 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1152000 next_dts_time:1.152
next_pts:1152000 next_pts_time:1.152 pkt_pts:9960983600000
pkt_pts_time:996098 pkt_dts:9960983600000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:11680000
pkt_pts_time:1.168 pkt_dts:11680000 pkt_dts_time:1.168 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1184000 next_dts_time:1.184
next_pts:1184000 next_pts_time:1.184 pkt_pts:9960983920000
pkt_pts_time:996098 pkt_dts:9960983920000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:12000000 pkt_pts_time:1.2
pkt_dts:12000000 pkt_dts_time:1.2 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1216000 next_dts_time:1.216
next_pts:1216000 next_pts_time:1.216 pkt_pts:9960984240000
pkt_pts_time:996098 pkt_dts:9960984240000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:12320000
pkt_pts_time:1.232 pkt_dts:12320000 pkt_dts_time:1.232 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1248000 next_dts_time:1.248
next_pts:1248000 next_pts_time:1.248 pkt_pts:9960984560000
pkt_pts_time:996098 pkt_dts:9960984560000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:12640000
pkt_pts_time:1.264 pkt_dts:12640000 pkt_dts_time:1.264 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1280000 next_dts_time:1.28
next_pts:1280000 next_pts_time:1.28 pkt_pts:9960984880000
pkt_pts_time:996098 pkt_dts:9960984880000 pkt_dts_time:996098
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:12960000
pkt_pts_time:1.296 pkt_dts:12960000 pkt_dts_time:1.296 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1312000 next_dts_time:1.312
next_pts:1312000 next_pts_time:1.312 pkt_pts:9960985200000
pkt_pts_time:996099 pkt_dts:9960985200000 pkt_dts_time:996099
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:13280000
pkt_pts_time:1.328 pkt_dts:13280000 pkt_dts_time:1.328 off:-996097192000
off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1344000 next_dts_time:1.344
next_pts:1344000 next_pts_time:1.344 pkt_pts:9960985520000
pkt_pts_time:996099 pkt_dts:9960985520000 pkt_dts_time:996099
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:13600000 pkt_pts_time:1.36
pkt_dts:13600000 pkt_dts_time:1.36 off:-996097192000 off_time:-996097
demuxer -> ist_index:0 type:video next_dts:1376000 next_dts_time:1.376
next_pts:1376000 next_pts_time:1.376 pkt_pts:9960986000000
pkt_pts_time:996099 pkt_dts:9960986000000 pkt_dts_time:996099
off:-996097192000 off_time:-996097
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:14080000
pkt_pts_time:1.408 pkt_dts:14080000 pkt_dts_time:1.408 off:-996097192000
off_time:-996097
frame=    0 fps=0.0 q=-1.0 size=       0kB time=00:00:00.00 bitrate=N/A
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)
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0000000006257280] Statistics: 4 seeks, 4 writeouts



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 johncrossman at berkeley.edu  Mon Jul 22 23:49:21 2013
From: johncrossman at berkeley.edu (John Crossman)
Date: Mon, 22 Jul 2013 14:49:21 -0700
Subject: [FFmpeg-user] Better way to merge audio and video? (ffmpeg
	1.2.1)
In-Reply-To: <20130722133840.700b818d@lrcd.com>
References: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com>
	<20130722133840.700b818d@lrcd.com>
Message-ID: <CANdKVX6BGL1HwamB50=iF9vvm2iTJ1hDqEz2VBRwMQvU_waNaA@mail.gmail.com>

On Mon, Jul 22, 2013 at 2:38 PM, Lou <lou at lrcd.com> wrote:

> > Is the following the best I can do? My current wait for process to finish
> > seems too long.
>
> What are you trying to do?
>

Merge the audio and video into a single video track (i.e., output.mp4)


> > *ffmpeg -i audio.mp2 -i video.mp4 -qscale 0 -shortest -strict
> experimental
> > output.mp4
> > *
> >
> > Also, should I replace  '-qscale 0'  with  '-q:v 0'?
>
> You did not include the complete console output. Note that the default
> encoder used for mp4 output (libx264 or mpeg4) depends on your ffmpeg
> configuration, and libx264 ignores "-qscale", but you did not provide
> enough information for me to give you any suggestions.
>

*Output is below. *Note: this is a job in progress so you don't get final
summary.* *I can send that once it is done.
*
*
/usr/bin/ffmpeg -i /export/mh_data/.../1013/audio.mp2 -i
/export/mh_data/.../1013/screen.mp4 -qscale 0 -shortest -strict
experimental
/export/mh_data/.../1013/screen_804358ef-3eff-41ef-9f90-fa24144a17ab-work2.mp4

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
[mp3 @ 0x12f5b60] max_analyze_duration 5000000 reached at 5015510
microseconds
[mp3 @ 0x12f5b60] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from
'/export/mh_data/opencast/shared/workspace/collection/1013/audio.mp2':
  Duration: 01:19:52.53, start: 0.000000, bitrate: 192 kb/s
    Stream #0:0: Audio: mp2, 44100 Hz, stereo, s16p, 192 kb/s
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from
'/export/mh_data/opencast/shared/workspace/collection/1013/screen.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 0
    compatible_brands: mp42mp41isomiso2
    creation_time   : 2013-07-22 19:31:00
  Duration: 01:19:59.64, start: 0.000000, bitrate: 1157 kb/s
    Stream #1:0(eng): Video: h264 (Constrained Baseline) (avc1 /
0x31637661), yuv420p, 1024x768 [SAR 1:1 DAR 4:3], 1151 kb/s, 100 fps, 100
tbr, 1k tbn, 1073741824 tbc
    Metadata:
      creation_time   : 2013-07-22 19:31:00
      handler_name    : VideoHandler
Please use -q:a or -q:v, -qscale is ambiguous
[libx264 @ 0x1307880] using SAR=1/1
[libx264 @ 0x1307880] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x1307880] profile High, level 4.2
[libx264 @ 0x1307880] 264 - core 130 - 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
'/export/mh_data/opencast/shared/workspace/collection/1013/screen_804358ef-3eff-41ef-9f90-fa24144a17ab-work2.mp4':
  Metadata:
    encoder         : Lavf54.63.104
    Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p,
1024x768 [SAR 1:1 DAR 4:3], q=-1--1, 12800 tbn, 100 tbc
    Metadata:
      creation_time   : 2013-07-22 19:31:00
      handler_name    : VideoHandler
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo,
fltp, 128 kb/s
Stream mapping:
  Stream #1:0 -> #0:0 (h264 -> libx264)
  Stream #0:0 -> #0:1 (mp2 -> aac)
Press [q] to stop, [?] for help
frame= 6820 fps=120 q=33.0 size=    7924kB time=00:01:07.70 bitrate=
958.7kbits/s

From bouke at videotoolshed.com  Tue Jul 23 00:00:16 2013
From: bouke at videotoolshed.com (Bouke (VideoToolShed))
Date: Tue, 23 Jul 2013 00:00:16 +0200
Subject: [FFmpeg-user] Better way to merge audio and video? (ffmpeg1.2.1)
References: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com><20130722133840.700b818d@lrcd.com>
	<CANdKVX6BGL1HwamB50=iF9vvm2iTJ1hDqEz2VBRwMQvU_waNaA@mail.gmail.com>
Message-ID: <263AE4F2A4A74408B3B80A4413AA42B0@HPKANTOOR>

----- Original Message ----- 
From: "John Crossman" <johncrossman at berkeley.edu>
To: "FFmpeg user questions" <ffmpeg-user at ffmpeg.org>
Sent: Monday, July 22, 2013 11:49 PM
Subject: Re: [FFmpeg-user] Better way to merge audio and video? 
(ffmpeg1.2.1)


> On Mon, Jul 22, 2013 at 2:38 PM, Lou <lou at lrcd.com> wrote:
>
>> > Is the following the best I can do? My current wait for process to 
>> > finish
>> > seems too long.
>>
>> What are you trying to do?
>>
>
> Merge the audio and video into a single video track (i.e., output.mp4)

But you are re-encoding the whole shebang.
Try -i audio -i video -vcodec copy -acodec copy outfile

Bouke

VideoToolShed
van Oldenbarneveltstraat 33
6512 AS  NIJMEGEN, the Netherlands
+31 24 3553311


>> > *ffmpeg -i audio.mp2 -i video.mp4 -qscale 0 -shortest -strict
>> experimental
>> > output.mp4
>> > *
>> >
>> > Also, should I replace  '-qscale 0'  with  '-q:v 0'?
>>
>> You did not include the complete console output. Note that the default
>> encoder used for mp4 output (libx264 or mpeg4) depends on your ffmpeg
>> configuration, and libx264 ignores "-qscale", but you did not provide
>> enough information for me to give you any suggestions.
>>
>
> *Output is below. *Note: this is a job in progress so you don't get final
> summary.* *I can send that once it is done.
> *
> *
> /usr/bin/ffmpeg -i /export/mh_data/.../1013/audio.mp2 -i
> /export/mh_data/.../1013/screen.mp4 -qscale 0 -shortest -strict
> experimental
> /export/mh_data/.../1013/screen_804358ef-3eff-41ef-9f90-fa24144a17ab-work2.mp4
>
> 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
> [mp3 @ 0x12f5b60] max_analyze_duration 5000000 reached at 5015510
> microseconds
> [mp3 @ 0x12f5b60] Estimating duration from bitrate, this may be inaccurate
> Input #0, mp3, from
> '/export/mh_data/opencast/shared/workspace/collection/1013/audio.mp2':
>  Duration: 01:19:52.53, start: 0.000000, bitrate: 192 kb/s
>    Stream #0:0: Audio: mp2, 44100 Hz, stereo, s16p, 192 kb/s
> Input #1, mov,mp4,m4a,3gp,3g2,mj2, from
> '/export/mh_data/opencast/shared/workspace/collection/1013/screen.mp4':
>  Metadata:
>    major_brand     : mp42
>    minor_version   : 0
>    compatible_brands: mp42mp41isomiso2
>    creation_time   : 2013-07-22 19:31:00
>  Duration: 01:19:59.64, start: 0.000000, bitrate: 1157 kb/s
>    Stream #1:0(eng): Video: h264 (Constrained Baseline) (avc1 /
> 0x31637661), yuv420p, 1024x768 [SAR 1:1 DAR 4:3], 1151 kb/s, 100 fps, 100
> tbr, 1k tbn, 1073741824 tbc
>    Metadata:
>      creation_time   : 2013-07-22 19:31:00
>      handler_name    : VideoHandler
> Please use -q:a or -q:v, -qscale is ambiguous
> [libx264 @ 0x1307880] using SAR=1/1
> [libx264 @ 0x1307880] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> [libx264 @ 0x1307880] profile High, level 4.2
> [libx264 @ 0x1307880] 264 - core 130 - 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
> '/export/mh_data/opencast/shared/workspace/collection/1013/screen_804358ef-3eff-41ef-9f90-fa24144a17ab-work2.mp4':
>  Metadata:
>    encoder         : Lavf54.63.104
>    Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p,
> 1024x768 [SAR 1:1 DAR 4:3], q=-1--1, 12800 tbn, 100 tbc
>    Metadata:
>      creation_time   : 2013-07-22 19:31:00
>      handler_name    : VideoHandler
>    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo,
> fltp, 128 kb/s
> Stream mapping:
>  Stream #1:0 -> #0:0 (h264 -> libx264)
>  Stream #0:0 -> #0:1 (mp2 -> aac)
> Press [q] to stop, [?] for help
> frame= 6820 fps=120 q=33.0 size=    7924kB time=00:01:07.70 bitrate=
> 958.7kbits/s
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user 


From wimbonner at gmail.com  Tue Jul 23 00:21:04 2013
From: wimbonner at gmail.com (William C Bonner)
Date: Mon, 22 Jul 2013 15:21:04 -0700
Subject: [FFmpeg-user] Streaming Formats, video overlays, other options?
Message-ID: <CALqHiHN_YLyhhSzCQA+0Ws31QDkaAxR_sR_SbtEv35AxRMEPrQ@mail.gmail.com>

I've got a question about streaming formats, video overlays, and what I
might search about for learning the possibilities. I perhaps know too much
and have become lost in the possibilities.

First, what I'd like to do: Use a low powered computer along with a camera
that produces H264 to generate video streams that can be optionally
overlayed at a receiving computer that has significantly more processing
power. I'd like the sender to send a video stream of the camera output, and
a second video stream of simple data related to the time / location /
direction / orientation of the camera. It's essentially a heads up display
overlay.

I know that DVDs produce their subtitles as a video overlay track, that can
be optionally enabled at the player and it seems to me that my data would
be a very simple video overlay compared to the underlying video stream. I
could easily produce the frames in the sender 30 times a second and believe
that they would compress much more efficiently than the camera video
stream, since most of my stream would simply be marked as transparent. I'm
hoping to not be required to transcode the camera video stream from h264
back to h264 with the overlay because so far my experience has been that my
sender cannot manage to keep up with the transcoding while it only loads
the CPU about 3% to simply do the demux/mux process.

Any pointers would be helpful. Doing this on linux, I was thinking to do
this using an FFMPEG command that got it's primary video stream from
/dev/video0 and got it's secondary input from a pipe where my program
output individual images in either GIF or PNG format, or whatever image
format might support transparent pixels.

Thanks. Wim.

From johncrossman at berkeley.edu  Tue Jul 23 00:33:11 2013
From: johncrossman at berkeley.edu (John Crossman)
Date: Mon, 22 Jul 2013 15:33:11 -0700
Subject: [FFmpeg-user] Better way to merge audio and video? (ffmpeg1.2.1)
In-Reply-To: <263AE4F2A4A74408B3B80A4413AA42B0@HPKANTOOR>
References: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com>
	<20130722133840.700b818d@lrcd.com>
	<CANdKVX6BGL1HwamB50=iF9vvm2iTJ1hDqEz2VBRwMQvU_waNaA@mail.gmail.com>
	<263AE4F2A4A74408B3B80A4413AA42B0@HPKANTOOR>
Message-ID: <CANdKVX5OCNnscqSLLWnwVxWcK1=qD-d_qdeZkeYUOsiz9CPnPQ@mail.gmail.com>

On Mon, Jul 22, 2013 at 3:00 PM, Bouke (VideoToolShed) <
bouke at videotoolshed.com> wrote:

> Merge the audio and video into a single video track (i.e., output.mp4)
>>
>
> But you are re-encoding the whole shebang.
> Try -i audio -i video -vcodec copy -acodec copy outfile
>

Thanks! After one sample run, the output looks good. Much faster.

Per Lou's comment, regardless of my operation I should never need '-qscale'
(or the equivalent '-q:v') if my configuration includes
*--enable-libx264 *(which
it does). Right?

Thanks again.

From gjholtslander at gmail.com  Tue Jul 23 00:51:10 2013
From: gjholtslander at gmail.com (Gordon Holtslander)
Date: Mon, 22 Jul 2013 16:51:10 -0600
Subject: [FFmpeg-user] Frame number repeatsllnd time resets when encoding
Message-ID: <CAD2Cw36Fu4B0URCAfE54q2=mc1XVA=uNHFGWh5UyAtO9LiXMeg@mail.gmail.com>

Dz lu pi poo bull pmlplhl I'll m loop
On Jul 22, 2013 11:47 AM, "Jason Palmer" <jpalmer at worldnow.com> wrote:

> Hi
>
> I am attempting to create a .mp4 file from an AJA Capture Source and I am
> seeing the following that I cannot explain:
>
>
> 1.       The audio is out of sync with the video.  I have tried the
> -async, -itsoffset, -ss flags and they did not work.
>
> 2.       The frame number starts at frame 167 (instead of 0) and it
> repeats itself towards the end of the recorded video.  In the output below,
> note how frame 871 is repeated multiple times.
>
> 3.       The time of the output of ffmpeg starts at 5.75 seconds (instead
> of 0 seconds) and starts over.  In the output below, note how once the
> output reaches 30 seconds (as specified by the -t flag), it starts over for
> about the same amount of time that ffmpeg takes to start the recording
> (~5.75 seconds).  It is also at this time when the frame number repeats
> itself.
>
> Interestingly, the 5.75 seconds that ffmpeg takes to start recording is
> approximately how long the audio/video is out of sync.  Note that I am
> copying the video stream (which is in h264 format) and the audio stream
> (which is in aac format).  Any help/explanation would be greatly
> appreciated.
>
> Here is the command I am using:
>
> FFMPEG.EXE -y -rtbufsize 100M -f dshow -i video="Virtual Capture Source"
> -acodec aac -f dshow -i audio="Virtual Capture Source" -map 0:0 -map 1:0
> -absf aac_adtstoasc -acodec copy -vcodec copy -vsync drop -r 29 -copyinkf
> -aspect 16:9 -t 00:00:30 out.mp4
>
> Here is my output:
>
> FFMPEG.EXE -y -rtbufsize 100M -f dshow -i video="Virtual Capture Sour
> ce" -acodec aac -f dshow -i audio="Virtual Capture Source" -map 0:0 -map
> 1:0 -ab
> sf aac_adtstoasc -acodec copy -vcodec copy -vsync drop -r 29 -copyinkf
> -aspect 1
> 6:9 -t 00:00:30 out.mp4
> ffmpeg started on 2013-07-22 at 11:37:50
> Report written to "log.txt"
> ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
>   built on Jul 15 2013 16:04:21 with gcc 4.8.0 (rubenvb-4.8.0)
>   configuration: --enable-static --disable-shared
> --prefix=/mingw/i686-w64-mingw
> 32 --arch=x86 --target-os=mingw32 --enable-gpl --enable-nonfree
> --enable-runtime
> -cpudetect --enable-openssl --enable-libfreetype --enable-fontconfig
> --enable-li
> bass --enable-zlib --enable-librtmp --enable-w32threads --enable-libx264
> --enabl
> e-libfdk-aac --enable-debug --disable-optimizations --disable-stripping
>   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
>   libpostproc    52.  3.100 / 52.  3.100
> [h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
> [h264 @ 05e8e5e0] non-existing PPS 0 referenced
> [h264 @ 05e8e5e0] decode_slice_header error
> [h264 @ 05e8e5e0] no frame!
> [h264 @ 05e8e5e0] non-existing SPS 0 referenced in buffering period
> [h264 @ 05e8e5e0] non-existing PPS 0 referenced
> [h264 @ 05e8e5e0] decode_slice_header error
> [h264 @ 05e8e5e0] no frame!
> [h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
> [h264 @ 05e8e5e0] non-existing PPS 0 referenced
> [h264 @ 05e8e5e0] decode_slice_header error
> [h264 @ 05e8e5e0] no frame!
> [h264 @ 05e8e5e0] non-existing SPS 0 referenced in buffering period
> [h264 @ 05e8e5e0] non-existing PPS 0 referenced
> [h264 @ 05e8e5e0] decode_slice_header error
> [h264 @ 05e8e5e0] no frame!
> [h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
> [h264 @ 05e8e5e0] non-existing PPS 0 referenced
> [h264 @ 05e8e5e0] decode_slice_header error
> [h264 @ 05e8e5e0] no frame!
> [h264 @ 05e8e5e0] non-existing SPS 0 referenced in buffering period
> [h264 @ 05e8e5e0] non-existing PPS 0 referenced
> [h264 @ 05e8e5e0] decode_slice_header error
> [h264 @ 05e8e5e0] no frame!
> [h264 @ 05e8e5e0] non-existing SPS 32 referenced in buffering period
> [h264 @ 05e8e5e0] non-existing PPS 0 referenced
> [h264 @ 05e8e5e0] decode_slice_header error
> [h264 @ 05e8e5e0] no frame!
> Input #0, dshow, from 'video=Virtual Capture Source':
>   Duration: N/A, start: 2341547.916000, bitrate: N/A
>     Stream #0:0: Video: h264 (Main), yuv420p, 854x480 [SAR 1:1 DAR
> 427:240], 4 t
> br, 10000k tbn, 58 tbc
> Input #1, dshow, from 'audio=Virtual Capture Source':
>   Duration: N/A, start: 2341552.949000, bitrate: N/A
>     Stream #1:0: Audio: aac, 48000 Hz, stereo, fltp
> Overriding aspect ratio with stream copy may produce invalid files
> Output #0, mp4, to out.mp4':
>   Metadata:
>     encoder         : Lavf55.12.100
>     Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 854x480
> [SAR 128
> 0:1281 DAR 16:9], q=2-31, 14848 tbn, 29 tbc
>     Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo
> Stream mapping:
>   Stream #0:0 -> #0:0 (copy)
>   Stream #1:0 -> #0:1 (copy)
> Press [q] to stop, [?] for help
> [mp4 @ 0767f040] Encoder did not produce proper pts, making some up.
> frame=  167 fps=0.0 q=-1.0 size=     154kB time=00:00:05.75 bitrate=
> 219.6kbits/
> frame=  182 fps=182 q=-1.0 size=     245kB time=00:00:06.27 bitrate=
> 319.9kbits/
> frame=  196 fps=130 q=-1.0 size=     326kB time=00:00:06.75 bitrate=
> 395.6kbits/
> frame=  211 fps=105 q=-1.0 size=     430kB time=00:00:07.27 bitrate=
> 483.7kbits/
> frame=  225 fps= 90 q=-1.0 size=     559kB time=00:00:07.75 bitrate=
> 590.0kbits/
> frame=  240 fps= 80 q=-1.0 size=     697kB time=00:00:08.27 bitrate=
> 689.8kbits/
> frame=  254 fps= 72 q=-1.0 size=     776kB time=00:00:08.75 bitrate=
> 725.6kbits/
> frame=  269 fps= 67 q=-1.0 size=     849kB time=00:00:09.27 bitrate=
> 750.0kbits/
> frame=  283 fps= 63 q=-1.0 size=     910kB time=00:00:09.75 bitrate=
> 763.9kbits/
> frame=  298 fps= 59 q=-1.0 size=     983kB time=00:00:10.27 bitrate=
> 784.0kbits/
> frame=  312 fps= 57 q=-1.0 size=    1074kB time=00:00:10.75 bitrate=
> 818.0kbits/
> frame=  327 fps= 54 q=-1.0 size=    1175kB time=00:00:11.27 bitrate=
> 854.0kbits/
> frame=  341 fps= 52 q=-1.0 size=    1272kB time=00:00:11.75 bitrate=
> 886.3kbits/
> frame=  356 fps= 51 q=-1.0 size=    1358kB time=00:00:12.27 bitrate=
> 905.9kbits/
> frame=  371 fps= 49 q=-1.0 size=    1470kB time=00:00:12.79 bitrate=
> 941.4kbits/
> frame=  385 fps= 48 q=-1.0 size=    1577kB time=00:00:13.27 bitrate=
> 973.1kbits/
> frame=  400 fps= 47 q=-1.0 size=    1677kB time=00:00:13.79 bitrate=
> 996.0kbits/
> frame=  414 fps= 46 q=-1.0 size=    1788kB time=00:00:14.27
> bitrate=1026.2kbits/
> frame=  429 fps= 45 q=-1.0 size=    1907kB time=00:00:14.79
> bitrate=1055.8kbits/
> frame=  443 fps= 44 q=-1.0 size=    2020kB time=00:00:15.27
> bitrate=1083.3kbits/
> frame=  458 fps= 43 q=-1.0 size=    2152kB time=00:00:15.79
> bitrate=1116.2kbits/
> frame=  472 fps= 43 q=-1.0 size=    2278kB time=00:00:16.27
> bitrate=1146.6kbits/
> frame=  487 fps= 42 q=-1.0 size=    2338kB time=00:00:16.79
> bitrate=1140.3kbits/
> frame=  501 fps= 42 q=-1.0 size=    2429kB time=00:00:17.27
> bitrate=1152.0kbits/
> frame=  516 fps= 41 q=-1.0 size=    2543kB time=00:00:17.79
> bitrate=1171.0kbits/
> frame=  530 fps= 41 q=-1.0 size=    2626kB time=00:00:18.27
> bitrate=1177.2kbits/
> frame=  545 fps= 40 q=-1.0 size=    2725kB time=00:00:18.79
> bitrate=1187.9kbits/
> frame=  559 fps= 40 q=-1.0 size=    2840kB time=00:00:19.27
> bitrate=1207.1kbits/
> frame=  574 fps= 39 q=-1.0 size=    2941kB time=00:00:19.79
> bitrate=1217.1kbits/
> frame=  588 fps= 39 q=-1.0 size=    3058kB time=00:00:20.27
> bitrate=1235.4kbits/
> frame=  603 fps= 39 q=-1.0 size=    3170kB time=00:00:20.79
> bitrate=1249.1kbits/
> frame=  617 fps= 38 q=-1.0 size=    3299kB time=00:00:21.27
> bitrate=1270.2kbits/
> frame=  632 fps= 38 q=-1.0 size=    3335kB time=00:00:21.79
> bitrate=1253.6kbits/
> frame=  646 fps= 38 q=-1.0 size=    3424kB time=00:00:22.27
> bitrate=1259.1kbits/
> frame=  661 fps= 38 q=-1.0 size=    3554kB time=00:00:22.79
> bitrate=1277.4kbits/
> frame=  675 fps= 37 q=-1.0 size=    3633kB time=00:00:23.27
> bitrate=1278.6kbits/
> frame=  690 fps= 37 q=-1.0 size=    3724kB time=00:00:23.79
> bitrate=1282.1kbits/
> frame=  704 fps= 37 q=-1.0 size=    3830kB time=00:00:24.27
> bitrate=1292.4kbits/
> frame=  719 fps= 37 q=-1.0 size=    3942kB time=00:00:24.79
> bitrate=1302.4kbits/
> frame=  733 fps= 37 q=-1.0 size=    4051kB time=00:00:25.27
> bitrate=1312.9kbits/
> frame=  748 fps= 36 q=-1.0 size=    4131kB time=00:00:25.79
> bitrate=1312.0kbits/
> frame=  762 fps= 36 q=-1.0 size=    4223kB time=00:00:26.27
> bitrate=1316.6kbits/
> frame=  777 fps= 36 q=-1.0 size=    4317kB time=00:00:26.79
> bitrate=1320.1kbits/
> frame=  791 fps= 36 q=-1.0 size=    4436kB time=00:00:27.27
> bitrate=1332.4kbits/
> frame=  806 fps= 36 q=-1.0 size=    4556kB time=00:00:27.79
> bitrate=1342.8kbits/
> frame=  820 fps= 36 q=-1.0 size=    4626kB time=00:00:28.27
> bitrate=1340.2kbits/
> frame=  835 fps= 35 q=-1.0 size=    4731kB time=00:00:28.79
> bitrate=1346.1kbits/
> frame=  849 fps= 35 q=-1.0 size=    4813kB time=00:00:29.27
> bitrate=1346.8kbits/
> frame=  864 fps= 35 q=-1.0 size=    4904kB time=00:00:29.79
> bitrate=1348.4kbits/
> frame=  871 fps= 35 q=-1.0 size=    4963kB time=00:00:25.28
> bitrate=1608.3kbits/
> frame=  871 fps= 34 q=-1.0 size=    5067kB time=00:00:25.79
> bitrate=1609.5kbits/
> frame=  871 fps= 33 q=-1.0 size=    5197kB time=00:00:26.28
> bitrate=1619.9kbits/
> frame=  871 fps= 33 q=-1.0 size=    5315kB time=00:00:26.77
> bitrate=1626.1kbits/
> frame=  871 fps= 32 q=-1.0 size=    5400kB time=00:00:27.28
> bitrate=1621.2kbits/
> frame=  871 fps= 32 q=-1.0 size=    5512kB time=00:00:27.77
> bitrate=1625.7kbits/
> frame=  871 fps= 31 q=-1.0 size=    5633kB time=00:00:28.28
> bitrate=1631.4kbits/
> frame=  871 fps= 30 q=-1.0 size=    5771kB time=00:00:28.77
> bitrate=1642.7kbits/
> frame=  871 fps= 30 q=-1.0 size=    5908kB time=00:00:29.29
> bitrate=1652.3kbits/
> frame=  871 fps= 29 q=-1.0 size=    6033kB time=00:00:29.78
> bitrate=1659.5kbits/
> frame=  871 fps= 29 q=-1.0 Lsize=    6094kB time=00:00:30.03
> bitrate=1662.3kbits
> /s
> video:5493kB audio:586kB subtitle:0 global headers:0kB muxing overhead
> 0.249619%
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From johncrossman at berkeley.edu  Tue Jul 23 01:53:26 2013
From: johncrossman at berkeley.edu (John Crossman)
Date: Mon, 22 Jul 2013 16:53:26 -0700
Subject: [FFmpeg-user] ffmpeg optimization question:
Message-ID: <CANdKVX6AaFYLc+7Cse9i2db=_O-KoFsm1HbTRp=3gsi_-nQJKA@mail.gmail.com>

The performance improvement I received from previous suggestions on merging
video and audio tracks makes me thing other ffmpeg operations could be
faster.

*The use case*. Our system publishes videos to YouTube and uses ffmpeg in
two separate steps:

   1. Join video1 and video2 sources into one side-by-side composition
   2. Encode merged video. We want to publish HD video to YouTube.

Here are the two ffmpeg (1.2.1) commands we use to achieve the above:

   1. ffmpeg -i *video1.mp4* -i  *video2.mp4*  -filter_complex
   "[1:v]pad=iw*2:ih[bg];[bg][0:v]overlay=W/2+((W/2-w)/2):(H-h)/2" -qscale:v 2
   -strict -2  *video3.mp4*
   2. ffmpeg -i *video3.mp4*  -acodec libmp3lame -ab 64k -ar 22050 -r 25
   -vcodec libx264 -level 41 -crf 20 -bufsize 20000k -maxrate 25000k -g 250 -s
   1280x720 -coder 1  -subq 7 -me_range 16 -keyint_min 25 -sc_threshold 40 *
   final.mp4*

My question: How can I optimize the command #2 above? I have sample stdout
below.

Thanks again.

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

/usr/bin/ffmpeg -i video.mp4  -acodec libmp3lame -ab 64k -ar 22050 -r 25
-vcodec libx264 -level 41 -crf 20 -bufsize 20000k -maxrate 25000k -g 250 -s
1280x720 -coder 1  -subq 7 -me_range 16 -keyint_min 25 -sc_threshold 40
result.mp4

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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.63.104
  Duration: 00:08:01.02, start: 0.000000, bitrate: 234 kb/s
    Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1024x768 [SAR 1:1 DAR 4:3], 28 kb/s, 100 fps, 100 tbr, 12800 tbn, 200 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: mp2 (mp4a / 0x6134706D), 44100 Hz, stereo,
s16p, 191 kb/s
    Metadata:
      handler_name    : SoundHandler
[libx264 @ 0x1ef40c0] using SAR=3/4
[libx264 @ 0x1ef40c0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x1ef40c0] profile High, level 4.1
[libx264 @ 0x1ef40c0] 264 - core 130 - 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=20.0 qcomp=0.60 qpmin=0
qpmax=69 qpstep=4 vbv_maxrate=25000 vbv_bufsize=20000 crf_max=0.0
nal_hrd=none ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'result.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,
1280x720 [SAR 3:4 DAR 4:3], q=-1--1, 12800 tbn, 25 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: mp3 (i[0][0][0] / 0x0069), 22050 Hz, stereo,
s16p, 64 kb/s
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (mp2 -> libmp3lame)
Press [q] to stop, [?] for help
frame=  321 fps= 36 q=32766.0 Lsize=     126kB time=00:00:12.76 bitrate=
 81.2kbits/s dup=0 drop=951
video:14kB audio:100kB subtitle:0 global headers:0kB muxing overhead
11.387343%
[libx264 @ 0x1ef40c0] frame I:2     Avg QP: 4.50  size:   216
[libx264 @ 0x1ef40c0] frame P:81    Avg QP: 6.14  size:    46
[libx264 @ 0x1ef40c0] frame B:238   Avg QP: 9.67  size:    38
[libx264 @ 0x1ef40c0] consecutive B-frames:  0.9%  0.6%  0.0% 98.4%
[libx264 @ 0x1ef40c0] mb I  I16..4: 100.0%  0.0%  0.0%
[libx264 @ 0x1ef40c0] mb P  I16..4:  0.0%  0.0%  0.0%  P16..4:  0.0%  0.0%
 0.0%  0.0%  0.0%    skip:100.0%
[libx264 @ 0x1ef40c0] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  0.0%  0.0%
 0.0%  direct: 0.0%  skip:100.0%
[libx264 @ 0x1ef40c0] 8x8 transform intra:0.0%
[libx264 @ 0x1ef40c0] coded y,uvDC,uvAC intra: 0.0% 0.0% 0.0% inter: 0.0%
0.0% 0.0%
[libx264 @ 0x1ef40c0] i16 v,h,dc,p: 98%  0%  2%  0%
[libx264 @ 0x1ef40c0] i8c dc,h,v,p: 100%  0%  0%  0%
[libx264 @ 0x1ef40c0] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x1ef40c0] kb/s:8.17

From lou at lrcd.com  Tue Jul 23 06:14:59 2013
From: lou at lrcd.com (Lou)
Date: Mon, 22 Jul 2013 20:14:59 -0800
Subject: [FFmpeg-user] Better way to merge audio and video? (ffmpeg1.2.1)
In-Reply-To: <CANdKVX5OCNnscqSLLWnwVxWcK1=qD-d_qdeZkeYUOsiz9CPnPQ@mail.gmail.com>
References: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com>
	<20130722133840.700b818d@lrcd.com>
	<CANdKVX6BGL1HwamB50=iF9vvm2iTJ1hDqEz2VBRwMQvU_waNaA@mail.gmail.com>
	<263AE4F2A4A74408B3B80A4413AA42B0@HPKANTOOR>
	<CANdKVX5OCNnscqSLLWnwVxWcK1=qD-d_qdeZkeYUOsiz9CPnPQ@mail.gmail.com>
Message-ID: <1374552899.13887.9223372036855943389.77D060E0@webmail.messagingengine.com>

On Mon, Jul 22, 2013, at 02:33 PM, John Crossman wrote:
> On Mon, Jul 22, 2013 at 3:00 PM, Bouke (VideoToolShed) <
> bouke at videotoolshed.com> wrote:
> 
> > Merge the audio and video into a single video track (i.e., output.mp4)
> >>
> >
> > But you are re-encoding the whole shebang.
> > Try -i audio -i video -vcodec copy -acodec copy outfile

Or simply:
ffmpeg -i audio -i video -c copy output

See description of "stream copy":
https://ffmpeg.org/ffmpeg.html#Stream-copy

Note that by default ffmpeg will only copy one stream of each type:
https://ffmpeg.org/ffmpeg.html#Stream-selection

You can use the -map option to include any or all streams if the default
behavior is not desired.

mp4 output container may not support all of your input formats, so
consider another container format such as Matroska (.mkv) if this is the
case. Or, of course, you can always re-encode if you need mp4.

> Thanks! After one sample run, the output looks good. Much faster.
> 
> Per Lou's comment, regardless of my operation I should never need
> '-qscale'
> (or the equivalent '-q:v') if my configuration includes
> *--enable-libx264 *(which
> it does). Right?

Since your console output indicates that libx264 is the default encoder
for mp4 output then -q:v will be ignored by this encoder resulting in
the default of "-crf 23" (also shown in the console output).

From colinbgames at gmail.com  Mon Jul 22 23:08:05 2013
From: colinbgames at gmail.com (Colin Barnette)
Date: Mon, 22 Jul 2013 17:08:05 -0400
Subject: [FFmpeg-user] Problem with concat at the stitch point!
Message-ID: <CAFGJ85-mNrO-iWeiH1z0R=AvbyJY2c2Eqg1zGvoC5jws+6WiLg@mail.gmail.com>

***Apologies for the double-send.. the new Gmail key shortcuts prematurely
sent this e-mail! ***

Hi all,

I have been using ffmpeg to encode videos of gameplay recorded with FRAPS.
 I encode the files to .avi at 11000k bitrate, all with the same settings.
I then use the concat function to stitch the files together.  This usually
is not a problem however a stitch point seems to have a problem in which
the dithering from frame to frame gets mangled and the picture becomes
blocky and strangely interpolated This continues for the rest of the video.

In this instance, the problem occurs precisely at the stitch point from
file 3 to file 4.

Is this behavior documented? Have I been somewhat clear in my explanation?
 I really am not sure what to do about this as I have tried to re-concat
the files to no avail!

Any clues or help would be very appreciated!

Thanks!

Here's an example of my batch script to first encode:

ffmpeg -i s7-1.avi -b:v 11000k 7-1.avi
ffmpeg -i s7-2.avi -b:v 11000k 7-2.avi
ffmpeg -i s7-3.avi -b:v 11000k 7-3.avi
ffmpeg -i s7-4.avi -b:v 11000k 7-4.avi

And the command I use to concat:

ffmpeg -f concat -i list.txt -c copy 7-Final.avi

And a portion of 'list.txt':

file '7-1.avi'
file '7-2.avi'
file '7-3.avi'
file '7-4.avi'

From jiachielee at live.com  Mon Jul 22 23:06:37 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 22 Jul 2013 14:06:37 -0700 (PDT)
Subject: [FFmpeg-user] iMovie created variable frame rates cause a/v
	sync problems
In-Reply-To: <gpmgoi$e5o$1@ger.gmane.org>
References: <gpmgoi$e5o$1@ger.gmane.org>
Message-ID: <1374527197109-4660121.post@n4.nabble.com>

I come across video of variable frame rate when I tried to encode video with
the constant frame rate of 

15 FPS into video with the constant frame rate of 24 FPS. I typically create
batch files to let command 

interpreter execute a series of command sequence, instead of entering only a
line of command once at a 

time.
ffmpeg -i "input.mkv" -r "24" -vcodec "libx264" -crf "0" "output.mkv"
Unknown to me, however, ffmpeg would encode the video into a video of
variable frame rate when the frame 

rate of the source is lower than output video. The unintended consequence
would be that the output video 

is out of sync with audio. Without changing the ?playback speed?, changing
frame rate itself will 

inevitably result in the change of video length, even though such difference
may be negligible. ?Out of 

sync? is obviously caused by unintended change of playback speed of the
output video.
Of course, a video of variable frame rate can be created by joining videos
of different (constant) frame 

rate together. The solution is obviously to avoid unintended change of
playback speed of the output 

video. If I can make the output video having constant frame rate, I thought
that the output video might 

be able to be in sync with audio.
If the input video is of constant frame rate, it is possible to convert it
to a series of image 

sequence.
mkdir "output"
ffmpeg -i "input.mkv" "output\f_%%1d.png"
What if the input video is of variable frame rate?
It turns out the image sequence cannot have variable frame rate. I
specifically joining three videos 

together; each of them with the length of one second; first is of 15 FPS;
second is of 24 FPS; third is 

of 30 FPS. Media Player Classic reports the joined video is of 15 FPS and
variable frame rate. The 

average frame rate of the joined video would be 23 FPS; but Media Player
Classic only reports 15 FPS, 

which is the frame rate for the first second. When I use the commands:
mkdir "output"
ffmpeg -i "input.mkv" "output\f_%%1d.png"
ffmpeg actually makes the image sequence of 15 FPS when no frame rate as
output option is being 

specified. In this case, it actually needs to drop frames to achieve the
frame rate of 15 FPS.
When it comes to suitable frame rate, it?s worth mentioning that the basic
notion of video is that human 

eyes can only discriminate up to 12-15 frames every second, subject to light
condition and other 

factors; the film industry commonly adopts 24 FPS as the standard frame
rate. 
The short answer would be:
mkdir "output"
ffmpeg -i "input.mkv" -r "24" "output\f_%%1d.png"
ffmpeg -r "24" -i "output\f_%%1d.png" -vcodec "libx264" -crf "0"
"output.mkv"
rmdir /s /q "output"
Here I use Constant Rate Factor of ?0?, which is lossless encoding of x264.




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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/iMovie-created-variable-frame-rates-cause-a-v-sync-problems-tp939895p4660121.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jiachielee at live.com  Tue Jul 23 04:26:34 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 22 Jul 2013 19:26:34 -0700 (PDT)
Subject: [FFmpeg-user] Better way to merge audio and video? (ffmpeg1.2.1)
In-Reply-To: <CANdKVX5OCNnscqSLLWnwVxWcK1=qD-d_qdeZkeYUOsiz9CPnPQ@mail.gmail.com>
References: <CANdKVX5L5iQTJzD+oJncoi8X=63wsSOc_ikv3iovJ6NfXRM=Xw@mail.gmail.com>
	<20130722133840.700b818d@lrcd.com>
	<CANdKVX6BGL1HwamB50=iF9vvm2iTJ1hDqEz2VBRwMQvU_waNaA@mail.gmail.com>
	<263AE4F2A4A74408B3B80A4413AA42B0@HPKANTOOR>
	<CANdKVX5OCNnscqSLLWnwVxWcK1=qD-d_qdeZkeYUOsiz9CPnPQ@mail.gmail.com>
Message-ID: <1374546394138-4660131.post@n4.nabble.com>

Personally, I recommend mkvtoolnix to serve your purpose. More specifically,
it allows to create, alter, split, join and inspect Matroska files without
re-encoding the video/audio. Matroska container allows virtually any number
of video/audio/subtitle track of any codec. Some popular media players, such
as Media Player Classic Home Cinema, allows you to play variety of
video/audio codecs. In your case, it can read mp4 files directly and put the
video/audio track into Matroska files. If you must use mp4 container for
some reasons, if you can copy the audio/video tracks into mp4 files,ffmpeg
-i "input.mkv" -codec "copy" "output.mp4"Note that mp4 files only support
certain audio/video codecs, such as H.264 or AAC.I would admit that ffmpeg
is an excellent encoding tool. I assume that you want to create a file with
a video track and audio track. You may recall how you can change languages,
subtitles, or video angles in DVD playback; it?s because of the existence of
multiple tracks of audio, video, and subtitle in VOB file containers. For
more information, go to http://bit.ly/1OBiH , http://bit.ly/14VDQ , and
http://bit.ly/2Xyp4 



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Better-way-to-merge-audio-and-video-ffmpeg-1-2-1-tp4660122p4660131.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From 99191957 at qq.com  Tue Jul 23 10:01:07 2013
From: 99191957 at qq.com (99191957 at qq.com)
Date: Tue, 23 Jul 2013 16:01:07 +0800
Subject: [FFmpeg-user] How to show the encoding or decoding latency?
Message-ID: <201307231601068321925@qq.com>

I want to make a comparasison between Inter Media SDK's sample decoder and ffmpeg, video format: .264- Yuv420.
But I don't know how to show decode latency when using ffmpeg. 
In order to be impartial, is there any more parameters should be set? 
thank you guys.



99191957 at qq.com

From francois.visagie at gmail.com  Tue Jul 23 11:57:15 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Tue, 23 Jul 2013 11:57:15 +0200
Subject: [FFmpeg-user] ffmpeg how to set stream layout
In-Reply-To: <CAFd2cZUe8WJ13D+XtUY6Cgr_au1-v33Dmzm23=fLoX2LrmHZGA@mail.gmail.com>
References: <CAFd2cZUe8WJ13D+XtUY6Cgr_au1-v33Dmzm23=fLoX2LrmHZGA@mail.gmail.com>
Message-ID: <000001ce878b$081ceb80$1856c280$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Soho Soho123
> Sent: 16 July 2013 07:24
> To: FFmpeg user questions and RTFMs
> Subject: [FFmpeg-user] ffmpeg how to set stream layout
> 
> Hi All,
> 
> I use ffmpeg command to generate audio and video stream.
> ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264
> -i /dev/video1 -f alsa -ar 44100 -ac 2 -i hw:0 -vcodec copy -acodec copy
> http://localhost:8090/feed1.ffm
> 
> I got the stream layout :
>  Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 44100 Hz, stereo, 1411 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
> 
> Can I modify the layout to
> Stream 0 is video , and stream 1 is audio?

Yes, use the -map option to enumerate streams in the order you want them to
appear in the output.

Cheers,
Francois

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


From soho123.2012 at gmail.com  Tue Jul 23 14:25:03 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 23 Jul 2013 20:25:03 +0800
Subject: [FFmpeg-user] ffmpeg how to set stream layout
In-Reply-To: <000001ce878b$081ceb80$1856c280$@gmail.com>
References: <CAFd2cZUe8WJ13D+XtUY6Cgr_au1-v33Dmzm23=fLoX2LrmHZGA@mail.gmail.com>
	<000001ce878b$081ceb80$1856c280$@gmail.com>
Message-ID: <CAFd2cZW_3pUhXJGAe2YxJa=4nDWTUGkrwWXmMVAfCcddjzA9vw@mail.gmail.com>

> I use ffmpeg command to generate audio and video stream.
> > ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format
> h264
> > -i /dev/video1 -f alsa -ar 44100 -ac 2 -i hw:0 -vcodec copy -acodec copy
> > http://localhost:8090/feed1.ffm
> >
> > I got the stream layout :
> >  Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 44100 Hz, stereo, 1411 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
> >
> > Can I modify the layout to
> > Stream 0 is video , and stream 1 is audio?
>
> Yes, use the -map option to enumerate streams in the order you want them to
> appear in the output.
>
> Cheers,
> Francois
>
> Hi,
Excuse me!
Do you know where is the example about "-map" option usage?

Thanks!

From sw.craftsman at gmail.com  Tue Jul 23 15:21:51 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Tue, 23 Jul 2013 18:51:51 +0530
Subject: [FFmpeg-user] Video bit rate missing from FFProbe output
Message-ID: <CA+XaXC=J7BMNZTF3orYpr_pNrDbmYdVmUycwJeWMa9b7NHGh5w@mail.gmail.com>

Hi,

I have upgraded FFMpeg / FFProbe from a very old version 0.6.1 to 1.2 some
time back. I just realize I no longer see video bit rate in the output for
formats like gxf, mpeg2, mss, mxf. It is coming for other formats as always.

(0.6.1 version)
------------------

C:\Rajiv\"C:\Users\Administrator\desktop\FFMpeg\0.6.1\ffprobe.exe" cc.mpg2

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, mpeg, from 'cc.mpg2':
Duration: 00:01:00.00, start: 0.224556, bitrate: 11993 kb/s
Stream #0:0[0x1e0]: Video: mpeg2video (4:2:2), yuv422p, 720x512 [SAR 128:135
DAR 4:3], 11386 kb/s, 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc
Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s

-------------------------------------------------------------
(1.2 version)
---------------
C:\Rajiv\"C:\Users\Administrator\desktop\FFMpeg\1.2\ffprobe.exe" cc.mpg2

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, mpeg, from 'cc.mpg2':
Duration: 00:01:00.00, start: 0.224556, bitrate: 11993 kb/s
Stream #0:0[0x1e0]: Video: mpeg2video (4:2:2), yuv422p, 720x512 [SAR 128:135
DAR 4:3], 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc
Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 384 kb/s

Is this a regression ?

Thanks

From oliver at fromme.com  Tue Jul 23 17:18:10 2013
From: oliver at fromme.com (Oliver Fromme)
Date: Tue, 23 Jul 2013 17:18:10 +0200 (CEST)
Subject: [FFmpeg-user] Problem with concat at the stitch point!
In-Reply-To: <CAFGJ85-mNrO-iWeiH1z0R=AvbyJY2c2Eqg1zGvoC5jws+6WiLg@mail.gmail.com>
Message-ID: <201307231518.r6NFIAu3083755@haluter.fromme.com>

Colin Barnette wrote:
 > I have been using ffmpeg to encode videos of gameplay recorded with FRAPS.
 >  I encode the files to .avi at 11000k bitrate, all with the same settings.
 > I then use the concat function to stitch the files together.  This usually
 > is not a problem however a stitch point seems to have a problem in which
 > the dithering from frame to frame gets mangled and the picture becomes
 > blocky and strangely interpolated This continues for the rest of the video.
 > 
 > In this instance, the problem occurs precisely at the stitch point from
 > file 3 to file 4.
 > 
 > Is this behavior documented? Have I been somewhat clear in my explanation?
 >  I really am not sure what to do about this as I have tried to re-concat
 > the files to no avail!
 > 
 > Any clues or help would be very appreciated!
 > 
 > Thanks!
 > 
 > Here's an example of my batch script to first encode:
 > 
 > ffmpeg -i s7-1.avi -b:v 11000k 7-1.avi
 > ffmpeg -i s7-2.avi -b:v 11000k 7-2.avi
 > ffmpeg -i s7-3.avi -b:v 11000k 7-3.avi
 > ffmpeg -i s7-4.avi -b:v 11000k 7-4.avi
 > 
 > And the command I use to concat:
 > 
 > ffmpeg -f concat -i list.txt -c copy 7-Final.avi
 > 
 > And a portion of 'list.txt':
 > 
 > file '7-1.avi'
 > file '7-2.avi'
 > file '7-3.avi'
 > file '7-4.avi'

When I concatenate files, I use a different syntax:

ffmpeg -i "concat:file1|file2|file3|file4" ...

I don't know if that makes any difference, but it might be
worth a try.

Also, it might work better to concatenate the source files
first and then encode the result to 11000 kbps, i.e.:

ffmpeg -i "concat:s7-1.avi|s7-2.avi|s7-3.avi|s7-4.avi" -b:v 11000k 7-Final.avi

Finally, please make sure that you use a reasonably recent
version of ffmpeg (the version from git head if possible).
In particular, do not use any version older than 1.2.

Best regards
   Oliver

-- 

From cehoyos at ag.or.at  Tue Jul 23 22:01:39 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 23 Jul 2013 20:01:39 +0000 (UTC)
Subject: [FFmpeg-user] Video bit rate missing from FFProbe output
References: <CA+XaXC=J7BMNZTF3orYpr_pNrDbmYdVmUycwJeWMa9b7NHGh5w@mail.gmail.com>
Message-ID: <loom.20130723T220119-727@post.gmane.org>

Shyam Sundar <sw.craftsman <at> gmail.com> writes:

> Input #0, mpeg, from 'cc.mpg2':
> Duration: 00:01:00.00, start: 0.224556, bitrate: 11993 kb/s
> Stream #0:0[0x1e0]: Video: mpeg2video (4:2:2), yuv422p, 
> 720x512 [SAR 128:135
> DAR 4:3], 11386 kb/s, 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc
> Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s

Please provide the sample.

Carl Eugen


From cehoyos at ag.or.at  Tue Jul 23 22:03:28 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 23 Jul 2013 20:03:28 +0000 (UTC)
Subject: [FFmpeg-user] Problem with concat at the stitch point!
References: <CAFGJ85-mNrO-iWeiH1z0R=AvbyJY2c2Eqg1zGvoC5jws+6WiLg@mail.gmail.com>
Message-ID: <loom.20130723T220220-136@post.gmane.org>

Colin Barnette <colinbgames <at> gmail.com> writes:

> ffmpeg -f concat -i list.txt -c copy 7-Final.avi
> 
> And a portion of 'list.txt':
> 
> file '7-1.avi'
> file '7-2.avi'

Are two (specific) files sufficient to reproduce 
the problem?
If yes, please provide command line including 
complete, uncut console output and upload the 
input files.

Carl Eugen


From cehoyos at ag.or.at  Tue Jul 23 22:05:17 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 23 Jul 2013 20:05:17 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg optimization question:
References: <CANdKVX6AaFYLc+7Cse9i2db=_O-KoFsm1HbTRp=3gsi_-nQJKA@mail.gmail.com>
Message-ID: <loom.20130723T220349-504@post.gmane.org>

John Crossman <johncrossman <at> berkeley.edu> writes:

> 1. ffmpeg -i video1.mp4 -i  *video2.mp4*  -filter_complex
> "[1:v]pad=iw*2:ih[bg];[bg][0:v]overlay=W/2+((W/2-w)/2):(H-h)/2" 
> -qscale:v 2 -strict -2  video3.mp4
> 2. ffmpeg -i video3.mp4  -acodec libmp3lame -ab 64k -ar 22050 
> -r 25 -vcodec libx264 -level 41 -crf 20 -bufsize 20000k 
> -maxrate 25000k -g 250 -s 1280x720 -coder 1  -subq 7 
> -me_range 16 -keyint_min 25 -sc_threshold 40 final.mp4
> 
> My question: How can I optimize the command #2 above?

Why are you using two commands?
Even if decoding is fast, it is certainly a huge overhead.

Carl Eugen


From edward.chopuryan at gmail.com  Tue Jul 23 11:08:18 2013
From: edward.chopuryan at gmail.com (edward20)
Date: Tue, 23 Jul 2013 02:08:18 -0700 (PDT)
Subject: [FFmpeg-user]
 =?utf-8?q?libavutil/common=2Eh=3A173=3A47=3A_error?=
 =?utf-8?q?=3A_=E2=80=98UINT64=5FC=E2=80=99_was_not_declared_in_this_scope?=
In-Reply-To: <CAPvYX16Z6ji6r93qxPo1SRScdNBVfQgWjAhT8HWv+NbgV-LkXw@mail.gmail.com>
References: <CAPvYX16Z6ji6r93qxPo1SRScdNBVfQgWjAhT8HWv+NbgV-LkXw@mail.gmail.com>
Message-ID: <1374570498433-4660135.post@n4.nabble.com>

Hi all, 

I am getting the same problem as yours. I have configured ffmpeg ./configure
--enable-shared --enable-version3 --enable-version3 

and I am getting the following error: 
/workspace1/avconv/avconv_master/libav/libavutil/common.h: In function
?int32_t av_clipl_int32_c(int64_t)?:
/workspace1/avconv/avconv_master/libav/libavutil/common.h:150:47: error:
?UINT64_C? was not declared in this scope

I'm using ubuntu 12.04. Please help me.

Thanks in advance.



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Sent from the FFmpeg-users mailing list archive at Nabble.com.

From edward.chopuryan at gmail.com  Tue Jul 23 13:26:10 2013
From: edward.chopuryan at gmail.com (edward20)
Date: Tue, 23 Jul 2013 04:26:10 -0700 (PDT)
Subject: [FFmpeg-user]
 =?utf-8?q?libavutil/common=2Eh=3A173=3A47=3A_error?=
 =?utf-8?q?=3A_=E2=80=98UINT64=5FC=E2=80=99_was_not_declared_in_this_scope?=
In-Reply-To: <1374570498433-4660135.post@n4.nabble.com>
References: <CAPvYX16Z6ji6r93qxPo1SRScdNBVfQgWjAhT8HWv+NbgV-LkXw@mail.gmail.com>
	<1374570498433-4660135.post@n4.nabble.com>
Message-ID: <1374578770523-4660137.post@n4.nabble.com>

I have figured out mine problem. I have main.cpp file and I was trying to
compile with gcc. After I had changed main.cpp to main.c compilation went
OK. 

If your source file has .cpp extension change it to .c. I hope it will solve
your problem too.

Thanks.



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From jiachielee at live.com  Tue Jul 23 14:00:43 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 23 Jul 2013 05:00:43 -0700 (PDT)
Subject: [FFmpeg-user] What is the (technical) difference between -r and
 the fps filter?
In-Reply-To: <CAN0LTaNAhqN4gLshVuLFryyci0ezdknPR_Pm8mYmngW1iSQkXw@mail.gmail.com>
References: <CAN0LTaNAhqN4gLshVuLFryyci0ezdknPR_Pm8mYmngW1iSQkXw@mail.gmail.com>
Message-ID: <1374580843559-4660138.post@n4.nabble.com>

Thanks for telling me the existence of fps video filter. As well as I know,
based on my own experiments, fps video filter can handle frame rate
conversion accurately; -r, on the other hand, is not exactly meant for frame
rate conversion. In other words, depending on usage scenarios, video filter
may not be replacement for -r.
ffmpeg allows us to encode a series of image sequence into a single video
file. For example,
ffmpeg -i "frames\f_%%1d.png" -vcodec "libx264" -crf "0" "output.mkv"
By definition, a video is a series of still images which, when shown on
screen, creates the illusion of moving images. Frame rate is frequency
(rate) at which an imaging device produces unique consecutive images called
frames. Such frame rate may be constant or variable. If a series of image
sequence is to be encoded into a single video with lossless encoding, we can
encode it back to image sequence for editing purposes. We can edit the video
frame by frame; or we can add or remove some frames. Lossy encoding, in
contrast, is irreversible process such that further encoding will further
compromise image quality. Videos meant for commercial release or
distribution are not meant for editing, which are different from ?master
copy?. As you may notice, I did not specify frame rate, so by default ffmpeg
chooses -r "25" for me. Film industry generally chooses 24 FPS in film
capturing and production. In that case,
ffmpeg -r "24" -i "frames\f_%%1d.png" -vcodec "libx264" -crf "0"
"output.mkv"
25 FPS is chosen by default because of the preset of ffmpeg as well as there
is no inherent frame rate for still images. Note that -r is placed before -i
instead of after -i, it is to tell that the still images or source is of the
frame rate of 24 FPS. -crf "0" is to tell x264 to encode the video in
?lossless mode?. Similarly, I can get the still images back:
mkdir "frames"
ffmpeg -i "output.mkv" "frames\f_%%1d.png"
In this case, I omitted -r both before and after -i. ffmpeg can detect the
frame rate from the source ?output.mkv?; omission of -i in the output option
means that the frame rate of the source is to be used. The conversion
between still images and a video (lossless encoding) is reversible. What if
-r is placed after -i?
ffmpeg -i "frames\f_%%1d.png" -r "24" -vcodec "libx264" -crf "0"
"output.mkv"
Of course, omission of -r before -i would mean that ffmpeg would assume that
the source (still images) is of 25 FPS, but the output video is of 24 FPS.
The achievement of 24 FPS, which is lower than the source, is only possible
if some frames are dropped. This is indeed frame rate conversion, from 25
FPS to 24 FPS. However, the algorithm of -r is different from video filter
"fps", which means ffmpeg will drop/duplicate different frames depending
whether -r or video filter "fps" or both are in use. In constant frame rate,
only certain length is allowed for a given frame rate; which means whenever
the frame rate conversion occurs, inevitably there will be changes in video
length as well; the change of video length is due to frame quantization or
rounding in calculation. Video filter "fps", unlike -r, cannot be used to
specify the frame rate for a series of still images sequence; it is meant
for frame rate conversion from a video with a given frame rate to another.
-r has different purposes from video filter "fps". When it comes to frame
rate conversion, how does -r handle compared with video filter "fps"? 
ffmpeg -i "input.mkv" -r "29.97" -vcodec "libx264" -crf "0" "output.mkv"
When I check the media info of output video, it reports the frame rate of
29.97 FPS and it is variable frame rate while the source I use is of the
frame rate of 24 FPS. What does it actually mean?
mkdir "frames"
ffmpeg -r "1" -i "output.mkv" "frames\f_%%1d.png"
Guess what I saw? The number of frames do not change after the encoding.
Variable frame rate means that the frame rate could by any and can be
changed while the video is being played. In this case, for the playback
speed to remain unchanged, no change in the number of frames would mean that
the actual frame rate does not change at all, assuming that the media player
can actually handle such media file correctly. If not, the video file may be
played at a different ?normal? speed than original intended. What really
changes is merely the media info being reported.
Why do I put -r "1" before -i? I want to see how many frames that the video
file actually has. If -r "1" is omitted, -r "29.97" would be assumed since
that is what being reported by media player. However, instead of extracting
every frame from the video, ffmpeg will add/drop frames to achieve 29.97 FPS
since the actual frame rate is reported as ?variable?; I will not be able to
see how many frames the output video has. By telling ffmpeg that the output
video has a constant frame rate, no matter what it is, thus ignoring the
original frame rate as contained in the source, ffmpeg would extracting all
frames from the output video.
Due to clever design of H.264 codec, its video can have changing frame rate,
which allows higher compression level. ffmpeg actually adds frames if I
choose to create a series of still images first.
mkdir "output"
ffmpeg -i "input.mkv" -r "29.97" "output\f_%%1d.png"
ffmpeg -r "29.97" -i "output\f_%%1d.png" -vcodec "libx264" -crf "0"
"output.mkv"
rmdir /s /q "output"
The argument is simple. ?Variable frame rate? simply does not exist for a
series of still images sequence. Thus, the only way to achieve 29.97 FPS
from the source of 24 FPS is to add frames. Video filter "fps" will force
the output video to have constant frame rate.
ffmpeg -i "input.mkv" -vf "fps=29.97" -vcodec "libx264" -crf "0"
"output.mkv"
When it comes to conversion to higher frame rate, video filter "fps" and -r
"29.97" actually creates video of the same length; but a closer inspection
makes me realize that ffmpeg duplicates different frames between video
filter "fps" and -r "29.97". When it comes to conversion to lower frame
rate, -r will increase the video length of 1/24 ? 1/12 second, depending on
which codec is being used; while video filter "fps" largely preserves the
video length, up to expected quantization, of course.
The bottom line is that -r and video filter "fps" are technically different
from each other. Video filter "fps" is meant for frame rate conversion. -r
has different uses as well.




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From lou at lrcd.com  Tue Jul 23 23:31:04 2013
From: lou at lrcd.com (Lou)
Date: Tue, 23 Jul 2013 13:31:04 -0800
Subject: [FFmpeg-user] ffmpeg optimization question:
In-Reply-To: <CANdKVX6AaFYLc+7Cse9i2db=_O-KoFsm1HbTRp=3gsi_-nQJKA@mail.gmail.com>
References: <CANdKVX6AaFYLc+7Cse9i2db=_O-KoFsm1HbTRp=3gsi_-nQJKA@mail.gmail.com>
Message-ID: <20130723133104.061b7053@lrcd.com>

On Mon, 22 Jul 2013 16:53:26 -0700
John Crossman <johncrossman at berkeley.edu> wrote:

>    1. ffmpeg -i *video1.mp4* -i  *video2.mp4*  -filter_complex
>    "[1:v]pad=iw*2:ih[bg];[bg][0:v]overlay=W/2+((W/2-w)/2):(H-h)/2" -qscale:v 2
>    -strict -2  *video3.mp4*

-qscale:v is ignored by libx264.

>    2. ffmpeg -i *video3.mp4*  -acodec libmp3lame -ab 64k -ar 22050 -r 25
>    -vcodec libx264 -level 41 -crf 20 -bufsize 20000k -maxrate 25000k -g 250 -s
>    1280x720 -coder 1  -subq 7 -me_range 16 -keyint_min 25 -sc_threshold 40 *
>    final.mp4*
> 
> My question: How can I optimize the command #2 above? I have sample stdout
> below.

As Carl already mentioned, and I may be missing something, but I don't
see the need for two commands when you can do it in one. If you're
uploading to YouTube:

ffmpeg -i input1 -i input2 -filter_complex pad;overlay -preset medium
-crf 18 -codec:a copy output.mkv

* No need for "-coder 1  -subq 7 -me_range 16 -keyint_min 25
  -sc_threshold 40"; that's what the presets are for.
  See https://trac.ffmpeg.org/wiki/x264EncodingGuide

* I see no reason for "-level 41 -bufsize 20000k -maxrate 25000k -g
  250 -s 1280x720"

From onemda at gmail.com  Tue Jul 23 23:37:31 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 23 Jul 2013 21:37:31 +0000
Subject: [FFmpeg-user] ffmpeg optimization question:
In-Reply-To: <loom.20130723T220349-504@post.gmane.org>
References: <CANdKVX6AaFYLc+7Cse9i2db=_O-KoFsm1HbTRp=3gsi_-nQJKA@mail.gmail.com>
	<loom.20130723T220349-504@post.gmane.org>
Message-ID: <CAPYw7P6LJoF79Fzu9+Sh6AJNG5JCALfGXP8tB1yRN0wH9-FF=w@mail.gmail.com>

On 7/23/13, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> John Crossman <johncrossman <at> berkeley.edu> writes:
>
>> 1. ffmpeg -i video1.mp4 -i  *video2.mp4*  -filter_complex
>> "[1:v]pad=iw*2:ih[bg];[bg][0:v]overlay=W/2+((W/2-w)/2):(H-h)/2"
>> -qscale:v 2 -strict -2  video3.mp4
>> 2. ffmpeg -i video3.mp4  -acodec libmp3lame -ab 64k -ar 22050
>> -r 25 -vcodec libx264 -level 41 -crf 20 -bufsize 20000k
>> -maxrate 25000k -g 250 -s 1280x720 -coder 1  -subq 7
>> -me_range 16 -keyint_min 25 -sc_threshold 40 final.mp4
>>
>> My question: How can I optimize the command #2 above?
>
> Why are you using two commands?
> Even if decoding is fast, it is certainly a huge overhead.

Decoding overhead is irrelevant here. He is encoding video twice
and that drops quality even more....

From johncrossman at berkeley.edu  Tue Jul 23 23:42:05 2013
From: johncrossman at berkeley.edu (John Crossman)
Date: Tue, 23 Jul 2013 14:42:05 -0700
Subject: [FFmpeg-user] ffmpeg optimization question:
In-Reply-To: <CAPYw7P6LJoF79Fzu9+Sh6AJNG5JCALfGXP8tB1yRN0wH9-FF=w@mail.gmail.com>
References: <CANdKVX6AaFYLc+7Cse9i2db=_O-KoFsm1HbTRp=3gsi_-nQJKA@mail.gmail.com>
	<loom.20130723T220349-504@post.gmane.org>
	<CAPYw7P6LJoF79Fzu9+Sh6AJNG5JCALfGXP8tB1yRN0wH9-FF=w@mail.gmail.com>
Message-ID: <CANdKVX5UPaSsoisZHOi3Uf3MMf84yN0Z3r72NhjkLxww7LLb4A@mail.gmail.com>

Thank you Carl, Lou and Paul. I'm testing now. Stay tuned.

From cehoyos at ag.or.at  Wed Jul 24 00:02:50 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 23 Jul 2013 22:02:50 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?libavutil/common=2Eh=3A173=3A47=3A_error?=
	=?utf-8?b?OiDigJhVSU5UNjRfQ+KAmSB3YXMgbm90IGRlY2xhcmVkIGluIHRoaXMg?=
	=?utf-8?q?scope?=
References: <CAPvYX16Z6ji6r93qxPo1SRScdNBVfQgWjAhT8HWv+NbgV-LkXw@mail.gmail.com>
	<1374570498433-4660135.post@n4.nabble.com>
Message-ID: <loom.20130723T235652-550@post.gmane.org>

edward20 <edward.chopuryan <at> gmail.com> writes:

> /workspace1/avconv/avconv_master/libav/libavutil/common.h

Please remember that since avconv contains a few hundred 
known bugs not reproducible with FFmpeg, some of them 
security relevant, we cannot support it here.
You can get supported versions at 
http://ffmpeg.org/download.html
Also consider reading 
http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html 
to learn more.

Regarding your problem: FFmpeg is a C project and 
provides C libraries and headers, if you want to 
use the library from C++, it is your responsibility 
to provide all necessary precautions, including to 
define __STDC_CONSTANT_MACROS

Carl Eugen


From deger at cobercafe.net  Wed Jul 24 03:41:27 2013
From: deger at cobercafe.net (Domnick Eger)
Date: Wed, 24 Jul 2013 01:41:27 +0000
Subject: [FFmpeg-user] RTMP to MJPEG
Message-ID: <CE147F60.37549%deger@cobercafe.net>

We been successful in previous releases of FFMPEG to stream our RTMP streams to MJPEG but as of the last repo update we are unable to make it work again. Here the syntax we been using but the version were currently using 0.6.5 seems to have issue with it. Any ideas ?

/usr/bin/ffmpeg -i rtmp://1.1.1.1/fsmjpeg/302 -s 320x240 -y -f image2 -vframes 1 /var/www/stream.jpg

From ajsinclair at gmail.com  Wed Jul 24 08:57:53 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Wed, 24 Jul 2013 16:57:53 +1000
Subject: [FFmpeg-user] librtmp with authentication
Message-ID: <CAOeAR15CMXqCZz2R3eOr-644CRC1GL=Hi1+VOsrCq7NR2iTmEQ@mail.gmail.com>

Hi,

Has anyone had recent success using the librtmp (--enable-librtmp) to
authenticate with an Adobe Media Server when pushing a stream?

I have it compiling in OK and running but I am getting an authentication
error.

./ffmpeg -f lavfi -i smptebars -c:v libx264 -b:v 500k -an -f flv 'rtmp://
p.ep130468.i.akamaientrypoint.net/EntryPointflashver=FMLE/3.0\20(compatible;\20FMSc/1.0)
live=true
pubUser='userremoved' pubPasswd='passwordremoved' playpath=event_1 at 130468'
-loglevel debug
ffmpeg version N-54152-g730e07f Copyright (c) 2000-2013 the FFmpeg
developers
  built on Jul 24 2013 16:08:47 with Apple LLVM version 4.2
(clang-425.0.28) (based on LLVM 3.2svn)
  configuration: --enable-gpl --enable-version3 --enable-nonfree
--enable-postproc --enable-libaacplus --enable-libcelt --enable-libfaac
--enable-libfdk-aac --enable-libfreetype --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-openssl --enable-libopus --enable-libschroedinger
--enable-libspeex --enable-libtheora --enable-libvo-aacenc
--enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid
--enable-libvidstab --prefix=/usr/local --enable-librtmp --cc=clang
  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


Here is a relevant example from my logs without giving passwords out.

RTMP_ClientPacket, received: invoke 158 bytes
(object begin)
Property: <Name:           no-name., STRING: _error>
Property: <Name:           no-name., NUMBER: 1.00>
Property: NULL
Property: <Name:           no-name., OBJECT>
(object begin)
Property: <Name:              level, STRING: error>
Property: <Name:               code, STRING: NetConnection.Connect.Rejected>
Property: <Name:        description, STRING: [ AccessManager.Reject ] : [
authmod=adobe ] : ?reason=authfailed>
(object end)
(object end)
HandleInvoke, server invoking <_error>
HandleInvoke, received error for method call <connect>
HandleInvoke, error description: [ AccessManager.Reject ] : [ authmod=adobe
] : ?reason=authfailed
PublisherAuth, Authentication failed: wrong password
RTMP_ReadPacket: fd=3
  0000:  03 00 00 00 00 00 12 14  00 00 00 00               ............

  0000:  02 00 05 63 6c 6f 73 65  00 00 00 00 00 00 00 00
...close........
  0010:  00 05                                              ..

RTMP_ClientPacket, received: invoke 18 bytes
(object begin)
Property: <Name:           no-name., STRING: close>
Property: <Name:           no-name., NUMBER: 0.00>
Property: NULL
(object end)
HandleInvoke, server invoking <close>

Now I am quite sure all the properties are correct as I can connect with
Flash Media Live Encoder OK. I do have packet captures of both if anyone
thinks they might be able to assist or point me in the direction to patch.

Thanks,
Andrew

From francois.visagie at gmail.com  Wed Jul 24 10:43:46 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Wed, 24 Jul 2013 10:43:46 +0200
Subject: [FFmpeg-user] ffmpeg how to set stream layout
In-Reply-To: <CAFd2cZW_3pUhXJGAe2YxJa=4nDWTUGkrwWXmMVAfCcddjzA9vw@mail.gmail.com>
References: <CAFd2cZUe8WJ13D+XtUY6Cgr_au1-v33Dmzm23=fLoX2LrmHZGA@mail.gmail.com>	<000001ce878b$081ceb80$1856c280$@gmail.com>
	<CAFd2cZW_3pUhXJGAe2YxJa=4nDWTUGkrwWXmMVAfCcddjzA9vw@mail.gmail.com>
Message-ID: <005401ce8849$eec28a40$cc479ec0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Soho Soho123
> Sent: 23 July 2013 14:25
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] ffmpeg how to set stream layout
> 
> > I use ffmpeg command to generate audio and video stream.
> > > ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480
> > > -input_format
> > h264
> > > -i /dev/video1 -f alsa -ar 44100 -ac 2 -i hw:0 -vcodec copy -acodec
> > > copy http://localhost:8090/feed1.ffm
> > >
> > > I got the stream layout :
> > >  Stream #0:0, 0, 1/1000000: Audio: pcm_s16be, 44100 Hz, stereo, 1411
> 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
> > >
> > > Can I modify the layout to
> > > Stream 0 is video , and stream 1 is audio?
> >
> > Yes, use the -map option to enumerate streams in the order you want
> > them to appear in the output.
> >
> > Cheers,
> > Francois
> >
> > Hi,
> Excuse me!
> Do you know where is the example about "-map" option usage?

http://www.ffmpeg.org/ffmpeg-all.html#Advanced-options

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


From cehoyos at ag.or.at  Wed Jul 24 11:11:55 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 24 Jul 2013 09:11:55 +0000 (UTC)
Subject: [FFmpeg-user] RTMP to MJPEG
References: <CE147F60.37549%deger@cobercafe.net>
Message-ID: <loom.20130724T111029-174@post.gmane.org>

Domnick Eger <deger <at> cobercafe.net> writes:

> We been successful in previous releases of FFMPEG to 
> stream our RTMP streams to MJPEG but as of the last 
> repo update we are unable to make it work again.

Please also add information about which version was 
still working.

> ffmpeg -i rtmp://1.1.1.1/fsmjpeg/302 -s 320x240 
> -y -f image2 -vframes 1 /var/www/stream.jpg

Complete, uncut console output missing.

Carl Eugen


From nicolas.george at normalesup.org  Wed Jul 24 11:14:16 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 24 Jul 2013 11:14:16 +0200
Subject: [FFmpeg-user] Problem with concat at the stitch point!
In-Reply-To: <201307231518.r6NFIAu3083755@haluter.fromme.com>
References: <CAFGJ85-mNrO-iWeiH1z0R=AvbyJY2c2Eqg1zGvoC5jws+6WiLg@mail.gmail.com>
	<201307231518.r6NFIAu3083755@haluter.fromme.com>
Message-ID: <20130724091416.GD11090@phare.normalesup.org>

Le quintidi 5 thermidor, an CCXXI, Oliver Fromme a ?crit?:
> ffmpeg -i "concat:file1|file2|file3|file4" ...
> 
> I don't know if that makes any difference, but it might be
> worth a try.

It makes a huge difference: this commands glues the sequences of bytes
together, while Colin Barnette's command glues the sequences of packets
together. In particular:

> ffmpeg -i "concat:s7-1.avi|s7-2.avi|s7-3.avi|s7-4.avi" -b:v 11000k 7-Final.avi

... AVI files can not be usefully concatenated at the byte stream level,
this command will just ignore s7-2.avi and the next ones.

Only a handful of formats, usually formats without a global header, can be
concatenated at the byte stream level.

Regards,

-- 
  Nicolas George
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From nicolas.george at normalesup.org  Wed Jul 24 11:17:43 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 24 Jul 2013 11:17:43 +0200
Subject: [FFmpeg-user] Problem with concat at the stitch point!
In-Reply-To: <CAFGJ85-mNrO-iWeiH1z0R=AvbyJY2c2Eqg1zGvoC5jws+6WiLg@mail.gmail.com>
References: <CAFGJ85-mNrO-iWeiH1z0R=AvbyJY2c2Eqg1zGvoC5jws+6WiLg@mail.gmail.com>
Message-ID: <20130724091743.GE11090@phare.normalesup.org>

Le quartidi 4 thermidor, an CCXXI, Colin Barnette a ?crit?:
> Here's an example of my batch script to first encode:
> 
> ffmpeg -i s7-1.avi -b:v 11000k 7-1.avi
> ffmpeg -i s7-2.avi -b:v 11000k 7-2.avi
> ffmpeg -i s7-3.avi -b:v 11000k 7-3.avi
> ffmpeg -i s7-4.avi -b:v 11000k 7-4.avi
> 
> And the command I use to concat:
> 
> ffmpeg -f concat -i list.txt -c copy 7-Final.avi
> 
> And a portion of 'list.txt':
> 
> file '7-1.avi'
> file '7-2.avi'
> file '7-3.avi'
> file '7-4.avi'

It should work fine, but I would suggest to try with a less brain-dead
format than AVI for the intermediate files before anything, for example NUT.

Note that since you are transcoding the packets anyway, you may be better
off using the concat filter, that works on decoded frames, than the concat
demuxer, that works on packets.

Regards,

-- 
  Nicolas George
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From 670292548 at qq.com  Wed Jul 24 12:20:32 2013
From: 670292548 at qq.com (=?ISO-8859-1?B?dW5kZWZpbmVk?=)
Date: Wed, 24 Jul 2013 18:20:32 +0800
Subject: [FFmpeg-user] help
Message-ID: <tencent_6CE926983B0244B351CB2105@qq.com>

Hi,
We would like to build a web site of live streaming. We push high-quality stream in the format of h264 & aac to AMS5, and intend to output 4 streams with different size but the same format to AMS5 using FFMpeg. But get an error: muxer does not support non seekable output. And then "Could not write header for output file #0 (incorrect codec parameters ?): Error number -1 occurred".


I use windows 7. cmd:
ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx264 -s 640*360 -f mp4 "rtmp://localhost/testFFMpeg/W3FHRHHT_360"


This cmd goes well:
ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx264 -f mp4 null.mp4
Here's the metadata:
{
data.duration : 50.60907029478458
data.avcprofile : 100
data.moovposition : 1310780
data.tags : 
data.audiosamplerate : 44100
data.width : 640
data.seekpoints : [object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
data.audiocodecid : mp4a
data.trackinfo : [object Object],[object Object]
data.videoframerate : 25
data.audiochannels : 2
data.videocodecid : avc1
data.aacaot : 2
data.height : 360
data.avclevel : 30

}


And this file can be output in the format of flv:
ffmpeg -re -y -i null.mp4 -f flv "rtmp://localhost/testFFMpeg/W3FHRHHT_360"


How can I output it in the format of mp4 or f4v?


Please help!




Best Regards,
Lane

From hoangit1237 at gmail.com  Wed Jul 24 04:07:20 2013
From: hoangit1237 at gmail.com (hoang nguyen huy)
Date: Wed, 24 Jul 2013 09:07:20 +0700
Subject: [FFmpeg-user] [Question] How to live streaming multicast
Message-ID: <CAMw5opTFzt9TFS+-yHgqnPd6PW9zZrF=AFO0k+UuLrob6bvjaA@mail.gmail.com>

Hi team !
You are working great job and thanks for this !
I have a problem, and I think that you may support me :
I have built an Android application based on FFMPEG, refer
http://www.roman10.net/how-to-build-android-applications-based-on-ffmpeg-by-an-example/.
Now, I want to live streaming multicast.
I can do this in client ? And I need convert multicast url to unicast url?
So, How to do this ?
I hope, I'll receive your reply soon.
Thanks you very much !

--
Best & Regard

Nguy?n Huy Ho?ng
IS-SOICT-HUT
Mobile: 0975 156 422
YM: hoangconan_0101 at yahoo.com
GMail: hoangit1237 at gmail.com

From jaswant.panwar at dotsquares.com  Wed Jul 24 15:38:25 2013
From: jaswant.panwar at dotsquares.com (Jaswant Panwar)
Date: Wed, 24 Jul 2013 19:08:25 +0530
Subject: [FFmpeg-user] Problem while compiling
Message-ID: <000e01ce8873$13b13440$3b139cc0$@panwar@dotsquares.com>

Hi,

 

I'm facing following error while compiling FFMPEG on Windows using Android
NDKr83 and Cygwin:

 

  ./configure_make_everything.sh

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/x264
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

Makefile:3: config.mak: No such file or directory

cat: config.h: No such file or directory

./configure

./configure: line 2: $'\r': command not found

./configure: line 59: syntax error near unexpected token `$'{\r''

'/configure: line 59: `log_check() {

Makefile:199: recipe for target `config.mak' failed

make: *** [config.mak] Error 2

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/freet
ype2 /cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

config.mk:25: builds/unix/unix-def.mk: No such file or directory

config.mk:26: builds/unix/unix-cc.mk: No such file or directory

make: *** No rule to make target `builds/unix/unix-cc.mk'.  Stop.

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/ffmpe
g /cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

Makefile:2: config.mak: No such file or directory

Makefile:47: /common.mak: No such file or directory

Makefile:88: /libavutil/Makefile: No such file or directory

Makefile:88: /library.mak: No such file or directory

Makefile:168: /doc/Makefile: No such file or directory

Makefile:169: /tests/Makefile: No such file or directory

make: *** No rule to make target `/tests/Makefile'.  Stop.

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/sox
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

make: *** No rule to make target `clean'.  Stop.

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/x264
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

./configure: line 2: $'\r': command not found

./configure: line 59: syntax error near unexpected token `$'{\r''

'/configure: line 59: `log_check() {

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/x264
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

Makefile:3: config.mak: No such file or directory

cat: config.h: No such file or directory

./configure

./configure: line 2: $'\r': command not found

./configure: line 59: syntax error near unexpected token `$'{\r''

'/configure: line 59: `log_check() {

Makefile:199: recipe for target `config.mak' failed

make: *** [config.mak] Error 2

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/freet
ype2 /cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

./autogen.sh: line 2: $'\r': command not found

./autogen.sh: line 11: $'\r': command not found

./autogen.sh: line 12: syntax error near unexpected token `$'\r''

'/autogen.sh: line 12: `run ()

./configure: line 15: $'\r': command not found

./configure: line 17: $'\r': command not found

./configure: line 73: syntax error near unexpected token `$'do\r''

'/configure: line 73: `  for x in "$@"; do

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/freet
ype2 /cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

make: ver: Command not found

 

FreeType build system -- automatic system detection

 

The following settings are used:

 

  platform                    unix

  compiler                    cc

  configuration directory     ./builds/unix

  configuration rules         ./builds/unix/unix.mk

 

If this does not correspond to your system or settings please remove the
file

`config.mk' from this directory then read the INSTALL file for help.

 

Otherwise, simply type `make' again to build the library,

or `make refdoc' to build the API reference (the latter needs python).

 

cd builds/unix; ./configure

/bin/sh: ./configure: No such file or directory

builds/unix/detect.mk:85: recipe for target `setup' failed

make: *** [setup] Error 127

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

./configure_ffmpeg.sh: line 14: patch: command not found

./configure_ffmpeg.sh: line 15: patch: command not found

./configure_ffmpeg.sh: line 16: patch: command not found

./configure_ffmpeg.sh: line 18: patch: command not found

./configure_ffmpeg.sh: line 20: patch: command not found

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/ffmpe
g /cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

d/android-ndk-r8e/toolchains/arm-linux-androideabi-4.6/prebuilt/cygwin_nt-6.
1-wow64-x86/bin/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.

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/ffmpe
g /cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

Makefile:2: config.mak: No such file or directory

Makefile:47: /common.mak: No such file or directory

Makefile:88: /libavutil/Makefile: No such file or directory

Makefile:88: /library.mak: No such file or directory

Makefile:168: /doc/Makefile: No such file or directory

Makefile:169: /tests/Makefile: No such file or directory

make: *** No rule to make target `/tests/Makefile'.  Stop.

Makefile:2: config.mak: No such file or directory

Makefile:47: /common.mak: No such file or directory

Makefile:88: /libavutil/Makefile: No such file or directory

Makefile:88: /library.mak: No such file or directory

Makefile:168: /doc/Makefile: No such file or directory

Makefile:169: /tests/Makefile: No such file or directory

make: *** No rule to make target `/tests/Makefile'.  Stop.

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/sox
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

./configure_sox.sh: line 11: patch: command not found

autoreconf-2.69: Entering directory `.'

autoreconf-2.69: configure.ac: not using Gettext

autoreconf-2.69: running: aclocal --force -I m4

autoreconf-2.69: configure.ac: tracing

autoreconf-2.69: configure.ac: not using Libtool

autoreconf-2.69: running: /usr/bin/autoconf-2.69 --force

autoreconf-2.69: running: /usr/bin/autoheader-2.69 --force

autoreconf-2.69: running: automake --add-missing --copy --force-missing

libgsm/Makefile.am:35: error: Libtool library used but 'LIBTOOL' is
undefined

libgsm/Makefile.am:35:   The usual way to define 'LIBTOOL' is to add
'LT_INIT'

libgsm/Makefile.am:35:   to 'configure.ac' and run 'aclocal' and 'autoconf'
again.

libgsm/Makefile.am:35:   If 'LT_INIT' is in 'configure.ac', make sure

libgsm/Makefile.am:35:   its definition is in aclocal's search path.

lpc10/Makefile.am:8: error: Libtool library used but 'LIBTOOL' is undefined

lpc10/Makefile.am:8:   The usual way to define 'LIBTOOL' is to add 'LT_INIT'

lpc10/Makefile.am:8:   to 'configure.ac' and run 'aclocal' and 'autoconf'
again.

lpc10/Makefile.am:8:   If 'LT_INIT' is in 'configure.ac', make sure

lpc10/Makefile.am:8:   its definition is in aclocal's search path.

autoreconf-2.69: automake failed with exit status: 1

configure: WARNING: unrecognized options: --with-sysroot, --enable-static,
--disable-shared

checking build system type... i686-pc-cygwin

checking host system type... arm-unknown-linux-androideabi

checking target system type... arm-unknown-linux-androideabi

checking for a BSD-compatible install... /usr/bin/install -c

checking whether build environment is sane... yes

checking for arm-linux-androideabi-strip...
d/android-ndk-r8e/toolchains/arm-linux-androideabi-4.6/prebuilt/cygwin_nt-6.
1-wow64-x86/bin/arm-linux-androideabi-strip

checking for a thread-safe mkdir -p... /usr/bin/mkdir -p

checking for gawk... gawk

checking whether make sets $(MAKE)... yes

checking for arm-linux-androideabi-gcc...
d/android-ndk-r8e/toolchains/arm-linux-androideabi-4.6/prebuilt/cygwin_nt-6.
1-wow64-x86/bin/arm-linux-androideabi-gcc
--sysroot=d/android-ndk-r8e/platforms/android-3/arch-arm

checking whether the C compiler works... no

configure: error: in
`/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/sox'
:

configure: error: C compiler cannot create executables

See `config.log' for more details

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg/sox
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

make: *** No targets specified and no makefile found.  Stop.

make: *** No rule to make target `install-strip'.  Stop.

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg
/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

/cygdrive/d/ffmpeg/23-july/android-ffmpeg-java/external/android-ffmpeg

 


From yosik at macam.ac.il  Wed Jul 24 08:59:08 2013
From: yosik at macam.ac.il (Yossi Klein)
Date: Wed, 24 Jul 2013 06:59:08 +0000
Subject: [FFmpeg-user] DirectShow -list_devices | programatically getting a
 list (friendly names) of connected video and audio devices
Message-ID: <6E859B8BAFFDDB4585AB5C57B6D774C383D10CBE@EXCH02.macamnet.local>

Hi,
Using: ffmpeg -list_devices true -f dshow -i dummy, I can get a list of connected devices.
I need to get access to the friendly names of the devices in that list programmatically (to paste it/insert it in another program).
Thanks in advance for any help.
Yos

From sw.craftsman at gmail.com  Wed Jul 24 20:03:15 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Wed, 24 Jul 2013 23:33:15 +0530
Subject: [FFmpeg-user] Video bit rate missing from FFProbe output
In-Reply-To: <loom.20130723T220119-727@post.gmane.org>
References: <CA+XaXC=J7BMNZTF3orYpr_pNrDbmYdVmUycwJeWMa9b7NHGh5w@mail.gmail.com>
	<loom.20130723T220119-727@post.gmane.org>
Message-ID: <CA+XaXCnhFAYePWHi12_zA_GDqXcqLsa8y9CdMokb1qrrLnh5CQ@mail.gmail.com>

I have uploaded it at http://files.videohelp.com/u/211860/cc.mpg2

Thanks


On Wed, Jul 24, 2013 at 1:31 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Shyam Sundar <sw.craftsman <at> gmail.com> writes:
>
> > Input #0, mpeg, from 'cc.mpg2':
> > Duration: 00:01:00.00, start: 0.224556, bitrate: 11993 kb/s
> > Stream #0:0[0x1e0]: Video: mpeg2video (4:2:2), yuv422p,
> > 720x512 [SAR 128:135
> > DAR 4:3], 11386 kb/s, 29.97 fps, 59.94 tbr, 90k tbn, 59.94 tbc
> > Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s
>
> Please provide the sample.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Thu Jul 25 00:07:05 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 24 Jul 2013 22:07:05 +0000 (UTC)
Subject: [FFmpeg-user] Problem while compiling
References: <33456.4657569446$1374685255@news.gmane.org>
Message-ID: <loom.20130725T000612-120@post.gmane.org>

Jaswant Panwar <jaswant.panwar <at> dotsquares.com> writes:

> I'm facing following error while compiling FFMPEG on 
> Windows using Android NDKr83 and Cygwin:
> 
>   ./configure_make_everything.sh

This script is not part of FFmpeg, please use 
the configure script delivered with current 
FFmpeg if you need support here.

Carl Eugen


From cehoyos at ag.or.at  Thu Jul 25 00:08:28 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 24 Jul 2013 22:08:28 +0000 (UTC)
Subject: [FFmpeg-user] Video bit rate missing from FFProbe output
References: <CA+XaXC=J7BMNZTF3orYpr_pNrDbmYdVmUycwJeWMa9b7NHGh5w@mail.gmail.com>
	<loom.20130723T220119-727@post.gmane.org>
	<CA+XaXCnhFAYePWHi12_zA_GDqXcqLsa8y9CdMokb1qrrLnh5CQ@mail.gmail.com>
Message-ID: <loom.20130725T000737-643@post.gmane.org>

Shyam Sundar <sw.craftsman <at> gmail.com> writes:

> I have uploaded it at http://files.videohelp.com/u/211860/cc.mpg2

This stream only contains a maximum bitrate, I sent 
a patch to show the maxrate if no bitrate is defined.

Please do not top-post here, Carl Eugen


From jpboard2 at yahoo.com  Thu Jul 25 01:57:08 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 24 Jul 2013 16:57:08 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
Message-ID: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>

I'm trying to use ffmpeg to edit AVI files.? The video is not mpeg-encoded yet
and each frame is stored in full.? I want to do frame-exact edits.? I really
want to specify the start and end frame numbers, but it appears ffmpeg doesn't
support that.? So, I have to specify a start and end time for each segment, and
then combine the segments with ffmpeg.? This is the command I'm using for each
segment:

ffmpeg -i in.avi -ss [start] -to end -vcodec copy -acodec copy out.avi

The 'start' and 'end' arguments are floating point numbers like 1.25, etc.? My
question is how ffmpeg interprets those numbers.? Since floating-point numbers
aren't exact, it isn't always possible to specify the exact position.? So what
does ffmpeg do when the 'start' argument doesn't fall exactly on a frame.
Does it start at the closest frame?? Does it start at the next frame after
that position?? Does it do something else?

Alternatively, if I have the exact start and end frame numbers, how can I then
convert that to exact start and end times?

Thanks for all your wonderful help.

From horsley1953 at gmail.com  Thu Jul 25 03:01:40 2013
From: horsley1953 at gmail.com (Tom Horsley)
Date: Wed, 24 Jul 2013 21:01:40 -0400
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <20130724210140.4cf72f11@zooty>

On Wed, 24 Jul 2013 16:57:08 -0700 (PDT)
James Board wrote:

> Alternatively, if I have the exact start and end frame numbers, how can I then
> convert that to exact start and end times?

My technique is to produce a low res version of the original file
that uses the drawtext filter to write the exact timestamp
on each frame, then I pick off the times from that video :-).

See:

http://home.comcast.net/~tomhorsley/game/ffmpeg.html#generatetimestamps

The one way that is guaranteed to give me timestamps that
ffmpeg believes (since it generated them).

From 670292548 at qq.com  Thu Jul 25 04:17:32 2013
From: 670292548 at qq.com (=?gb18030?B?dW5kZWZpbmVk?=)
Date: Thu, 25 Jul 2013 10:17:32 +0800
Subject: [FFmpeg-user] help
Message-ID: <tencent_2924C8593E943A9624DCFDD0@qq.com>

here's the uncut log:


C:\Users\CODE-02\ffmpeg>ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx
264 -s 640*360 -f mp4 "rtmp://localhost/testFFMpeg/W3FHRHHT_360"
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
rtmp server sent error
Metadata:
  duration              0.00
  fileSize              0.00
  width                 640.00
  height                360.00
  videocodecid          avc1
  videodatarate         500.00
  framerate             25.00
  audiocodecid          mp4a
  audiodatarate         64.00
  audiosamplerate       44100.00
  audiosamplesize       16.00
  audiochannels         2.00
  stereo                TRUE
  encoder               Open Broadcaster Software v0.522b
Input #0, flv, from 'rtmp://localhost/testFFMpeg/W3FHRHHT live=1':
  Metadata:
    fileSize        : 0
    audiochannels   : 2
    encoder         : Open Broadcaster Software v0.522b
  Duration: 00:00:00.00, start: 0.017000, bitrate: N/A
    Stream #0:0: Video: h264 (High), yuv420p, 640x360, 512 kb/s, 25 tbr, 1k tbn, 2k tbc
    Stream #0:1: Audio: aac, 44100 Hz, stereo, fltp, 65 kb/s
[libx264 @ 0000000004c00020] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0000000004c00020] profile High, level 3.0
[libx264 @ 0000000004c00020] 264 - core 135 r2345 f0c1c53 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.vid
eolan.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 lookah
ead_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_lookahe
ad=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 @ 0000000004af00a0] muxer does not support non seekable output
Output #0, mp4, to 'rtmp://localhost/testFFMpeg/W3FHRHHT_360':
  Metadata:
    fileSize        : 0
    audiochannels   : 2
    encoder         : Lavf55.12.102
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 640x360, q=-1--1, 90k tbn, 25 tbc
    Stream #0:1: Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (aac -> libvo_aacenc)
Could not write header for output file #0 (incorrect codec parameters ?): Error number -1 occurred




------------------ Original ------------------
From: "undefined"<670292548 at qq.com>; 
Date: 2013?7?24?(???) ??6:20
To: "ffmpeg-user"; 
Subject: [FFmpeg-user] help



Hi,
We would like to build a web site of live streaming. We push high-quality stream in the format of h264 & aac to AMS5, and intend to output 4 streams with different size but the same format to AMS5 using FFMpeg. But get an error: muxer does not support non seekable output. And then "Could not write header for output file #0 (incorrect codec parameters ?): Error number -1 occurred".


I use windows 7. cmd:
ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx264 -s 640*360 -f mp4 "rtmp://localhost/testFFMpeg/W3FHRHHT_360"


This cmd goes well:
ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx264 -f mp4 null.mp4
Here's the metadata:
{
data.duration : 50.60907029478458
data.avcprofile : 100
data.moovposition : 1310780
data.tags : 
data.audiosamplerate : 44100
data.width : 640
data.seekpoints : [object Object],[object Object],[object Object],[object Object],[object Object],[object Object]
data.audiocodecid : mp4a
data.trackinfo : [object Object],[object Object]
data.videoframerate : 25
data.audiochannels : 2
data.videocodecid : avc1
data.aacaot : 2
data.height : 360
data.avclevel : 30

}


And this file can be output in the format of flv:
ffmpeg -re -y -i null.mp4 -f flv "rtmp://localhost/testFFMpeg/W3FHRHHT_360"


How can I output it in the format of mp4 or f4v?


Please help!




Best Regards,
Lane
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
.

From dave at dericed.com  Thu Jul 25 04:39:18 2013
From: dave at dericed.com (Dave Rice)
Date: Wed, 24 Jul 2013 22:39:18 -0400
Subject: [FFmpeg-user] encoding bff input to mov labels the output as
	progressive
Message-ID: <8B50D84E-9161-45D5-B217-550EB5DC3516@dericed.com>

Hi all,

I am encoding bff v210 video to other codecs (uncomrpessed, ffv1, prores, j2k) in quicktime. Also the input is identified as bff the output is identified as progressive unless I force it with -vf setfield=bff. For the test below I'm using ffmpeg 2.0 built on July 19th, 2013.

Here is a command to create a sample bff input:

ffmpeg -f lavfi -i mandelbrot=s=720x486:r=ntsc -t 5 -c:v v210 -vf setfield=bff v210bff.mov

Checking with mediainfo:

mediainfo --inform="Video;%ScanType%,%ScanOrder%" v210bff.mov
Interlaced,BFF

When I encode like (same codec in and out):
ffmpeg -i v210bff.mov -c:v v210 v210bff-2-v210.mov

Then the output shows this in mediainfo:
mediainfo --inform="Video;%ScanType%,%ScanOrder%" v210bff-2-v210.mov 
Progressive,

I can force the output to bff with:
ffmpeg -i v210bff.mov -c:v v210 -vf setfield=bff v210bff-2-v210forced-bff.mov

but from reading past threads of the quicktime 'fiel' atom it seems that preserving the field order is the intent although not the default behavior that I'm getting. If I don't check to see if the input is bff (using ffmbc or mediainfo) and then include setfield=bff then the resulting progressive output causes playback issues on interlaced monitors.

I can put this into the tracker but wanted to check that my test process and expectations are sane.

Dave Rice

From soho123.2012 at gmail.com  Thu Jul 25 05:36:03 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Thu, 25 Jul 2013 11:36:03 +0800
Subject: [FFmpeg-user] ffmpeg how to set http timeout
Message-ID: <CAFd2cZXY6QjVXryfW8qJUF9ev_8BjhMv8tXUPhat=rzs4BZa7Q@mail.gmail.com>

Hi all,

If ffmpeg I use :
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format
mjpeg -i /dev/video0 -vcodec
 copy http://localhost:8080/feed1.ffm

how to set timeout for http (tcp )connect?

thanks!

From tevans.uk at googlemail.com  Thu Jul 25 10:59:47 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Thu, 25 Jul 2013 09:59:47 +0100
Subject: [FFmpeg-user] help
In-Reply-To: <tencent_2924C8593E943A9624DCFDD0@qq.com>
References: <tencent_2924C8593E943A9624DCFDD0@qq.com>
Message-ID: <CAFHbX1JtgxdnbSyHVb2ngcZwTj8qUPOU3AxAJFddXS-922ErWg@mail.gmail.com>

On Thu, Jul 25, 2013 at 3:17 AM, undefined <670292548 at qq.com> wrote:
> here's the uncut log:
>
>
> C:\Users\CODE-02\ffmpeg>ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx
> 264 -s 640*360 -f mp4 "rtmp://localhost/testFFMpeg/W3FHRHHT_360"
> ?
> [mp4 @ 0000000004af00a0] muxer does not support non seekable output

mp4 requires a seekable output, since it writes a MOOV atom at the
start of the file once it finishes encoding the file. Therefore, this
is not a suitable format for feeding data to rtmp.

Cheers

Tom

From Gergely.Lukacsy at streamnet.hu  Thu Jul 25 10:45:51 2013
From: Gergely.Lukacsy at streamnet.hu (gergely)
Date: Thu, 25 Jul 2013 01:45:51 -0700 (PDT)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
 playback of video streams
In-Reply-To: <loom.20130228T094518-529@post.gmane.org>
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
Message-ID: <1374741951675-4660169.post@n4.nabble.com>

*Hi Eugen,*
I reproduced the same issue here with plain ffmpeg, so I can provide you the
commandline output.

I'm trying to concatenate and convert some .MTS files produced by our
Panasonic AG-HMC150 Camera at the moment, and I got the same mmco errors as
the OP.
However, the converter started to convert the files, so it seems that's not
a critical error. I'm curious, because I have never saw this error.

*Output:*

> C:\VSQ\CONFORG\20130718\recordings>ffmpeg -i 00000.mts -i 00001.mts -i
> 00002.mts -i 00003.mts -i 00004.mts -i 00005.mts -filter_complex "[0:0]
> [0:1] [1:0] [1:1] [2:0] [2:1] [3:0] [3:1] [4:0] [4:1] [5:0] [5:1]
> concat=n=6:v=1:a=1 [v] [a]" -map [v] -map [a] -strict experimental -c:v
> libx264 -b:v 2200k -profile:v main -preset:v slow -c:a aac -b:a 128k -ac 2
> -f mp4 video_0_merged.mp4
> ffmpeg version N-54762-g38b701a Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Jul 15 2013 18:47:20 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. 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. 81.101 /  3. 81.101
>   libswscale      2.  3.100 /  2.  3.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  3.100 / 52.  3.100
> [h264 @ 00000000002fe260] mmco: unref short failure
> [mpegts @ 0000000000367fe0] PES packet size mismatch
> Input #0, mpegts, from '00000.mts':
>   Duration: 00:26:16.72, start: 3307.145500, bitrate: 21754 kb/s
>   Program 1
>     Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
>     Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo,
> fltp, 384 kb/s
> [h264 @ 00000000043302e0] mmco: unref short failure
> Input #1, mpegts, from '00001.mts':
>   Duration: 00:26:25.39, start: 4883.433500, bitrate: 21638 kb/s
>   Program 1
>     Stream #1:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
>     Stream #1:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo,
> fltp, 384 kb/s
> [h264 @ 0000000005533d20] mmco: unref short failure
> Input #2, mpegts, from '00002.mts':
>   Duration: 00:26:24.88, start: 6468.425500, bitrate: 21647 kb/s
>   Program 1
>     Stream #2:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
>     Stream #2:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo,
> fltp, 384 kb/s
> [h264 @ 000000000037ca00] mmco: unref short failure
> Input #3, mpegts, from '00003.mts':
>   Duration: 00:26:25.36, start: 8052.905500, bitrate: 21635 kb/s
>   Program 1
>     Stream #3:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
>     Stream #3:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo,
> fltp, 384 kb/s
> [h264 @ 00000000055368c0] mmco: unref short failure
> Input #4, mpegts, from '00004.mts':
>   Duration: 00:26:24.84, start: 9637.897500, bitrate: 21644 kb/s
>   Program 1
>     Stream #4:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
>     Stream #4:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo,
> fltp, 384 kb/s
> [h264 @ 0000000004e76020] mmco: unref short failure
> Input #5, mpegts, from '00005.mts':
>   Duration: 00:00:56.61, start: 11222.313500, bitrate: 20949 kb/s
>   Program 1
>     Stream #5:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
>     Stream #5:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo,
> fltp, 384 kb/s
> [libx264 @ 000000000036fd40] using SAR=1/1
> [libx264 @ 000000000036fd40] using cpu capabilities: MMX2 SSE2Fast SSSE3
> SSE4.2 AVX
> [libx264 @ 000000000036fd40] profile Main, level 3.2
> [libx264 @ 000000000036fd40] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4
> AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html -
> options: cabac=1 ref=5 deblock=1:0:0 analyse=0x1:0x111 me=umh subme=8
> 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=2 sliced_threads=0 nr=0 decimate=1 interlaced=0
> bluray_compat=0 constrained_intra=0 bframes=3 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=50 rc=abr mbtree=1 bitrate=2200
> ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> Output #0, mp4, to 'video_0_merged.mp4':
>   Metadata:
>     encoder         : Lavf55.12.102
>     Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 2200 kb/s, 12800 tbn, 50 tbc
>     Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo,
> fltp, 128 kb/s
> Stream mapping:
>   Stream #0:0 (h264) -> concat:in0:v0
>   Stream #0:1 (ac3) -> concat:in0:a0
>   Stream #1:0 (h264) -> concat:in1:v0
>   Stream #1:1 (ac3) -> concat:in1:a0
>   Stream #2:0 (h264) -> concat:in2:v0
>   Stream #2:1 (ac3) -> concat:in2:a0
>   Stream #3:0 (h264) -> concat:in3:v0
>   Stream #3:1 (ac3) -> concat:in3:a0
>   Stream #4:0 (h264) -> concat:in4:v0
>   Stream #4:1 (ac3) -> concat:in4:a0
>   Stream #5:0 (h264) -> concat:in5:v0
>   Stream #5:1 (ac3) -> concat:in5:a0
>   concat:out:v0 -> Stream #0:0 (libx264)
>   concat:out:a0 -> Stream #0:1 (aac)
> Press [q] to stop, [?] for help
> [h264 @ 0000000005d39220] mmco: unref short failure
> frame= 2289 fps= 51 q=28.0 size=   12645kB time=00:00:45.54
> bitrate=2274.3kbits/s dup=23 drop=0





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From cehoyos at ag.or.at  Thu Jul 25 11:58:35 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 25 Jul 2013 09:58:35 +0000 (UTC)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
	playback of video streams
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
	<1374741951675-4660169.post@n4.nabble.com>
Message-ID: <loom.20130725T115702-217@post.gmane.org>

gergely <Gergely.Lukacsy <at> streamnet.hu> writes:

> > [h264  <at>  00000000002fe260] mmco: unref short failure

Can you reproduce the message with one of the following 
commands?
$ ffmpeg -i 00000.mts
$ ffmpeg -i 00001.mts
$ ffmpeg -i 00002.mts
$ ffmpeg -i 00003.mts
$ ffmpeg -i 00004.mts
$ ffmpeg -i 00005.mts

Carl Eugen


From sw.craftsman at gmail.com  Thu Jul 25 13:10:00 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Thu, 25 Jul 2013 16:40:00 +0530
Subject: [FFmpeg-user] Video bit rate missing from FFProbe output
In-Reply-To: <loom.20130725T000737-643@post.gmane.org>
References: <CA+XaXC=J7BMNZTF3orYpr_pNrDbmYdVmUycwJeWMa9b7NHGh5w@mail.gmail.com>
	<loom.20130723T220119-727@post.gmane.org>
	<CA+XaXCnhFAYePWHi12_zA_GDqXcqLsa8y9CdMokb1qrrLnh5CQ@mail.gmail.com>
	<loom.20130725T000737-643@post.gmane.org>
Message-ID: <CA+XaXC=za6GKxgvfjX2tRTbmU4BZ7nGoAUKvOvnxypNbK56Tmw@mail.gmail.com>

Hi Carl,

I am a new user ... can you tell me where can I get that patch from ? I
will need static binary.

I have many videos like this wherein latest FFProbe is not showing video
bit-rate unlike earlier version, and when I just open it in Media-Info, I
can see the video-bit rate there. I guess all those videos may just have
the max bit-rate (like this video).

Was this deliberately removed from the latest FFProbe version ?

Thanks



On Thu, Jul 25, 2013 at 3:38 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Shyam Sundar <sw.craftsman <at> gmail.com> writes:
>
> > I have uploaded it at http://files.videohelp.com/u/211860/cc.mpg2
>
> This stream only contains a maximum bitrate, I sent
> a patch to show the maxrate if no bitrate is defined.
>
> Please do not top-post here, Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From lulebo at gmail.com  Thu Jul 25 13:12:08 2013
From: lulebo at gmail.com (Carl Lindqvist)
Date: Thu, 25 Jul 2013 13:12:08 +0200
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <20130724210140.4cf72f11@zooty>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<20130724210140.4cf72f11@zooty>
Message-ID: <CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>

I've just started experimenting using the select filter to do this.
You can use select in combination with between and the "n" parameter
to select a bunch of frames to cut/keep. Right now I use aselect to do
the same thing on the audio. Not sure if it's needed, haven't done
much testing. Here is an example that will encode frames 0-100 and
130-200 to the output:

ffmpeg.exe -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'm not yet sure if between 0-100 means that it will keep frame 0 and
100 or if it means 1 to 99, but I think it is 0 to 100.

/Carl

From Gergely.Lukacsy at streamnet.hu  Thu Jul 25 16:02:17 2013
From: Gergely.Lukacsy at streamnet.hu (gergely)
Date: Thu, 25 Jul 2013 07:02:17 -0700 (PDT)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
 playback of video streams
In-Reply-To: <loom.20130725T115702-217@post.gmane.org>
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
	<1374741951675-4660169.post@n4.nabble.com>
	<loom.20130725T115702-217@post.gmane.org>
Message-ID: <1374760937649-4660175.post@n4.nabble.com>

Yes, I get the same message for *all inputs*, even using the newest static
version. In the meantime, the conversion ended /successfully/.




Carl Eugen Hoyos wrote
> gergely 
> <Gergely.Lukacsy <at>
>  streamnet.hu> writes:
> 
>> > [h264  
> <at>
>   00000000002fe260] mmco: unref short failure
> 
> Can you reproduce the message with one of the following 
> commands?
> $ ffmpeg -i 00000.mts
> $ ffmpeg -i 00001.mts
> $ ffmpeg -i 00002.mts
> $ ffmpeg -i 00003.mts
> $ ffmpeg -i 00004.mts
> $ ffmpeg -i 00005.mts
> 
> Carl Eugen
> 
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

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





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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/mmco-unref-short-failure-error-and-freeze-playback-of-video-streams-tp4657614p4660175.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From cehoyos at ag.or.at  Thu Jul 25 16:20:08 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 25 Jul 2013 14:20:08 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg how to set http timeout
References: <CAFd2cZXY6QjVXryfW8qJUF9ev_8BjhMv8tXUPhat=rzs4BZa7Q@mail.gmail.com>
Message-ID: <loom.20130725T161935-294@post.gmane.org>

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

> ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 
> -input_format mjpeg -i /dev/video0 
> -vcodec copy http://localhost:8080/feed1.ffm
> 
> how to set timeout for http (tcp )connect?

Did you read http://ffmpeg.org/ffmpeg-all.html#http ?

Carl Eugen


From cehoyos at ag.or.at  Thu Jul 25 16:22:03 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 25 Jul 2013 14:22:03 +0000 (UTC)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <loom.20130725T162056-240@post.gmane.org>

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

> I want to do frame-exact edits.? I really want 
> to specify the start and end frame numbers

As explained by Carl Lindqvist use the select filter 
to specify frame numbers, note that "between" is not 
the only function it supports.

Carl Eugen


From jpboard2 at yahoo.com  Thu Jul 25 16:52:21 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 25 Jul 2013 07:52:21 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <20130724210140.4cf72f11@zooty>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<20130724210140.4cf72f11@zooty>
Message-ID: <1374763941.71962.YahooMailNeo@web164704.mail.gq1.yahoo.com>

> My technique is to produce a low res version of the original file

> that uses the drawtext filter to write the exact timestamp
> on each frame, then I pick off the times from that video :-).

That's interesting.? Is there a way to use the drawtext filter to
write frame numbers on each frame?

Thanks.

From jpboard2 at yahoo.com  Thu Jul 25 16:59:44 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 25 Jul 2013 07:59:44 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>	<20130724210140.4cf72f11@zooty>
	<CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
Message-ID: <1374764384.36258.YahooMailNeo@web164702.mail.gq1.yahoo.com>

> I've just started experimenting using the select filter to do this.

> You can use select in combination with between and the "n" parameter
> to select a bunch of frames to cut/keep. Right now I use aselect to do
> the same thing on the audio. Not sure if it's needed, haven't done
> much testing. Here is an example that will encode frames 0-100 and
> 130-200 to the output:

> ffmpeg.exe -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'm not yet sure if between 0-100 means that it will keep frame 0 and
> 100 or if it means 1 to 99, but I think it is 0 to 100.

This sounds exactly like what I've been looking for.

I could not find any documentation for the 'select' filter in
http://www.ffmpeg.org/ffmpeg-all.html.? However it might be
in there, but it's hard to find because if you search for the word
'select' it appears hundreds of times.

Is there documentation for select?? The reason I'm asking
is that your ffmpeg command has a lot of arguments which I
don't understand and I'm not sure if I really need them.? My
input and output file will be AVI, and both will be some raw
format where each frame is stored in full (mpeg encode later).

From jpboard2 at yahoo.com  Thu Jul 25 17:06:12 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 25 Jul 2013 08:06:12 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374764384.36258.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>	<20130724210140.4cf72f11@zooty>
	<CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
	<1374764384.36258.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <1374764772.39989.YahooMailNeo@web164701.mail.gq1.yahoo.com>

> I could not find any documentation for the 'select' filter in

> http://www.ffmpeg.org/ffmpeg-all.html.? However it might be
> in there, but it's hard to find because if you search for the word
> 'select' it appears hundreds of times.

Never mind, I found it.? It's in section 40 Multimedia Filters, 
not section 37 Video Filters.

However, that documentation doesn't mention 'between'.? Is there
current documentation?

From cehoyos at ag.or.at  Thu Jul 25 17:14:12 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 25 Jul 2013 15:14:12 +0000 (UTC)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
	playback of video streams
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
	<1374741951675-4660169.post@n4.nabble.com>
	<loom.20130725T115702-217@post.gmane.org>
	<1374760937649-4660175.post@n4.nabble.com>
Message-ID: <loom.20130725T171350-173@post.gmane.org>

gergely <Gergely.Lukacsy <at> streamnet.hu> writes:

> Yes, I get the same message for all inputs

Please provide a sample.

Please do not top-post here, Carl Eugen


From jpboard2 at yahoo.com  Thu Jul 25 17:26:35 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 25 Jul 2013 08:26:35 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>	<20130724210140.4cf72f11@zooty>
	<CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
Message-ID: <1374765995.43447.YahooMailNeo@web164706.mail.gq1.yahoo.com>

> ffmpeg.exe -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 tried this, and it didn't work.? It complained about a missing ')' or too many
args in 'between(n'
??? ffmpeg -i IN.avi -vf select = 'between(n,32,42)' OUT.avi

I also tried this, and it produced an output file, but that was the same as the
original: the select filter didn't do anything:

ffmpeg -i IN.avi -vf select = 'between(n,32,42)' -vcodec copy -acodec copy OUT.avi

So, what's an example of something that would work for me?? I have raw pixels
as input and I want raw pixels as output (no mpeg encoding).? 

I don't know what options like 'setpts=N/FRAME_RATE/TB' mean.? I'm
more used to mencoder, not ffmpeg, but trying to learn.

From 670292548 at qq.com  Thu Jul 25 17:40:45 2013
From: 670292548 at qq.com (=?gb18030?B?dW5kZWZpbmVk?=)
Date: Thu, 25 Jul 2013 23:40:45 +0800
Subject: [FFmpeg-user] =?gb18030?b?u9i4tKO6ICBoZWxw?=
Message-ID: <tencent_5CA0E8233FB045883BBC1A61@qq.com>

Hi,Tom 
Thank you very much! I've been tested it for a whole day. And I know a little about the formats and codecs. I tried this cmd and it ran well: 
ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx264 -s 640*360 -f flv "rtmp://localhost/testFFMpeg/W3FHRHHT_360"
What I can't understand is that format flv shouldn't support h264 & AAC. The input live stream, published by OBS, is encoded in h264 & AAC. Why did the log say that it was formatted with flv? Like this:
Input #0, flv, from 'rtmp://localhost/testFFMpeg/W3FHRHHT live=1'.
And the output stream encoded in h264 & AAC but format with flv can play smoothly, and be recorded in a file formatted with mp4 or f4v by AMS5 normally. Does it mean that the live streams which run on AMS5 are in a box of flv, but encoded in h264 & AAC?


Best Regards,
Lane 


------------------ ???? ------------------
???: "Tom Evans"<tevans.uk at googlemail.com>;
????: 2013?7?25?(???) ??4:59
???: "FFmpeg user questions"<ffmpeg-user at ffmpeg.org>; 

??: Re: [FFmpeg-user] help



On Thu, Jul 25, 2013 at 3:17 AM, undefined <670292548 at qq.com> wrote:
> here's the uncut log:
>
>
> C:\Users\CODE-02\ffmpeg>ffmpeg -re -y -i "rtmp://localhost/testFFMpeg/W3FHRHHT live=1" -acodec libvo_aacenc -vcodec libx
> 264 -s 640*360 -f mp4 "rtmp://localhost/testFFMpeg/W3FHRHHT_360"
> ?
> [mp4 @ 0000000004af00a0] muxer does not support non seekable output

mp4 requires a seekable output, since it writes a MOOV atom at the
start of the file once it finishes encoding the file. Therefore, this
is not a suitable format for feeding data to rtmp.

Cheers

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

From lucielemonnier at hotmail.fr  Thu Jul 25 17:42:17 2013
From: lucielemonnier at hotmail.fr (lucie Lemonnier)
Date: Thu, 25 Jul 2013 17:42:17 +0200
Subject: [FFmpeg-user] ffmpeg --disable-w32threads doesn't enable pthread
Message-ID: <DUB111-W84D3FDB0F4CAC43300258B9690@phx.gbl>

Hello,
I want to compile ffmpeg with pthread but when I add --disable-w32threads, my compilator doesn't add pthread.
Why?
I use Visual Studio 2010 (--toolchain = msvc) with msys.bat.
Can you help me, please?
Lucie
 		 	   		  

From tevans.uk at googlemail.com  Thu Jul 25 17:50:29 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Thu, 25 Jul 2013 16:50:29 +0100
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374765995.43447.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<20130724210140.4cf72f11@zooty>
	<CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
	<1374765995.43447.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1KrR=p6_YOSV3LAuJkCKVkj6ufK-KUEeMN0HJtzm4pD=g@mail.gmail.com>

On Thu, Jul 25, 2013 at 4:26 PM, James Board <jpboard2 at yahoo.com> wrote:
>> ffmpeg.exe -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 tried this, and it didn't work.  It complained about a missing ')' or too many
> args in 'between(n'
>     ffmpeg -i IN.avi -vf select = 'between(n,32,42)' OUT.avi
>

Filters are tricky. You need to pass a single argument to '-vf'. From
your command line, it looks like you are passing the argument
"select", there should not be spaces between "select", "=" and the
filter.

Secondly, filters are graphs - a connected chain of inputs and
outputs. You chain two filters together with a comma, and therefore,
in filter specs, commas that do not chain two filters together must be
escaped or quoted.

Finally, this is all describing what ffmpeg needs to see it. If your
shell needs extra quoting, then you need to add that in as well.

So, for your example, either of these will work:

  ffmpeg -i IN -vf select='between(n\,32\,42)' OUT

or

  ffmpeg -i IN -vf "select='between(n,32,42)'" OUT

Read this section on filtering:

http://trac.ffmpeg.org/wiki/FilteringGuide

Cheers

Tom

From lucielemonnier at hotmail.fr  Thu Jul 25 18:19:37 2013
From: lucielemonnier at hotmail.fr (lucie Lemonnier)
Date: Thu, 25 Jul 2013 18:19:37 +0200
Subject: [FFmpeg-user] ffmpeg --disable-w32threads doesn't enable pthread
In-Reply-To: <DUB111-W84D3FDB0F4CAC43300258B9690@phx.gbl>
References: <DUB111-W84D3FDB0F4CAC43300258B9690@phx.gbl>
Message-ID: <DUB111-W32D4E97F9E698EF816B41AB9690@phx.gbl>

I found the issue, i have pthreadVC2.lib instead of pthreadGC2.lib.

> From: lucielemonnier at hotmail.fr
> To: ffmpeg-user at ffmpeg.org
> Date: Thu, 25 Jul 2013 17:42:17 +0200
> Subject: [FFmpeg-user] ffmpeg --disable-w32threads doesn't enable pthread
> 
> Hello,
> I want to compile ffmpeg with pthread but when I add --disable-w32threads, my compilator doesn't add pthread.
> Why?
> I use Visual Studio 2010 (--toolchain = msvc) with msys.bat.
> Can you help me, please?
> Lucie
>  		 	   		  
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
 		 	   		  

From lou at lrcd.com  Thu Jul 25 20:32:19 2013
From: lou at lrcd.com (Lou)
Date: Thu, 25 Jul 2013 10:32:19 -0800
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374764772.39989.YahooMailNeo@web164701.mail.gq1.yahoo.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<20130724210140.4cf72f11@zooty>
	<CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
	<1374764384.36258.YahooMailNeo@web164702.mail.gq1.yahoo.com>
	<1374764772.39989.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <20130725103219.27b0a839@lrcd.com>

On Thu, 25 Jul 2013 08:06:12 -0700 (PDT)
James Board <jpboard2 at yahoo.com> wrote:

> However, that documentation doesn't mention 'between'.? Is there
> current documentation?

http://ffmpeg.org/ffmpeg-utils.html#Expression-Evaluation

From zachjacobs at gmail.com  Thu Jul 25 21:43:12 2013
From: zachjacobs at gmail.com (Zach Jacobs)
Date: Thu, 25 Jul 2013 15:43:12 -0400
Subject: [FFmpeg-user] How to set X264 options using FFmpeg libraries
Message-ID: <CACE3me6E9+O=d7JxzJ+C7tvaY+tYQA+pH+T34Ej56m0N-1GQxw@mail.gmail.com>

Hello,

I am trying to control the bit rate of a file using x264 with the crf and
maximum bit rate parameters.  I am following the x264 encoding guide
here<https://trac.ffmpeg.org/wiki/x264EncodingGuide#CRFwithmaximumbitrate>.
 I tried the following from the command line:

ffmpeg -i input.m4v -c:v libx264 -crf 20 -maxrate 400k -bufsize 1835k
output.ts

This works very well and the resulting bit rate is usually very close to
400k.  However when I try to do the same thing by using the FFmpeg
libraries, the encoder seems to ignore the maxrate parameter.  Here is how
I am setting up my CodecContext:

                AVDictionary* opt = null;
                FFmpegInvoke.av_dict_set(&opt, "preset", "slow", 0);
                FFmpegInvoke.av_dict_set(&opt, "crf", "20.0", 0);
                FFmpegInvoke.av_dict_set(&opt, "maxrate", "400k", 0);
                FFmpegInvoke.av_dict_set(&opt, "bufsize", "1835k", 0);

                videoCodec =
FFmpegInvoke.avcodec_find_encoder(AVCodecID.AV_CODEC_ID_H264);

                if (videoCodec == null)
                    throw new Exception("FFMPEG Encoder Error: Codec Not
Found");

                videoCodecContext =
FFmpegInvoke.avcodec_alloc_context3(videoCodec);
                videoCodecContext->width = encoderInfo.Settings.VideoWidth;
                videoCodecContext->height =
encoderInfo.Settings.VideoHeight;

                videoCodecContext->pix_fmt =
AVPixelFormat.AV_PIX_FMT_YUV420P;
                videoCodecContext->time_base.num = 1;
                videoCodecContext->time_base.den = 1000;

                videoCodecContext->codec_type =
AVMediaType.AVMEDIA_TYPE_VIDEO;
                videoCodecContext->codec_id = AVCodecID.AV_CODEC_ID_H264;

                if (codecFlags != -1)
                    videoCodecContext->flags |= codecFlags;

                if (FFmpegInvoke.avcodec_open2(videoCodecContext,
videoCodec, &opt) < 0)
                    throw new Exception("FFmpeg Encoder Error: could not
open video codec");

opt is empty after avcodec_open2 is called indicating all of my dictionary
options that I tried to set were found and processed.  However, the output
video bit rate is no where near 400k maxrate that I specified.  If I
increase maxrate, the bit rate increases but it still isn't anywhere close
to what it should be.  The output file is roughly 319 MB.

>From the command line, the video bit rate is 395k and the resulting video
size is only 42MB.  Am I missing a setting?

Thanks for your time!

Zach

From dut.samrat at gmail.com  Thu Jul 25 13:27:17 2013
From: dut.samrat at gmail.com (samrat)
Date: Thu, 25 Jul 2013 04:27:17 -0700 (PDT)
Subject: [FFmpeg-user] MP3 file max bit rate issue
Message-ID: <1374751637912-4660174.post@n4.nabble.com>

I have tried to create a mp3 file with some specific parameter like maximum
bitrate , duration ,sampling rate, channel. For this I have used ffmpeg
Libmp3lame library with the following command ;

ffmpeg -i OLD1.mp3 -codec:a libmp3lame -ar 32000 -ab 56k -ac 2 -q:a 1 -t
00:03:00.000 -y NEW-1.mp3

Here we need to create a mp3 file with maxbit rate 56kbps. So I used -ab 56K
but unfortunately It creates with 64K max bit rate.
So I want to know , is there any other command to create the file with
proper maxbit rate?

and another question is what is the command in ffmpeg to get the max bit
rate?

Answer of the above two questions are relly helpful to me.

Thanks



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From hevesyagyeman at yahoo.com  Thu Jul 25 05:53:58 2013
From: hevesyagyeman at yahoo.com (hevesyagyeman)
Date: Wed, 24 Jul 2013 20:53:58 -0700 (PDT)
Subject: [FFmpeg-user] libx264 mp4 encoded video won't play back in
	chrome/quicktime
In-Reply-To: <CAHJN-tktqtUEfKUZbt-0EdphsXWYs=HmDTvVZoX_qGbM6SXGDg@mail.gmail.com>
References: <CAHJN-tktqtUEfKUZbt-0EdphsXWYs=HmDTvVZoX_qGbM6SXGDg@mail.gmail.com>
Message-ID: <1374724438056-4660168.post@n4.nabble.com>

Acting as a video container format, MP4 is the package and can contain video
and audio components encoded with different types (codecs). However,
QuickTime just supports a few video and audio codecs in MP4 format. That is,
if the MP4 video contains QuickTime unsupported audio or video codecs,
QuickTime will refuse to play this kind of MP4 files.

In order to solve MP4 won't play in QuickTime issue, we had better  convert
QuickTime unplayable MP4 to QuickTime supported format.
<http://www.idealshare.net/mp4-quicktime-solution.html>  




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Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jiachielee at live.com  Thu Jul 25 16:05:51 2013
From: jiachielee at live.com (thljcl)
Date: Thu, 25 Jul 2013 07:05:51 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <1374761151615-4660176.post@n4.nabble.com>

Let?s see if I get your question correctly. You have a video with
uncompressed file format. You want to 

edit some frames; but even if you found the exact frame number, ffmpeg might
not be able to extract the 

frame you want. Assuming that I understood your question correctly, here is
what I would do if I were 

you or in similar situation.

Assume that your video is of constant frame rate, frame number will then be
directly proportional to the 

time. I?m not really sure which video codec is being used to store video
information. I would assume 

that you might be in favor of lossless compression compared with lossy
compression, especially when you 

still need to edit the videos. You can transcode the video to H.264 lossless
if it is not already in 

H.264 lossless.

ffmpeg -i "input.avi" -vcodec "libx264" -crf "0" "input.mkv"

You can convert the entire video into a series of image sequence but they
may end up costing a lot of 

storage space. More efficiently, you can just get a small part of video
which contains the frames you 

want to edit. You can do that with mkvtoolnix. It split, merge, remove
tracks, and add tracks to MKV 

file, without re-encoding the video/audio/subtitle; but when it comes to
accuracy of splitting video 

files based on duration or time, there will be discrepancies due to
limitation of accuracy. While it 

cannot split the frame you want exactly, it can split video to a small part,
which you can later convert 

that small part to a series of images sequence, instead of entire videos.

mkdir ?frames?
ffmpeg -i "input.mkv" "frames\f_%%1d.png"

After you edited the frames you want,

ffmpeg -r "24" -i "frames\f_%%1d.png" -vcodec "libx264" -crf "0"
"output-1.mkv"

Here I assume that the frame rate of your video is 24 FPS.

You can then join the video back using mkvtoolnix.

If your video source is of variable frame rate, you need to how the frame
rate changes over time in 

order to preserve the original frame rate. The only way to know exactly is
that you actually specify how 

the frame rate changes yourself. If you do know how the frame rate changes,
To convert the video part to image sequence without changing frame rate,

mkdir "frames"
ffmpeg -r "1" "input.mkv" "frames\f_%%1d.png"

To change it back to videos, you need to do it part by part, which means
that each part is of constant 

frame rate; when they are joined together, its frame rate is of course
variable.
If you do not know how the frame rate changes but do know that frame rate is
variable due to the setting 

of recording device or other reasons, you do not want to change the playback
speed, you need to change 

the frame rate to a constant frame rate you find suitable, for example,

ffmpeg -i "input.avi" -vf "fps=24" -vcodec "libx264" -crf "0" "input.mkv"

Using video filter to change frame rate will of course result in the drop or
add of frames to achieve 

the constant frame rate you designated. Personally, I generally use 24 FPS,
the industry standard 

adopted by film industry.




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From cehoyos at ag.or.at  Thu Jul 25 21:55:42 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 25 Jul 2013 19:55:42 +0000 (UTC)
Subject: [FFmpeg-user] MP3 file max bit rate issue
References: <1374751637912-4660174.post@n4.nabble.com>
Message-ID: <loom.20130725T215354-986@post.gmane.org>

samrat <dut.samrat <at> gmail.com> writes:

> ffmpeg -i OLD1.mp3 -codec:a libmp3lame -ar 32000 -ab 56k 
> -ac 2 -q:a 1 -t 00:03:00.000 -y NEW-1.mp3
> 
> Here we need to create a mp3 file with maxbit rate 56kbps. 
> So I used -ab 56K but unfortunately It creates with 64K 
> max bit rate.

(Complete, uncut console output missing.)

You specified both a quality and a bitrate (which 
makes no sense), please remove the quality setting 
and report back.

Carl Eugen


From onemda at gmail.com  Thu Jul 25 22:30:04 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Thu, 25 Jul 2013 20:30:04 +0000
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374761151615-4660176.post@n4.nabble.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<1374761151615-4660176.post@n4.nabble.com>
Message-ID: <CAPYw7P5psCgt1xcqV5iCFx3cSXBMgFchL5UW6_q65vXxmPwRzg@mail.gmail.com>

On 7/25/13, thljcl <jiachielee at live.com> wrote:
> Let?s see if I get your question correctly. You have a video with
> uncompressed file format. You want to
>
> edit some frames; but even if you found the exact frame number, ffmpeg might
> not be able to extract the
>
> frame you want. Assuming that I understood your question correctly, here is
> what I would do if I were
>
> you or in similar situation.
>
> Assume that your video is of constant frame rate, frame number will then be
> directly proportional to the
>
> time. I?m not really sure which video codec is being used to store video
> information. I would assume
>
> that you might be in favor of lossless compression compared with lossy
> compression, especially when you
>
> still need to edit the videos. You can transcode the video to H.264 lossless
> if it is not already in
>
> H.264 lossless.
>
> ffmpeg -i "input.avi" -vcodec "libx264" -crf "0" "input.mkv"
>
> You can convert the entire video into a series of image sequence but they
> may end up costing a lot of
>
> storage space. More efficiently, you can just get a small part of video
> which contains the frames you
>
> want to edit. You can do that with mkvtoolnix. It split, merge, remove
> tracks, and add tracks to MKV
>
> file, without re-encoding the video/audio/subtitle; but when it comes to
> accuracy of splitting video
>
> files based on duration or time, there will be discrepancies due to
> limitation of accuracy. While it
>
> cannot split the frame you want exactly, it can split video to a small part,
> which you can later convert
>
> that small part to a series of images sequence, instead of entire videos.
>
> mkdir ?frames?
> ffmpeg -i "input.mkv" "frames\f_%%1d.png"
>
> After you edited the frames you want,
>
> ffmpeg -r "24" -i "frames\f_%%1d.png" -vcodec "libx264" -crf "0"
> "output-1.mkv"
>
> Here I assume that the frame rate of your video is 24 FPS.
>
> You can then join the video back using mkvtoolnix.
>
> If your video source is of variable frame rate, you need to how the frame
> rate changes over time in
>
> order to preserve the original frame rate. The only way to know exactly is
> that you actually specify how
>
> the frame rate changes yourself. If you do know how the frame rate changes,
> To convert the video part to image sequence without changing frame rate,
>
> mkdir "frames"
> ffmpeg -r "1" "input.mkv" "frames\f_%%1d.png"
>
> To change it back to videos, you need to do it part by part, which means
> that each part is of constant
>
> frame rate; when they are joined together, its frame rate is of course
> variable.
> If you do not know how the frame rate changes but do know that frame rate is
> variable due to the setting
>
> of recording device or other reasons, you do not want to change the playback
> speed, you need to change
>
> the frame rate to a constant frame rate you find suitable, for example,
>
> ffmpeg -i "input.avi" -vf "fps=24" -vcodec "libx264" -crf "0" "input.mkv"
>
> Using video filter to change frame rate will of course result in the drop or
> add of frames to achieve
>
> the constant frame rate you designated. Personally, I generally use 24 FPS,
> the industry standard
>
> adopted by film industry.

That last line says everything.....

From cehoyos at ag.or.at  Fri Jul 26 01:16:02 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 25 Jul 2013 23:16:02 +0000 (UTC)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<1374761151615-4660176.post@n4.nabble.com>
Message-ID: <loom.20130726T011459-433@post.gmane.org>

thljcl <jiachielee <at> live.com> writes:

> You want to edit some frames; but even if you 
> found the exact frame number, ffmpeg might 
> not be able to extract the frame you want.

Why do you think so?
I do not remember a bug report about the select 
filter, if there is a problem, please report!

Carl Eugen


From jpboard2 at yahoo.com  Fri Jul 26 04:09:59 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 25 Jul 2013 19:09:59 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374761151615-4660176.post@n4.nabble.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<1374761151615-4660176.post@n4.nabble.com>
Message-ID: <1374804599.8768.YahooMailNeo@web164701.mail.gq1.yahoo.com>

> From: thljcl <jiachielee at live.com>
> To: ffmpeg-user at ffmpeg.org 
> Sent: Thursday, July 25, 2013 10:05 AM
> Subject: Re: [FFmpeg-user] Cut-edits with ffmpeg
 

Thanks for your thoughtful reply.? But I was hoping for something a bit simpler.

Actually, the 'Select' filter seems to do everything I wanted: multiple segments, and 
frame numbers instead of start/stop times.? I wonder why nobody mentioned this
the other times I asked.

Anyway, I think I figured out how to get the Select filter working.? However, I didn't
check the audio sync yet.

From jpboard2 at yahoo.com  Fri Jul 26 04:11:00 2013
From: jpboard2 at yahoo.com (James Board)
Date: Thu, 25 Jul 2013 19:11:00 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <20130725103219.27b0a839@lrcd.com>
References: <1374710228.49172.YahooMailNeo@web164705.mail.gq1.yahoo.com>	<20130724210140.4cf72f11@zooty>	<CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>	<1374764384.36258.YahooMailNeo@web164702.mail.gq1.yahoo.com>	<1374764772.39989.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<20130725103219.27b0a839@lrcd.com>
Message-ID: <1374804660.66169.YahooMailNeo@web164703.mail.gq1.yahoo.com>

>> However, that documentation doesn't mention 'between'.? Is there

>> current documentation?
>
>http://ffmpeg.org/ffmpeg-utils.html#Expression-Evaluation

Yes, there it is.? Thank you.? I didn't know that capability existed.

From soho123.2012 at gmail.com  Fri Jul 26 04:42:49 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Fri, 26 Jul 2013 10:42:49 +0800
Subject: [FFmpeg-user] ffmpeg how to set http timeout
In-Reply-To: <loom.20130725T161935-294@post.gmane.org>
References: <CAFd2cZXY6QjVXryfW8qJUF9ev_8BjhMv8tXUPhat=rzs4BZa7Q@mail.gmail.com>
	<loom.20130725T161935-294@post.gmane.org>
Message-ID: <CAFd2cZU_gQjvMbbKJF7n9M0egdMVkm_8k53-OHcp3SphD8nfTA@mail.gmail.com>

> Did you read http://ffmpeg.org/ffmpeg-all.html#http ?
>

It is not clear about how to use "timeout" option.
for example:
the ffmpeg command line:
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format h264
-i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm

when I try the command:
ffmpeg -loglevel debug -f video4linux2 -r 30 -s 1280x720 -input_format h264
-i /dev/video1 -vcodec copy http://localhost:8090/feed2.ffm?timeout=3000000

then ffmpeg says : Invalid argument


Could you kindly help to explain where I can put the option "timeout"?

From ajsinclair at gmail.com  Fri Jul 26 06:11:40 2013
From: ajsinclair at gmail.com (Andrew Sinclair)
Date: Fri, 26 Jul 2013 14:11:40 +1000
Subject: [FFmpeg-user] librtmp with authentication
In-Reply-To: <CAOeAR15CMXqCZz2R3eOr-644CRC1GL=Hi1+VOsrCq7NR2iTmEQ@mail.gmail.com>
References: <CAOeAR15CMXqCZz2R3eOr-644CRC1GL=Hi1+VOsrCq7NR2iTmEQ@mail.gmail.com>
Message-ID: <CAOeAR178oZQZ26daTXVBUYdBRMWWPTo=99_k8W+At9DOVgiBag@mail.gmail.com>

On 24 July 2013 16:57, Andrew Sinclair <ajsinclair at gmail.com> wrote:

> Hi,
>
> Has anyone had recent success using the librtmp (--enable-librtmp) to
> authenticate with an Adobe Media Server when pushing a stream?
>
>
> Turns out I don't need librtmp to authenticate and you can just use http
style authentication options e.g.
ffmpeg -f lavfi -i smptebars -c:v libx264 -b:v 500k -an -f flv
rtmp://<USERNAME>:<PASSWORD>@myrtmpserver.com/Entrypoint/stream_1 at 2003030

Note that if anyone else is looking to do this it worked with Akamai's
Adobe/Flash Media servers for.

From andycivil at gmail.com  Fri Jul 26 08:08:46 2013
From: andycivil at gmail.com (Andy Civil)
Date: Fri, 26 Jul 2013 02:08:46 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
Message-ID: <51F2126E.40200@gmail.com>

Short description: I'm moaning that the default output for "ntsc-dvd" is 
progressive, even if I give it enough information to make it interlaced.

Long description:

First, an aside on interlacing. I know that in general, people here don't like 
interlacing and attempt to deinterlace at the earliest opportunity. I can see a 
day when interlacing is pointless, when all CRT displays are on the scrap heap 
and no one uses DVDs any more. I used to think that interlacing was a viable 
alternative to compression for saving data, but I now realise that where 
software compression is available, the advantage is moot because compression can 
reduce the data rate more efficiently than the interlacing process, and although 
interlacing might appear to halve the amount of data required, it simply makes 
the datastream less compressible so you gain nothing in the end.

Having said that, we still live in a world where people use DVDs to store and 
share movies. And like it or not, we have to work within the constraints of what 
is a valid DVD-video stream. Within that constraint, a video of 30fps interlaced 
(60 fields per second) does indeed look better than a straight 30p because there 
is more temporal information. Pointing out that a 60p movie might compress to 
smaller than the 30i is irrelevant, because 60p is not a valid mode for DVD-video.

I am in the position of wanting to make DVD videos of material I have shot with 
an HD camera. After days of head-scratching, I have found that the interlacing 
of HD video is opposite in field order to that required by DVD. Further, the 
video editor that I'm using to edit my videos seems to do a poor job of scaling 
the interlaced content (seriously). I therefore turned to trusty FFmpeg to solve 
the quality problem. I got the editor to output a rather large temporary file in 
HD resolution and the full 60p. (The quality is OK because it seems able to 
de-interlace fine.) I then asked FFmpeg to convert this to DVD-video. It's a 
simple command:
c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-07-hdp.mp4 -target ntsc-dvd -acodec mp2 
-ac 2 -b:a 128000 output-dvd.mpg

Now here's my complaint: this works fine, but it makes a 30p video, simply 
throwing away half of the frames. 30p is a valid mode for DVD-video, but it's 
not as smooth as it could be. I found that I had to specifically add the flags 
-flags ildct+ilme
to make it create a proper interlaced video. (Which it does, for sure.)

Therefore the default action is not the best it can do (with a 60p source). Is 
that a fault, or simply a limitation of the 'preset' system (not knowing if the 
source video has enough frames to make interlaced or not)?

--
Andy


From francois.visagie at gmail.com  Fri Jul 26 09:14:55 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Fri, 26 Jul 2013 09:14:55 +0200
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F2126E.40200@gmail.com>
References: <51F2126E.40200@gmail.com>
Message-ID: <000e01ce89cf$d8387fa0$88a97ee0$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Andy Civil
> Sent: 26 July 2013 08:09
> To: ffmpeg-user at ffmpeg.org
> Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
> 
> Short description: I'm moaning that the default output for "ntsc-dvd" is
> progressive, even if I give it enough information to make it interlaced.
> 
> Long description:
> 
> First, an aside on interlacing. I know that in general, people here don't
like
> interlacing and attempt to deinterlace at the earliest opportunity. I can
see a
> day when interlacing is pointless, when all CRT displays are on the scrap
heap
> and no one uses DVDs any more. I used to think that interlacing was a
viable
> alternative to compression for saving data, but I now realise that where
> software compression is available, the advantage is moot because
> compression can
> reduce the data rate more efficiently than the interlacing process, and
> although
> interlacing might appear to halve the amount of data required, it simply
> makes
> the datastream less compressible so you gain nothing in the end.
> 
> Having said that, we still live in a world where people use DVDs to store
and
> share movies. And like it or not, we have to work within the constraints
of
> what
> is a valid DVD-video stream.

Progressive input to the authoring process isn't a problem, the DVD standard
allows that. I own quite a few progressively encoded commercial DVDs.

> Within that constraint, a video of 30fps
> interlaced
> (60 fields per second) does indeed look better than a straight 30p because
> there
> is more temporal information. Pointing out that a 60p movie might compress
> to
> smaller than the 30i is irrelevant, because 60p is not a valid mode for
DVD-
> video.
> 
> I am in the position of wanting to make DVD videos of material I have shot
> with
> an HD camera. After days of head-scratching, I have found that the
> interlacing
> of HD video is opposite in field order to that required by DVD.

If you want to encode interlaced, field order isn't a problem either; the
DVD standard accepts either. For ffmpeg just set the correct field order
with '-top[:stream_specifier] {0|1}'. If you really want to reverse field
order, that's possible with tools like Avisynth, but let's leave that for
another day for now ;-).

>  Further, the
> video editor that I'm using to edit my videos seems to do a poor job of
scaling
> the interlaced content (seriously).

Well picked up, few people realise or notice that. It's a consequence that
applies to scaling interlaced content in general, not only to your editor.
Fields need to be scaled separately to preserve their vertical structure,
otherwise you get vertical blending of field content. Either separate fields
or rather properly deinterlace first.

>  I therefore turned to trusty FFmpeg to
> solve
> the quality problem. I got the editor to output a rather large temporary
file in
> HD resolution and the full 60p. (The quality is OK because it seems able
to
> de-interlace fine.) I then asked FFmpeg to convert this to DVD-video. It's
a
> simple command:
> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-07-hdp.mp4 -target ntsc-dvd -
> acodec mp2
> -ac 2 -b:a 128000 output-dvd.mpg

FWIW, I think the '-acodec mp2' is redundant (at least my command line
doesn't use it). See what happens without it.

> Now here's my complaint: this works fine, but it makes a 30p video, simply
> throwing away half of the frames. 30p is a valid mode for DVD-video, but
it's
> not as smooth as it could be. I found that I had to specifically add the
flags
> -flags ildct+ilme
> to make it create a proper interlaced video. (Which it does, for sure.)

Because the DVD standard allows either interlaced or progressive video, the
'-target [{pal|ntsc}-]dvd' option does not specify the ildct+ilme flags by
default.

> Therefore the default action is not the best it can do (with a 60p
source). Is
> that a fault, or simply a limitation of the 'preset' system (not knowing
if the
> source video has enough frames to make interlaced or not)?

To prevent frame decimation as you're experiencing, feed ffmpeg either 30fps
progressive or 30fps interlaced, with flags set appropriately for the
latter. If you want to go from 60fps progressive to 30fps interlaced before
encoding, you have to reinterlace first. I don't know if that can be done in
ffmpeg. VirtualDub's 'interlace' filter works, but I usually use the
following Avisynth script fragment for jobs like this (going from 60fps
progressive to 30fps interlaced):

# For top field first material
AssumeTFF()
# For bottom field first material
# AssumeBFF()
SeparateFields()
SelectEvery(4, 0, 3)
Weave()

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


From cehoyos at ag.or.at  Fri Jul 26 10:51:18 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 26 Jul 2013 08:51:18 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
Message-ID: <loom.20130726T104114-810@post.gmane.org>

Andy Civil <andycivil <at> gmail.com> writes:

> Short description: I'm moaning that the default output 
> for "ntsc-dvd" is progressive, even if I give it 
> enough information to make it interlaced.

> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-07-hdp.mp4 
> -target ntsc-dvd -acodec mp2 -ac 2 -b:a 128000 
> output-dvd.mpg

(Complete, uncut console output missing.)

Nothing about this command specifies that your input 
is interlaced so why on earth should FFmpeg use 
interlaced encoding (that of course hurts 
compression)?

Or in other words: You did not give enough information 
to FFmpeg that you want interlaced content, so how 
should it guess?

[...]

> I can see a day when interlacing is pointless, 
> when all CRT displays are on the scrap

I use a CRT (all the time) and I certainly don't 
want interlaced content.

[...]

> a video of 30fps interlaced (60 fields per second) 
> does indeed look better than a straight 30p because 
> there is more temporal information

I consider this "wrong" (please consider that an 
euphemism).
If it is course perfectly fine that you disagree 
but please don't claim that this is a universal 
truth (especially since a large percentage of the 
60p material that can be used as input to your 
command line contains duplicated frames and 
never had more temporal information).

> I found that I had to specifically add the flags 
> -flags ildct+ilme to make it create a proper 
> interlaced video.

As such this is simply not true.
The flags make the encoder use interlaced algorithms 
which should only only be used on interlaced 
material (and which of course should be used for 
interlaced sources). In practice, it is often used 
for progressive material (and interlaced pictures 
were of course sometimes encoded without those flags).

Please note that FFmpeg supports producing 
interlaced content with the appropriate filter.

Carl Eugen


From Gergely.Lukacsy at streamnet.hu  Fri Jul 26 11:50:32 2013
From: Gergely.Lukacsy at streamnet.hu (gergely)
Date: Fri, 26 Jul 2013 02:50:32 -0700 (PDT)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
 playback of video streams
In-Reply-To: <loom.20130725T171350-173@post.gmane.org>
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
	<1374741951675-4660169.post@n4.nabble.com>
	<loom.20130725T115702-217@post.gmane.org>
	<1374760937649-4660175.post@n4.nabble.com>
	<loom.20130725T171350-173@post.gmane.org>
Message-ID: <1374832232027-4660200.post@n4.nabble.com>

Carl Eugen:
> Please provide a sample.

Here you go: http://dload.streamnet.hu/4090982482cb4a163ff480ba480b90e3/
I've chosen to smallest one, it's 141.3 MB.



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/mmco-unref-short-failure-error-and-freeze-playback-of-video-streams-tp4657614p4660200.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From cehoyos at ag.or.at  Fri Jul 26 12:45:57 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 26 Jul 2013 10:45:57 +0000 (UTC)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
	playback of video streams
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
	<1374741951675-4660169.post@n4.nabble.com>
	<loom.20130725T115702-217@post.gmane.org>
	<1374760937649-4660175.post@n4.nabble.com>
	<loom.20130725T171350-173@post.gmane.org>
	<1374832232027-4660200.post@n4.nabble.com>
Message-ID: <loom.20130726T124519-81@post.gmane.org>

gergely <Gergely.Lukacsy <at> streamnet.hu> writes:

> Carl Eugen:
> > Please provide a sample.
> 
> Here you go: 
> http://dload.streamnet.hu/4090982482cb4a163ff480ba480b90e3/

Is this the original recording from a camera or 
was the sample cut somehow (in the beginning)?

Carl Eugen


From expandafter at yahoo.com  Fri Jul 26 13:09:41 2013
From: expandafter at yahoo.com (S. Fisher)
Date: Fri, 26 Jul 2013 04:09:41 -0700 (PDT)
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
Message-ID: <1374836981.63746.YahooMailBasic@web163805.mail.gq1.yahoo.com>

On Thu, 7/25/13, Carl Lindqvist <lulebo at gmail.com> wrote:

> I've just started experimenting using the select filter to do this.
> You can use select in combination with between and the "n" parameter
> to select a bunch of frames to cut/keep. Right now I use aselect to do
> the same thing on the audio. Not sure if it's needed, haven't done
> much testing. Here is an example that will encode frames 0-100 and
> 130-200 to the output:
> 
> ffmpeg.exe -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'm not yet sure if between 0-100 means that it will keep frame 0 and
> 100 or if it means 1 to 99, but I think it is 0 to 100.


Look at my post of Jul 18 again:

> It can be done using the filters select, aselect, setpts, and asetpts.
> 
> Let's say you want to keep only two segments: the one starting at
> 5 seconds and ending at 20 seconds, and the one between 40 seconds
> and 52 seconds.
> 
> ffmpeg -i "sample.avi" -vf select='-between(t,5,20)-between(t,40,52)',setpts=N/FRAME_RATE/TB
>  -af aselect='-between(t,5,20)-between(t,40,52)',asetpts=N/SR/TB sample.mp4

When you say
  between(n,130,200)
the numbers 130 and 200 refer to seconds, not frames.

The "aselect" is necessary.


I've since decided that negative numbers aren't a good idea.

ffmpeg -i "sample.avi" -vf select='between(t,5,20)+between(t,40,52)',setpts=N/FRAME_RATE/TB
 -af aselect='between(t,5,20)+between(t,40,52)',asetpts=N/SR/TB sample.mp4







From tevans.uk at googlemail.com  Fri Jul 26 13:24:18 2013
From: tevans.uk at googlemail.com (Tom Evans)
Date: Fri, 26 Jul 2013 12:24:18 +0100
Subject: [FFmpeg-user] Cut-edits with ffmpeg
In-Reply-To: <1374836981.63746.YahooMailBasic@web163805.mail.gq1.yahoo.com>
References: <CAJ6A8tJSjV-wObzgZY8khq_eCXde+VxDzj83aD=otNaNAJ-NMA@mail.gmail.com>
	<1374836981.63746.YahooMailBasic@web163805.mail.gq1.yahoo.com>
Message-ID: <CAFHbX1KoT7QqVNnvPRVAE_LwSdWTz2sgTEBR9iwX2W-iLMuaTQ@mail.gmail.com>

On Fri, Jul 26, 2013 at 12:09 PM, S. Fisher <expandafter at yahoo.com> wrote:
> Look at my post of Jul 18 again:
>
>> It can be done using the filters select, aselect, setpts, and asetpts.
>>
>> Let's say you want to keep only two segments: the one starting at
>> 5 seconds and ending at 20 seconds, and the one between 40 seconds
>> and 52 seconds.
>>
>> ffmpeg -i "sample.avi" -vf select='-between(t,5,20)-between(t,40,52)',setpts=N/FRAME_RATE/TB
>>  -af aselect='-between(t,5,20)-between(t,40,52)',asetpts=N/SR/TB sample.mp4
>
> When you say
>   between(n,130,200)
> the numbers 130 and 200 refer to seconds, not frames.

No, when you say "between(n,130,200)" the numbers refer to frames :)

It's when you say "between(t,130,200)", as you do in your example,
that is when it refers to timestamps.

Cheers

Tom

From Gergely.Lukacsy at streamnet.hu  Fri Jul 26 15:10:04 2013
From: Gergely.Lukacsy at streamnet.hu (gergely)
Date: Fri, 26 Jul 2013 06:10:04 -0700 (PDT)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
 playback of video streams
In-Reply-To: <loom.20130726T124519-81@post.gmane.org>
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
	<1374741951675-4660169.post@n4.nabble.com>
	<loom.20130725T115702-217@post.gmane.org>
	<1374760937649-4660175.post@n4.nabble.com>
	<loom.20130725T171350-173@post.gmane.org>
	<1374832232027-4660200.post@n4.nabble.com>
	<loom.20130726T124519-81@post.gmane.org>
Message-ID: <1374844204565-4660205.post@n4.nabble.com>

Carl Eugen Hoyos:

> Is this the original recording from a camera or 
> was the sample cut somehow (in the beginning)?
>
> Carl Eugen

Yes, it's one of the original files. (The size of the other chunks are 4
Gigabytes, so I gave you the smallest.)

Regards, G.



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/mmco-unref-short-failure-error-and-freeze-playback-of-video-streams-tp4657614p4660205.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From cehoyos at ag.or.at  Fri Jul 26 15:36:02 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 26 Jul 2013 13:36:02 +0000 (UTC)
Subject: [FFmpeg-user] mmco: unref short failure error and freeze
	playback of video streams
References: <CAAyZe_H=m4kxsCbLydpFQfKCWS3gZstrdhsJM6vQvSqgG=NMdA@mail.gmail.com>
	<loom.20130228T094518-529@post.gmane.org>
	<1374741951675-4660169.post@n4.nabble.com>
	<loom.20130725T115702-217@post.gmane.org>
	<1374760937649-4660175.post@n4.nabble.com>
	<loom.20130725T171350-173@post.gmane.org>
	<1374832232027-4660200.post@n4.nabble.com>
	<loom.20130726T124519-81@post.gmane.org>
	<1374844204565-4660205.post@n4.nabble.com>
Message-ID: <loom.20130726T153448-583@post.gmane.org>

gergely <Gergely.Lukacsy <at> streamnet.hu> writes:

> Carl Eugen Hoyos:
> 
> > Is this the original recording from a camera or 
> > was the sample cut somehow (in the beginning)?
> 
> Yes, it's one of the original files.

I opened ticket #2811

> (The size of the other chunks are 4
> Gigabytes, so I gave you the smallest.)

Thank you!

Carl Eugen


From zachjacobs at gmail.com  Fri Jul 26 19:27:07 2013
From: zachjacobs at gmail.com (Zach Jacobs)
Date: Fri, 26 Jul 2013 13:27:07 -0400
Subject: [FFmpeg-user] How to set X264 options using FFmpeg libraries
In-Reply-To: <CACE3me6E9+O=d7JxzJ+C7tvaY+tYQA+pH+T34Ej56m0N-1GQxw@mail.gmail.com>
References: <CACE3me6E9+O=d7JxzJ+C7tvaY+tYQA+pH+T34Ej56m0N-1GQxw@mail.gmail.com>
Message-ID: <CACE3me7TZXLmA2s0ixTu77rhh4iPeVX8cWFoPdQ=vEzZDun+fQ@mail.gmail.com>

No one can help?  Did I not provide enough information?  I would really
like to get this issue solved.
Thanks!


On Thu, Jul 25, 2013 at 3:43 PM, Zach Jacobs <zachjacobs at gmail.com> wrote:

> Hello,
>
> I am trying to control the bit rate of a file using x264 with the crf and
> maximum bit rate parameters.  I am following the x264 encoding guide here<https://trac.ffmpeg.org/wiki/x264EncodingGuide#CRFwithmaximumbitrate>.
>  I tried the following from the command line:
>
> ffmpeg -i input.m4v -c:v libx264 -crf 20 -maxrate 400k -bufsize 1835k
> output.ts
>
> This works very well and the resulting bit rate is usually very close to
> 400k.  However when I try to do the same thing by using the FFmpeg
> libraries, the encoder seems to ignore the maxrate parameter.  Here is how
> I am setting up my CodecContext:
>
>                 AVDictionary* opt = null;
>                 FFmpegInvoke.av_dict_set(&opt, "preset", "slow", 0);
>                 FFmpegInvoke.av_dict_set(&opt, "crf", "20.0", 0);
>                 FFmpegInvoke.av_dict_set(&opt, "maxrate", "400k", 0);
>                 FFmpegInvoke.av_dict_set(&opt, "bufsize", "1835k", 0);
>
>                 videoCodec =
> FFmpegInvoke.avcodec_find_encoder(AVCodecID.AV_CODEC_ID_H264);
>
>                 if (videoCodec == null)
>                     throw new Exception("FFMPEG Encoder Error: Codec Not
> Found");
>
>                 videoCodecContext =
> FFmpegInvoke.avcodec_alloc_context3(videoCodec);
>                 videoCodecContext->width = encoderInfo.Settings.VideoWidth;
>                 videoCodecContext->height =
> encoderInfo.Settings.VideoHeight;
>
>                 videoCodecContext->pix_fmt =
> AVPixelFormat.AV_PIX_FMT_YUV420P;
>                 videoCodecContext->time_base.num = 1;
>                 videoCodecContext->time_base.den = 1000;
>
>                 videoCodecContext->codec_type =
> AVMediaType.AVMEDIA_TYPE_VIDEO;
>                 videoCodecContext->codec_id = AVCodecID.AV_CODEC_ID_H264;
>
>                 if (codecFlags != -1)
>                     videoCodecContext->flags |= codecFlags;
>
>                 if (FFmpegInvoke.avcodec_open2(videoCodecContext,
> videoCodec, &opt) < 0)
>                     throw new Exception("FFmpeg Encoder Error: could not
> open video codec");
>
> opt is empty after avcodec_open2 is called indicating all of my dictionary
> options that I tried to set were found and processed.  However, the output
> video bit rate is no where near 400k maxrate that I specified.  If I
> increase maxrate, the bit rate increases but it still isn't anywhere close
> to what it should be.  The output file is roughly 319 MB.
>
> From the command line, the video bit rate is 395k and the resulting video
> size is only 42MB.  Am I missing a setting?
>
> Thanks for your time!
>
> Zach
>



-- 
Zach Jacobs
Cell  859.620.3767

From dave at dericed.com  Fri Jul 26 19:41:40 2013
From: dave at dericed.com (Dave Rice)
Date: Fri, 26 Jul 2013 13:41:40 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F2126E.40200@gmail.com>
References: <51F2126E.40200@gmail.com>
Message-ID: <5C074109-FA42-479A-8C24-2D6F4DB90C40@dericed.com>


On Jul 26, 2013, at 2:08 AM, Andy Civil wrote:

> Short description: I'm moaning that the default output for "ntsc-dvd" is progressive, even if I give it enough information to make it interlaced.
> 
> Long description:
> 
> First, an aside on interlacing. I know that in general, people here don't like interlacing and attempt to deinterlace at the earliest opportunity. I can see a day when interlacing is pointless, when all CRT displays are on the scrap heap and no one uses DVDs any more. I used to think that interlacing was a viable alternative to compression for saving data, but I now realise that where software compression is available, the advantage is moot because compression can reduce the data rate more efficiently than the interlacing process, and although interlacing might appear to halve the amount of data required, it simply makes the datastream less compressible so you gain nothing in the end.
> 
> Having said that, we still live in a world where people use DVDs to store and share movies. And like it or not, we have to work within the constraints of what is a valid DVD-video stream. Within that constraint, a video of 30fps interlaced (60 fields per second) does indeed look better than a straight 30p because there is more temporal information. Pointing out that a 60p movie might compress to smaller than the 30i is irrelevant, because 60p is not a valid mode for DVD-video.
> 
> I am in the position of wanting to make DVD videos of material I have shot with an HD camera. After days of head-scratching, I have found that the interlacing of HD video is opposite in field order to that required by DVD. Further, the video editor that I'm using to edit my videos seems to do a poor job of scaling the interlaced content (seriously). I therefore turned to trusty FFmpeg to solve the quality problem. I got the editor to output a rather large temporary file in HD resolution and the full 60p. (The quality is OK because it seems able to de-interlace fine.) I then asked FFmpeg to convert this to DVD-video. It's a simple command:
> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-07-hdp.mp4 -target ntsc-dvd -acodec mp2 -ac 2 -b:a 128000 output-dvd.mpg
> 
> Now here's my complaint: this works fine, but it makes a 30p video, simply throwing away half of the frames. 30p is a valid mode for DVD-video, but it's not as smooth as it could be. I found that I had to specifically add the flags -flags ildct+ilme
> to make it create a proper interlaced video. (Which it does, for sure.)
> 
> Therefore the default action is not the best it can do (with a 60p source). Is that a fault, or simply a limitation of the 'preset' system (not knowing if the source video has enough frames to make interlaced or not)?


Could you try:
c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-07-hdp.mp4 -target ntsc-dvd -vf "fieldorder=bff" -acodec mp2 -ac 2 -b:a 128000 output-dvd.mpg
Dave Rice

From edward.chopuryan at gmail.com  Fri Jul 26 14:00:30 2013
From: edward.chopuryan at gmail.com (edward20)
Date: Fri, 26 Jul 2013 05:00:30 -0700 (PDT)
Subject: [FFmpeg-user] Decode H264 and save to yuv file
Message-ID: <1374840030366-4660204.post@n4.nabble.com>

Hello everyone,

I am new here and I am trying to decode video using ffmpeg c API and save as
yuv file with YUV420P pixel format.

All examples that I have seen they converting extracted frames to RGB or PPM
and then saving them to file.
But I want to save as YUV. I don't know do I have to convert frames after
decoding them or maybe after decoding the frames are already YUV frames. Any
help will be appreciated. 

Thanks in advance

Edward.



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From onemda at gmail.com  Fri Jul 26 22:00:41 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 26 Jul 2013 20:00:41 +0000
Subject: [FFmpeg-user] Decode H264 and save to yuv file
In-Reply-To: <1374840030366-4660204.post@n4.nabble.com>
References: <1374840030366-4660204.post@n4.nabble.com>
Message-ID: <CAPYw7P47t=__hBOzMLQbMo6LVJq6uqNSDvukNFn4k31-ns2YEQ@mail.gmail.com>

On 7/26/13, edward20 <edward.chopuryan at gmail.com> wrote:
> Hello everyone,
>
> I am new here and I am trying to decode video using ffmpeg c API and save
> as
> yuv file with YUV420P pixel format.
>
> All examples that I have seen they converting extracted frames to RGB or
> PPM
> and then saving them to file.
> But I want to save as YUV. I don't know do I have to convert frames after
> decoding them or maybe after decoding the frames are already YUV frames.
> Any
> help will be appreciated.

Usually videos are in yuv420 and decoder outputs it in yuv420 planar
format. So after decoding there is nothing left to do.

But perhaps you need packed yuv format?

From onemda at gmail.com  Sat Jul 27 00:54:14 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Fri, 26 Jul 2013 22:54:14 +0000
Subject: [FFmpeg-user] How to set X264 options using FFmpeg libraries
In-Reply-To: <CACE3me7TZXLmA2s0ixTu77rhh4iPeVX8cWFoPdQ=vEzZDun+fQ@mail.gmail.com>
References: <CACE3me6E9+O=d7JxzJ+C7tvaY+tYQA+pH+T34Ej56m0N-1GQxw@mail.gmail.com>
	<CACE3me7TZXLmA2s0ixTu77rhh4iPeVX8cWFoPdQ=vEzZDun+fQ@mail.gmail.com>
Message-ID: <CAPYw7P6QQkO2H11P70j_1E4EHCyUiiBNDRzubN7NtB4Q5SSUeg@mail.gmail.com>

On 7/26/13, Zach Jacobs <zachjacobs at gmail.com> wrote:
> No one can help?  Did I not provide enough information?  I would really
> like to get this issue solved.

x264-params

This is mentioned in documentation.

From andycivil at gmail.com  Sat Jul 27 08:33:58 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 02:33:58 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130726T104114-810@post.gmane.org>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
Message-ID: <51F369D6.8090704@gmail.com>

On 2013-07-26 4:51 AM, Carl Eugen Hoyos wrote:
> Andy Civil<andycivil<at>  gmail.com>  writes:
>
>> Short description: I'm moaning that the default output
>> for "ntsc-dvd" is progressive, even if I give it
>> enough information to make it interlaced.
>
>> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-07-hdp.mp4
>> -target ntsc-dvd -acodec mp2 -ac 2 -b:a 128000
>> output-dvd.mpg
>
> (Complete, uncut console output missing.)

(Deliberate. Analysis unnecessary distraction.)

> Nothing about this command specifies that your input
> is interlaced so why on earth should FFmpeg use
> interlaced encoding (that of course hurts
> compression)?

Because I went to the trouble of supplying it with 60fps so that it could.

> Or in other words: You did not give enough information
> to FFmpeg that you want interlaced content, so how
> should it guess?

Because DVDs are normally interlaced, interlaced video is better*, and I gave it 
enough input information to produce interlaced.

(* not normally, of course, but within the constraints of what is a valid 
DVD-video stream.)

> I use a CRT (all the time) and I certainly don't
> want interlaced content.

Ha ha - touch?! I mean that the days when interlacing was a viable alternative 
to compression for data rate reduction are over, and the only need we have for 
interlacing now are the embarrassing hold-overs from those days :-) like the 
DVD-video specification.

>> a video of 30fps interlaced (60 fields per second)
>> does indeed look better than a straight 30p because
>> there is more temporal information
>
> I consider this "wrong" (please consider that an
> euphemism).
> If it is course perfectly fine that you disagree
> but please don't claim that this is a universal
> truth

Well, I gave my reasoning and my experience bears it out, so forgive me sir if I 
consider it the truth until a seed of doubt can be sown. In my opinion, the only 
way that interlaced could end up looking worse, would be if the data rate was 
inadequate and the compression (harder on interlaced content because there's 
more information) couldn't keep up.

> (especially since a large percentage of the
> 60p material that can be used as input to your
> command line contains duplicated frames and
> never had more temporal information).

All the content came from my video camera - this isn't a pirated movie - and 
there are no duplicated frames at all.

>
>> I found that I had to specifically add the flags
>> -flags ildct+ilme to make it create a proper
>> interlaced video.
>
> As such this is simply not true.

Sadly, you're right. According to "gspot" the output is interlaced, but it's 
actually 30p. The flags are set, but the actual frames are progressive.

> Please note that FFmpeg supports producing
> interlaced content with the appropriate filter.

I will follow up with another post asking for hints as to the form of the 
command. For this I will definitely include your famous "uncut console output" 
so that you can see what I'm starting from.

Thank you.

--
Andy

From andycivil at gmail.com  Sat Jul 27 08:45:11 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 02:45:11 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130726T104114-810@post.gmane.org>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
Message-ID: <51F36C77.8070604@gmail.com>

On 2013-07-26 4:51 AM, Carl Eugen Hoyos wrote:

> Please note that FFmpeg supports producing
> interlaced content with the appropriate filter.

Could you please tell me how I could produce DVD-video which is interlaced? This 
is what I'm giving it:

E:\Convert Sony (+vegas) to DVD>c:\ffmpeg\bin\ffprobe -i E:\temp\2013-06-06-hdp.
mp4
ffprobe version N-54822-g07704c6 Copyright (c) 2007-2013 the FFmpeg developers
   built on Jul 19 2013 18:01:39 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. 81.101 /  3. 81.101
   libswscale      2.  4.100 /  2.  4.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'E:\temp\2013-06-06-hdp.mp4':
   Metadata:
     major_brand     : mp42
     minor_version   : 18743402
     compatible_brands: mp42isom
     creation_time   : 2013-07-21 03:19:29
   Duration: 00:10:59.78, start: 0.000000, bitrate: 10074 kb/s
     Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 12
8 kb/s
     Metadata:
       creation_time   : 2013-07-21 03:19:29
       handler_name    : Sound Media Handler
     Stream #0:1(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080
  [SAR 1:1 DAR 16:9], 9939 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 119.88 tbc
     Metadata:
       creation_time   : 2013-07-21 03:19:29
       handler_name    : Video Media Handler

E:\Convert Sony (+vegas) to DVD>

Thank you.

--
Andy

From andycivil at gmail.com  Sat Jul 27 09:09:14 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 03:09:14 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <5C074109-FA42-479A-8C24-2D6F4DB90C40@dericed.com>
References: <51F2126E.40200@gmail.com>
	<5C074109-FA42-479A-8C24-2D6F4DB90C40@dericed.com>
Message-ID: <51F3721A.7040500@gmail.com>

On 2013-07-26 1:41 PM, Dave Rice wrote:

> Could you try:
> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-07-hdp.mp4 -target ntsc-dvd -vf "fieldorder=bff" -acodec mp2 -ac 2 -b:a 128000 output-dvd.mpg
> Dave Rice

It 'works', but the output is progressive (30p)

E:\Convert Sony (+vegas) to DVD>c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.m
p4 -target ntsc-dvd -vf "fieldorder=bff" -acodec mp2 -ac 2 -b:a 128000 -frames:v
  1000 2013-06-06-i.mpg
ffmpeg version N-54822-g07704c6 Copyright (c) 2000-2013 the FFmpeg developers
   built on Jul 19 2013 18:01:39 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. 81.101 /  3. 81.101
   libswscale      2.  4.100 /  2.  4.100
   libswresample   0. 17.102 /  0. 17.102
   libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'E:\temp\2013-06-06-hdp.mp4':
   Metadata:
     major_brand     : mp42
     minor_version   : 18743402
     compatible_brands: mp42isom
     creation_time   : 2013-07-21 03:19:29
   Duration: 00:10:59.78, start: 0.000000, bitrate: 10074 kb/s
     Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 12
8 kb/s
     Metadata:
       creation_time   : 2013-07-21 03:19:29
       handler_name    : Sound Media Handler
     Stream #0:1(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080
  [SAR 1:1 DAR 16:9], 9939 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 119.88 tbc
     Metadata:
       creation_time   : 2013-07-21 03:19:29
       handler_name    : Video Media Handler
File '2013-06-06-i.mpg' already exists. Overwrite ? [y/N] y
Output #0, dvd, to '2013-06-06-i.mpg':
   Metadata:
     major_brand     : mp42
     minor_version   : 18743402
     compatible_brands: mp42isom
     encoder         : Lavf55.12.102
     Stream #0:0(eng): Video: mpeg2video, yuv420p, 720x480 [SAR 32:27 DAR 16:9],
q=2-31, 6000 kb/s, 90k tbn, 29.97 tbc
     Metadata:
       creation_time   : 2013-07-21 03:19:29
       handler_name    : Video Media Handler
     Stream #0:1(eng): Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s
     Metadata:
       creation_time   : 2013-07-21 03:19:29
       handler_name    : Sound Media Handler
Stream mapping:
   Stream #0:1 -> #0:0 (h264 -> mpeg2video)
   Stream #0:0 -> #0:1 (aac -> mp2)
Press [q] to stop, [?] for help
frame=   12 fps=0.0 q=2.0 size=     122kB time=00:00:00.33 bitrate=2995.3kbits/s
frame=   22 fps= 22 q=2.0 size=     544kB time=00:00:00.66 bitrate=6678.0kbits/s
frame=   31 fps= 20 q=2.0 size=     758kB time=00:00:00.99 bitrate=6222.1kbits/s
frame=   40 fps= 20 q=2.0 size=     796kB time=00:00:01.26 bitrate=5142.9kbits/s

--
Andy

From cehoyos at ag.or.at  Sat Jul 27 10:45:22 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 27 Jul 2013 08:45:22 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?encoding_bff_input_to_mov_labels_the_outp?=
	=?utf-8?q?ut_as=09progressive?=
References: <8B50D84E-9161-45D5-B217-550EB5DC3516@dericed.com>
Message-ID: <loom.20130727T104318-882@post.gmane.org>

Dave Rice <dave <at> dericed.com> writes:

> I am encoding bff v210 video to other codecs 
> (uncomrpessed, ffv1, prores, j2k) in quicktime. Also 
> the input is identified as bff the output is identified 
> as progressive unless I force it with -vf setfield=bff.

I just sent a patch, could you test it?

Thank you for the report, Carl Eugen


From cehoyos at ag.or.at  Sat Jul 27 10:50:30 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 27 Jul 2013 08:50:30 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
Message-ID: <loom.20130727T104545-441@post.gmane.org>

Andy Civil <andycivil <at> gmail.com> writes:

> Because DVDs are normally interlaced

This obviously depends on the definition of 
"normally" but I don't think it is true.
(And as explained it would be very bad 
because people working with video see the 
artefacts on every display.)

[...]

> > Please note that FFmpeg supports producing
> > interlaced content with the appropriate filter.

See http://ffmpeg.org/ffmpeg-filters.html#tinterlace
I suspect that you still have to specify the 
encoding flags, this could indeed be seen as a bug 
but it is not what you reported originally.

Carl Eugen


From bobm-ffmpeg at burner.com  Sat Jul 27 11:12:18 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Sat, 27 Jul 2013 03:12:18 -0600
Subject: [FFmpeg-user] uyvy422 psuedo 10-bit?
Message-ID: <51F38EF2.9090207@burner.com>

I'm doing some playing around with Quicktimes using Avid's codecs of
which it has a couple;  "Avid 1:1" which is 8 bit, and "Avid Packed"
which is 10-bit.

ffmpeg is able to read both of these, but just identifies the pixel
format for both as "uyvy422", which -pix_fmts shows as 16bpp, yet it
reads the 10-bit version as well... though I'd expect to see a different
pixel format at 20bpp.

Any insight/explanation as to what is happening in this case?  Is it
actually reading the 10-bit and scaling it immediately to 8-bit to shove
into the uyvy422 pixel format?  Just trying to understand what is
happening internally.


ffprobe from the 'Avid 1:1' (8-bit) source:
------------------------------------

D:\>d:\ffmpeg\ffprobe.exe Avid-11-709.mov
ffprobe version N-54960-gf3f4e13 Copyright (c) 2007-2013 the FFmpeg
developers
  built on Jul 26 2013 19:21:01 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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Avid-11-709.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2013-07-27 09:47:35
  Duration: 00:00:01.00, start: 0.000000, bitrate: 796692 kb/s
    Stream #0:0(eng): Video: rawvideo (AV1x / 0x78315641), uyvy422,
1920x1080, 796645 kb/s, SAR 1920:1920 DAR 16:9, 23.98 fps, 23.98 tbr,
24k tbn, 24k tbc
    Metadata:
      creation_time   : 2013-07-27 09:47:35
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
    Stream #0:1(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
    Metadata:
      creation_time   : 2013-07-27 09:47:36
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
Unsupported codec with id 0 for input stream 1


ffprobe from the 'Avid Packed' source:
--------------------------------------

D:\>d:\ffmpeg\ffprobe.exe AvidPacked-709.mov
ffprobe version N-54960-gf3f4e13 Copyright (c) 2007-2013 the FFmpeg
developers
  built on Jul 26 2013 19:21:01 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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'AvidPacked-709.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2013-07-27 09:34:50
  Duration: 00:00:01.00, start: 0.000000, bitrate: 994380 kb/s
    Stream #0:0(eng): Video: rawvideo (AVup / 0x70755641), uyvy422,
1920x1080, 994333 kb/s, SAR 1920:1920 DAR 16:9, 23.98 fps, 23.98 tbr,
24k tbn, 24k tbc
    Metadata:
      creation_time   : 2013-07-27 09:34:50
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
    Stream #0:1(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
    Metadata:
      creation_time   : 2013-07-27 09:34:52
      handler_name    : Apple Alias Data Handler
      timecode        : 00:00:00:00
Unsupported codec with id 0 for input stream 1

From cehoyos at ag.or.at  Sat Jul 27 12:17:47 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 27 Jul 2013 10:17:47 +0000 (UTC)
Subject: [FFmpeg-user] uyvy422 psuedo 10-bit?
References: <51F38EF2.9090207@burner.com>
Message-ID: <loom.20130727T121406-38@post.gmane.org>

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

> I'm doing some playing around with Quicktimes using 
> Avid's codecs of which it has a couple;  "Avid 1:1" 
> which is 8 bit, and "Avid Packed" which is 10-bit.
> 
> ffmpeg is able to read both of these, but just 
> identifies the pixel format for both as "uyvy422", 
> which -pix_fmts shows as 16bpp, yet it reads the 10-bit 
> version as well... though I'd expect to see a different
> pixel format at 20bpp.

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

Unrelated to your question: Are you sure that the samples 
are shown visually correct? I always wondered if 
there are fields that have to handled specifically.

Carl Eugen


From francois.visagie at gmail.com  Sat Jul 27 13:17:25 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Sat, 27 Jul 2013 13:17:25 +0200
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130727T104545-441@post.gmane.org>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
Message-ID: <000001ce8aba$e2ea0220$a8be0660$@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: 27 July 2013 10:51
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
> 
> Andy Civil <andycivil <at> gmail.com> writes:
> 
> > Because DVDs are normally interlaced
> 
> This obviously depends on the definition of "normally" but I don't think
it is
> true.
> (And as explained it would be very bad
> because people working with video see the artefacts on every display.)
> 
> [...]
> 
> > > Please note that FFmpeg supports producing interlaced content with
> > > the appropriate filter.
> 
> See http://ffmpeg.org/ffmpeg-filters.html#tinterlace

That's very useful information, thanks.

> I suspect that you still have to specify the encoding flags, this could
indeed be
> seen as a bug but it is not what you reported originally.
> 
> Carl Eugen
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From werner.robitza at gmail.com  Sat Jul 27 14:12:01 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Sat, 27 Jul 2013 14:12:01 +0200
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
Message-ID: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>

As a frequent troubleshooter or non-working ffmpeg commands, it has
occurred to me countless times that users ended up with videos that were,
for no obvious reason, "not playable".

The reason for those broken videos was that the original color space of the
input video was RGB (e.g. when piping PNG images or uncompressed video).
Using x264 without any further options would result in the video being
encoded as YUV, but without chroma subsampling, i.e. yuv444p, which most
players cannot handle.

The fix for these problems is to add -pix_fmt yuv420p, but it's hard to
spot this as a new (or unexperienced) user.

Wouldn't it make sense, when converting from RGB, to automatically use
yuv420p?
Or are there any good reasons against doing so? I would assume that users
who know they need non-subsampled video also know how to force the
conversion.

Werner

From onemda at gmail.com  Sat Jul 27 14:46:43 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 12:46:43 +0000
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
Message-ID: <CAPYw7P5EbF+-+0H31OdskHaJf7gTde_eTAtVGe5MVS6ER9j7pg@mail.gmail.com>

On 7/27/13, Werner Robitza <werner.robitza at gmail.com> wrote:
> As a frequent troubleshooter or non-working ffmpeg commands, it has
> occurred to me countless times that users ended up with videos that were,
> for no obvious reason, "not playable".
>
> The reason for those broken videos was that the original color space of the
> input video was RGB (e.g. when piping PNG images or uncompressed video).
> Using x264 without any further options would result in the video being
> encoded as YUV, but without chroma subsampling, i.e. yuv444p, which most
> players cannot handle.
>
> The fix for these problems is to add -pix_fmt yuv420p, but it's hard to
> spot this as a new (or unexperienced) user.
>
> Wouldn't it make sense, when converting from RGB, to automatically use
> yuv420p?

That is bad idea, you are forcing users that want to keep quality
to add -pix_fmt yuv444p

Instead this message was added in code and is displayed:

No pixel format specified, yuv444p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.

Perhaps it should be in BIG _R_E_D_ blinking letters with popup window?

> Or are there any good reasons against doing so? I would assume that users
> who know they need non-subsampled video also know how to force the
> conversion.
>
> Werner
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From werner.robitza at gmail.com  Sat Jul 27 17:06:10 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Sat, 27 Jul 2013 17:06:10 +0200
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <CAPYw7P5EbF+-+0H31OdskHaJf7gTde_eTAtVGe5MVS6ER9j7pg@mail.gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<CAPYw7P5EbF+-+0H31OdskHaJf7gTde_eTAtVGe5MVS6ER9j7pg@mail.gmail.com>
Message-ID: <CAN0LTaMC50L0rXV2SbH9YtgL_HoPaaoi6pcvR57R9=VJwNNZBw@mail.gmail.com>

> Instead this message was added in code and is displayed:
>
> No pixel format specified, yuv444p for H.264 encoding chosen.
> Use -pix_fmt yuv420p for compatibility with outdated media players.
>
> Perhaps it should be in BIG _R_E_D_ blinking letters with popup window?
>

Did you really just implement that or is it already there? Because I don't
see it in a recent-ish post-2.0 build.
I don't think it needs to be anything more than a warning, because the
conversion will work regardless.

By the way, I wouldn't go as far to say "outdated media players" ? can you
name a few players that support it?

From stefasab at gmail.com  Sat Jul 27 17:32:22 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Sat, 27 Jul 2013 17:32:22 +0200
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <CAN0LTaMC50L0rXV2SbH9YtgL_HoPaaoi6pcvR57R9=VJwNNZBw@mail.gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<CAPYw7P5EbF+-+0H31OdskHaJf7gTde_eTAtVGe5MVS6ER9j7pg@mail.gmail.com>
	<CAN0LTaMC50L0rXV2SbH9YtgL_HoPaaoi6pcvR57R9=VJwNNZBw@mail.gmail.com>
Message-ID: <20130727153222.GE13126@barisone>

On date Saturday 2013-07-27 17:06:10 +0200, Werner Robitza wrote:
> > Instead this message was added in code and is displayed:
> >
> > No pixel format specified, yuv444p for H.264 encoding chosen.
> > Use -pix_fmt yuv420p for compatibility with outdated media players.
> >
> > Perhaps it should be in BIG _R_E_D_ blinking letters with popup window?
> >
> 
> Did you really just implement that or is it already there? Because I don't
> see it in a recent-ish post-2.0 build.

Try for example:
ffmpeg -f lavfi -i testsrc=d=10,format=rgb24 -c:v libx264 -y out.mp4

> I don't think it needs to be anything more than a warning, because the
> conversion will work regardless.

It's INFO level, and thus it's not highlighted in yellow, we could
raise the message to the WARNING level.

> By the way, I wouldn't go as far to say "outdated media players" ? can you
> name a few players that support it?

ffplay, VLC, MPlayer, in general all up-to-date FFmpeg-based players
(that is most Linux/FLOSS media players) just to make an example.
-- 
FFmpeg = Fast and Fantastic Meaningless Portable Everlasting Geek

From jpboard2 at yahoo.com  Sat Jul 27 18:00:56 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 27 Jul 2013 09:00:56 -0700 (PDT)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a 1080p 60
	Hz Computer Monitor
Message-ID: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>

I have video I captured from a DVR at 29.97 fps.? The video was telecined.
I inverse telecined it and reduced the fps to 24000/1001 (about 24 fps).? I'm playing
the video on my LCD TV screen via the HDMI out connector of my video card.
My TV says the says this connection displays video at 1080p 60 Hz.


Now I'm wondering if the inverse telecine step makes any sense for me.? If I converted
my video to 24 fps, and the TV displays 1080p at 60 Hz, what is actually happening?
I assume the video card is sending 60 full frames to the TV every second.? But if the
video has only 24 frames per second, how does it do this?? Does the video card
telecine it?? Does the TV telecine it.? If so, does it even make sense for me to
inverse telecine the video?? Can someone make sense of this for me?

From werner.robitza at gmail.com  Sat Jul 27 18:05:08 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Sat, 27 Jul 2013 18:05:08 +0200
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <20130727153222.GE13126@barisone>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<CAPYw7P5EbF+-+0H31OdskHaJf7gTde_eTAtVGe5MVS6ER9j7pg@mail.gmail.com>
	<CAN0LTaMC50L0rXV2SbH9YtgL_HoPaaoi6pcvR57R9=VJwNNZBw@mail.gmail.com>
	<20130727153222.GE13126@barisone>
Message-ID: <CAN0LTaMfUxo03V7EPZCyMHGQPksf8QA_DEy4avB2tO9nJ3=CgA@mail.gmail.com>

>
> It's INFO level, and thus it's not highlighted in yellow, we could
> raise the message to the WARNING level.
>

At least to me that would make sense, given the number of times I've helped
users troubleshoot that particular problem.


> > By the way, I wouldn't go as far to say "outdated media players" ? can
> you
> > name a few players that support it?
>
> ffplay, VLC, MPlayer, in general all up-to-date FFmpeg-based players
> (that is most Linux/FLOSS media players) just to make an example.
>

Latest stable VLC 2.0.7 on OS X simply crashes with a seg fault for all
yuv444p videos I generated with x264 and recent ffmpeg, but that's probably
for the VLC people to investigate.
Thanks for the info though! That clears up a few points.

From dave at dericed.com  Sat Jul 27 18:28:25 2013
From: dave at dericed.com (Dave Rice)
Date: Sat, 27 Jul 2013 12:28:25 -0400
Subject: [FFmpeg-user] encoding bff input to mov labels the output
	as	progressive
In-Reply-To: <loom.20130727T104318-882@post.gmane.org>
References: <8B50D84E-9161-45D5-B217-550EB5DC3516@dericed.com>
	<loom.20130727T104318-882@post.gmane.org>
Message-ID: <EE1B16F9-8A1F-4720-91FF-F665AADFF0BD@dericed.com>


On Jul 27, 2013, at 4:45 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Dave Rice <dave <at> dericed.com> writes:
> 
>> I am encoding bff v210 video to other codecs 
>> (uncomrpessed, ffv1, prores, j2k) in quicktime. Also 
>> the input is identified as bff the output is identified 
>> as progressive unless I force it with -vf setfield=bff.
> 
> I just sent a patch, could you test it?

That worked for me. Output is now bff without needing to be forced. Thanks!
Dave Rice

From spirian at qq.com  Sat Jul 27 04:03:52 2013
From: spirian at qq.com (=?ISO-8859-1?B?U3Bpcmlhbg==?=)
Date: Sat, 27 Jul 2013 10:03:52 +0800
Subject: [FFmpeg-user] Decode H264 and save to yuv file
Message-ID: <tencent_05C78B781D6F668440CD3187@qq.com>

Normally, Decode means converting an codec format to .yuv or rgb24 format.
for ffmpeg : ffmepg -i x1.avi x2.yuv  x1=input x2=output   




------------------ Original ------------------
From:  "Paul B Mahol"<onemda at gmail.com>;
Date:  Sat, Jul 27, 2013 04:00 AM
To:  "FFmpeg user questions"<ffmpeg-user at ffmpeg.org>; 

Subject:  Re: [FFmpeg-user] Decode H264 and save to yuv file



On 7/26/13, edward20 <edward.chopuryan at gmail.com> wrote:
> Hello everyone,
>
> I am new here and I am trying to decode video using ffmpeg c API and save
> as
> yuv file with YUV420P pixel format.
>
> All examples that I have seen they converting extracted frames to RGB or
> PPM
> and then saving them to file.
> But I want to save as YUV. I don't know do I have to convert frames after
> decoding them or maybe after decoding the frames are already YUV frames.
> Any
> help will be appreciated.

Usually videos are in yuv420 and decoder outputs it in yuv420 planar
format. So after decoding there is nothing left to do.

But perhaps you need packed yuv format?
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
.

From jpboard2 at yahoo.com  Sat Jul 27 19:19:16 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 27 Jul 2013 10:19:16 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
Message-ID: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>

With ffmpeg, I generate an md5 file with the following command:

??? ffmpeg -i IN.avi -mdf OUT.md5

The OUT.md5 file contains checksums for both the video and audio packets.? The
video checksums makes sense (to me): there is one for each frame in the input (the
AVI file stores each frame in full, so there's no inter-frame compression).
However, there's about twice as many audio packets as video packets.? Why is
that?? How can I tell which audio packet corresponds to a video frame?? How
are the audio packets created?

From cehoyos at ag.or.at  Sat Jul 27 19:22:20 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 27 Jul 2013 17:22:20 +0000 (UTC)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <loom.20130727T192056-984@post.gmane.org>

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

> With ffmpeg, I generate an md5 file with the 
> following command:
> 
> ??? ffmpeg -i IN.avi -mdf OUT.md5

...
Unrecognized option 'mdf'.

> How can I tell which audio packet corresponds 
> to a video frame?

I may misunderstand but I don't think that audio 
packets generally "correspond" to video frames.

Carl Eugen


From cehoyos at ag.or.at  Sat Jul 27 19:27:41 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 27 Jul 2013 17:27:41 +0000 (UTC)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a 1080p
	60 Hz Computer Monitor
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <loom.20130727T192450-485@post.gmane.org>

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

> Now I'm wondering if the inverse telecine step 
> makes any sense for me.

Only you can answer if you saw ugly combing artefacts 
before inverse telecine that disappeared after.

Carl Eugen


From jpboard2 at yahoo.com  Sat Jul 27 19:48:04 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 27 Jul 2013 10:48:04 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <loom.20130727T192056-984@post.gmane.org>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<loom.20130727T192056-984@post.gmane.org>
Message-ID: <1374947284.89290.YahooMailNeo@web164703.mail.gq1.yahoo.com>

>Unrecognized option 'mdf'.

>
>> How can I tell which audio packet corresponds 
>> to a video frame?
>
>I may misunderstand but I don't think that audio 
>packets generally "correspond" to video frames.
>
>Carl Eugen

Sorry, I should have typed '-f framemd5'.? That's generates a file containing the
audio and video checksums.

What I'd like to know is how the audio packets work.? They obviously don't
correspond to video frames in a 1-to-1 fashion, since there are more audio
packets then video packets.? And since the number of audio packets isn't
an integer multiple of the number of frames, then they can't correspond
to video packets in an n-to-1 fashion.? However, they must corresponds ]
to frames in some way.? Does that make any sense?

From jpboard2 at yahoo.com  Sat Jul 27 19:53:09 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 27 Jul 2013 10:53:09 -0700 (PDT)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a
	1080p	60 Hz Computer Monitor
In-Reply-To: <loom.20130727T192450-485@post.gmane.org>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<loom.20130727T192450-485@post.gmane.org>
Message-ID: <1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>

>Only you can answer if you saw ugly combing artefacts 

>before inverse telecine that disappeared after.

Yes, there were ugly combing artifacts that did disappear when
I inverse telecined it.

I want to know if will I be introducing other artifacts due to
the fact that the inverse telecined video is 24 fps, while the display
is 1080p at 60 Hz.? For example, will motion now be less smooth?

How does it work?? How does a video which has 24 fps get displayed on
a TV with displays at 60 fps?? Does it duplicate frames?? Or does it
telecine it?? Something else?? That's the question.

From seanm at goldeneraproductions.org  Sat Jul 27 20:17:28 2013
From: seanm at goldeneraproductions.org (Sean McShane)
Date: Sat, 27 Jul 2013 11:17:28 -0700
Subject: [FFmpeg-user] vc1 and wmv3 encode capability
Message-ID: <003d01ce8af5$8d6f8640$a84e92c0$@org>

Is there any plan to add encode capabilities in addition to the existing
decode capabilities for vc1 and wmv3 codecs in ffmpeg?

 

-Sean

 




From nicolas.george at normalesup.org  Sat Jul 27 20:33:14 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sat, 27 Jul 2013 20:33:14 +0200
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <20130727183314.GA27349@phare.normalesup.org>

Le nonidi 9 thermidor, an CCXXI, James Board a ?crit?:
> The video checksums makes sense (to me): there is one for each frame in
> (the AVI file stores each frame in full, so there's no inter-frame
> compression).

This is not true.

> However, there's about twice as many audio packets as video packets. Why
> is that?

Most audio codecs have a specific frame size (for specified settings): a
number of samples they encode at once. If the input file uses one such
codec, then you get the corresponding frames. Otherwise, you get whatever
frames the tool that produced the AVI decided to make, maybe corresponding
to a sound device buffer size or something.

Regards,

-- 
  Nicolas George
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From onemda at gmail.com  Sat Jul 27 21:06:28 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 19:06:28 +0000
Subject: [FFmpeg-user] vc1 and wmv3 encode capability
In-Reply-To: <003d01ce8af5$8d6f8640$a84e92c0$@org>
References: <003d01ce8af5$8d6f8640$a84e92c0$@org>
Message-ID: <CAPYw7P6pnk+zY0CL=ikcgt3ecigBx6Db+1nmBPQtduCKzvG=VQ@mail.gmail.com>

On 7/27/13, Sean McShane <seanm at goldeneraproductions.org> wrote:
> Is there any plan to add encode capabilities in addition to the existing
> decode capabilities for vc1 and wmv3 codecs in ffmpeg?

There are no such plans.

From nicolas.george at normalesup.org  Sat Jul 27 20:33:14 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sat, 27 Jul 2013 20:33:14 +0200
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <20130727183314.GA27349@phare.normalesup.org>

Le nonidi 9 thermidor, an CCXXI, James Board a ?crit?:
> The video checksums makes sense (to me): there is one for each frame in
> (the AVI file stores each frame in full, so there's no inter-frame
> compression).

This is not true.

> However, there's about twice as many audio packets as video packets. Why
> is that?

Most audio codecs have a specific frame size (for specified settings): a
number of samples they encode at once. If the input file uses one such
codec, then you get the corresponding frames. Otherwise, you get whatever
frames the tool that produced the AVI decided to make, maybe corresponding
to a sound device buffer size or something.

Regards,

-- 
  Nicolas George
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http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From seanm at goldeneraproductions.org  Sat Jul 27 20:17:28 2013
From: seanm at goldeneraproductions.org (Sean McShane)
Date: Sat, 27 Jul 2013 11:17:28 -0700
Subject: [FFmpeg-user]  vc1 and wmv3 encode capability
Message-ID: <003d01ce8af5$8d6f8640$a84e92c0$@org>

Is there any plan to add encode capabilities in addition to the existing
decode capabilities for vc1 and wmv3 codecs in ffmpeg?

 

-Sean

 



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



From onemda at gmail.com  Sat Jul 27 21:06:28 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 19:06:28 +0000
Subject: [FFmpeg-user] vc1 and wmv3 encode capability
In-Reply-To: <003d01ce8af5$8d6f8640$a84e92c0$@org>
References: <003d01ce8af5$8d6f8640$a84e92c0$@org>
Message-ID: <CAPYw7P6pnk+zY0CL=ikcgt3ecigBx6Db+1nmBPQtduCKzvG=VQ@mail.gmail.com>

On 7/27/13, Sean McShane <seanm at goldeneraproductions.org> wrote:
> Is there any plan to add encode capabilities in addition to the existing
> decode capabilities for vc1 and wmv3 codecs in ffmpeg?

There are no such plans.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user



From nicolas.george at normalesup.org  Sat Jul 27 20:33:14 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sat, 27 Jul 2013 20:33:14 +0200
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <20130727183314.GA27349@phare.normalesup.org>

Le nonidi 9 thermidor, an CCXXI, James Board a ?crit?:
> The video checksums makes sense (to me): there is one for each frame in
> (the AVI file stores each frame in full, so there's no inter-frame
> compression).

This is not true.

> However, there's about twice as many audio packets as video packets. Why
> is that?

Most audio codecs have a specific frame size (for specified settings): a
number of samples they encode at once. If the input file uses one such
codec, then you get the corresponding frames. Otherwise, you get whatever
frames the tool that produced the AVI decided to make, maybe corresponding
to a sound device buffer size or something.

Regards,

-- 
  Nicolas George
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_______________________________________________
ffmpeg-user mailing list
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http://ffmpeg.org/mailman/listinfo/ffmpeg-user
-------------- next part --------------
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From nicolas.george at normalesup.org  Sat Jul 27 20:33:14 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sat, 27 Jul 2013 20:33:14 +0200
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <20130727183314.GA27349@phare.normalesup.org>

Le nonidi 9 thermidor, an CCXXI, James Board a ?crit?:
> The video checksums makes sense (to me): there is one for each frame in
> (the AVI file stores each frame in full, so there's no inter-frame
> compression).

This is not true.

> However, there's about twice as many audio packets as video packets. Why
> is that?

Most audio codecs have a specific frame size (for specified settings): a
number of samples they encode at once. If the input file uses one such
codec, then you get the corresponding frames. Otherwise, you get whatever
frames the tool that produced the AVI decided to make, maybe corresponding
to a sound device buffer size or something.

Regards,

-- 
  Nicolas George
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_______________________________________________
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http://ffmpeg.org/mailman/listinfo/ffmpeg-user
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ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
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From onemda at gmail.com  Sat Jul 27 21:06:28 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 19:06:28 +0000
Subject: [FFmpeg-user] vc1 and wmv3 encode capability
In-Reply-To: <003d01ce8af5$8d6f8640$a84e92c0$@org>
References: <003d01ce8af5$8d6f8640$a84e92c0$@org>
Message-ID: <CAPYw7P6pnk+zY0CL=ikcgt3ecigBx6Db+1nmBPQtduCKzvG=VQ@mail.gmail.com>

On 7/27/13, Sean McShane <seanm at goldeneraproductions.org> wrote:
> Is there any plan to add encode capabilities in addition to the existing
> decode capabilities for vc1 and wmv3 codecs in ffmpeg?

There are no such plans.
_______________________________________________
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 seanm at goldeneraproductions.org  Sat Jul 27 20:17:28 2013
From: seanm at goldeneraproductions.org (Sean McShane)
Date: Sat, 27 Jul 2013 11:17:28 -0700
Subject: [FFmpeg-user]   vc1 and wmv3 encode capability
Message-ID: <003d01ce8af5$8d6f8640$a84e92c0$@org>

Is there any plan to add encode capabilities in addition to the existing
decode capabilities for vc1 and wmv3 codecs in ffmpeg?

 

-Sean

 



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From seanm at goldeneraproductions.org  Sat Jul 27 20:17:28 2013
From: seanm at goldeneraproductions.org (Sean McShane)
Date: Sat, 27 Jul 2013 11:17:28 -0700
Subject: [FFmpeg-user]    vc1 and wmv3 encode capability
Message-ID: <003d01ce8af5$8d6f8640$a84e92c0$@org>

Is there any plan to add encode capabilities in addition to the existing
decode capabilities for vc1 and wmv3 codecs in ffmpeg?

 

-Sean

 



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


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From onemda at gmail.com  Sat Jul 27 21:06:28 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 19:06:28 +0000
Subject: [FFmpeg-user] vc1 and wmv3 encode capability
In-Reply-To: <003d01ce8af5$8d6f8640$a84e92c0$@org>
References: <003d01ce8af5$8d6f8640$a84e92c0$@org>
Message-ID: <CAPYw7P6pnk+zY0CL=ikcgt3ecigBx6Db+1nmBPQtduCKzvG=VQ@mail.gmail.com>

On 7/27/13, Sean McShane <seanm at goldeneraproductions.org> wrote:
> Is there any plan to add encode capabilities in addition to the existing
> decode capabilities for vc1 and wmv3 codecs in ffmpeg?

There are no such plans.
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ffmpeg-user at ffmpeg.org
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From nicolas.george at normalesup.org  Sat Jul 27 20:33:14 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Sat, 27 Jul 2013 20:33:14 +0200
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <20130727183314.GA27349@phare.normalesup.org>

Le nonidi 9 thermidor, an CCXXI, James Board a ?crit?:
> The video checksums makes sense (to me): there is one for each frame in
> (the AVI file stores each frame in full, so there's no inter-frame
> compression).

This is not true.

> However, there's about twice as many audio packets as video packets. Why
> is that?

Most audio codecs have a specific frame size (for specified settings): a
number of samples they encode at once. If the input file uses one such
codec, then you get the corresponding frames. Otherwise, you get whatever
frames the tool that produced the AVI decided to make, maybe corresponding
to a sound device buffer size or something.

Regards,

-- 
  Nicolas George
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From seanm at goldeneraproductions.org  Sat Jul 27 20:17:28 2013
From: seanm at goldeneraproductions.org (Sean McShane)
Date: Sat, 27 Jul 2013 11:17:28 -0700
Subject: [FFmpeg-user]     vc1 and wmv3 encode capability
Message-ID: <003d01ce8af5$8d6f8640$a84e92c0$@org>

Is there any plan to add encode capabilities in addition to the existing
decode capabilities for vc1 and wmv3 codecs in ffmpeg?

 

-Sean

 



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


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


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


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



From onemda at gmail.com  Sat Jul 27 21:06:28 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 19:06:28 +0000
Subject: [FFmpeg-user] vc1 and wmv3 encode capability
In-Reply-To: <003d01ce8af5$8d6f8640$a84e92c0$@org>
References: <003d01ce8af5$8d6f8640$a84e92c0$@org>
Message-ID: <CAPYw7P6pnk+zY0CL=ikcgt3ecigBx6Db+1nmBPQtduCKzvG=VQ@mail.gmail.com>

On 7/27/13, Sean McShane <seanm at goldeneraproductions.org> wrote:
> Is there any plan to add encode capabilities in addition to the existing
> decode capabilities for vc1 and wmv3 codecs in ffmpeg?

There are no such plans.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user


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


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From yugiohjcj-mailinglist at laposte.net  Sat Jul 27 21:34:14 2013
From: yugiohjcj-mailinglist at laposte.net (YuGiOhJCJ Mailing-List)
Date: Sat, 27 Jul 2013 21:34:14 +0200
Subject: [FFmpeg-user] recording audio from speakers
Message-ID: <20130727213414.64cf90feddcafdcca6d883be@laposte.net>

Hello,

I am recording my desktop with ffmpeg 1.2.
The problem is I am recording audio from microphone whereas I would like to record audio from speakers.

This is my command line:
$ ffmpeg -f alsa -ac 2 -i hw:0,0 -f x11grab -s 640x480 -r 25 -i :0.0 /tmp/out.mpg

This is information you may need to help me:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: VT1708S Analog [VT1708S Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 2: VT1708S HP [VT1708S HP]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 7: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 8: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 9: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

$ aplay -L
null
    Discard all samples (playback) or generate zero samples (capture)
default:CARD=PCH
    HDA Intel PCH, VT1708S Analog
    Default Audio Device
sysdefault:CARD=PCH
    HDA Intel PCH, VT1708S Analog
    Default Audio Device
front:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    Front speakers
surround40:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
hdmi:CARD=NVidia,DEV=0
    HDA NVidia, HDMI 0
    HDMI Audio Output
hdmi:CARD=NVidia,DEV=1
    HDA NVidia, HDMI 0
    HDMI Audio Output
hdmi:CARD=NVidia,DEV=2
    HDA NVidia, HDMI 0
    HDMI Audio Output
hdmi:CARD=NVidia,DEV=3
    HDA NVidia, HDMI 0
    HDMI Audio Output
$ amixer 
Simple mixer control 'Master',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 42
  Mono: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Headphone',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 42
  Mono:
  Front Left: Playback 42 [100%] [0.00dB] [on]
  Front Right: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [on]
  Front Right: Playback 31 [100%] [12.00dB] [on]
Simple mixer control 'Front',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 42
  Mono:
  Front Left: Playback 42 [100%] [0.00dB] [on]
  Front Right: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Front Mic',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [on]
  Front Right: Playback 31 [100%] [12.00dB] [on]
Simple mixer control 'Front Mic Boost',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 3
  Front Left: 3 [100%] [30.75dB]
  Front Right: 3 [100%] [30.75dB]
Simple mixer control 'Surround',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 42
  Mono:
  Front Left: Playback 42 [100%] [0.00dB] [on]
  Front Right: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Center',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 42
  Mono: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'LFE',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 42
  Mono: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Line',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [on]
  Front Right: Playback 31 [100%] [12.00dB] [on]
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 31
  Front Left: Capture 31 [100%] [30.00dB] [on]
  Front Right: Capture 31 [100%] [30.00dB] [on]
Simple mixer control 'Capture',1
  Capabilities: cvolume cswitch
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 31
  Front Left: Capture 31 [100%] [30.00dB] [off]
  Front Right: Capture 31 [100%] [30.00dB] [off]
Simple mixer control 'Digital',0
  Capabilities: cvolume
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 120
  Front Left: Capture 55 [46%] [-2.50dB]
  Front Right: Capture 55 [46%] [-2.50dB]
Simple mixer control 'Dynamic Power-Control',0
  Capabilities: enum
  Items: 'Disabled' 'Enabled'
  Item0: 'Disabled'
Simple mixer control 'Independent HP',0
  Capabilities: enum
  Items: 'OFF' 'ON'
  Item0: 'OFF'
Simple mixer control 'Input Source',0
  Capabilities: cenum
  Items: 'Rear Mic' 'Front Mic' 'Line' 'Stereo Mixer'
  Item0: 'Front Mic'
Simple mixer control 'Loopback Mixing',0
  Capabilities: enum
  Items: 'Disabled' 'Enabled'
  Item0: 'Disabled'
Simple mixer control 'Rear Mic',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [off]
  Front Right: Playback 31 [100%] [12.00dB] [off]
Simple mixer control 'Rear Mic Boost',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 3
  Front Left: 0 [0%] [0.00dB]
  Front Right: 0 [0%] [0.00dB]
Simple mixer control 'Smart 5.1',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [off]

I am using a headset plugged on the 2 front jacks (green and red).

How can I capture my audio speakers with ffmpeg instead of recording from microphone ?

Thank you.
Best regards.

From yugiohjcj-mailinglist at laposte.net  Sat Jul 27 21:34:14 2013
From: yugiohjcj-mailinglist at laposte.net (YuGiOhJCJ Mailing-List)
Date: Sat, 27 Jul 2013 21:34:14 +0200
Subject: [FFmpeg-user]  recording audio from speakers
Message-ID: <20130727213414.64cf90feddcafdcca6d883be@laposte.net>

Hello,

I am recording my desktop with ffmpeg 1.2.
The problem is I am recording audio from microphone whereas I would like to record audio from speakers.

This is my command line:
$ ffmpeg -f alsa -ac 2 -i hw:0,0 -f x11grab -s 640x480 -r 25 -i :0.0 /tmp/out.mpg

This is information you may need to help me:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: VT1708S Analog [VT1708S Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 2: VT1708S HP [VT1708S HP]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 7: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 8: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 9: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

$ aplay -L
null
    Discard all samples (playback) or generate zero samples (capture)
default:CARD=PCH
    HDA Intel PCH, VT1708S Analog
    Default Audio Device
sysdefault:CARD=PCH
    HDA Intel PCH, VT1708S Analog
    Default Audio Device
front:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    Front speakers
surround40:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
    HDA Intel PCH, VT1708S Analog
    7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
hdmi:CARD=NVidia,DEV=0
    HDA NVidia, HDMI 0
    HDMI Audio Output
hdmi:CARD=NVidia,DEV=1
    HDA NVidia, HDMI 0
    HDMI Audio Output
hdmi:CARD=NVidia,DEV=2
    HDA NVidia, HDMI 0
    HDMI Audio Output
hdmi:CARD=NVidia,DEV=3
    HDA NVidia, HDMI 0
    HDMI Audio Output
$ amixer 
Simple mixer control 'Master',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 42
  Mono: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Headphone',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 42
  Mono:
  Front Left: Playback 42 [100%] [0.00dB] [on]
  Front Right: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'PCM',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [on]
  Front Right: Playback 31 [100%] [12.00dB] [on]
Simple mixer control 'Front',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 42
  Mono:
  Front Left: Playback 42 [100%] [0.00dB] [on]
  Front Right: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Front Mic',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [on]
  Front Right: Playback 31 [100%] [12.00dB] [on]
Simple mixer control 'Front Mic Boost',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 3
  Front Left: 3 [100%] [30.75dB]
  Front Right: 3 [100%] [30.75dB]
Simple mixer control 'Surround',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 42
  Mono:
  Front Left: Playback 42 [100%] [0.00dB] [on]
  Front Right: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Center',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 42
  Mono: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'LFE',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 42
  Mono: Playback 42 [100%] [0.00dB] [on]
Simple mixer control 'Line',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [on]
  Front Right: Playback 31 [100%] [12.00dB] [on]
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 31
  Front Left: Capture 31 [100%] [30.00dB] [on]
  Front Right: Capture 31 [100%] [30.00dB] [on]
Simple mixer control 'Capture',1
  Capabilities: cvolume cswitch
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 31
  Front Left: Capture 31 [100%] [30.00dB] [off]
  Front Right: Capture 31 [100%] [30.00dB] [off]
Simple mixer control 'Digital',0
  Capabilities: cvolume
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 120
  Front Left: Capture 55 [46%] [-2.50dB]
  Front Right: Capture 55 [46%] [-2.50dB]
Simple mixer control 'Dynamic Power-Control',0
  Capabilities: enum
  Items: 'Disabled' 'Enabled'
  Item0: 'Disabled'
Simple mixer control 'Independent HP',0
  Capabilities: enum
  Items: 'OFF' 'ON'
  Item0: 'OFF'
Simple mixer control 'Input Source',0
  Capabilities: cenum
  Items: 'Rear Mic' 'Front Mic' 'Line' 'Stereo Mixer'
  Item0: 'Front Mic'
Simple mixer control 'Loopback Mixing',0
  Capabilities: enum
  Items: 'Disabled' 'Enabled'
  Item0: 'Disabled'
Simple mixer control 'Rear Mic',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 31 [100%] [12.00dB] [off]
  Front Right: Playback 31 [100%] [12.00dB] [off]
Simple mixer control 'Rear Mic Boost',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 3
  Front Left: 0 [0%] [0.00dB]
  Front Right: 0 [0%] [0.00dB]
Simple mixer control 'Smart 5.1',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [off]

I am using a headset plugged on the 2 front jacks (green and red).

How can I capture my audio speakers with ffmpeg instead of recording from microphone ?

Thank you.
Best regards.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user



From jpboard2 at yahoo.com  Sat Jul 27 22:11:26 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sat, 27 Jul 2013 13:11:26 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <20130727183314.GA27349@phare.normalesup.org>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org> 
Message-ID: <1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>

>> The video checksums makes sense (to me): there is one for each frame in

>> (the AVI file stores each frame in full, so there's no inter-frame
>> compression).
>
>This is not true.

I think you mean it isn't true in general.? But for my example, where there's no
inter-frame compression, it is true.? I counted the frames and I counted the
checksums, and they are equal.

>Most audio
 codecs have a specific frame size (for specified settings): a
>number of samples they encode at once. If the input file uses one such

So if, for example, I wanted all of the audio samples that correspond to
a specific frame, I might have to look at multiple audio packets, and then
figure out which audio samples in each packet belong to which frame?

From onemda at gmail.com  Sat Jul 27 22:59:26 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 20:59:26 +0000
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org>
	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>

On 7/27/13, James Board <jpboard2 at yahoo.com> wrote:
>>> The video checksums makes sense (to me): there is one for each frame in
>
>>> (the AVI file stores each frame in full, so there's no inter-frame
>>> compression).
>>
>>This is not true.
>
> I think you mean it isn't true in general.  But for my example, where
> there's no
> inter-frame compression, it is true.  I counted the frames and I counted
> the
> checksums, and they are equal.
>
>>Most audio
>  codecs have a specific frame size (for specified settings): a
>>number of samples they encode at once. If the input file uses one such
>
> So if, for example, I wanted all of the audio samples that correspond to
> a specific frame, I might have to look at multiple audio packets, and then
> figure out which audio samples in each packet belong to which frame?

Something like that, note that there are containers (not AVI) that actually do
that.

From andycivil at gmail.com  Sat Jul 27 23:09:33 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 17:09:33 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130727T104545-441@post.gmane.org>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
Message-ID: <51F4370D.3050204@gmail.com>

On 2013-07-27 4:50 AM, Carl Eugen Hoyos wrote:

>
> See http://ffmpeg.org/ffmpeg-filters.html#tinterlace
> I suspect that you still have to specify the
> encoding flags, this could indeed be seen as a bug
> but it is not what you reported originally.

I'm going to need help with this. I tried adding:
-vf tinterlace=interleave_bottom
-flags ildct+ilme
and it's doing something, but it's not right.

The way I'm analysing my output, is that I view it in VLC. What should happen is 
that I see the 'comb' effect on any moving edges, but if I choose to turn 
de-interlacing on, that comb effect disappears.

With the addition of those two options, what I get is that the two fields are 
'fuzzed together' (like a 'blend' deinterlace had been used).

I also use the analysis tool "gspot" to tell me if the video is labelled as 
interlaced. It seems that where those flags are set, it is.

There's something specific that I don't understand. When I specify "format 
ntsc-DVD" that must set up a filter chain of some kind. But when I add the 
'tinterlace' filter, that's probably going to be added in the wrong place in the 
chain, isn't it?

Is there any way that I can find out exactly what "format ntsc-DVD" does, and 
specify it all manually, inserting the "tinterlace" filter in exactly the right 
point in the chain?

(I know I haven't posted any "uncut console output"; since I've tried so many 
experiments, I'm not sure which ones would be relevant. If someone suggests a 
command line for me, I'll be only to happy to cut and paste the "uncut console 
output" that I get.)

--
Andy

From andycivil at gmail.com  Sat Jul 27 23:20:56 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 17:20:56 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130727T104545-441@post.gmane.org>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
Message-ID: <51F439B8.2050807@gmail.com>

*never mind*

Sorry, I think I found the answer by trial and error.

c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.mp4 -vf 
scale=720:480,tinterlace=interleave_bottom -flags ildct+ilme -target ntsc-dvd 
-acodec mp2 -ac 2 -b:a 128000 2013-06-06-i.mpg

This LOOKS right in VLC. It seems that if I put the filter definition first, and 
scale before interleaving, I get a good output.

Thanks for all the help.

--
Andy

From onemda at gmail.com  Sat Jul 27 23:27:43 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 21:27:43 +0000
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F439B8.2050807@gmail.com>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
Message-ID: <CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>

On 7/27/13, Andy Civil <andycivil at gmail.com> wrote:
> *never mind*
>
> Sorry, I think I found the answer by trial and error.
>
> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.mp4 -vf
> scale=720:480,tinterlace=interleave_bottom -flags ildct+ilme -target
> ntsc-dvd
> -acodec mp2 -ac 2 -b:a 128000 2013-06-06-i.mpg
>
> This LOOKS right in VLC. It seems that if I put the filter definition first,
> and
> scale before interleaving, I get a good output.

Yes, scaling interlaced content is no no.

> Thanks for all the help.
>
> --
> Andy
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From adf.lists at gmail.com  Sun Jul 28 00:30:37 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sat, 27 Jul 2013 23:30:37 +0100
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
Message-ID: <51F44A0D.2030302@gmail.com>

Paul B Mahol wrote:
> On 7/27/13, Andy Civil <andycivil at gmail.com> wrote:
>> *never mind*
>>
>> Sorry, I think I found the answer by trial and error.
>>
>> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.mp4 -vf
>> scale=720:480,tinterlace=interleave_bottom -flags ildct+ilme -target
>> ntsc-dvd
>> -acodec mp2 -ac 2 -b:a 128000 2013-06-06-i.mpg
>>
>> This LOOKS right in VLC. It seems that if I put the filter definition first,
>> and
>> scale before interleaving, I get a good output.
>
> Yes, scaling interlaced content is no no.

I agree that weave scaled is one of the most annoying things you may see 
"out there" but -

ffmpeg seems to be able to scale interlaced as long as you tell it -

-vf scale=720:576:interl=1 -vcodec mpeg2video -flags +ildct+ilme -target 
pal-dvd  -aspect 16:9 -b:v 8000k

looks like it works on a 1080i h264 sample I just tested.

One wierd thing is that I got that from a post ages ago and it had an 
extra argument -

-vf scale=720:576:interl=1,fieldorder=tff

now whether I use tff or bff it has the effect of trashing the chroma, 
which may not be instantly obvious depending on content.

Of course my sample was 420 - I don't know what the OPs is, so that 
would be something to consider if he wanted to scale direct instead of 
de then re interlacing.

709 vs 601 colour is also something to consider as is the fact that if 
you scale down video you should ideally low pass filter it first - maybe 
more so for interlaced verticaly, I guess the anamophism sort of handles 
the kell factor horizontally.

I haven't got a clue how to do either with ffmpeg - TBH my test still 
looks OK anyway. I guess deinterlacing (on playback) helps soften and my 
sample is UK DVB-T broadcast which generally seems to be softer than 
kell factor would dictate to start with.





From adf.lists at gmail.com  Sun Jul 28 00:57:03 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sat, 27 Jul 2013 23:57:03 +0100
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F439B8.2050807@gmail.com>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
Message-ID: <51F4503F.6050603@gmail.com>

Andy Civil wrote:
> *never mind*
>
> Sorry, I think I found the answer by trial and error.
>
> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.mp4 -vf
> scale=720:480,tinterlace=interleave_bottom -flags ildct+ilme -target
> ntsc-dvd -acodec mp2 -ac 2 -b:a 128000 2013-06-06-i.mpg
>
> This LOOKS right in VLC. It seems that if I put the filter definition
> first, and scale before interleaving, I get a good output.

It may be perfect, I don't know, but I do know that playing with 
interlaced can be a pain.

Even if something looks OK in VLC there could be hidden issues - eg if 
VLC is deinterlacing to frame rate you may miss field order being wrong.

The content also plays a role, try getting a shot with something bright 
red moving fast horizontally, or do a fast pan past similar.

After making the ntsc, just as a test, recode the sample to field rate 
progressive like -

ffmpeg -i ntsc.mpg -an -sn -vf yadif=1 -vcodec libx264 -crf 22 -r 60 
test.mkv

(maybe -r should be 59.94)

Play back and frame step the fast horizontal chroma motion/pan to check 
that it doesn't skip back and forward and also that the chroma is 
planted on the luma!



From andycivil at gmail.com  Sun Jul 28 01:04:19 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 19:04:19 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F44A0D.2030302@gmail.com>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F439B8.2050807@gmail.com>	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com>
Message-ID: <51F451F3.5030002@gmail.com>

On 2013-07-27 6:30 PM, Andy Furniss wrote:

>
> ffmpeg seems to be able to scale interlaced as long as you tell it -
>
> -vf scale=720:576:interl=1  ...

I was a long hold-out on MEncoder because for years it was the only one that had 
interlace-aware scaling, but now, FFmpeg does it all.

> -target pal-dvd

I think life may be slightly different for you, in PAL world. I won't quote what 
I think the difference is, for fear of getting it wrong, but I am sure that it's 
different.

> One wierd thing is that I got that from a post ages ago and it had an extra
> argument -
>
> -vf scale=720:576:interl=1,fieldorder=tff

Field order is the thing. I was going bananas trying to get this working until I 
realised that my HD camera was producing TFF video, and I needed BFF video for 
the DVD. The ridiculousness of this situation was what prompted me to get my 
movie editor to deinterlace into 59.94p for my intermediate file, and then 
asking FFmpeg to scale and interleave.

> Of course my sample was 420 - I don't know what the OPs is, so that would be
> something to consider if he wanted to scale direct instead of de then re
> interlacing.

Because of the field switch, I don't think I can 'scale direct'. Let's face it, 
even if I thought I was doing it, most likely it would be converted to 
progressive, scaled, and re-interlaced in the background anyway! I might as well 
have it out in the open and control it. That's why I like command line tools ;-)

--
Andy

From adf.lists at gmail.com  Sun Jul 28 01:19:56 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sun, 28 Jul 2013 00:19:56 +0100
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F451F3.5030002@gmail.com>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F439B8.2050807@gmail.com>	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com> <51F451F3.5030002@gmail.com>
Message-ID: <51F4559C.1020705@gmail.com>

Andy Civil wrote:
> On 2013-07-27 6:30 PM, Andy Furniss wrote:
>
>>
>> ffmpeg seems to be able to scale interlaced as long as you tell it -
>>
>> -vf scale=720:576:interl=1  ...
>
> I was a long hold-out on MEncoder because for years it was the only one
> that had interlace-aware scaling, but now, FFmpeg does it all.
>
>> -target pal-dvd
>
> I think life may be slightly different for you, in PAL world. I won't
> quote what I think the difference is, for fear of getting it wrong, but
> I am sure that it's different.

I don't know to be honest.

>
>> One wierd thing is that I got that from a post ages ago and it had an
>> extra
>> argument -
>>
>> -vf scale=720:576:interl=1,fieldorder=tff
>
> Field order is the thing. I was going bananas trying to get this working
> until I realised that my HD camera was producing TFF video, and I needed
> BFF video for the DVD. The ridiculousness of this situation was what
> prompted me to get my movie editor to deinterlace into 59.94p for my
> intermediate file, and then asking FFmpeg to scale and interleave

I don't think that's true as has been mentioned elsewhere, dvd uses 
mpeg2 and mpeg2 will handle bff or tff as long as it's flagged correctly.

>> Of course my sample was 420 - I don't know what the OPs is, so that
>> would be
>> something to consider if he wanted to scale direct instead of de then re
>> interlacing.
>
> Because of the field switch, I don't think I can 'scale direct'. Let's
> face it, even if I thought I was doing it, most likely it would be
> converted to progressive, scaled, and re-interlaced in the background
> anyway! I might as well have it out in the open and control it. That's
> why I like command line tools ;-)

Fair enough, it's working results that count I suppose :-)


From onemda at gmail.com  Sat Jul 27 23:27:43 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 21:27:43 +0000
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F439B8.2050807@gmail.com>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
Message-ID: <CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>

On 7/27/13, Andy Civil <andycivil at gmail.com> wrote:
> *never mind*
>
> Sorry, I think I found the answer by trial and error.
>
> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.mp4 -vf
> scale=720:480,tinterlace=interleave_bottom -flags ildct+ilme -target
> ntsc-dvd
> -acodec mp2 -ac 2 -b:a 128000 2013-06-06-i.mpg
>
> This LOOKS right in VLC. It seems that if I put the filter definition first,
> and
> scale before interleaving, I get a good output.

Yes, scaling interlaced content is no no.

> Thanks for all the help.
>
> --
> Andy
> _______________________________________________
> 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  Sat Jul 27 22:59:26 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sat, 27 Jul 2013 20:59:26 +0000
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org>
	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>

On 7/27/13, James Board <jpboard2 at yahoo.com> wrote:
>>> The video checksums makes sense (to me): there is one for each frame in
>
>>> (the AVI file stores each frame in full, so there's no inter-frame
>>> compression).
>>
>>This is not true.
>
> I think you mean it isn't true in general.  But for my example, where
> there's no
> inter-frame compression, it is true.  I counted the frames and I counted
> the
> checksums, and they are equal.
>
>>Most audio
>  codecs have a specific frame size (for specified settings): a
>>number of samples they encode at once. If the input file uses one such
>
> So if, for example, I wanted all of the audio samples that correspond to
> a specific frame, I might have to look at multiple audio packets, and then
> figure out which audio samples in each packet belong to which frame?

Something like that, note that there are containers (not AVI) that actually do
that.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user



From andycivil at gmail.com  Sat Jul 27 23:20:56 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 17:20:56 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130727T104545-441@post.gmane.org>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
Message-ID: <51F439B8.2050807@gmail.com>

*never mind*

Sorry, I think I found the answer by trial and error.

c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.mp4 -vf 
scale=720:480,tinterlace=interleave_bottom -flags ildct+ilme -target ntsc-dvd 
-acodec mp2 -ac 2 -b:a 128000 2013-06-06-i.mpg

This LOOKS right in VLC. It seems that if I put the filter definition first, and 
scale before interleaving, I get a good output.

Thanks for all the help.

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



From andycivil at gmail.com  Sat Jul 27 23:09:33 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 17:09:33 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130727T104545-441@post.gmane.org>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
Message-ID: <51F4370D.3050204@gmail.com>

On 2013-07-27 4:50 AM, Carl Eugen Hoyos wrote:

>
> See http://ffmpeg.org/ffmpeg-filters.html#tinterlace
> I suspect that you still have to specify the
> encoding flags, this could indeed be seen as a bug
> but it is not what you reported originally.

I'm going to need help with this. I tried adding:
-vf tinterlace=interleave_bottom
-flags ildct+ilme
and it's doing something, but it's not right.

The way I'm analysing my output, is that I view it in VLC. What should happen is 
that I see the 'comb' effect on any moving edges, but if I choose to turn 
de-interlacing on, that comb effect disappears.

With the addition of those two options, what I get is that the two fields are 
'fuzzed together' (like a 'blend' deinterlace had been used).

I also use the analysis tool "gspot" to tell me if the video is labelled as 
interlaced. It seems that where those flags are set, it is.

There's something specific that I don't understand. When I specify "format 
ntsc-DVD" that must set up a filter chain of some kind. But when I add the 
'tinterlace' filter, that's probably going to be added in the wrong place in the 
chain, isn't it?

Is there any way that I can find out exactly what "format ntsc-DVD" does, and 
specify it all manually, inserting the "tinterlace" filter in exactly the right 
point in the chain?

(I know I haven't posted any "uncut console output"; since I've tried so many 
experiments, I'm not sure which ones would be relevant. If someone suggests a 
command line for me, I'll be only to happy to cut and paste the "uncut console 
output" that I get.)

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



From adf.lists at gmail.com  Sun Jul 28 00:30:37 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sat, 27 Jul 2013 23:30:37 +0100
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
Message-ID: <51F44A0D.2030302@gmail.com>

Paul B Mahol wrote:
> On 7/27/13, Andy Civil <andycivil at gmail.com> wrote:
>> *never mind*
>>
>> Sorry, I think I found the answer by trial and error.
>>
>> c:\ffmpeg\bin\ffmpeg -i E:\temp\2013-06-06-hdp.mp4 -vf
>> scale=720:480,tinterlace=interleave_bottom -flags ildct+ilme -target
>> ntsc-dvd
>> -acodec mp2 -ac 2 -b:a 128000 2013-06-06-i.mpg
>>
>> This LOOKS right in VLC. It seems that if I put the filter definition first,
>> and
>> scale before interleaving, I get a good output.
>
> Yes, scaling interlaced content is no no.

I agree that weave scaled is one of the most annoying things you may see 
"out there" but -

ffmpeg seems to be able to scale interlaced as long as you tell it -

-vf scale=720:576:interl=1 -vcodec mpeg2video -flags +ildct+ilme -target 
pal-dvd  -aspect 16:9 -b:v 8000k

looks like it works on a 1080i h264 sample I just tested.

One wierd thing is that I got that from a post ages ago and it had an 
extra argument -

-vf scale=720:576:interl=1,fieldorder=tff

now whether I use tff or bff it has the effect of trashing the chroma, 
which may not be instantly obvious depending on content.

Of course my sample was 420 - I don't know what the OPs is, so that 
would be something to consider if he wanted to scale direct instead of 
de then re interlacing.

709 vs 601 colour is also something to consider as is the fact that if 
you scale down video you should ideally low pass filter it first - maybe 
more so for interlaced verticaly, I guess the anamophism sort of handles 
the kell factor horizontally.

I haven't got a clue how to do either with ffmpeg - TBH my test still 
looks OK anyway. I guess deinterlacing (on playback) helps soften and my 
sample is UK DVB-T broadcast which generally seems to be softer than 
kell factor would dictate to start with.




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



From cehoyos at ag.or.at  Sun Jul 28 03:04:58 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 01:04:58 +0000 (UTC)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a 1080p
	60 Hz Computer Monitor
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<loom.20130727T192450-485@post.gmane.org>
	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <loom.20130728T030129-819@post.gmane.org>

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

> there were ugly combing artifacts that did disappear 
> when I inverse telecined it.
> 
> I want to know if will I be introducing other artifacts 
> due to the fact that the inverse telecined video is 24 
> fps, while the display is 1080p at 60 Hz.?

> For example, will motion now be less smooth?

Of course.

> How does it work?? How does a video which has 24 fps 
> get displayed on a TV with displays at 60 fps??

(How is this related to FFmpeg? And I would expect 
that the answer depends on your display system.)

> Does it duplicate frames?

I would expect so.

>?Or does it telecine it?

Hopefully not;-)

> Something else?

Theoretically yes, I am on CRT so certainly not for me.

Carl Eugen


From cehoyos at ag.or.at  Sun Jul 28 03:07:46 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 01:07:46 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F439B8.2050807@gmail.com>	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com> <51F451F3.5030002@gmail.com>
Message-ID: <loom.20130728T030525-766@post.gmane.org>

Andy Civil <andycivil <at> gmail.com> writes:

> > -vf scale=720:576:interl=1  ...
> 
> I was a long hold-out on MEncoder because for years it 
> was the only one that had interlace-aware scaling

Interlace-aware scaling that was part of FFmpeg 
for several years, yes.

[...]

> Because of the field switch, I don't think I can 
> 'scale direct'.

> Let's face it, even if I thought I was doing it, most 
> likely it would be converted to progressive, scaled, 
> and re-interlaced in the background anyway!

Of course not!
A simple performance test will tell.

Carl Eugen


From cehoyos at ag.or.at  Sun Jul 28 03:12:27 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 01:12:27 +0000 (UTC)
Subject: [FFmpeg-user] encoding bff input to mov labels the output as
	progressive
References: <8B50D84E-9161-45D5-B217-550EB5DC3516@dericed.com>
	<loom.20130727T104318-882@post.gmane.org>
	<EE1B16F9-8A1F-4720-91FF-F665AADFF0BD@dericed.com>
Message-ID: <loom.20130728T031138-677@post.gmane.org>

Dave Rice <dave <at> dericed.com> writes:

> > I just sent a patch, could you test it?
> 
> That worked for me. Output is now bff without 
> needing to be forced. Thanks!

The patch was merged, thank you for testing!
(I suspect there are other codecs that also 
need this patch.)

Carl Eugen


From brown at mrvideo.vidiot.com  Sun Jul 28 03:16:58 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Sat, 27 Jul 2013 20:16:58 -0500
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F4370D.3050204@gmail.com>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
Message-ID: <20130728011658.GP27428@mrvideo.vidiot.com>

On Sat, Jul 27, 2013 at 05:09:33PM -0400, Andy Civil wrote:
> The way I'm analysing my output, is that I view it in VLC. What should 
> happen is that I see the 'comb' effect on any moving edges, but if I choose 
> to turn de-interlacing on, that comb effect disappears.

With interlaced video, you better see the "comb" effect with motion video.
But, there is fine print: 1) if the source was from pure interlaced video,
i.e., a live event; 2) the source is 2:3 pulldown video, where 3 frames will
not have the effect and 2 frames will.

You won't have the effect if the source video was progressive.  You might not
get the effect if converting 25 fps source to 29.97 fps.  The 25 fps source
might be progressive in nature (Doctor Who for example).  It depends on how
the conversion to 29.97 fps is done and if told to do interlaced.

There are lots of factors involved with video.

Enjoy.

MB
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From cehoyos at ag.or.at  Sun Jul 28 03:10:17 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 01:10:17 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com>
Message-ID: <loom.20130728T030907-289@post.gmane.org>

Andy Furniss <adf.lists <at> gmail.com> writes:

> One wierd thing is that I got that from a post ages 
> ago and it had an extra argument -
> 
> -vf scale=720:576:interl=1,fieldorder=tff
> 
> now whether I use tff or bff it has the effect of 
> trashing the chroma, which may not be instantly 
> obvious depending on content.

This sounds (very) important, please add command line 
and complete, uncut console output so I can test.

Thank you, Carl Eugen


From andycivil at gmail.com  Sun Jul 28 03:56:35 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 21:56:35 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <20130728011658.GP27428@mrvideo.vidiot.com>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
Message-ID: <51F47A53.40404@gmail.com>

On 2013-07-27 9:16 PM, Mike Brown wrote:
> On Sat, Jul 27, 2013 at 05:09:33PM -0400, Andy Civil wrote:
>> The way I'm analysing my output, is that I view it in VLC. What should
>> happen is that I see the 'comb' effect on any moving edges, but if I choose
>> to turn de-interlacing on, that comb effect disappears.
>
> With interlaced video, you better see the "comb" effect with motion video.
> But, there is fine print: 1) if the source was from pure interlaced video,
> i.e., a live event; 2) the source is 2:3 pulldown video, where 3 frames will
> not have the effect and 2 frames will.
>
> You won't have the effect if the source video was progressive.  You might not
> get the effect if converting 25 fps source to 29.97 fps.  The 25 fps source
> might be progressive in nature (Doctor Who for example).  It depends on how
> the conversion to 29.97 fps is done and if told to do interlaced.

The video starts from my Sony HX5 as 1920 x 1080 x 30i. It's then deinterlaced 
by a commercial video editor to 1920 x 1080 x 60p. Every frame is different.

--
Andy

From brown at mrvideo.vidiot.com  Sun Jul 28 04:12:59 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Sat, 27 Jul 2013 21:12:59 -0500
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F47A53.40404@gmail.com>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
Message-ID: <20130728021259.GQ27428@mrvideo.vidiot.com>

On Sat, Jul 27, 2013 at 09:56:35PM -0400, Andy Civil wrote:
> The video starts from my Sony HX5 as 1920 x 1080 x 30i. It's then 
> deinterlaced by a commercial video editor to 1920 x 1080 x 60p. Every frame 
> is different.

I'm curious as to why you would want to do that.

I have to admit that I do not know what all ffmpeg can do.  But, IMHO, the
best way to go from 1080i to 480i is to rescale each 1080i field to the
480i field.  By going to 60p first, each of the frames is a "guess" as to
what a full frame would look like from a half frame.

By going directly from 540 lines to 240 lines, you skip the guessing game.

It is what I would try to do.  But, that is me.

MB
-- 
e-mail: vidiot at vidiot.com | vidiot at vidiot.net            /~\ The ASCII
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Visit - URL: http://vidiot.com/                           X  Against
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From andycivil at gmail.com  Sun Jul 28 04:14:37 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 22:14:37 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130728T030525-766@post.gmane.org>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F439B8.2050807@gmail.com>	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>	<51F44A0D.2030302@gmail.com>
	<51F451F3.5030002@gmail.com>
	<loom.20130728T030525-766@post.gmane.org>
Message-ID: <51F47E8D.1000304@gmail.com>

On 2013-07-27 9:07 PM, Carl Eugen Hoyos wrote:
> Andy Civil<andycivil<at>  gmail.com>  writes:
>
>>> -vf scale=720:576:interl=1  ...
>>
>> I was a long hold-out on MEncoder because for years it
>> was the only one that had interlace-aware scaling
>
> Interlace-aware scaling that was part of FFmpeg
> for several years, yes.

It was 2008 that I tried to get FFmpeg to do it, and I was directed to MEncoder.
<http://ffmpeg-users.933282.n4.nabble.com/Can-t-seem-to-shrink-and-preserve-interlacing-td935065.html>


>> Because of the field switch, I don't think I can
>> 'scale direct'.
>
>> Let's face it, even if I thought I was doing it, most
>> likely it would be converted to progressive, scaled,
>> and re-interlaced in the background anyway!
>
> Of course not!
> A simple performance test will tell.

Ha ha - OK I stand corrected.

Thank you for all the help.

--
Andy


From cehoyos at ag.or.at  Sun Jul 28 04:22:01 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 02:22:01 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
Message-ID: <loom.20130728T042057-293@post.gmane.org>

Mike Brown <brown <at> mrvideo.vidiot.com> writes:

> I have to admit that I do not know what all ffmpeg can do.

(Not necessarily related to your post:)
It can do both advanced deinterlacing and interlaced scaling.

[...]

> It is what I would try to do.  But, that is me.

What you described is actually what everybody would do;-)

Carl Eugen


From andycivil at gmail.com  Sun Jul 28 04:31:56 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 22:31:56 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <20130728021259.GQ27428@mrvideo.vidiot.com>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
Message-ID: <51F4829C.6070902@gmail.com>

On 2013-07-27 10:12 PM, Mike Brown wrote:
> On Sat, Jul 27, 2013 at 09:56:35PM -0400, Andy Civil wrote:
>> The video starts from my Sony HX5 as 1920 x 1080 x 30i. It's then
>> deinterlaced by a commercial video editor to 1920 x 1080 x 60p. Every frame
>> is different.
>
> I'm curious as to why you would want to do that.

Because I first tried the interlace-aware scaler, and I was unable to make it 
also switch from TFF input to BFF output. (My camera produces TFF video, being 
the HD standard, but a BFF DVD-video is more common. Yes, I know that you can 
also have a TFF DVD-video, but I want my output to be as 'normal' as possible.)

>
> I have to admit that I do not know what all ffmpeg can do.  But, IMHO, the
> best way to go from 1080i to 480i is to rescale each 1080i field to the
> 480i field.  By going to 60p first, each of the frames is a "guess" as to
> what a full frame would look like from a half frame.
>
> By going directly from 540 lines to 240 lines, you skip the guessing game.

Because it's being scaled, some lines will be directly taken from the input, 
some will be guesses, depending on how they fall, no? You can only eliminate 
guesses if the output size is a factor of the input size e.g. 480 -> 240.

I'm not saying that I'm doing it the best way, nor that my intermediate file is 
the smallest, but it works and I'm confident in the quality of the output. 
(However, I do realise that if I accepted 30p as being 'smooth enough' then I 
could probably get a sharper picture, due to improved compression characteristics.)

--
Andy

From cehoyos at ag.or.at  Sun Jul 28 04:40:58 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 02:40:58 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
	<51F4829C.6070902@gmail.com>
Message-ID: <loom.20130728T043707-807@post.gmane.org>

Andy Civil <andycivil <at> gmail.com> writes:

> > By going directly from 540 lines to 240 lines, you 
> > skip the guessing game.
> 
> Because it's being scaled, some lines will be 
> directly taken from the input, some will be 
> guesses, depending on how they fall, no?

That would not even be true for non-interlaced-aware 
scaling afaict (and I suspect not even for a "simple" 
implementation), and it is of course not true for 
the interlace-aware scaling that FFmpeg provides.

(The idea is to scale two half-resolution images to
half the resolution of the output image.)

Carl Eugen


From cehoyos at ag.or.at  Sun Jul 28 04:45:21 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 02:45:21 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F439B8.2050807@gmail.com>	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>	<51F44A0D.2030302@gmail.com>
	<51F451F3.5030002@gmail.com>
	<loom.20130728T030525-766@post.gmane.org>
	<51F47E8D.1000304@gmail.com>
Message-ID: <loom.20130728T044202-800@post.gmane.org>

Andy Civil <andycivil <at> gmail.com> writes:

> On 2013-07-27 9:07 PM, Carl Eugen Hoyos wrote:
> > Andy Civil<andycivil<at>  gmail.com>  writes:
> >
> >>> -vf scale=720:576:interl=1  ...
> >>
> >> I was a long hold-out on MEncoder because for years 
> >> it was the only one that had interlace-aware scaling
> >
> > Interlace-aware scaling that was part of FFmpeg
> > for several years, yes.
> 
> It was 2008 that I tried to get FFmpeg to do it, and 
> I was directed to MEncoder.

Sorry, I was not clear above.
It appears (iirc) that while MEncoder used the same 
scaler that is now (for several years) an integral 
part of FFmpeg, the interface that ffmpeg (the cli) 
originally provided did not allow the users to 
specify interlaced scaling, MPlayer's scale filter 
did.

I would not rule out that the reason for the FFmpeg 
limitation was that no bug report was ever opened...

Carl Eugen


From andycivil at gmail.com  Sun Jul 28 05:09:14 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sat, 27 Jul 2013 23:09:14 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130728T043707-807@post.gmane.org>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>	<20130728021259.GQ27428@mrvideo.vidiot.com>	<51F4829C.6070902@gmail.com>
	<loom.20130728T043707-807@post.gmane.org>
Message-ID: <51F48B5A.9000502@gmail.com>

On 2013-07-27 10:40 PM, Carl Eugen Hoyos wrote:
> Andy Civil<andycivil<at>  gmail.com>  writes:
>
>> Because it's being scaled, some lines will be
>> directly taken from the input, some will be
>> guesses, depending on how they fall, no?
>
> That would not even be true for non-interlaced-aware
> scaling afaict (and I suspect not even for a "simple"
> implementation), and it is of course not true for
> the interlace-aware scaling that FFmpeg provides.
>
> (The idea is to scale two half-resolution images to
> half the resolution of the output image.)

I suspect that it's my attempt to describe the process which is actually at 
fault; but in any case your time is too valuable for me to waste it arguing...

--
Andy

From brown at mrvideo.vidiot.com  Sun Jul 28 07:28:48 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Sun, 28 Jul 2013 00:28:48 -0500
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F4829C.6070902@gmail.com>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
	<51F4829C.6070902@gmail.com>
Message-ID: <20130728052848.GR27428@mrvideo.vidiot.com>

On Sat, Jul 27, 2013 at 10:31:56PM -0400, Andy Civil wrote:
> Because I first tried the interlace-aware scaler, and I was unable to make 
> it also switch from TFF input to BFF output. (My camera produces TFF video, 
> being the HD standard, but a BFF DVD-video is more common. Yes, I know that 
> you can also have a TFF DVD-video, but I want my output to be as 'normal' 
> as possible.)

Interesting.  I never knew that the DVD MPEG-2 video was supposed to be
BFF.  I alawys thought it was TFF.  But... since pretty much all movies are
placed on DVD as 23.976p, with pulldown flags, the DVD player will convert
the MPEG-2 to analog NTSC 2:3 output.

As far as being normal as possible... the person who plays the DVD could
care less which field is first, as long as it plays correctly on the analog
TV.  If played on a computer, or any other display, it isn't going to make
a difference.

IMHO, you are overthinking the result.  As the saying goes... KISS.

> Because it's being scaled, some lines will be directly taken from the 
> input, some will be guesses, depending on how they fall, no? You can only 
> eliminate guesses if the output size is a factor of the input size e.g. 480 
> -> 240.

Right.  But, you skip one of the guessing games instead of having two.

MB
-- 
e-mail: vidiot at vidiot.com | vidiot at vidiot.net            /~\ The ASCII
        6082066843 at email.uscc.net (140 char limit)       \ / Ribbon Campaign
Visit - URL: http://vidiot.com/                           X  Against
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From andycivil at gmail.com  Sun Jul 28 08:05:39 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sun, 28 Jul 2013 02:05:39 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <20130728052848.GR27428@mrvideo.vidiot.com>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>	<20130728021259.GQ27428@mrvideo.vidiot.com>	<51F4829C.6070902@gmail.com>
	<20130728052848.GR27428@mrvideo.vidiot.com>
Message-ID: <51F4B4B3.6080003@gmail.com>

On 2013-07-28 1:28 AM, Mike Brown wrote:
>
> Interesting.  I never knew that the DVD MPEG-2 video was supposed to be
> BFF.  I alawys thought it was TFF.  But... since pretty much all movies are
> placed on DVD as 23.976p, with pulldown flags, the DVD player will convert
> the MPEG-2 to analog NTSC 2:3 output.

I inferred it from the Wikipedia page on NTSC
"By convention an NTSC video frame is considered to start with an even field 
followed by an odd field."
http://en.wikipedia.org/wiki/480i

> As far as being normal as possible... the person who plays the DVD could
> care less which field is first, as long as it plays correctly on the analog
> TV.  If played on a computer, or any other display, it isn't going to make
> a difference.

I guess you mean "couldn't care less"?

You may be right, but this video is actually going to be played on Grandma's LCD 
television in Britain (A PAL country). I have observed that NTSC DVDs I make 
work correctly in her DVD player, but I don't know exactly how. Therefore, I 
don't dare mess with the NTSC standard, in case the equipment assumes anything. 
I could make a DVD which is essentially NTSC but TFF and it could work on my 
computer, but her DVD player might ignore the TFF flag and assume it's BFF with 
disasterous results.

--
Andy

From werner.robitza at gmail.com  Sun Jul 28 08:35:10 2013
From: werner.robitza at gmail.com (Werner Robitza)
Date: Sun, 28 Jul 2013 08:35:10 +0200
Subject: [FFmpeg-user] libx264 mp4 encoded video won't play back in
	chrome/quicktime
In-Reply-To: <1374724438056-4660168.post@n4.nabble.com>
References: <CAHJN-tktqtUEfKUZbt-0EdphsXWYs=HmDTvVZoX_qGbM6SXGDg@mail.gmail.com>
	<1374724438056-4660168.post@n4.nabble.com>
Message-ID: <CAN0LTaMcsZ8ijcBSDDfUZHU0ScPdM2FsnqRBAG282jZnR5Wovg@mail.gmail.com>

On Thu, Jul 25, 2013 at 5:53 AM, hevesyagyeman <hevesyagyeman at yahoo.com>wrote:

> Acting as a video container format, MP4 is the package and can contain
> video
> and audio components encoded with different types (codecs). However,
> QuickTime just supports a few video and audio codecs in MP4 format. That
> is,
> if the MP4 video contains QuickTime unsupported audio or video codecs,
> QuickTime will refuse to play this kind of MP4 files.
>
> In order to solve MP4 won't play in QuickTime issue, we had better  convert
> QuickTime unplayable MP4 to QuickTime supported format.
> <http://www.idealshare.net/mp4-quicktime-solution.html>
>

Can someone remove this spammer from the mailing list?

The link is rather fishy, and just look at the text.

Thanks

From cehoyos at ag.or.at  Sun Jul 28 10:46:18 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 08:46:18 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
	<51F4829C.6070902@gmail.com>
	<20130728052848.GR27428@mrvideo.vidiot.com>
Message-ID: <loom.20130728T104212-627@post.gmane.org>

Mike Brown <brown <at> mrvideo.vidiot.com> writes:

> I never knew that the DVD MPEG-2 video was supposed to be
> BFF. I alawys thought it was TFF.

Afair, both are possible and allowed and both are seen in 
the wild.

> But... since pretty much all movies are placed on DVD as 
> 23.976p, with pulldown flags, the DVD player will convert
> the MPEG-2 to analog NTSC 2:3 output.

The world is not NTSC-only...

[...]

> > Because it's being scaled, some lines will be directly 
> > taken from the input, some will be guesses, depending 
> > on how they fall, no? You can only eliminate guesses if 
> > the output size is a factor of the input size e.g. 480 
> > -> 240.

(As said, this is simply not correct.)

> Right.  But, you skip one of the guessing games instead 
> of having two.

Depending on the definition of "guessing", I don't think 
scaling counts as guessing, you simply have to choose one 
of several algorithms.

Carl Eugen


From brown at mrvideo.vidiot.com  Sun Jul 28 10:50:26 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Sun, 28 Jul 2013 03:50:26 -0500
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F4B4B3.6080003@gmail.com>
References: <loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
	<51F4829C.6070902@gmail.com>
	<20130728052848.GR27428@mrvideo.vidiot.com>
	<51F4B4B3.6080003@gmail.com>
Message-ID: <20130728085026.GT27428@mrvideo.vidiot.com>

On Sun, Jul 28, 2013 at 02:05:39AM -0400, Andy Civil wrote:
> On 2013-07-28 1:28 AM, Mike Brown wrote:
>
> I inferred it from the Wikipedia page on NTSC
> "By convention an NTSC video frame is considered to start with an even 
> field followed by an odd field."
> http://en.wikipedia.org/wiki/480i

The DVD player will take the MPEG-2 video and output it as correct NTSC
analog.

>> As far as being normal as possible... the person who plays the DVD could
>> care less which field is first, as long as it plays correctly on the analog
>> TV.  If played on a computer, or any other display, it isn't going to make
>> a difference.
>
> I guess you mean "couldn't care less"?

I believe that the "not" should not be there.  I've heard it both ways.
Not sure which is actuall correct.  To me it seems to be a double negative.

> You may be right, but this video is actually going to be played on 
> Grandma's LCD television in Britain (A PAL country). I have observed that 
> NTSC DVDs I make work correctly in her DVD player, but I don't know exactly 
> how. Therefore, I don't dare mess with the NTSC standard, in case the 
> equipment assumes anything. I could make a DVD which is essentially NTSC 
> but TFF and it could work on my computer, but her DVD player might ignore 
> the TFF flag and assume it's BFF with disasterous results.

That is because pretty much every U.K. (and I believe all of Europe) DVD
player will handle 480i as well as 576i video.  Been standard for ages.
The MPEG-2 chipset does the conversion.  I have a JVC DVD player that will
handle both as well and output as NTSC or PAL, plus it is region neutered.
I seriously doubt that she will have trouble.

MB
-- 
e-mail: vidiot at vidiot.com | vidiot at vidiot.net            /~\ The ASCII
        6082066843 at email.uscc.net (140 char limit)       \ / Ribbon Campaign
Visit - URL: http://vidiot.com/                           X  Against
             http://vidiot.net/                          / \ HTML Email
"Take your hand off your tool, Marco!" - Det. Beckett - Castle - 10/08/12

From brown at mrvideo.vidiot.com  Sun Jul 28 10:53:52 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Sun, 28 Jul 2013 03:53:52 -0500
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130728T104212-627@post.gmane.org>
References: <loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
	<51F4829C.6070902@gmail.com>
	<20130728052848.GR27428@mrvideo.vidiot.com>
	<loom.20130728T104212-627@post.gmane.org>
Message-ID: <20130728085352.GU27428@mrvideo.vidiot.com>

On Sun, Jul 28, 2013 at 08:46:18AM +0000, Carl Eugen Hoyos wrote:
> Mike Brown <brown <at> mrvideo.vidiot.com> writes:
> > But... since pretty much all movies are placed on DVD as 
> > 23.976p, with pulldown flags, the DVD player will convert
> > the MPEG-2 to analog NTSC 2:3 output.
> 
> The world is not NTSC-only...

Sorry, I should have indicated stuff for Region 1.  The discussion has been
about 480i, so I didn't mention it.
> > Right.  But, you skip one of the guessing games instead 
> > of having two.
> 
> Depending on the definition of "guessing", I don't think 
> scaling counts as guessing, you simply have to choose one 
> of several algorithms.

Guessing may indeed be the incorrect word to use.  The point is why do it
twice.  Skip the middle step.

MB
-- 
e-mail: vidiot at vidiot.com | vidiot at vidiot.net            /~\ The ASCII
        6082066843 at email.uscc.net (140 char limit)       \ / Ribbon Campaign
Visit - URL: http://vidiot.com/                           X  Against
             http://vidiot.net/                          / \ HTML Email
"Take your hand off your tool, Marco!" - Det. Beckett - Castle - 10/08/12

From adf.lists at gmail.com  Sun Jul 28 12:45:12 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Sun, 28 Jul 2013 11:45:12 +0100
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <loom.20130728T030907-289@post.gmane.org>
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com>
	<loom.20130728T030907-289@post.gmane.org>
Message-ID: <51F4F638.6000201@gmail.com>

Carl Eugen Hoyos wrote:
> Andy Furniss <adf.lists <at> gmail.com> writes:
>
>> One wierd thing is that I got that from a post ages
>> ago and it had an extra argument -
>>
>> -vf scale=720:576:interl=1,fieldorder=tff
>>
>> now whether I use tff or bff it has the effect of
>> trashing the chroma, which may not be instantly
>> obvious depending on content.
>
> This sounds (very) important, please add command line
> and complete, uncut console output so I can test.
>
> Thank you, Carl Eugen

Apologies for the indirect link, I changed ISP and don't have webspace 
anymore - but if you want the source (only 14 meg) it's here -

http://www.filedropper.com/h264-1080i

There is a fast pan past some hoardings that shows the issue well with 
mplayer -vf yadif=1.


ffmpeg -i h264-1080i.ts -an -sn -vf 
scale=720:576:interl=1,fieldorder=tff -vcodec mpeg2video -flags 
+ildct+ilme -target pal-dvd -aspect 16:9 -b:v 8000k tff.mpg
ffmpeg version N-54994-gedab63a Copyright (c) 2000-2013 the FFmpeg 
developers
   built on Jul 28 2013 10:41:25 with gcc 4.6.3 (GCC for Cross-LFS 
4.6.3.20121105)
   configuration: --prefix=/usr --enable-gpl --enable-libx264 
--enable-x11grab
   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
[h264 @ 0x26b82e0] mmco: unref short failure
     Last message repeated 2 times
[NULL @ 0x26bc260] start time is not set in estimate_timings_from_pts
[mpegts @ 0x2695b60] PES packet size mismatch
     Last message repeated 1 times
[mpegts @ 0x2695b60] Could not find codec parameters for stream 4 
(Unknown: none ([5][0][0][0] / 0x0005)): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' 
options
Input #0, mpegts, from 'h264-1080i.ts':
   Duration: 00:00:16.56, start: 44603.868144, bitrate: 7177 kb/s
   Program 17472
   Program 17540
     Stream #0:0[0x19c9]: Video: h264 (High) ([27][0][0][0] / 0x001B), 
yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
     Stream #0:1[0x19ca](eng): Audio: aac_latm ([17][0][0][0] / 0x0011), 
48000 Hz, stereo, fltp
     Stream #0:2[0x19ce](eng): Audio: aac_latm ([17][0][0][0] / 0x0011), 
48000 Hz, stereo, fltp (visual impaired)
     Stream #0:3[0x19cd](eng): Subtitle: dvb_subtitle ([6][0][0][0] / 
0x0006)
     Stream #0:4[0x19fa]: Unknown: none ([5][0][0][0] / 0x0005)
   Program 17603
   Program 17664
   Program 19648
   Program 20288
Output #0, dvd, to 'tff.mpg':
   Metadata:
     encoder         : Lavf55.12.102
     Stream #0:0: Video: mpeg2video, yuv420p, 720x576 [SAR 64:45 DAR 
16:9], q=2-31, 8000 kb/s, 90k tbn, 25 tbc
Stream mapping:
   Stream #0:0 -> #0:0 (h264 -> mpeg2video)
Press [q] to stop, [?] for help
[h264 @ 0x2824660] mmco: unref short failure
     Last message repeated 1 times
[h264 @ 0x27df940] mmco: unref short failure
[h264 @ 0x2892fc0] error while decoding MB 71 6, bytestream 
(-34)e=8144.7kbits/s
[h264 @ 0x2892fc0] concealing 7418 DC, 7418 AC, 7418 MV errors in I frame
frame=  385 fps= 57 q=2.0 Lsize=   15040kB time=00:00:15.36 
bitrate=8021.3kbits/s dup=7 drop=0
video:14788kB audio:0kB subtitle:0 global headers:0kB muxing overhead 
1.703735%




From basteon at gmail.com  Sun Jul 28 13:33:56 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:33:56 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <loom.20130718T095657-197@post.gmane.org>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
Message-ID: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>

Hello,
Some time ado I've been using axis cameras arm based. What I can see your
ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
the camera or PC? Surely I fix that by using v4l on the camera


On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Andy <aseto97 <at> yahoo.com> writes:
>
> > I did git checkout 2e4ec89 and my ffmpeg source is
> > back to Sep.26,2012. Did the build, tried to stream
> > the camera again, and got the same error.
>
> Your problem is apparently not the same as ticket 1734.
>
> > Do you have any idea why it is not working?
>
> No, but consider opening a ticket adding the -fdebug ts
> output as requested by Michael in 1734.
>
> > --disable-optimizations
>
> Probably unrelated:
> --disable-optimizations is a configure option that
> should only be used in special circumstances (debugging
> a crash that shows no useful backtrace with gdb or
> speeding up compilation when bisecting), I suspect
> you should remove it.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From basteon at gmail.com  Sun Jul 28 13:52:37 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:52:37 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
	<CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
Message-ID: <CAEJHKkUpNZgudHQdQxHAn0OX5LVCX0BqB-0PbJuNMnWduFVuFw@mail.gmail.com>

here,
Since 3yrs I work with them I found one available camera on the network.
I'll newer tell the ip, but that works for some axis camera with ffmpeg
version 0.10.3 ...
  Metadata:
    title           : Media Presentation
  Duration: N/A, start: 0.009063, bitrate: N/A
    Stream #0:0: Video: h264 (Baseline), yuvj420p, 800x450 [SAR 1:1 DAR
16:9], 30.08 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: aac, 16000 Hz, mono, s16
File '/tmp/out.avi' already exists. Overwrite ? [y/N] y
Incompatible pixel format 'yuvj420p' for codec 'mpeg4', auto-selecting
format 'yuv420p'
[buffer @ 0x276ce40] w:800 h:450 pixfmt:yuvj420p tb:1/1000000 sar:1/1
sws_param:
[buffersink @ 0x276c700] auto-inserting filter 'auto-inserted scale 0'
between the filter 'src' and the filter 'out'
[scale @ 0x276bde0] w:800 h:450 fmt:yuvj420p -> w:800 h:450 fmt:yuv420p
flags:0x4
Output #0, avi, to '/tmp/out.avi':
  Metadata:
    INAM            : Media Presentation
    ISFT            : Lavf53.32.100
    Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 800x450 [SAR
1:1 DAR 16:9], q=2-31, 200 kb/s, 30.08 tbn, 30.08 tbc
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 16000 Hz, mono, s16, 128
kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg4)
  Stream #0:1 -> #0:1 (aac -> libmp3lame)
Press [q] to stop, [?] for help
frame= 2298 fps= 23 q=31.0 Lsize=    4575kB time=00:01:36.84 bitrate=
387.0kbits/s dup=0 drop=63


On 28 July 2013 11:33, basteon <basteon at gmail.com> wrote:

> Hello,
> Some time ado I've been using axis cameras arm based. What I can see your
> ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
> the camera or PC? Surely I fix that by using v4l on the camera
>
>
> On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
>
>> Andy <aseto97 <at> yahoo.com> writes:
>>
>> > I did git checkout 2e4ec89 and my ffmpeg source is
>> > back to Sep.26,2012. Did the build, tried to stream
>> > the camera again, and got the same error.
>>
>> Your problem is apparently not the same as ticket 1734.
>>
>> > Do you have any idea why it is not working?
>>
>> No, but consider opening a ticket adding the -fdebug ts
>> output as requested by Michael in 1734.
>>
>> > --disable-optimizations
>>
>> Probably unrelated:
>> --disable-optimizations is a configure option that
>> should only be used in special circumstances (debugging
>> a crash that shows no useful backtrace with gdb or
>> speeding up compilation when bisecting), I suspect
>> you should remove it.
>>
>> Carl Eugen
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
>

From jpboard2 at yahoo.com  Sun Jul 28 14:58:40 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 05:58:40 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<20130727183314.GA27349@phare.normalesup.org>	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
Message-ID: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> So if, for example, I wanted all of the audio samples that correspond to

>> a specific frame, I might have to look at multiple audio packets, and then
>> figure out which audio samples in each packet belong to which frame?
>
>Something like that, note that there are containers (not AVI) that actually do
>that.

Thank you.? Which containers do this?? Which containers store all the
audio packets for a single video frame in the same audio packet?? Or
which containers make it easier to determine which audio sample belongs
to whichever video?

From jpboard2 at yahoo.com  Sun Jul 28 15:01:42 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 06:01:42 -0700 (PDT)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a
	1080p	60 Hz Computer Monitor
In-Reply-To: <loom.20130728T030129-819@post.gmane.org>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
Message-ID: <1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>



>> How does it work?? How does a video which has 24 fps 
>> get displayed on a TV with displays at 60 fps??
>
>(How is this related to FFmpeg? And I would expect 
>that the answer depends on your display system.)

This is a video question.? FFmpeg is a video project that
produces libraries and programs for handling video and audio.

Seems like someone here probably has thought through this.

From cehoyos at ag.or.at  Sun Jul 28 15:48:54 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 13:48:54 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com>
	<loom.20130728T030907-289@post.gmane.org>
	<51F4F638.6000201@gmail.com>
Message-ID: <loom.20130728T153837-674@post.gmane.org>

Andy Furniss <adf.lists <at> gmail.com> writes:

> Apologies for the indirect link, I changed ISP and 
> don't have webspace anymore - but if you want the 
> source (only 14 meg) it's here -

Don't worry!

> http://www.filedropper.com/h264-1080i

Tested while watching Budapest;-))

> There is a fast pan past some hoardings that shows 
> the issue well with  mplayer -vf yadif=1.
> 
> ffmpeg -i h264-1080i.ts -an -sn 
> -vf scale=720:576:interl=1,fieldorder=tff 
> -vcodec mpeg2video -flags +ildct+ilme -target pal-dvd 
> -aspect 16:9 -b:v 8000k tff.mpg

The following works fine here:

ffmpeg -i ~/Downloads/h264-1080i.ts -an -sn 
-vf scale=720:576:interl=1,setfield=tff 
-flags +ildct+ilme -target pal-dvd -aspect 16:9 
-b:v 8000k tff.mpg

I don't know why adding fieldorder=tff breaks encoding 
so horribly...

Carl Eugen


From andycivil at gmail.com  Sun Jul 28 16:19:32 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sun, 28 Jul 2013 10:19:32 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <20130728085026.GT27428@mrvideo.vidiot.com>
References: <loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>	<20130728021259.GQ27428@mrvideo.vidiot.com>	<51F4829C.6070902@gmail.com>	<20130728052848.GR27428@mrvideo.vidiot.com>	<51F4B4B3.6080003@gmail.com>
	<20130728085026.GT27428@mrvideo.vidiot.com>
Message-ID: <51F52874.1020502@gmail.com>

On 2013-07-28 4:50 AM, Mike Brown wrote:

>>
>> I guess you mean "couldn't care less"?
>
> I believe that the "not" should not be there.  I've heard it both ways.
> Not sure which is actuall correct.  To me it seems to be a double negative.
>

It's short for "I care so little that I couldn't care any less".

Imagine you were expecting someone to say "I couldn't agree more" but it came 
out as "I could agree more", wouldn't that grate on you?

No one cares about incorrect apostrophes or misplaced their/they're/there any 
more, but that one is particularly annoying because it pretty much says the 
opposite of what you're actually trying to say.

--
Andy

From jamesmichaelford at gmail.com  Sun Jul 28 02:41:06 2013
From: jamesmichaelford at gmail.com (James Ford)
Date: Sun, 28 Jul 2013 08:41:06 +0800
Subject: [FFmpeg-user] Rotation Property
Message-ID: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>

I have some videos that have an incorrect rotation property. This means
that some viewers will rotate it and some will not.

I am looking for a way to reset the rotation property to zero so that the
video will NOT  be rotated.

One of my problems is that all of my searches lead me to ways to rotate the
video. To be clear: I am NOT looking to rotate the video, only to change
the property

Thanks, everyone.

From cehoyos at ag.or.at  Sun Jul 28 18:27:24 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:27:24 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
Message-ID: <loom.20130728T182550-211@post.gmane.org>

James Ford <jamesmichaelford <at> gmail.com> writes:

> I am looking for a way to reset the rotation property 
> to zero so that the video will NOT  be rotated.

(Command line and complete, uncut console output missing.)

Please try -map_metadata -1

Carl Eugen


From cehoyos at ag.or.at  Sun Jul 28 18:30:52 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:30:52 +0000 (UTC)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a 1080p
	60 Hz Computer Monitor
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
	<1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <loom.20130728T182747-704@post.gmane.org>

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

> >> How does it work?? How does a video which has 24 fps 
> >> get displayed on a TV with displays at 60 fps??
> >
> >(How is this related to FFmpeg? And I would expect 
> >that the answer depends on your display system.)
> 
> This is a video question.? FFmpeg is a video project that
> produces libraries and programs for handling video and audio.

Sure.
And somebody may be able to answer if James added information 
about his operating system, his video hardware, his 
video drivers, his X / Windows / OSX display settings and his 
display.

Carl Eugen


From cehoyos at ag.or.at  Sun Jul 28 18:51:51 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:51:51 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
	<loom.20130728T182550-211@post.gmane.org>
Message-ID: <loom.20130728T185129-213@post.gmane.org>

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

> James Ford <jamesmichaelford <at> gmail.com> writes:
> 
> > I am looking for a way to reset the rotation property 
> > to zero so that the video will NOT  be rotated.

> Please try -map_metadata -1

Probably even better:
-metadata rotate=0

Carl Eugen


From onemda at gmail.com  Sun Jul 28 20:41:42 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sun, 28 Jul 2013 18:41:42 +0000
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org>
	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
	<1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P5gmzXSk0vh15+=o=Hfgd=W_4MTY1ZT44JwNMEbWz9mEQ@mail.gmail.com>

On 7/28/13, James Board <jpboard2 at yahoo.com> wrote:
>>> So if, for example, I wanted all of the audio samples that correspond to
>
>>> a specific frame, I might have to look at multiple audio packets, and
>>> then
>>> figure out which audio samples in each packet belong to which frame?
>>
>>Something like that, note that there are containers (not AVI) that actually
>> do
>>that.
>
> Thank you.  Which containers do this?  Which containers store all the
> audio packets for a single video frame in the same audio packet?  Or
> which containers make it easier to determine which audio sample belongs
> to whichever video?

Simple mxf files. Also MPEG-TS with SMPTE S302M audio.

From jpboard2 at yahoo.com  Sun Jul 28 14:58:40 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 05:58:40 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<20130727183314.GA27349@phare.normalesup.org>	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
Message-ID: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> So if, for example, I wanted all of the audio samples that correspond to

>> a specific frame, I might have to look at multiple audio packets, and then
>> figure out which audio samples in each packet belong to which frame?
>
>Something like that, note that there are containers (not AVI) that actually do
>that.

Thank you.? Which containers do this?? Which containers store all the
audio packets for a single video frame in the same audio packet?? Or
which containers make it easier to determine which audio sample belongs
to whichever video?
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user



From basteon at gmail.com  Sun Jul 28 13:33:56 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:33:56 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <loom.20130718T095657-197@post.gmane.org>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
Message-ID: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>

Hello,
Some time ado I've been using axis cameras arm based. What I can see your
ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
the camera or PC? Surely I fix that by using v4l on the camera


On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Andy <aseto97 <at> yahoo.com> writes:
>
> > I did git checkout 2e4ec89 and my ffmpeg source is
> > back to Sep.26,2012. Did the build, tried to stream
> > the camera again, and got the same error.
>
> Your problem is apparently not the same as ticket 1734.
>
> > Do you have any idea why it is not working?
>
> No, but consider opening a ticket adding the -fdebug ts
> output as requested by Michael in 1734.
>
> > --disable-optimizations
>
> Probably unrelated:
> --disable-optimizations is a configure option that
> should only be used in special circumstances (debugging
> a crash that shows no useful backtrace with gdb or
> speeding up compilation when bisecting), I suspect
> you should remove it.
>
> Carl Eugen
>
> _______________________________________________
> 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 basteon at gmail.com  Sun Jul 28 13:52:37 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:52:37 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
	<CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
Message-ID: <CAEJHKkUpNZgudHQdQxHAn0OX5LVCX0BqB-0PbJuNMnWduFVuFw@mail.gmail.com>

here,
Since 3yrs I work with them I found one available camera on the network.
I'll newer tell the ip, but that works for some axis camera with ffmpeg
version 0.10.3 ...
  Metadata:
    title           : Media Presentation
  Duration: N/A, start: 0.009063, bitrate: N/A
    Stream #0:0: Video: h264 (Baseline), yuvj420p, 800x450 [SAR 1:1 DAR
16:9], 30.08 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: aac, 16000 Hz, mono, s16
File '/tmp/out.avi' already exists. Overwrite ? [y/N] y
Incompatible pixel format 'yuvj420p' for codec 'mpeg4', auto-selecting
format 'yuv420p'
[buffer @ 0x276ce40] w:800 h:450 pixfmt:yuvj420p tb:1/1000000 sar:1/1
sws_param:
[buffersink @ 0x276c700] auto-inserting filter 'auto-inserted scale 0'
between the filter 'src' and the filter 'out'
[scale @ 0x276bde0] w:800 h:450 fmt:yuvj420p -> w:800 h:450 fmt:yuv420p
flags:0x4
Output #0, avi, to '/tmp/out.avi':
  Metadata:
    INAM            : Media Presentation
    ISFT            : Lavf53.32.100
    Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 800x450 [SAR
1:1 DAR 16:9], q=2-31, 200 kb/s, 30.08 tbn, 30.08 tbc
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 16000 Hz, mono, s16, 128
kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg4)
  Stream #0:1 -> #0:1 (aac -> libmp3lame)
Press [q] to stop, [?] for help
frame= 2298 fps= 23 q=31.0 Lsize=    4575kB time=00:01:36.84 bitrate=
387.0kbits/s dup=0 drop=63


On 28 July 2013 11:33, basteon <basteon at gmail.com> wrote:

> Hello,
> Some time ado I've been using axis cameras arm based. What I can see your
> ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
> the camera or PC? Surely I fix that by using v4l on the camera
>
>
> On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
>
>> Andy <aseto97 <at> yahoo.com> writes:
>>
>> > I did git checkout 2e4ec89 and my ffmpeg source is
>> > back to Sep.26,2012. Did the build, tried to stream
>> > the camera again, and got the same error.
>>
>> Your problem is apparently not the same as ticket 1734.
>>
>> > Do you have any idea why it is not working?
>>
>> No, but consider opening a ticket adding the -fdebug ts
>> output as requested by Michael in 1734.
>>
>> > --disable-optimizations
>>
>> Probably unrelated:
>> --disable-optimizations is a configure option that
>> should only be used in special circumstances (debugging
>> a crash that shows no useful backtrace with gdb or
>> speeding up compilation when bisecting), I suspect
>> you should remove it.
>>
>> Carl Eugen
>>
>> _______________________________________________
>> 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  Sun Jul 28 18:30:52 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:30:52 +0000 (UTC)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a
	1080p	60 Hz Computer Monitor
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
	<1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <loom.20130728T182747-704@post.gmane.org>

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

> >> How does it work?? How does a video which has 24 fps 
> >> get displayed on a TV with displays at 60 fps??
> >
> >(How is this related to FFmpeg? And I would expect 
> >that the answer depends on your display system.)
> 
> This is a video question.? FFmpeg is a video project that
> produces libraries and programs for handling video and audio.

Sure.
And somebody may be able to answer if James added information 
about his operating system, his video hardware, his 
video drivers, his X / Windows / OSX display settings and his 
display.

Carl Eugen

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



From cehoyos at ag.or.at  Sun Jul 28 18:51:51 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:51:51 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
	<loom.20130728T182550-211@post.gmane.org>
Message-ID: <loom.20130728T185129-213@post.gmane.org>

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

> James Ford <jamesmichaelford <at> gmail.com> writes:
> 
> > I am looking for a way to reset the rotation property 
> > to zero so that the video will NOT  be rotated.

> Please try -map_metadata -1

Probably even better:
-metadata rotate=0

Carl Eugen

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



From andycivil at gmail.com  Sun Jul 28 16:19:32 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sun, 28 Jul 2013 10:19:32 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <20130728085026.GT27428@mrvideo.vidiot.com>
References: <loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>	<20130728021259.GQ27428@mrvideo.vidiot.com>	<51F4829C.6070902@gmail.com>	<20130728052848.GR27428@mrvideo.vidiot.com>	<51F4B4B3.6080003@gmail.com>
	<20130728085026.GT27428@mrvideo.vidiot.com>
Message-ID: <51F52874.1020502@gmail.com>

On 2013-07-28 4:50 AM, Mike Brown wrote:

>>
>> I guess you mean "couldn't care less"?
>
> I believe that the "not" should not be there.  I've heard it both ways.
> Not sure which is actuall correct.  To me it seems to be a double negative.
>

It's short for "I care so little that I couldn't care any less".

Imagine you were expecting someone to say "I couldn't agree more" but it came 
out as "I could agree more", wouldn't that grate on you?

No one cares about incorrect apostrophes or misplaced their/they're/there any 
more, but that one is particularly annoying because it pretty much says the 
opposite of what you're actually trying to say.

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



From onemda at gmail.com  Sun Jul 28 20:41:42 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sun, 28 Jul 2013 18:41:42 +0000
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org>
	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
	<1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P5gmzXSk0vh15+=o=Hfgd=W_4MTY1ZT44JwNMEbWz9mEQ@mail.gmail.com>

On 7/28/13, James Board <jpboard2 at yahoo.com> wrote:
>>> So if, for example, I wanted all of the audio samples that correspond to
>
>>> a specific frame, I might have to look at multiple audio packets, and
>>> then
>>> figure out which audio samples in each packet belong to which frame?
>>
>>Something like that, note that there are containers (not AVI) that actually
>> do
>>that.
>
> Thank you.  Which containers do this?  Which containers store all the
> audio packets for a single video frame in the same audio packet?  Or
> which containers make it easier to determine which audio sample belongs
> to whichever video?

Simple mxf files. Also MPEG-TS with SMPTE S302M audio.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user



From cehoyos at ag.or.at  Sun Jul 28 15:48:54 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 13:48:54 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com>
	<loom.20130728T030907-289@post.gmane.org>
	<51F4F638.6000201@gmail.com>
Message-ID: <loom.20130728T153837-674@post.gmane.org>

Andy Furniss <adf.lists <at> gmail.com> writes:

> Apologies for the indirect link, I changed ISP and 
> don't have webspace anymore - but if you want the 
> source (only 14 meg) it's here -

Don't worry!

> http://www.filedropper.com/h264-1080i

Tested while watching Budapest;-))

> There is a fast pan past some hoardings that shows 
> the issue well with  mplayer -vf yadif=1.
> 
> ffmpeg -i h264-1080i.ts -an -sn 
> -vf scale=720:576:interl=1,fieldorder=tff 
> -vcodec mpeg2video -flags +ildct+ilme -target pal-dvd 
> -aspect 16:9 -b:v 8000k tff.mpg

The following works fine here:

ffmpeg -i ~/Downloads/h264-1080i.ts -an -sn 
-vf scale=720:576:interl=1,setfield=tff 
-flags +ildct+ilme -target pal-dvd -aspect 16:9 
-b:v 8000k tff.mpg

I don't know why adding fieldorder=tff breaks encoding 
so horribly...

Carl Eugen

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



From jpboard2 at yahoo.com  Sun Jul 28 15:01:42 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 06:01:42 -0700 (PDT)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on
	a	1080p	60 Hz Computer Monitor
In-Reply-To: <loom.20130728T030129-819@post.gmane.org>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
Message-ID: <1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>



>> How does it work?? How does a video which has 24 fps 
>> get displayed on a TV with displays at 60 fps??
>
>(How is this related to FFmpeg? And I would expect 
>that the answer depends on your display system.)

This is a video question.? FFmpeg is a video project that
produces libraries and programs for handling video and audio.

Seems like someone here probably has thought through this.
_______________________________________________
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ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user



From cehoyos at ag.or.at  Sun Jul 28 18:27:24 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:27:24 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
Message-ID: <loom.20130728T182550-211@post.gmane.org>

James Ford <jamesmichaelford <at> gmail.com> writes:

> I am looking for a way to reset the rotation property 
> to zero so that the video will NOT  be rotated.

(Command line and complete, uncut console output missing.)

Please try -map_metadata -1

Carl Eugen

_______________________________________________
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ffmpeg-user at ffmpeg.org
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From basteon at gmail.com  Sun Jul 28 13:52:37 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:52:37 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
	<CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
Message-ID: <CAEJHKkUpNZgudHQdQxHAn0OX5LVCX0BqB-0PbJuNMnWduFVuFw@mail.gmail.com>

here,
Since 3yrs I work with them I found one available camera on the network.
I'll newer tell the ip, but that works for some axis camera with ffmpeg
version 0.10.3 ...
  Metadata:
    title           : Media Presentation
  Duration: N/A, start: 0.009063, bitrate: N/A
    Stream #0:0: Video: h264 (Baseline), yuvj420p, 800x450 [SAR 1:1 DAR
16:9], 30.08 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: aac, 16000 Hz, mono, s16
File '/tmp/out.avi' already exists. Overwrite ? [y/N] y
Incompatible pixel format 'yuvj420p' for codec 'mpeg4', auto-selecting
format 'yuv420p'
[buffer @ 0x276ce40] w:800 h:450 pixfmt:yuvj420p tb:1/1000000 sar:1/1
sws_param:
[buffersink @ 0x276c700] auto-inserting filter 'auto-inserted scale 0'
between the filter 'src' and the filter 'out'
[scale @ 0x276bde0] w:800 h:450 fmt:yuvj420p -> w:800 h:450 fmt:yuv420p
flags:0x4
Output #0, avi, to '/tmp/out.avi':
  Metadata:
    INAM            : Media Presentation
    ISFT            : Lavf53.32.100
    Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 800x450 [SAR
1:1 DAR 16:9], q=2-31, 200 kb/s, 30.08 tbn, 30.08 tbc
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 16000 Hz, mono, s16, 128
kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg4)
  Stream #0:1 -> #0:1 (aac -> libmp3lame)
Press [q] to stop, [?] for help
frame= 2298 fps= 23 q=31.0 Lsize=    4575kB time=00:01:36.84 bitrate=
387.0kbits/s dup=0 drop=63


On 28 July 2013 11:33, basteon <basteon at gmail.com> wrote:

> Hello,
> Some time ado I've been using axis cameras arm based. What I can see your
> ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
> the camera or PC? Surely I fix that by using v4l on the camera
>
>
> On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
>
>> Andy <aseto97 <at> yahoo.com> writes:
>>
>> > I did git checkout 2e4ec89 and my ffmpeg source is
>> > back to Sep.26,2012. Did the build, tried to stream
>> > the camera again, and got the same error.
>>
>> Your problem is apparently not the same as ticket 1734.
>>
>> > Do you have any idea why it is not working?
>>
>> No, but consider opening a ticket adding the -fdebug ts
>> output as requested by Michael in 1734.
>>
>> > --disable-optimizations
>>
>> Probably unrelated:
>> --disable-optimizations is a configure option that
>> should only be used in special circumstances (debugging
>> a crash that shows no useful backtrace with gdb or
>> speeding up compilation when bisecting), I suspect
>> you should remove it.
>>
>> Carl Eugen
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
>
_______________________________________________
ffmpeg-user mailing list
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http://ffmpeg.org/mailman/listinfo/ffmpeg-user


_______________________________________________
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ffmpeg-user at ffmpeg.org
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From jpboard2 at yahoo.com  Sun Jul 28 14:58:40 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 05:58:40 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<20130727183314.GA27349@phare.normalesup.org>	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
Message-ID: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> So if, for example, I wanted all of the audio samples that correspond to

>> a specific frame, I might have to look at multiple audio packets, and then
>> figure out which audio samples in each packet belong to which frame?
>
>Something like that, note that there are containers (not AVI) that actually do
>that.

Thank you.? Which containers do this?? Which containers store all the
audio packets for a single video frame in the same audio packet?? Or
which containers make it easier to determine which audio sample belongs
to whichever video?
_______________________________________________
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From andycivil at gmail.com  Sun Jul 28 16:19:32 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sun, 28 Jul 2013 10:19:32 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <20130728085026.GT27428@mrvideo.vidiot.com>
References: <loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>	<20130728021259.GQ27428@mrvideo.vidiot.com>	<51F4829C.6070902@gmail.com>	<20130728052848.GR27428@mrvideo.vidiot.com>	<51F4B4B3.6080003@gmail.com>
	<20130728085026.GT27428@mrvideo.vidiot.com>
Message-ID: <51F52874.1020502@gmail.com>

On 2013-07-28 4:50 AM, Mike Brown wrote:

>>
>> I guess you mean "couldn't care less"?
>
> I believe that the "not" should not be there.  I've heard it both ways.
> Not sure which is actuall correct.  To me it seems to be a double negative.
>

It's short for "I care so little that I couldn't care any less".

Imagine you were expecting someone to say "I couldn't agree more" but it came 
out as "I could agree more", wouldn't that grate on you?

No one cares about incorrect apostrophes or misplaced their/they're/there any 
more, but that one is particularly annoying because it pretty much says the 
opposite of what you're actually trying to say.

--
Andy
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From cehoyos at ag.or.at  Sun Jul 28 18:51:51 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:51:51 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
	<loom.20130728T182550-211@post.gmane.org>
Message-ID: <loom.20130728T185129-213@post.gmane.org>

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

> James Ford <jamesmichaelford <at> gmail.com> writes:
> 
> > I am looking for a way to reset the rotation property 
> > to zero so that the video will NOT  be rotated.

> Please try -map_metadata -1

Probably even better:
-metadata rotate=0

Carl Eugen

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



From cehoyos at ag.or.at  Sun Jul 28 18:27:24 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:27:24 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
Message-ID: <loom.20130728T182550-211@post.gmane.org>

James Ford <jamesmichaelford <at> gmail.com> writes:

> I am looking for a way to reset the rotation property 
> to zero so that the video will NOT  be rotated.

(Command line and complete, uncut console output missing.)

Please try -map_metadata -1

Carl Eugen

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


_______________________________________________
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ffmpeg-user at ffmpeg.org
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From jpboard2 at yahoo.com  Sun Jul 28 15:01:42 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 06:01:42 -0700 (PDT)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video
	on	a	1080p	60 Hz Computer Monitor
In-Reply-To: <loom.20130728T030129-819@post.gmane.org>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
Message-ID: <1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>



>> How does it work?? How does a video which has 24 fps 
>> get displayed on a TV with displays at 60 fps??
>
>(How is this related to FFmpeg? And I would expect 
>that the answer depends on your display system.)

This is a video question.? FFmpeg is a video project that
produces libraries and programs for handling video and audio.

Seems like someone here probably has thought through this.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user


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



From cehoyos at ag.or.at  Sun Jul 28 15:48:54 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 13:48:54 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com>
	<loom.20130728T030907-289@post.gmane.org>
	<51F4F638.6000201@gmail.com>
Message-ID: <loom.20130728T153837-674@post.gmane.org>

Andy Furniss <adf.lists <at> gmail.com> writes:

> Apologies for the indirect link, I changed ISP and 
> don't have webspace anymore - but if you want the 
> source (only 14 meg) it's here -

Don't worry!

> http://www.filedropper.com/h264-1080i

Tested while watching Budapest;-))

> There is a fast pan past some hoardings that shows 
> the issue well with  mplayer -vf yadif=1.
> 
> ffmpeg -i h264-1080i.ts -an -sn 
> -vf scale=720:576:interl=1,fieldorder=tff 
> -vcodec mpeg2video -flags +ildct+ilme -target pal-dvd 
> -aspect 16:9 -b:v 8000k tff.mpg

The following works fine here:

ffmpeg -i ~/Downloads/h264-1080i.ts -an -sn 
-vf scale=720:576:interl=1,setfield=tff 
-flags +ildct+ilme -target pal-dvd -aspect 16:9 
-b:v 8000k tff.mpg

I don't know why adding fieldorder=tff breaks encoding 
so horribly...

Carl Eugen

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From basteon at gmail.com  Sun Jul 28 13:33:56 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:33:56 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <loom.20130718T095657-197@post.gmane.org>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
Message-ID: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>

Hello,
Some time ado I've been using axis cameras arm based. What I can see your
ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
the camera or PC? Surely I fix that by using v4l on the camera


On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Andy <aseto97 <at> yahoo.com> writes:
>
> > I did git checkout 2e4ec89 and my ffmpeg source is
> > back to Sep.26,2012. Did the build, tried to stream
> > the camera again, and got the same error.
>
> Your problem is apparently not the same as ticket 1734.
>
> > Do you have any idea why it is not working?
>
> No, but consider opening a ticket adding the -fdebug ts
> output as requested by Michael in 1734.
>
> > --disable-optimizations
>
> Probably unrelated:
> --disable-optimizations is a configure option that
> should only be used in special circumstances (debugging
> a crash that shows no useful backtrace with gdb or
> speeding up compilation when bisecting), I suspect
> you should remove it.
>
> Carl Eugen
>
> _______________________________________________
> 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


_______________________________________________
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ffmpeg-user at ffmpeg.org
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From cehoyos at ag.or.at  Sun Jul 28 18:30:52 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:30:52 +0000 (UTC)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on
	a	1080p	60 Hz Computer Monitor
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
	<1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <loom.20130728T182747-704@post.gmane.org>

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

> >> How does it work?? How does a video which has 24 fps 
> >> get displayed on a TV with displays at 60 fps??
> >
> >(How is this related to FFmpeg? And I would expect 
> >that the answer depends on your display system.)
> 
> This is a video question.? FFmpeg is a video project that
> produces libraries and programs for handling video and audio.

Sure.
And somebody may be able to answer if James added information 
about his operating system, his video hardware, his 
video drivers, his X / Windows / OSX display settings and his 
display.

Carl Eugen

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From onemda at gmail.com  Sun Jul 28 20:41:42 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sun, 28 Jul 2013 18:41:42 +0000
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org>
	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
	<1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P5gmzXSk0vh15+=o=Hfgd=W_4MTY1ZT44JwNMEbWz9mEQ@mail.gmail.com>

On 7/28/13, James Board <jpboard2 at yahoo.com> wrote:
>>> So if, for example, I wanted all of the audio samples that correspond to
>
>>> a specific frame, I might have to look at multiple audio packets, and
>>> then
>>> figure out which audio samples in each packet belong to which frame?
>>
>>Something like that, note that there are containers (not AVI) that actually
>> do
>>that.
>
> Thank you.  Which containers do this?  Which containers store all the
> audio packets for a single video frame in the same audio packet?  Or
> which containers make it easier to determine which audio sample belongs
> to whichever video?

Simple mxf files. Also MPEG-TS with SMPTE S302M audio.
_______________________________________________
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From onemda at gmail.com  Sun Jul 28 20:41:42 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sun, 28 Jul 2013 18:41:42 +0000
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org>
	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
	<1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P5gmzXSk0vh15+=o=Hfgd=W_4MTY1ZT44JwNMEbWz9mEQ@mail.gmail.com>

On 7/28/13, James Board <jpboard2 at yahoo.com> wrote:
>>> So if, for example, I wanted all of the audio samples that correspond to
>
>>> a specific frame, I might have to look at multiple audio packets, and
>>> then
>>> figure out which audio samples in each packet belong to which frame?
>>
>>Something like that, note that there are containers (not AVI) that actually
>> do
>>that.
>
> Thank you.  Which containers do this?  Which containers store all the
> audio packets for a single video frame in the same audio packet?  Or
> which containers make it easier to determine which audio sample belongs
> to whichever video?

Simple mxf files. Also MPEG-TS with SMPTE S302M audio.
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user


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


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



From basteon at gmail.com  Sun Jul 28 13:33:56 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:33:56 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <loom.20130718T095657-197@post.gmane.org>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
Message-ID: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>

Hello,
Some time ado I've been using axis cameras arm based. What I can see your
ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
the camera or PC? Surely I fix that by using v4l on the camera


On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Andy <aseto97 <at> yahoo.com> writes:
>
> > I did git checkout 2e4ec89 and my ffmpeg source is
> > back to Sep.26,2012. Did the build, tried to stream
> > the camera again, and got the same error.
>
> Your problem is apparently not the same as ticket 1734.
>
> > Do you have any idea why it is not working?
>
> No, but consider opening a ticket adding the -fdebug ts
> output as requested by Michael in 1734.
>
> > --disable-optimizations
>
> Probably unrelated:
> --disable-optimizations is a configure option that
> should only be used in special circumstances (debugging
> a crash that shows no useful backtrace with gdb or
> speeding up compilation when bisecting), I suspect
> you should remove it.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
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From jpboard2 at yahoo.com  Sun Jul 28 14:58:40 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 05:58:40 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<20130727183314.GA27349@phare.normalesup.org>	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
Message-ID: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> So if, for example, I wanted all of the audio samples that correspond to

>> a specific frame, I might have to look at multiple audio packets, and then
>> figure out which audio samples in each packet belong to which frame?
>
>Something like that, note that there are containers (not AVI) that actually do
>that.

Thank you.? Which containers do this?? Which containers store all the
audio packets for a single video frame in the same audio packet?? Or
which containers make it easier to determine which audio sample belongs
to whichever video?
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From pb at das-werkstatt.com  Sun Jul 28 22:54:28 2013
From: pb at das-werkstatt.com (Peter B.)
Date: Sun, 28 Jul 2013 22:54:28 +0200
Subject: [FFmpeg-user] FFmpeg + Mini-supercomputer "Parallela"?
Message-ID: <51F58504.7030805@das-werkstatt.com>

Hello everyone,

I've recently been told about the "Parallela: Supercomputing for
everyone" [1]:
An Open Hardware mini-supercomputer for 99$. It's equipped with 2 ARM
CPUs and 16 RISC Co-Processors (even with floating point unit).

Has anyone here heard of that project and could give me any feedback on
whether ffmpeg will run on these computers?


Thanks,
Pb

== References:
[1] http://www.parallella.org/

From cehoyos at ag.or.at  Sun Jul 28 15:48:54 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 13:48:54 +0000 (UTC)
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
References: <51F2126E.40200@gmail.com>
	<loom.20130726T104114-810@post.gmane.org>
	<51F369D6.8090704@gmail.com>
	<loom.20130727T104545-441@post.gmane.org>
	<51F439B8.2050807@gmail.com>
	<CAPYw7P5YU1EK79yh1AXAA6owAB=QNaGiMEgBwfqA9FGHwRE9sg@mail.gmail.com>
	<51F44A0D.2030302@gmail.com>
	<loom.20130728T030907-289@post.gmane.org>
	<51F4F638.6000201@gmail.com>
Message-ID: <loom.20130728T153837-674@post.gmane.org>

Andy Furniss <adf.lists <at> gmail.com> writes:

> Apologies for the indirect link, I changed ISP and 
> don't have webspace anymore - but if you want the 
> source (only 14 meg) it's here -

Don't worry!

> http://www.filedropper.com/h264-1080i

Tested while watching Budapest;-))

> There is a fast pan past some hoardings that shows 
> the issue well with  mplayer -vf yadif=1.
> 
> ffmpeg -i h264-1080i.ts -an -sn 
> -vf scale=720:576:interl=1,fieldorder=tff 
> -vcodec mpeg2video -flags +ildct+ilme -target pal-dvd 
> -aspect 16:9 -b:v 8000k tff.mpg

The following works fine here:

ffmpeg -i ~/Downloads/h264-1080i.ts -an -sn 
-vf scale=720:576:interl=1,setfield=tff 
-flags +ildct+ilme -target pal-dvd -aspect 16:9 
-b:v 8000k tff.mpg

I don't know why adding fieldorder=tff breaks encoding 
so horribly...

Carl Eugen

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From cehoyos at ag.or.at  Sun Jul 28 18:27:24 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:27:24 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
Message-ID: <loom.20130728T182550-211@post.gmane.org>

James Ford <jamesmichaelford <at> gmail.com> writes:

> I am looking for a way to reset the rotation property 
> to zero so that the video will NOT  be rotated.

(Command line and complete, uncut console output missing.)

Please try -map_metadata -1

Carl Eugen

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From jpboard2 at yahoo.com  Sun Jul 28 15:01:42 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 06:01:42 -0700 (PDT)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps
	video	on	a	1080p	60 Hz Computer Monitor
In-Reply-To: <loom.20130728T030129-819@post.gmane.org>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
Message-ID: <1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>



>> How does it work?? How does a video which has 24 fps 
>> get displayed on a TV with displays at 60 fps??
>
>(How is this related to FFmpeg? And I would expect 
>that the answer depends on your display system.)

This is a video question.? FFmpeg is a video project that
produces libraries and programs for handling video and audio.

Seems like someone here probably has thought through this.
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From basteon at gmail.com  Sun Jul 28 13:52:37 2013
From: basteon at gmail.com (basteon)
Date: Sun, 28 Jul 2013 11:52:37 +0000
Subject: [FFmpeg-user] Problems dumping H264 stream from an Axis camera
In-Reply-To: <CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
References: <1374091803.51199.YahooMailNeo@web121803.mail.ne1.yahoo.com>
	<loom.20130717T231205-344@post.gmane.org>
	<1374096453409-4660062.post@n4.nabble.com>
	<1374100341199-4660063.post@n4.nabble.com>
	<loom.20130718T095657-197@post.gmane.org>
	<CAEJHKkWAzHyMGshhEW6K+0odhX65eY5WmCUkJ1RP1CUcnPfeDQ@mail.gmail.com>
Message-ID: <CAEJHKkUpNZgudHQdQxHAn0OX5LVCX0BqB-0PbJuNMnWduFVuFw@mail.gmail.com>

here,
Since 3yrs I work with them I found one available camera on the network.
I'll newer tell the ip, but that works for some axis camera with ffmpeg
version 0.10.3 ...
  Metadata:
    title           : Media Presentation
  Duration: N/A, start: 0.009063, bitrate: N/A
    Stream #0:0: Video: h264 (Baseline), yuvj420p, 800x450 [SAR 1:1 DAR
16:9], 30.08 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: aac, 16000 Hz, mono, s16
File '/tmp/out.avi' already exists. Overwrite ? [y/N] y
Incompatible pixel format 'yuvj420p' for codec 'mpeg4', auto-selecting
format 'yuv420p'
[buffer @ 0x276ce40] w:800 h:450 pixfmt:yuvj420p tb:1/1000000 sar:1/1
sws_param:
[buffersink @ 0x276c700] auto-inserting filter 'auto-inserted scale 0'
between the filter 'src' and the filter 'out'
[scale @ 0x276bde0] w:800 h:450 fmt:yuvj420p -> w:800 h:450 fmt:yuv420p
flags:0x4
Output #0, avi, to '/tmp/out.avi':
  Metadata:
    INAM            : Media Presentation
    ISFT            : Lavf53.32.100
    Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 800x450 [SAR
1:1 DAR 16:9], q=2-31, 200 kb/s, 30.08 tbn, 30.08 tbc
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 16000 Hz, mono, s16, 128
kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg4)
  Stream #0:1 -> #0:1 (aac -> libmp3lame)
Press [q] to stop, [?] for help
frame= 2298 fps= 23 q=31.0 Lsize=    4575kB time=00:01:36.84 bitrate=
387.0kbits/s dup=0 drop=63


On 28 July 2013 11:33, basteon <basteon at gmail.com> wrote:

> Hello,
> Some time ado I've been using axis cameras arm based. What I can see your
> ffmpeg build for ppc platform.(--arch=ppc --cpu=e500v2) Is that you run on
> the camera or PC? Surely I fix that by using v4l on the camera
>
>
> On 18 July 2013 08:00, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
>
>> Andy <aseto97 <at> yahoo.com> writes:
>>
>> > I did git checkout 2e4ec89 and my ffmpeg source is
>> > back to Sep.26,2012. Did the build, tried to stream
>> > the camera again, and got the same error.
>>
>> Your problem is apparently not the same as ticket 1734.
>>
>> > Do you have any idea why it is not working?
>>
>> No, but consider opening a ticket adding the -fdebug ts
>> output as requested by Michael in 1734.
>>
>> > --disable-optimizations
>>
>> Probably unrelated:
>> --disable-optimizations is a configure option that
>> should only be used in special circumstances (debugging
>> a crash that shows no useful backtrace with gdb or
>> speeding up compilation when bisecting), I suspect
>> you should remove it.
>>
>> Carl Eugen
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
>
_______________________________________________
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http://ffmpeg.org/mailman/listinfo/ffmpeg-user


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From cehoyos at ag.or.at  Sun Jul 28 18:30:52 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:30:52 +0000 (UTC)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video
	on	a	1080p	60 Hz Computer Monitor
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
	<1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <loom.20130728T182747-704@post.gmane.org>

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

> >> How does it work?? How does a video which has 24 fps 
> >> get displayed on a TV with displays at 60 fps??
> >
> >(How is this related to FFmpeg? And I would expect 
> >that the answer depends on your display system.)
> 
> This is a video question.? FFmpeg is a video project that
> produces libraries and programs for handling video and audio.

Sure.
And somebody may be able to answer if James added information 
about his operating system, his video hardware, his 
video drivers, his X / Windows / OSX display settings and his 
display.

Carl Eugen

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From cehoyos at ag.or.at  Sun Jul 28 18:51:51 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 16:51:51 +0000 (UTC)
Subject: [FFmpeg-user] Rotation Property
References: <CA+O03eYucqggSJ2mC+ONV1hFp8s2tRfmvPodYHcogWRqPZXoRg@mail.gmail.com>
	<loom.20130728T182550-211@post.gmane.org>
Message-ID: <loom.20130728T185129-213@post.gmane.org>

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

> James Ford <jamesmichaelford <at> gmail.com> writes:
> 
> > I am looking for a way to reset the rotation property 
> > to zero so that the video will NOT  be rotated.

> Please try -map_metadata -1

Probably even better:
-metadata rotate=0

Carl Eugen

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From onemda at gmail.com  Sun Jul 28 20:41:42 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Sun, 28 Jul 2013 18:41:42 +0000
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<20130727183314.GA27349@phare.normalesup.org>
	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
	<1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P5gmzXSk0vh15+=o=Hfgd=W_4MTY1ZT44JwNMEbWz9mEQ@mail.gmail.com>

On 7/28/13, James Board <jpboard2 at yahoo.com> wrote:
>>> So if, for example, I wanted all of the audio samples that correspond to
>
>>> a specific frame, I might have to look at multiple audio packets, and
>>> then
>>> figure out which audio samples in each packet belong to which frame?
>>
>>Something like that, note that there are containers (not AVI) that actually
>> do
>>that.
>
> Thank you.  Which containers do this?  Which containers store all the
> audio packets for a single video frame in the same audio packet?  Or
> which containers make it easier to determine which audio sample belongs
> to whichever video?

Simple mxf files. Also MPEG-TS with SMPTE S302M audio.
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From jpboard2 at yahoo.com  Sun Jul 28 14:58:40 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 05:58:40 -0700 (PDT)
Subject: [FFmpeg-user] Audio Packets, Video Packets, md5 output
In-Reply-To: <CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
References: <1374945556.31898.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<20130727183314.GA27349@phare.normalesup.org>	<1374955886.351.YahooMailNeo@web164705.mail.gq1.yahoo.com>
	<CAPYw7P41PAL2D1OZ79f8RbakqY_i9Ean03EK=FOp3SnQQB_WhQ@mail.gmail.com>
Message-ID: <1375016320.97084.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> So if, for example, I wanted all of the audio samples that correspond to

>> a specific frame, I might have to look at multiple audio packets, and then
>> figure out which audio samples in each packet belong to which frame?
>
>Something like that, note that there are containers (not AVI) that actually do
>that.

Thank you.? Which containers do this?? Which containers store all the
audio packets for a single video frame in the same audio packet?? Or
which containers make it easier to determine which audio sample belongs
to whichever video?
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From pb at das-werkstatt.com  Sun Jul 28 22:54:28 2013
From: pb at das-werkstatt.com (Peter B.)
Date: Sun, 28 Jul 2013 22:54:28 +0200
Subject: [FFmpeg-user]  FFmpeg + Mini-supercomputer "Parallela"?
Message-ID: <51F58504.7030805@das-werkstatt.com>

Hello everyone,

I've recently been told about the "Parallela: Supercomputing for
everyone" [1]:
An Open Hardware mini-supercomputer for 99$. It's equipped with 2 ARM
CPUs and 16 RISC Co-Processors (even with floating point unit).

Has anyone here heard of that project and could give me any feedback on
whether ffmpeg will run on these computers?


Thanks,
Pb

== References:
[1] http://www.parallella.org/
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
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From jpboard2 at yahoo.com  Sun Jul 28 23:15:22 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 14:15:22 -0700 (PDT)
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a
	1080p	60 Hz Computer Monitor
In-Reply-To: <loom.20130728T182747-704@post.gmane.org>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>	<loom.20130728T030129-819@post.gmane.org>	<1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<loom.20130728T182747-704@post.gmane.org>
Message-ID: <1375046122.2209.YahooMailNeo@web164705.mail.gq1.yahoo.com>

> >> How does it work?? How does a video which has 24 fps 

> >> get displayed on a TV with displays at 60 fps??

>And somebody may be able to answer if James added information 
>about his operating system, his video hardware, his 
>video drivers, his X / Windows / OSX display settings and his 
>display.

I'm running CentOS Linux.? I have a modern Nvidia graphics card
and I set the vsync to the LCD TV display, where the video
is being played.? The LCD TV is a no-name brand.

I'm wondering if someone dealt with this issue before.
You need to inverse telecine to get rid of the combing
effects, but when you do that, your video is now at 24
fps, not 30 fps.? This makes motion un-smooth since most
displays are a multiple of 30 Hz.? Surely that issue has
been discussed, right?

Do some displays allow you to set 24 Hz or a multiple
of 24 Hz?

Thanks for all your help.

From jpboard2 at yahoo.com  Sun Jul 28 23:22:43 2013
From: jpboard2 at yahoo.com (James Board)
Date: Sun, 28 Jul 2013 14:22:43 -0700 (PDT)
Subject: [FFmpeg-user] Set audio to null
Message-ID: <1375046563.53850.YahooMailNeo@web164704.mail.gq1.yahoo.com>

Hi, this is an easy question, but I could not find an
answer in the ffmpeg documentation (which is huge!).
Can I set the audio stream to null with ffmpeg?? This would
be the equivalent of '-ao null' in mencoder?

Thank you!

ps.? I apologize in advance for whatever I did wrong.

From cehoyos at ag.or.at  Sun Jul 28 23:26:29 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 21:26:29 +0000 (UTC)
Subject: [FFmpeg-user] Set audio to null
References: <1375046563.53850.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <loom.20130728T232552-299@post.gmane.org>

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

> Hi, this is an easy question, but I could not find an
> answer in the ffmpeg documentation (which is huge!).

http://ffmpeg.org/ffmpeg-all.html#Stream-selection

Carl Eugen


From cehoyos at ag.or.at  Sun Jul 28 23:30:13 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 28 Jul 2013 21:30:13 +0000 (UTC)
Subject: [FFmpeg-user] FFmpeg + Mini-supercomputer "Parallela"?
References: <51F58504.7030805@das-werkstatt.com>
Message-ID: <loom.20130728T232710-991@post.gmane.org>

Peter B. <pb <at> das-werkstatt.com> writes:

> An Open Hardware mini-supercomputer for 99$. It's 
> equipped with 2 ARM CPUs and 16 RISC Co-Processors 
> (even with floating point unit).

I suspect this will be spectacularly slow running 
FFmpeg compared to any new x86_64.

> Has anyone here heard of that project and could 
> give me any feedback on whether ffmpeg will run on 
> these computers?

Only if the system is supported by gcc or another 
C89 / C99 compiler.

Carl Eugen


From andycivil at gmail.com  Sun Jul 28 23:40:00 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sun, 28 Jul 2013 17:40:00 -0400
Subject: [FFmpeg-user] Set audio to null
In-Reply-To: <1375046563.53850.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375046563.53850.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <51F58FB0.8030808@gmail.com>

On 2013-07-28 5:22 PM, James Board wrote:

> ... the equivalent of '-ao null' in mencoder?

'-an'

--
Andy

From brown at mrvideo.vidiot.com  Mon Jul 29 03:20:47 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Sun, 28 Jul 2013 20:20:47 -0500
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F52874.1020502@gmail.com>
References: <loom.20130727T104545-441@post.gmane.org>
	<51F4370D.3050204@gmail.com>
	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
	<20130728021259.GQ27428@mrvideo.vidiot.com>
	<51F4829C.6070902@gmail.com>
	<20130728052848.GR27428@mrvideo.vidiot.com>
	<51F4B4B3.6080003@gmail.com>
	<20130728085026.GT27428@mrvideo.vidiot.com>
	<51F52874.1020502@gmail.com>
Message-ID: <20130729012047.GZ27428@mrvideo.vidiot.com>

On Sun, Jul 28, 2013 at 10:19:32AM -0400, Andy Civil wrote:
> No one cares about incorrect apostrophes or misplaced their/they're/there 
> any more,

I do care when someone messes up the their/they're/there, and others of
its kind.  We are supposed to be an educated world, yet it happens constantly.

But, that is off-topic.

As a test, have you sent both versions of a disc overseas, i.e., a TFF and
a BFF, to if they'll play?

MB
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From brown at mrvideo.vidiot.com  Mon Jul 29 03:28:47 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Sun, 28 Jul 2013 20:28:47 -0500
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a
	1080p	60 Hz Computer Monitor
In-Reply-To: <1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<loom.20130727T192450-485@post.gmane.org>
	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<loom.20130728T030129-819@post.gmane.org>
	<1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
Message-ID: <20130729012847.GA27428@mrvideo.vidiot.com>

On Sun, Jul 28, 2013 at 06:01:42AM -0700, James Board wrote:
> 
> >> How does it work?? How does a video which has 24 fps 
> >> get displayed on a TV with displays at 60 fps??
> >
> >(How is this related to FFmpeg? And I would expect 
> >that the answer depends on your display system.)
> 
> This is a video question.? FFmpeg is a video project that
> produces libraries and programs for handling video and audio.
> 
> Seems like someone here probably has thought through this.

While it is indeed a video question, the playback of 23.976 fps video has
nothing to do with ffmpeg.  It is handled by the playing device and the
display device.

MB
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From andycivil at gmail.com  Mon Jul 29 03:38:23 2013
From: andycivil at gmail.com (Andy Civil)
Date: Sun, 28 Jul 2013 21:38:23 -0400
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <20130729012047.GZ27428@mrvideo.vidiot.com>
References: <loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>	<51F47A53.40404@gmail.com>	<20130728021259.GQ27428@mrvideo.vidiot.com>	<51F4829C.6070902@gmail.com>	<20130728052848.GR27428@mrvideo.vidiot.com>	<51F4B4B3.6080003@gmail.com>	<20130728085026.GT27428@mrvideo.vidiot.com>	<51F52874.1020502@gmail.com>
	<20130729012047.GZ27428@mrvideo.vidiot.com>
Message-ID: <51F5C78F.8050401@gmail.com>

On 2013-07-28 9:20 PM, Mike Brown wrote:
n
>
> As a test, have you sent both versions of a disc overseas, i.e., a TFF and
> a BFF, to if they'll play?

I haven't, but it's a good idea. Yet I don't want to burden them with tests - or 
long explanations as to why there are two DVDs with the same content - so I'll 
do the test myself next time I travel.

--
Andy

From bobm-ffmpeg at burner.com  Mon Jul 29 05:35:46 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Sun, 28 Jul 2013 21:35:46 -0600
Subject: [FFmpeg-user] OT: duplicate emails
Message-ID: <51F5E312.4020404@burner.com>

Is it just me, or is the list going nuts and sending like 4 and 5 copies
of the same message to everyone the last couple of days?  I don't think
it's a local problem to me as it seems like the mailing list footer gets
put on another time for each copy of the message I get..


From andycivil at gmail.com  Mon Jul 29 06:01:56 2013
From: andycivil at gmail.com (Andy Civil)
Date: Mon, 29 Jul 2013 00:01:56 -0400
Subject: [FFmpeg-user] OT: duplicate emails
In-Reply-To: <51F5E312.4020404@burner.com>
References: <51F5E312.4020404@burner.com>
Message-ID: <51F5E934.8010906@gmail.com>

On 2013-07-28 11:35 PM, Bob Maple wrote:
> Is it just me, or is the list going nuts and sending like 4 and 5 copies
> of the same message to everyone the last couple of days?  I don't think
> it's a local problem to me as it seems like the mailing list footer gets
> put on another time for each copy of the message I get..

Only getting one copy here...

--
Andy

From brown at mrvideo.vidiot.com  Mon Jul 29 07:03:55 2013
From: brown at mrvideo.vidiot.com (Mike Brown)
Date: Mon, 29 Jul 2013 00:03:55 -0500
Subject: [FFmpeg-user] OT: duplicate emails
In-Reply-To: <51F5E312.4020404@burner.com>
References: <51F5E312.4020404@burner.com>
Message-ID: <20130729050355.GB27428@mrvideo.vidiot.com>

On Sun, Jul 28, 2013 at 09:35:46PM -0600, Bob Maple wrote:
> Is it just me, or is the list going nuts and sending like 4 and 5 copies
> of the same message to everyone the last couple of days?  I don't think
> it's a local problem to me as it seems like the mailing list footer gets
> put on another time for each copy of the message I get..

It wasn't just you.

MB
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From jiachielee at live.com  Sun Jul 28 17:46:54 2013
From: jiachielee at live.com (thljcl)
Date: Sun, 28 Jul 2013 08:46:54 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
Message-ID: <1375026414031-4660271.post@n4.nabble.com>

When it comes to quality preservation, video editing, or lossless encoding;
yes, I would agree that ?yuv444p? makes plenty of sense. ?yuv444p? is a way
of re-encoding the entire 24-bit RGB information, which all color
information is preserved without any loss due to chroma subsampling. Lossy
encoding would cause the ?generational loss? of quality. Chroma subsampling
or any other lossy encoding algorithms does not make sense for repeating
video editing process.
For video/film producers, it makes sense to keep the master copy of the
videos, for future editing or remastering process. That said, videos often
undergo lossy compression before commercial or non-commercial public
distribution. Lossy compression is done to achieve smaller file size, at the
expense of image quality. Lossy compression that is done over high-quality
source can preserve fairly reasonably high quality.
Ffmpeg is not the only tool I ever use. Most tools would have presets for
lossy compression, without giving us an option to choose our own settings.
Since most videos being distributed have undergone lossy compression of
certain levels, it makes sense for the many media players support those
popular formats. 
If the sources have already underwent lossy compression, ffmpeg will choose
the ?same color space? by default. For example, it would choose YUV420p by
default for JPEG image as sources. If the sources are PNG image, it would
choose YUV444p. Of course, the user can choose his own pixel format by
-pix_fmt. 
There are free tools widely available to play H.264 encoded in YUV444p, such
as Media Player Classic Home Cinema included in K-Lite Codec Pack Mega.
Windows Media Player can play H.264 encoded in YUV444p with K-Lite Codec
Pack Mega. I?m not familiar with Mac OS X.
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. My personal preference is that I would like to play highest-quality
video possible.




--
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From jiachielee at live.com  Sun Jul 28 21:55:54 2013
From: jiachielee at live.com (thljcl)
Date: Sun, 28 Jul 2013 12:55:54 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
Message-ID: <1375041354497-4660277.post@n4.nabble.com>

Create slide show from still images is not exactly as simple as I originally
anticipated. After knowing that ffmpeg is able to convert still images into
video of lossless compression using encoders such as x264, I have a basic
idea of how to create a slide show from a bunch of still images. In a way,
creating slide show is exactly the opposite of creating the illusion of
moving pictures. Human eyes perceive motion when the similar images are
being displayed in sequence rapidly. For a slide show, however, each image
or slide is being displayed for a couple of seconds instead of 1/24 seconds.
Let?s start from the very beginning.
 The basic command to convert a couple of still images to H.264 lossless
video is:
ffmpeg -r "24" -i "f_%%1d.png" -vcodec "libx264" -crf "0" "output.mkv"
The input frames are being set to the frame rate of 24 FPS.
f_%%1d.png represents images with the file name starting with f_1.png,
f_2.png, f_3.png ?
In Windows 8, you select a couple of images with whatever file names, when
you rename those files together, they would be renamed to filename (1).png,
filename (2).png?
Each frame would be displayed for 1/24 second.
On ffmpeg wiki at http://bit.ly/14YRtSv, there is a guide for creating a
video slideshow with ffmpeg. If that guide works perfectly for me, I would
not bother to write here.
Say if you want to create a slide show with each slide to be displayed for 5
seconds, one way that would work, though not practical, is to manually
duplicate the same image. You would need 120 copies of the same image for
the same slide to be displayed for 5 seconds at 24 frame per second. 
The problem with the ffmpeg wiki?s guide is that ffmpeg drops frames,
especially at the beginning or ending of the videos. Dropping frames causes
the dropped slide or changed length. What I want to do is to dictate how
ffmpeg duplicate frames automatically as I want.
Here I have 135 images at http://sdrv.ms/11onHnh. Among them, 9 of them have
the resolution of 3456 ? 1080 pixels; the rest of them have the resolution
of 1920 ? 1080 pixels. To create a slideshow of 1280 ? 720 pixels, I need to
resize all of them to have the same resolution.

ffmpeg -r "1" -i "wallpapers\sources\1_%%1d.jpg" -pix_fmt "yuv444p" -vcodec
"libx264" -crf "0" "wallpapers\1.mkv"
ffmpeg -r "1" -i "wallpapers\sources\2_%%1d.jpg" -pix_fmt "yuv444p" -vf
"scale=1920:600,pad=1920:1080:0:240" -vcodec "libx264" -crf "0"
"wallpapers\2.mkv"
ffmpeg -r "1" -i "wallpapers\sources\3_%%1d.jpg" -pix_fmt "yuv444p" -vcodec
"libx264" -crf "0" "wallpapers\3.mkv"

I merge the three videos using mkvtoolnix. I extract frames from the
combined video.

mkdir "wallpapers\frames"
ffmpeg -i "wallpapers\t.mkv" -vf "scale=1280:720" "wallpapers\frames\f
(%%1d).png"

>From individual frames, I wrote a batch file to duplicate frames and create
a slideshow (H.264 lossless). Each slide is to be displayed for 10 seconds.

cd /d "wallpapers\frames"
copy "f (135).png" "f (136).png"
cd /d "%programfiles%\ffmpeg\bin"
ffmpeg -r "1" -i "wallpapers\frames\f (%%1d).png" -vcodec "libx264" -crf "0"
-vf "fps=10" "wallpapers\1.mkv"
cd /d "wallpapers\frames"
del "f (136).png"
cd /d "%programfiles%\ffmpeg\bin"
ffmpeg -r "1" -i "wallpapers\1.mkv" -vf "fps=24" -vcodec "libx264" -crf "0"
"wallpapers\2.mkv"
cd /d "wallpapers"
del "1.mkv"
ren "2.mkv" "slideshow.mkv"
cd /d "%programfiles%\ffmpeg\bin"

In duplicating the frames, ffmpeg will ignore the last frame. That?s why I
duplicate the last frame. Likewise, I can extract all the frames from the
last show. Ffmpeg will duplicate the first frame. Similarly, I delete the
first frame. Below is the lines I wrote to extract the frames from the
slideshow:

ffmpeg -i "wallpapers\slideshow.mkv" -vf "fps=1" -vcodec "libx264" -crf "0"
"wallpapers\1.mkv"
ffmpeg -r "10" -i "wallpapers\1.mkv" -vf "fps=1" -vcodec "libx264" -crf "0"
"wallpapers\2.mkv"
cd /d "wallpapers"
del "1.mkv"
ren "2.mkv" "1.mkv"
cd /d "%programfiles%\ffmpeg\bin"
mkdir "wallpapers\frames_1"
ffmpeg -i "wallpapers\1.mkv" "wallpapers\frames_1\f_%%1d.png"
cd /d "wallpapers\frames_1"
del "f_1.png"
cd /d "%programfiles%\ffmpeg\bin\wallpapers"
del "1.mkv"
cd /d "%programfiles%\ffmpeg\bin"
ffmpeg -r "1" -i "wallpapers\frames_1\f_%%1d.png" -vcodec "libx264" -crf "0"
"wallpapers\1.mkv"
rmdir /s /q "wallpapers\frames_1"
mkdir "wallpapers\frames_1"
ffmpeg -i "wallpapers\1.mkv" "wallpapers\frames_1\f_%%1d.png"
cd /d "wallpapers"
del "1.mkv"
cd /d "%programfiles%\ffmpeg\bin"

The slideshow I created is at http://sdrv.ms/15Xa2TU.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277.html
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From francois.visagie at gmail.com  Mon Jul 29 08:27:39 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Mon, 29 Jul 2013 08:27:39 +0200
Subject: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
In-Reply-To: <51F47A53.40404@gmail.com>
References: <51F2126E.40200@gmail.com>	<loom.20130726T104114-810@post.gmane.org>	<51F369D6.8090704@gmail.com>	<loom.20130727T104545-441@post.gmane.org>	<51F4370D.3050204@gmail.com>	<20130728011658.GP27428@mrvideo.vidiot.com>
	<51F47A53.40404@gmail.com>
Message-ID: <003601ce8c24$bebaaf60$3c300e20$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Andy Civil
> Sent: 28 July 2013 03:57
> To: FFmpeg user questions
> Subject: Re: [FFmpeg-user] ntsc-dvd doesn't default to interlaced.
> 
> On 2013-07-27 9:16 PM, Mike Brown wrote:
> > On Sat, Jul 27, 2013 at 05:09:33PM -0400, Andy Civil wrote:
> >> The way I'm analysing my output, is that I view it in VLC. What
> >> should happen is that I see the 'comb' effect on any moving edges,
> >> but if I choose to turn de-interlacing on, that comb effect disappears.
> >
> > With interlaced video, you better see the "comb" effect with motion
video.
> > But, there is fine print: 1) if the source was from pure interlaced
> > video, i.e., a live event; 2) the source is 2:3 pulldown video, where
> > 3 frames will not have the effect and 2 frames will.
> >
> > You won't have the effect if the source video was progressive.  You
> > might not get the effect if converting 25 fps source to 29.97 fps.
> > The 25 fps source might be progressive in nature (Doctor Who for
> > example).  It depends on how the conversion to 29.97 fps is done and if
> told to do interlaced.
> 
> The video starts from my Sony HX5 as 1920 x 1080 x 30i. It's then
deinterlaced
> by a commercial video editor to 1920 x 1080 x 60p. Every frame is
different.

In that case (converting HD to SD), you may want to consider converting from
BT.709 to BT.601 colour primaries (as earlier suggested by Andy Furniss).
The effect is subtle; it's a matter of how "perfectly" you want to do
things.

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


From onemda at gmail.com  Mon Jul 29 10:59:06 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 29 Jul 2013 08:59:06 +0000
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375026414031-4660271.post@n4.nabble.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
Message-ID: <CAPYw7P4BxmwdjTUjBubtRa9Jbgh6bau4A5H3QRR1__-X51-DqA@mail.gmail.com>

On 7/28/13, thljcl <jiachielee at live.com> wrote:

[...]

> To me personally, I kind of like the way ffmpeg handle color space
> conversion. Ffmpeg does have issues within its algorithm.

What are you talking about?

From onemda at gmail.com  Mon Jul 29 11:15:52 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 29 Jul 2013 09:15:52 +0000
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375041354497-4660277.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>
Message-ID: <CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>

Have you some questions?

If you are still not aware this list is not for propagating your
view of world but for user questions.

On 7/28/13, thljcl <jiachielee at live.com> wrote:
> Create slide show from still images is not exactly as simple as I originally
> anticipated. After knowing that ffmpeg is able to convert still images into
> video of lossless compression using encoders such as x264, I have a basic
> idea of how to create a slide show from a bunch of still images. In a way,
> creating slide show is exactly the opposite of creating the illusion of
> moving pictures. Human eyes perceive motion when the similar images are
> being displayed in sequence rapidly. For a slide show, however, each image
> or slide is being displayed for a couple of seconds instead of 1/24 seconds.
> Let's start from the very beginning.
>  The basic command to convert a couple of still images to H.264 lossless
> video is:
> ffmpeg -r "24" -i "f_%%1d.png" -vcodec "libx264" -crf "0" "output.mkv"
> The input frames are being set to the frame rate of 24 FPS.
> f_%%1d.png represents images with the file name starting with f_1.png,
> f_2.png, f_3.png ...
> In Windows 8, you select a couple of images with whatever file names, when
> you rename those files together, they would be renamed to filename (1).png,
> filename (2).png...
> Each frame would be displayed for 1/24 second.
> On ffmpeg wiki at http://bit.ly/14YRtSv, there is a guide for creating a
> video slideshow with ffmpeg. If that guide works perfectly for me, I would
> not bother to write here.

So instead of modifying that wiki, you posted your way how things should
be done here.

> Say if you want to create a slide show with each slide to be displayed for 5
> seconds, one way that would work, though not practical, is to manually
> duplicate the same image. You would need 120 copies of the same image for
> the same slide to be displayed for 5 seconds at 24 frame per second.
> The problem with the ffmpeg wiki's guide is that ffmpeg drops frames,
> especially at the beginning or ending of the videos. Dropping frames causes
> the dropped slide or changed length. What I want to do is to dictate how
> ffmpeg duplicate frames automatically as I want.
> Here I have 135 images at http://sdrv.ms/11onHnh. Among them, 9 of them have
> the resolution of 3456  * 1080 pixels; the rest of them have the resolution
> of 1920  * 1080 pixels. To create a slideshow of 1280  * 720 pixels, I need to
> resize all of them to have the same resolution.
>
> ffmpeg -r "1" -i "wallpapers\sources\1_%%1d.jpg" -pix_fmt "yuv444p" -vcodec
> "libx264" -crf "0" "wallpapers\1.mkv"
> ffmpeg -r "1" -i "wallpapers\sources\2_%%1d.jpg" -pix_fmt "yuv444p" -vf
> "scale=1920:600,pad=1920:1080:0:240" -vcodec "libx264" -crf "0"
> "wallpapers\2.mkv"
> ffmpeg -r "1" -i "wallpapers\sources\3_%%1d.jpg" -pix_fmt "yuv444p" -vcodec
> "libx264" -crf "0" "wallpapers\3.mkv"

Using -pix_fmt should only be required if some jpgs are yuv444p but
others are not.
Otherwise you are doing extra conversion for no real benefit.

>
> I merge the three videos using mkvtoolnix. I extract frames from the
> combined video.

If you use mkvtoolnix that is really bad, because this mailing list is not
about mkvtoolnix. So your way is not better in any way.

Above 3 videos could be merged with ffmpeg too, get learn that first
before posting next time.

>
> mkdir "wallpapers\frames"
> ffmpeg -i "wallpapers\t.mkv" -vf "scale=1280:720" "wallpapers\frames\f
> (%%1d).png"
>
> From individual frames, I wrote a batch file to duplicate frames and create
> a slideshow (H.264 lossless). Each slide is to be displayed for 10 seconds.
>
> cd /d "wallpapers\frames"
> copy "f (135).png" "f (136).png"
> cd /d "%programfiles%\ffmpeg\bin"
> ffmpeg -r "1" -i "wallpapers\frames\f (%%1d).png" -vcodec "libx264" -crf "0"
> -vf "fps=10" "wallpapers\1.mkv"
> cd /d "wallpapers\frames"
> del "f (136).png"
> cd /d "%programfiles%\ffmpeg\bin"
> ffmpeg -r "1" -i "wallpapers\1.mkv" -vf "fps=24" -vcodec "libx264" -crf "0"
> "wallpapers\2.mkv"
> cd /d "wallpapers"
> del "1.mkv"
> ren "2.mkv" "slideshow.mkv"
> cd /d "%programfiles%\ffmpeg\bin"
>
> In duplicating the frames, ffmpeg will ignore the last frame. That's why I
> duplicate the last frame. Likewise, I can extract all the frames from the
> last show. Ffmpeg will duplicate the first frame. Similarly, I delete the
> first frame. Below is the lines I wrote to extract the frames from the
> slideshow:

This sounds like bug, you give extremly good practice: never ever report bug
just find workarounds: slower, complicated, using non-ffmpeg tools are
better ....

>
> ffmpeg -i "wallpapers\slideshow.mkv" -vf "fps=1" -vcodec "libx264" -crf "0"
> "wallpapers\1.mkv"
> ffmpeg -r "10" -i "wallpapers\1.mkv" -vf "fps=1" -vcodec "libx264" -crf "0"
> "wallpapers\2.mkv"
> cd /d "wallpapers"
> del "1.mkv"
> ren "2.mkv" "1.mkv"
> cd /d "%programfiles%\ffmpeg\bin"
> mkdir "wallpapers\frames_1"
> ffmpeg -i "wallpapers\1.mkv" "wallpapers\frames_1\f_%%1d.png"
> cd /d "wallpapers\frames_1"
> del "f_1.png"
> cd /d "%programfiles%\ffmpeg\bin\wallpapers"
> del "1.mkv"
> cd /d "%programfiles%\ffmpeg\bin"
> ffmpeg -r "1" -i "wallpapers\frames_1\f_%%1d.png" -vcodec "libx264" -crf "0"
> "wallpapers\1.mkv"
> rmdir /s /q "wallpapers\frames_1"
> mkdir "wallpapers\frames_1"
> ffmpeg -i "wallpapers\1.mkv" "wallpapers\frames_1\f_%%1d.png"
> cd /d "wallpapers"
> del "1.mkv"
> cd /d "%programfiles%\ffmpeg\bin"
>
> The slideshow I created is at http://sdrv.ms/15Xa2TU.

In short this is perfect explanation how things should not be done.

From adf.lists at gmail.com  Mon Jul 29 12:24:55 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Mon, 29 Jul 2013 11:24:55 +0100
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375026414031-4660271.post@n4.nabble.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
Message-ID: <51F642F7.3080307@gmail.com>

thljcl wrote:
> When it comes to quality preservation, video editing, or lossless encoding;
> yes, I would agree that ?yuv444p? makes plenty of sense. ?yuv444p? is a way
> of re-encoding the entire 24-bit RGB information, which all color
> information is preserved without any loss due to chroma subsampling.

Well the subsampling is self evident, but I don't think 8 bit yuv can 
represent all of rgb (though not all of rgb is perceptible anyway).

http://www.poynton.com/PDFs/Merging_RGB_and_422.pdf


From barsnick at gmx.net  Mon Jul 29 17:48:13 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Mon, 29 Jul 2013 17:48:13 +0200
Subject: [FFmpeg-user] OT: duplicate emails
In-Reply-To: <51F5E312.4020404@burner.com>
References: <51F5E312.4020404@burner.com>
Message-ID: <20130729154813.GA5471@sunshine.barsnick.net>

On Sun, Jul 28, 2013 at 21:35:46 -0600, Bob Maple wrote:
> Is it just me, or is the list going nuts and sending like 4 and 5 copies

It's not just you, I got them too.

But I believe it's not the list itself going wild, because I can see
some loops in the Received: header getting longer and longer.
Curiously, it's not an endless loop (presumably because sendmail et.al.
limit it to something like 20), and not for all emails. *shrug*

Moritz

From edward.chopuryan at gmail.com  Mon Jul 29 08:34:33 2013
From: edward.chopuryan at gmail.com (edward20)
Date: Sun, 28 Jul 2013 23:34:33 -0700 (PDT)
Subject: [FFmpeg-user] Decode H264 and save to yuv file
In-Reply-To: <tencent_05C78B781D6F668440CD3187@qq.com>
References: <1374840030366-4660204.post@n4.nabble.com>
	<tencent_05C78B781D6F668440CD3187@qq.com>
Message-ID: <1375079673569-4660291.post@n4.nabble.com>

Spirian wrote
> Normally, Decode means converting an codec format to .yuv or rgb24 format.
> for ffmpeg : ffmepg -i x1.avi x2.yuv  x1=input x2=output   
> 
> 
> 
> 
> ------------------ Original ------------------
> From:  "Paul B Mahol"&lt;

> onemda@

> &gt;;
> Date:  Sat, Jul 27, 2013 04:00 AM
> To:  "FFmpeg user questions"&lt;

> ffmpeg-user@

> &gt;; 
> 
> Subject:  Re: [FFmpeg-user] Decode H264 and save to yuv file
> 
> 
> 
> On 7/26/13, edward20 &lt;

> edward.chopuryan@

> &gt; wrote:
>> Hello everyone,
>>
>> I am new here and I am trying to decode video using ffmpeg c API and save
>> as
>> yuv file with YUV420P pixel format.
>>
>> All examples that I have seen they converting extracted frames to RGB or
>> PPM
>> and then saving them to file.
>> But I want to save as YUV. I don't know do I have to convert frames after
>> decoding them or maybe after decoding the frames are already YUV frames.
>> Any
>> help will be appreciated.
> 
> Usually videos are in yuv420 and decoder outputs it in yuv420 planar
> format. So after decoding there is nothing left to do.
> 
> But perhaps you need packed yuv format?
> _______________________________________________
> ffmpeg-user mailing list

> ffmpeg-user@

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

> ffmpeg-user@

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

Thanks for your reply but actually I need an example of C code how to save
frames in file after decoding video and extracting frames. Can you provide
me an example?



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From jiachielee at live.com  Mon Jul 29 13:14:04 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 29 Jul 2013 04:14:04 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <51F642F7.3080307@gmail.com>
References: <CAN0LTaNk6arV0LLyaaficiJR-z2Q-F39wR98O1xk+KNqDyiJ0w@mail.gmail.com>
	<1375026414031-4660271.post@n4.nabble.com>
	<51F642F7.3080307@gmail.com>
Message-ID: <1375096444082-4660295.post@n4.nabble.com>

YCbCr is not an absolute color space. It?s just a way to represent all color
information in RGB in different set of ?coordinates?. Mathematically
speaking, YCbCr 4:4:4: is related to RGB color space by a single linear
transformation matrix, just like Lorentz Transformation or Rotation in
Euclidean Space. Now I know that there are different standards with regards
of the transformation matrix being used; but all of them are non-singular,
or reversible. That said, when it comes to calculation by computer, there
will always be rounding involved. However, no rounding error would be
introduced in this case because number of colors in RGB color space is
limited. For 24-bit sRGB color space, there are only 16,777,216 colors;
whereas the usual coordinate transformation involves vector space which can
be seen as a set of infinite elements of coordinates.
That said, by encoding RGB color space in Y, Cb, and Cr channels, we can
later reduce the number of bits being used for encoding color information in
Cb and Cr channels; that?s the basic idea behind chroma subsampling. In case
you wonder how a chroma subsampling scheme affects the number of bits to be
used for a single pixel on average:
LAGS NAME            NB_COMPONENTS BITS_PER_PIXEL
-----
IO... yuv420p                3            12
IO... yuyv422                3            16
IO... rgb24                  3            24
IO... bgr24                  3            24
IO... yuv422p                3            16
IO... yuv444p                3            24
IO... yuv410p                3             9
IO... yuv411p                3            12
IO... gray                   1             8
IO..B monow                  1             1
IO..B monob                  1             1
I..P. pal8                   1             8
IO... yuvj420p               3            12
IO... yuvj422p               3            16
IO... yuvj444p               3            24
..H.. xvmcmc                 0             0
..H.. xvmcidct               0             0
IO... uyvy422                3            16
..... uyyvyy411              3            12
IO... bgr8                   3             8
.O..B bgr4                   3             4
IO... bgr4_byte              3             4
IO... rgb8                   3             8
.O..B rgb4                   3             4
IO... rgb4_byte              3             4
IO... nv12                   3            12
IO... nv21                   3            12
IO... argb                   4            32
IO... rgba                   4            32
IO... abgr                   4            32
IO... bgra                   4            32
IO... gray16be               1            16
IO... gray16le               1            16
IO... yuv440p                3            16
IO... yuvj440p               3            16
IO... yuva420p               4            20
..H.. vdpau_h264             0             0
..H.. vdpau_mpeg1            0             0
..H.. vdpau_mpeg2            0             0
..H.. vdpau_wmv3             0             0
..H.. vdpau_vc1              0             0
IO... rgb48be                3            48
IO... rgb48le                3            48
IO... rgb565be               3            16
IO... rgb565le               3            16
IO... rgb555be               3            15
IO... rgb555le               3            15
IO... bgr565be               3            16
IO... bgr565le               3            16
IO... bgr555be               3            15
IO... bgr555le               3            15
..H.. vaapi_moco             0             0
..H.. vaapi_idct             0             0
..H.. vaapi_vld              0             0
IO... yuv420p16le            3            24
IO... yuv420p16be            3            24
IO... yuv422p16le            3            32
IO... yuv422p16be            3            32
IO... yuv444p16le            3            48
IO... yuv444p16be            3            48
..H.. vdpau_mpeg4            0             0
..H.. dxva2_vld              0             0
IO... rgb444le               3            12
IO... rgb444be               3            12
IO... bgr444le               3            12
IO... bgr444be               3            12
I.... gray8a                 2            16
IO... bgr48be                3            48
IO... bgr48le                3            48
IO... yuv420p9be             3            13
IO... yuv420p9le             3            13
IO... yuv420p10be            3            15
IO... yuv420p10le            3            15
IO... yuv422p10be            3            20
IO... yuv422p10le            3            20
IO... yuv444p9be             3            27
IO... yuv444p9le             3            27
IO... yuv444p10be            3            30
IO... yuv444p10le            3            30
IO... yuv422p9be             3            18
IO... yuv422p9le             3            18
..H.. vda_vld                0             0
IO... gbrp                   3            24
IO... gbrp9be                3            27
IO... gbrp9le                3            27
IO... gbrp10be               3            30
IO... gbrp10le               3            30
I.... gbrp16be               3            48
I.... gbrp16le               3            48
IO... yuva420p9be            4            22
IO... yuva420p9le            4            22
IO... yuva422p9be            4            27
IO... yuva422p9le            4            27
IO... yuva444p9be            4            36
IO... yuva444p9le            4            36
IO... yuva420p10be           4            25
IO... yuva420p10le           4            25
IO... yuva422p10be           4            30
IO... yuva422p10le           4            30
IO... yuva444p10be           4            40
IO... yuva444p10le           4            40
IO... yuva420p16be           4            40
IO... yuva420p16le           4            40
IO... yuva422p16be           4            48
IO... yuva422p16le           4            48
IO... yuva444p16be           4            64
IO... yuva444p16le           4            64
..H.. vdpau                  0             0
IO... xyz12le                3            36
IO... xyz12be                3            36
IO... rgba64be               4            64
IO... rgba64le               4            64
..... bgra64be               4            64
..... bgra64le               4            64
IO... 0rgb                   3            24
IO... rgb0                   3            24
IO... 0bgr                   3            24
IO... bgr0                   3            24
IO... yuva444p               4            32
IO... yuva422p               4            24
IO... yuv420p12be            3            18
IO... yuv420p12le            3            18
IO... yuv420p14be            3            21
IO... yuv420p14le            3            21
IO... yuv422p12be            3            24
IO... yuv422p12le            3            24
IO... yuv422p14be            3            28
IO... yuv422p14le            3            28
IO... yuv444p12be            3            36
IO... yuv444p12le            3            36
IO... yuv444p14be            3            42
IO... yuv444p14le            3            42
IO... gbrp12be               3            36
IO... gbrp12le               3            36
IO... gbrp14be               3            42
IO... gbrp14le               3            42
IO... gbrap                  4            32
I.... gbrap16be              4            64
I.... gbrap16le              4            64
IO... yuvj411p               3            12
The short answer: YUV444 can fully represent all color information in RGB.
 




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From jiachielee at live.com  Mon Jul 29 13:59:34 2013
From: jiachielee at live.com (thljcl)
Date: Mon, 29 Jul 2013 04:59:34 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
Message-ID: <1375099174769-4660296.post@n4.nabble.com>

Yes, I?m aware of the fact that ffmpeg can be used to concatenate videos
Physical concatenation protocol. 
Allow to read and seek from many resource in sequence as if they were a
unique resource. 
A URL accepted by this protocol has the syntax: 
 	concat:URL1|URL2|...|URLN

where URL1, URL2, ..., URLN are the urls of the resource to be concatenated,
each one possibly specifying a distinct protocol. 
For example to read a sequence of files ?split1.mpeg?, ?split2.mpeg?,
?split3.mpeg? with ffplay use the command: 
 	ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

Note that you may need to escape the character "|" which is special for many
shells. 
But I can make my own choice to use the tools I prefer, can I not?
With regards to reporting bugs to ffmpeg developers, I do not in this case
discover a bug, so to speak. If it sounds like a bug to you; suite yourself
to report it. I?m sure that you can test it to find the bugs yourself.
Bear in mind, frame rate conversion from one constant frame rate to another
always involves adding or dropping frames. As users, we might not know the
algorithms behind the video filter ?fps?; that?s why I specifically perform
a number of tests to find out how it decides which frames it adds or drops.
I would consider the frame rate conversion is ?flawed? if such a condition
is true: the conversion process is not reversible. Well, allows me to
elaborate further.
If the original source is of the frame rate of lower frame rate, when its
frame rate is to be converted to higher frame rate, it would involve the
duplication of frames; of course the question of which frames would be
duplicated depends on the algorithms designed by the developers. When the
video of higher frame rate with duplicated frames undergoes frame rate
conversion so that the output would have the original frame rates, only
duplicated frames would be dropped. Even though the length of the videos
would change due to frame rate conversion, repeated conversion from one to
another and vice versa would not change the video length for a specific
frame rate. Of course, if the source of higher frame rate with no duplicated
frames undergoes frame rate conversion to achieve lower frame rate, such a
process is irreversible, as expected.
Based on my own tests, ffmpeg passes my expectation of how frame rate
conversion should be handled. That?s why I do not discover a bug. However,
if you find bugs from my experience, feel free to report it and publish it.
I have to say that the basic idea behind the duplication of frames to create
a slideshow involves a feature in ffmpeg which is frame rate conversion. In
effect, we use frame rate conversion algorithms to cause ffmpeg to duplicate
frames for us, instead of manually copying frames to create a slideshow.
I?m not absolutely sure that my way of creating a slideshow is particularly
good or superior, that?s why I did not attempt to edit the ffmpeg wiki.
No, it?s not meant for propagating my views of world. In fact, similar
questions have been asked before at http://bit.ly/13rjKL3 and
http://bit.ly/1cgluyH. But while I believe that my experience may be helpful
to somebody, it does not directly answer their questions. Speaking of
questions the users may ask, it?s indeed a question. It is a question I
would ask myself as well as other possibly much more skillful users of
ffmpeg. I gave my ?workaround?. But in no way I suggested that somehow I
gave a perfect answer. If I sound like such, I apologize in advance.
Presumably, somebody else may have better answers for what I saw or what I
want to achieve. Possibly, my understanding of how ffmpeg works is
misguided.
That's why I specifically try to extract all the original non-duplicated
frames from the created slideshow, which is successful, I have to say.



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From support at hioxindia.com  Mon Jul 29 13:16:34 2013
From: support at hioxindia.com (support at hioxindia.com)
Date: Mon, 29 Jul 2013 16:46:34 +0530
Subject: [FFmpeg-user] FFMPEG Installation error in Centos
Message-ID: <2a3d593bd2c20ee7981da6793c54a756.squirrel@hioxindia.com>

Dear support Team,

I was trying to install with configure libvpx in FFMPEG but showing some
warnings,

root at host [/usr/local/ffmpeg]# ./configure --prefix=/usr --enable-gpl 
--enable-version3 --enable-nonfree  --disable-static --enable-shared
--enable-x11grab --enable-libfaac --enable-libfreetype --enable-libmp3lame
--enable-libopenjpeg --enable-libpulse --enable-libspeex
--enable-libtheora --enable-libvorbis --enable-libvpx  --enable-openssl
--disable-debug
Unable to create and execute files in /tmp.  Set the TMPDIR environment
variable to another directory and make sure that it is not mounted noexec.
Sanity 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.

My  git version 1.8.2.1.

Kindly assist this issue and give me the solution,

Please get back to us in case of any further support.

Thanks and Regards,
Hiox India,




From adf.lists at gmail.com  Mon Jul 29 18:31:06 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Mon, 29 Jul 2013 17:31:06 +0100
Subject: [FFmpeg-user] Playing Inverse Telecined 24 fps video on a 1080p
 60 Hz Computer Monitor
In-Reply-To: <1375046122.2209.YahooMailNeo@web164705.mail.gq1.yahoo.com>
References: <1374940856.48259.YahooMailNeo@web164701.mail.gq1.yahoo.com>	<loom.20130727T192450-485@post.gmane.org>	<1374947589.92778.YahooMailNeo@web164703.mail.gq1.yahoo.com>	<loom.20130728T030129-819@post.gmane.org>	<1375016502.98169.YahooMailNeo@web164701.mail.gq1.yahoo.com>
	<loom.20130728T182747-704@post.gmane.org>
	<1375046122.2209.YahooMailNeo@web164705.mail.gq1.yahoo.com>
Message-ID: <51F698CA.6030009@gmail.com>

James Board wrote:

> I'm wondering if someone dealt with this issue before.
> You need to inverse telecine to get rid of the combing
> effects, but when you do that, your video is now at 24
> fps, not 30 fps.  This makes motion un-smooth since most
> displays are a multiple of 30 Hz.

Yes that has always been the case with film on ntsc.

Don't think of 30fps though, it's 60, by which I mean 3/2 pull down 
historically worked on interlaced displays so 60 (/1.001) fields/sec.

When you convert to 24 and display on a progressive 60Hz screen you end 
up with exactly the same - not better, not worse, than it has always 
been for viewing film in an ntsc country.

> Do some displays allow you to set 24 Hz or a multiple
> of 24 Hz?

Yes some modern TVs do 24.



From onemda at gmail.com  Mon Jul 29 18:31:34 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 29 Jul 2013 16:31:34 +0000
Subject: [FFmpeg-user] FFMPEG Installation error in Centos
In-Reply-To: <2a3d593bd2c20ee7981da6793c54a756.squirrel@hioxindia.com>
References: <2a3d593bd2c20ee7981da6793c54a756.squirrel@hioxindia.com>
Message-ID: <CAPYw7P5p+=bDoxEnO274MYMhafSAuyRHBohLyvHdJNqC2rivHA@mail.gmail.com>

On 7/29/13, support at hioxindia.com <support at hioxindia.com> wrote:
> Dear support Team,
>
> I was trying to install with configure libvpx in FFMPEG but showing some
> warnings,
>
> root at host [/usr/local/ffmpeg]# ./configure --prefix=/usr --enable-gpl
> --enable-version3 --enable-nonfree  --disable-static --enable-shared
> --enable-x11grab --enable-libfaac --enable-libfreetype --enable-libmp3lame
> --enable-libopenjpeg --enable-libpulse --enable-libspeex
> --enable-libtheora --enable-libvorbis --enable-libvpx  --enable-openssl
> --disable-debug
> Unable to create and execute files in /tmp.  Set the TMPDIR environment
> variable to another directory and make sure that it is not mounted noexec.
> Sanity test failed.

Doesn't message above explain it all.

>
> 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.
>
> My  git version 1.8.2.1.
>
> Kindly assist this issue and give me the solution,
>
> Please get back to us in case of any further support.
>
> Thanks and Regards,
> Hiox India,

From onemda at gmail.com  Mon Jul 29 19:08:12 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Mon, 29 Jul 2013 17:08:12 +0000
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375099174769-4660296.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
Message-ID: <CAPYw7P4+rKbiq=veuRQyJauVEL5My7o8EjVVo+-Bda0Vuwn6mg@mail.gmail.com>

On 7/29/13, thljcl <jiachielee at live.com> wrote:
> Yes, I'm aware of the fact that ffmpeg can be used to concatenate videos
> Physical concatenation protocol.
> Allow to read and seek from many resource in sequence as if they were a
> unique resource.
> A URL accepted by this protocol has the syntax:
>  	concat:URL1|URL2|...|URLN
>
> where URL1, URL2, ..., URLN are the urls of the resource to be concatenated,
> each one possibly specifying a distinct protocol.
> For example to read a sequence of files `split1.mpeg', `split2.mpeg',
> `split3.mpeg' with ffplay use the command:
>  	ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
>
> Note that you may need to escape the character "|" which is special for many
> shells.
> But I can make my own choice to use the tools I prefer, can I not?

Yes, but not telling rest of world your sub-optimal solution.

Also there are better ways to concatenate videos, but it appears you are
aware of only this one.

> With regards to reporting bugs to ffmpeg developers, I do not in this case
> discover a bug, so to speak. If it sounds like a bug to you; suite yourself
> to report it. I'm sure that you can test it to find the bugs yourself.
> Bear in mind, frame rate conversion from one constant frame rate to another
> always involves adding or dropping frames. As users, we might not know the
> algorithms behind the video filter "fps"; that's why I specifically perform
> a number of tests to find out how it decides which frames it adds or drops.
> I would consider the frame rate conversion is "flawed" if such a condition
> is true: the conversion process is not reversible. Well, allows me to
> elaborate further.
> If the original source is of the frame rate of lower frame rate, when its
> frame rate is to be converted to higher frame rate, it would involve the
> duplication of frames; of course the question of which frames would be
> duplicated depends on the algorithms designed by the developers. When the
> video of higher frame rate with duplicated frames undergoes frame rate
> conversion so that the output would have the original frame rates, only
> duplicated frames would be dropped. Even though the length of the videos
> would change due to frame rate conversion, repeated conversion from one to
> another and vice versa would not change the video length for a specific
> frame rate. Of course, if the source of higher frame rate with no duplicated
> frames undergoes frame rate conversion to achieve lower frame rate, such a
> process is irreversible, as expected.
> Based on my own tests, ffmpeg passes my expectation of how frame rate
> conversion should be handled. That's why I do not discover a bug. However,
> if you find bugs from my experience, feel free to report it and publish it.
> I have to say that the basic idea behind the duplication of frames to create
> a slideshow involves a feature in ffmpeg which is frame rate conversion. In
> effect, we use frame rate conversion algorithms to cause ffmpeg to duplicate
> frames for us, instead of manually copying frames to create a slideshow.
> I'm not absolutely sure that my way of creating a slideshow is particularly
> good or superior, that's why I did not attempt to edit the ffmpeg wiki.
> No, it's not meant for propagating my views of world. In fact, similar
> questions have been asked before at http://bit.ly/13rjKL3 and
> http://bit.ly/1cgluyH. But while I believe that my experience may be helpful
> to somebody, it does not directly answer their questions. Speaking of
> questions the users may ask, it's indeed a question. It is a question I
> would ask myself as well as other possibly much more skillful users of
> ffmpeg. I gave my "workaround". But in no way I suggested that somehow I
> gave a perfect answer. If I sound like such, I apologize in advance.
> Presumably, somebody else may have better answers for what I saw or what I
> want to achieve. Possibly, my understanding of how ffmpeg works is
> misguided.
> That's why I specifically try to extract all the original non-duplicated
> frames from the created slideshow, which is successful, I have to say.


Have you read documentation of fps filter?
Did you tried changing its parameters?

From adf.lists at gmail.com  Mon Jul 29 19:17:13 2013
From: adf.lists at gmail.com (Andy Furniss)
Date: Mon, 29 Jul 2013 18:17:13 +0100
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375096444082-4660295.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>
Message-ID: <51F6A399.4020400@gmail.com>

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


From bobm-ffmpeg at burner.com  Mon Jul 29 20:49:46 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Mon, 29 Jul 2013 12:49:46 -0600
Subject: [FFmpeg-user] OT: duplicate emails
In-Reply-To: <20130729154813.GA5471@sunshine.barsnick.net>
References: <51F5E312.4020404@burner.com>
	<20130729154813.GA5471@sunshine.barsnick.net>
Message-ID: <51F6B94A.8030702@burner.com>

On 7/29/2013 9:48 AM, Moritz Barsnick wrote:

> It's not just you, I got them too.

Seems to have stopped for now, at least. Thanks for the confirms.

Bob

From compuvision.sr at gmail.com  Mon Jul 29 22:21:34 2013
From: compuvision.sr at gmail.com (Albert Scholtalbers)
Date: Mon, 29 Jul 2013 17:21:34 -0300
Subject: [FFmpeg-user] OT: duplicate emails
In-Reply-To: <51F5E312.4020404@burner.com>
References: <51F5E312.4020404@burner.com>
Message-ID: <CAECKL3qTZ9uHGc4xHqRiNZ25DoDP-J_DEVAi9vDPkNQ1S-yToQ@mail.gmail.com>

No problem here. You could also use the Gui in stead of the mailing list
On 29 Jul 2013 00:36, "Bob Maple" <bobm-ffmpeg at burner.com> wrote:

> Is it just me, or is the list going nuts and sending like 4 and 5 copies
> of the same message to everyone the last couple of days?  I don't think
> it's a local problem to me as it seems like the mailing list footer gets
> put on another time for each copy of the message I get..
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From d4rkw0lf at gmx.com  Mon Jul 29 18:56:04 2013
From: d4rkw0lf at gmx.com (Sean Shaffer)
Date: Mon, 29 Jul 2013 12:56:04 -0400
Subject: [FFmpeg-user] 2nd Attempt: error enabling libx264
Message-ID: <51F69EA4.8080609@gmx.com>

Attempting to compile ffmpeg-2.0 from source dated 2013-07-09

Also compiled libmp3lame, yasm and libx264 from source.

libmp3lame (3.99.5)
yasm (1.2.0)
libx264 (unknown, sourced from git 2013-07-26)

configure fails; indicates libx264 is not found or wrong version.

[output]
$ ./configure --enable-gpl --enable-libx264 --enable-libmp3lame
--enable-nonfree --enable-libvorbis --enable-x11grab
ERROR: libx264 must be installed and version must be >= 0.118.

[/output]

System is :
Ubuntu 8.04 LTS
GCC 4.2.4 (GCC) (Ubuntu 4.2.4-1ubuntu4)
LD GNU ld (GNU Binutils for Ubuntu) 2.18.0.20080103

ffmpeg will configure if I do not try to enable libx264
config.log is attached as per configure's instruction (7z archived)

-- 
Sean
KB3IMT
Public Key http://pgp.mit.edu:11371/pks/lookup?op=get&search=0xBD3DC5FB
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From e.sansone at hotmail.it  Mon Jul 29 18:13:30 2013
From: e.sansone at hotmail.it (emanuele)
Date: Mon, 29 Jul 2013 18:13:30 +0200
Subject: [FFmpeg-user] Cross-compiling FFMPEG for Blackberry 10 platform
Message-ID: <BLU0-SMTP238F7DCAF4AEB3C6A89F17A93550@phx.gbl>

Good evening,

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.

Hope that someone can help me in configuring for successful cross compilation of the source code

Thanks for the attention
Best Regards
Emanuele Sansone
---------------------------------------
E-mail e.sansone at hotmail.it


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From bored_to_death85 at yahoo.com  Mon Jul 29 22:55:11 2013
From: bored_to_death85 at yahoo.com (M. V.)
Date: Mon, 29 Jul 2013 13:55:11 -0700 (PDT)
Subject: [FFmpeg-user] problem with converting rtp packets to an audio format
Message-ID: <1375131311.43048.YahooMailNeo@web165006.mail.bf1.yahoo.com>

hi guys,

I'm new to ffmpeg. I'm developing a program which receives and recognizes rtp (+sip/sdp) packets from interface and writes rtp content (just rtp part, not udp and ip header) to a file. now I wanna use "ffmpeg"? to convert this file to an audio format. for this I use:
??? #ffmpeg -f rtp -i /locaton/of/my/rtp/file output.au
and I receive this error:

-----------------------------------------------------------------------------
[rtp @ 0x2d62880] Guessing on RTP content - if not received properly you need an SDP file describing it
Guessed Channel Layout for? Input Stream #0.0 : mono
Input #0, rtp, from '/location/to/my/rtp/file':
? Duration: N/A, bitrate: 64 kb/s
??? Stream #0:0: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
Output #0, au, to '/output.au':
? Metadata:
??? encoder???????? : Lavf55.12.100
??? Stream #0:0: Audio: pcm_s16be ([3][0][0][0] / 0x0003), 8000 Hz, mono, s16, 128 kb/s
Stream mapping:
? Stream #0:0 -> #0:0 (pcm_alaw -> pcm_s16be)/location/to/my/rtp/file: Connection timed out
size=?????? 0kB time=-577014:-32:-22.-77 bitrate=N/A??? 
size=?????? 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)
-----------------------------------------------------------------------------


so what am I doing wrong?
- should I write the whole rtp packets (with ethernet/ip/udp headers) to file or just rtp-section of packets are enough?
- do I also need to write all related sdp (and sip) packets to a separate file?
- if so, how can I introduce this file to ffmpeg?

I googled a lot and checked documentation, but didn't find an answer. (most of what I found was about playing rtp from ffmpeg)


any help would be much appreciated.

thank you.

From WPoll at nhnz.tv  Tue Jul 30 00:50:31 2013
From: WPoll at nhnz.tv (Wayne Poll)
Date: Mon, 29 Jul 2013 22:50:31 +0000
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <51E804EB.4060805@ringsidecreative.com>
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>
	<51E7F54D.2050708@ringsidecreative.com>
	<CAFHbX1KC3Z-dK3GG8U8_2=KL7uyMsdPTQzHbc6Tjk3N0Zqjwcw@mail.gmail.com>
	<51E804EB.4060805@ringsidecreative.com>
Message-ID: <8CDA830CFEB43147A86E9A144320F7C605CD27E5@NHNZDNEX01.dn.nhnz.tv>

Sorry this is so late as a response (I've been away) but the fix is easy Sam; you can exclude the ninth track like this....

./ffmpeg -i input.mxf -map 0 -map -0:9:0 -c copy -c:v prores -profile:v 3 output.mov

The "-map 0 -map -0:9:0" will tell ffmpeg to copy all streams EXCEPT stream 0:9 - which in my case is a metadata track (possibly TC - not sure).

HTH someone...

Wayne Poll?|?Technical & IT Systems Manager?|
NHNZ | 5 Melville Street, Dunedin 9016, New Zealand |
p. +64 3 479 9920 | m. +64 27 223 4171 | f. +64 3 479 9917 | www.nhnz.tv | Facebook? 


-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Sam Marrocco
Sent: Friday, 19 July 2013 3:08
To: FFmpeg user questions
Subject: Re: [FFmpeg-user] Is mixing of audio channels happening by default in ffmpeg?


On 7/18/2013 10:34 AM, Tom Evans wrote:





-map 0 means add every input stream in to the output. You've said what you want done with video, encode to prores_ks, and what to do with audio, copy, you just need to either say don't add all streams (just first video and all audio) or copy all input streams, including metadata, and re-encode the video.

You can do both with -map, eg to not copy the metadata track:  "-map 0:v -map 0:a  -c:a copy -c:v prores_ks"
or to copy everything, but re-encode the video: "-map 0 -c copy -c:v prores_ks"

-c:v is an alias for -vcodec, -c:a for -acodec, in case you were wondering.





Thanks, Tom.
in order to get the video, 8 channels of audio and the ninth track TC (which can't be specified specifically) what you stated might work as:

-map 0:v -map 0:a

If I'm understanding this seems to indicate "copy all video tracks across, using my pre-specified video codec settings" and "copy across all audio tracks, using my pre-specified audio codec settings". The timecode seems to be brought across by default without specifying.

I'll test further, but this seems to do the job.

sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

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

From andycivil at gmail.com  Tue Jul 30 01:31:29 2013
From: andycivil at gmail.com (Andy Civil)
Date: Mon, 29 Jul 2013 19:31:29 -0400
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375099174769-4660296.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
Message-ID: <51F6FB51.2080707@gmail.com>

On 2013-07-29 7:59 AM, thljcl wrote:

> ... I believe that my experience may be helpful to somebody  ...

I am interested to read your experiences, and also this thread becomes a 
historical record for others who search for answers in the future.

I have previously posted a method to create a slideshow with FFmpeg, 
specifically to allow for slides of different lengths (e.g. when doing a 
presentation with a voiceover) which is not covered in the Wiki or any other 
article that I know. However, my method is 'awkward' and I'm interested to look 
at yours to see if I can combine them to get a simpler method which still allows 
for different times. (Concatenation was not a possibility when I worked out my 
method.)

Thank you for posting.

--
Andy

From bzb.dev001 at gmail.com  Tue Jul 30 04:12:52 2013
From: bzb.dev001 at gmail.com (bzb.dev001)
Date: Mon, 29 Jul 2013 22:12:52 -0400
Subject: [FFmpeg-user] Can't get ffplay to read a stream from ffserver.
	Configuration problem?
Message-ID: <51F72124.602@gmail.com>


My system is running Lubuntu 12.04 64bit

$ uname -a
Linux myhostname 3.2.0-49-generic #75-Ubuntu SMP Tue Jun 18 17:39:32 UTC 
2013 x86_64 x86_64 x86_64 GNU/Linux


I downloaded the snapshot of ffmpeg-2.0 sources on July 29, 2013.  And 
built it as follows.

$ ./configure --prefix=/somelocation --enable-libmp3lame 
--enable-libx264 --enable-gpl
$ make
$ sudo make install


I proceeded to configure the ffserver.conf as shown below.


ffserver.conf
-------------
Port 8090
BindAddress 0.0.0.0
MaxHTTPConnections 5
MaxClients 5
MaxBandwidth 1000
CustomLog -
NoDaemon
<Feed videofeed1.ffm>
File /tmp/videofeed1.ffm
FileMaxSize 200K
Launch ffmpeg
ACL allow 127.0.0.1
ACL allow 192.168.1.0 192.168.1.255
</Feed>
<Stream videostream1.mpg>
Feed videofeed1.ffm
Format mpeg
AudioBitRate 32
AudioChannels 1
AudioSampleRate 22050
VideoBitRate 64
VideoBufferSize 40
VideoFrameRate 3
VideoSize 160x128
VideoGopSize 12
ACL allow localhost
ACL allow 192.168.1.0 192.168.1.255
</Stream>


The ffserver is started.

$ ffserver -d -f /somelocation/ffserver.conf
ffserver version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
   built on Jul 29 2013 18:57:52 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
   configuration: --prefix=/somelocation --enable-libmp3lame 
--enable-libx264 --enable-gpl
   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
   libpostproc    52.  3.100 / 52.  3.100
Mon Jul 29 21:24:26 2013 FFserver started.
Mon Jul 29 21:24:26 2013 Launch command line: ffmpeg 
http://127.0.0.1:8090/videofeed1.ffm
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
   built on Jul 29 2013 18:57:52 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
   configuration: --prefix=/somelocation --enable-libmp3lame 
--enable-libx264 --enable-gpl
   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
   libpostproc    52.  3.100 / 52.  3.100
Mon Jul 29 21:24:26 2013 127.0.0.1 - - New connection: GET /videofeed1.ffm
Mon Jul 29 21:24:26 2013 127.0.0.1 - - [GET] "/videofeed1.ffm HTTP/1.1" 
200 4175
Missing audio stream which is required by this ffm
videofeed1.ffm: Pid 24078 exited with status 256 after 0 seconds


I opened another console on the same system to upload a feed to the 
ffserver.


$ ffmpeg -i testvideo.mp4 http://localhost:8090/videofeed1.ffm
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
   built on Jul 29 2013 18:57:52 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
   configuration: --prefix=/somelocation --enable-libmp3lame 
--enable-libx264 --enable-gpl
   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
   libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testvideo.mp4':
   Metadata:
     major_brand     : mp42
     minor_version   : 0
     compatible_brands: isommp42
     creation_time   : 2011-04-11 13:04:38
   Duration: 00:02:10.72, start: 0.000000, bitrate: 394 kb/s
     Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 
0x31637661), yuv420p, 636x360, 293 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc
     Metadata:
       creation_time   : 1970-01-01 00:00:00
       handler_name    : VideoHandler
     Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 95 kb/s
     Metadata:
       creation_time   : 2011-04-11 13:04:39
       handler_name    : (C) 2007 Google Inc. v08.13.2007.
Output #0, ffm, to 'http://localhost:8090/videofeed1.ffm':
   Metadata:
     major_brand     : mp42
     minor_version   : 0
     compatible_brands: isommp42
     creation_time   : now
     encoder         : Lavf55.12.100
     Stream #0:0(und): Audio: mp2, 22050 Hz, mono, s16, 32 kb/s
     Metadata:
       creation_time   : 2011-04-11 13:04:39
       handler_name    : (C) 2007 Google Inc. v08.13.2007.
     Stream #0:1(und): Video: mpeg1video, yuv420p, 160x128, q=2-31, 64 
kb/s, 1000k tbn, 25 tbc
     Metadata:
       creation_time   : 1970-01-01 00:00:00
       handler_name    : VideoHandler
Stream mapping:
   Stream #0:1 -> #0:0 (aac -> mp2)
   Stream #0:0 -> #0:1 (h264 -> mpeg1video)
Press [q] to stop, [?] for help
frame= 3268 fps=201 q=9.3 Lsize=    1876kB time=00:02:10.69 bitrate= 
117.6kbits/s
video:1249kB audio:511kB subtitle:0 global headers:0kB muxing overhead 
6.598953%


I verified that /tmp/videofeed1.ffm was created by ffserver.
The ffserver console output showed the following additional output

...
Mon Jul 29 21:25:25 2013 127.0.0.1 - - New connection: GET /videofeed1.ffm
Mon Jul 29 21:25:25 2013 127.0.0.1 - - [GET] "/videofeed1.ffm HTTP/1.1" 
200 4175
Mon Jul 29 21:25:25 2013 127.0.0.1 - - New connection: POST /videofeed1.ffm
Mon Jul 29 21:25:44 2013 127.0.0.1 - - [POST] "/videofeed1.ffm HTTP/1.1" 
200 1921024


I then tried to read the stream using ffplay.


$ ffplay http://localhost:8090/videostream1.mpg
ffplay version 2.0 Copyright (c) 2003-2013 the FFmpeg developers
   built on Jul 29 2013 18:57:52 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
   configuration: --prefix=/somelocation --enable-libmp3lame 
--enable-libx264 --enable-gpl
   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
   libpostproc    52.  3.100 / 52.  3.100
     nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0


I observed that counters from ffplay showed no progress.  Ie. the ffplay 
console output remained static.  Nothing played.
The corresponding ffserver console output showed the following 
additional output.

...
Mon Jul 29 21:26:35 2013 127.0.0.1 - - New connection: GET /videostream1.mpg



From rose at cs.stanford.edu  Tue Jul 30 03:21:01 2013
From: rose at cs.stanford.edu (Rose Perrone)
Date: Mon, 29 Jul 2013 18:21:01 -0700
Subject: [FFmpeg-user] libx264 not found
Message-ID: <CAJtDrUduV1pFpU9VeVkmGVUmyY0GU6t43M6SkOJ3C6x1+hRa5g@mail.gmail.com>

I'm trying to use the encoder h264 with ffmpeg. I installed x264 from
source and with few, and I configured ffmpeg with ./configure --enable-gpl
--enable-shared --extra-ldflags="-ldl"  --enable-libx264. Yet the
config.log tells me fatal error: 'x264' file not found, and I get the error
libx264 not found.

From sw.craftsman at gmail.com  Tue Jul 30 09:35:11 2013
From: sw.craftsman at gmail.com (Shyam Sundar)
Date: Tue, 30 Jul 2013 13:05:11 +0530
Subject: [FFmpeg-user] Video bit rate missing from FFProbe output
In-Reply-To: <CA+XaXC=za6GKxgvfjX2tRTbmU4BZ7nGoAUKvOvnxypNbK56Tmw@mail.gmail.com>
References: <CA+XaXC=J7BMNZTF3orYpr_pNrDbmYdVmUycwJeWMa9b7NHGh5w@mail.gmail.com>
	<loom.20130723T220119-727@post.gmane.org>
	<CA+XaXCnhFAYePWHi12_zA_GDqXcqLsa8y9CdMokb1qrrLnh5CQ@mail.gmail.com>
	<loom.20130725T000737-643@post.gmane.org>
	<CA+XaXC=za6GKxgvfjX2tRTbmU4BZ7nGoAUKvOvnxypNbK56Tmw@mail.gmail.com>
Message-ID: <CA+XaXCkWtrpbmVaO6dNwGCWCC_z7xhPH3crohbaFCxB7W9i1ww@mail.gmail.com>

Hi Carl,

S
 orry for top-posting in the last response ... but I didn't fully get what
you said about the patch (to get max bit-rate if no bit-rate is defined) in
your last response.

- Can you pls let me know where can I get that patch from ?
- Has this behavior changed intentionally in latest FFProbe (as it used to
show the bit-rate in old versions) ?
- Is this fix going to be a part of major release in near future ?

Thanks a lot

From jiachielee at live.com  Tue Jul 30 09:20:58 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 00:20:58 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <51F6FB51.2080707@gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
Message-ID: <1375168858849-4660312.post@n4.nabble.com>

I?m sorry. I have not read your previous post. I do not which method you have
tried. Yes, I can create a slideshow with slides of different length. I can
even create a sample and share it over SkyDrive with you temporarily before
I remove it. What I would like to know is how you create your slideshow with
slides of different length. Would you mind to show me the link that leads to
your previous post? Or, would you mind to re-write your experience? Your
efforts are truly appreciated. Thank you.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277p4660312.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From bobm-ffmpeg at burner.com  Tue Jul 30 10:09:50 2013
From: bobm-ffmpeg at burner.com (Bob Maple)
Date: Tue, 30 Jul 2013 02:09:50 -0600
Subject: [FFmpeg-user] libx264 not found
In-Reply-To: <CAJtDrUduV1pFpU9VeVkmGVUmyY0GU6t43M6SkOJ3C6x1+hRa5g@mail.gmail.com>
References: <CAJtDrUduV1pFpU9VeVkmGVUmyY0GU6t43M6SkOJ3C6x1+hRa5g@mail.gmail.com>
Message-ID: <51F774CE.8030907@burner.com>

On 7/29/2013 7:21 PM, Rose Perrone wrote:

> I'm trying to use the encoder h264 with ffmpeg. I installed x264 from
> source and with few, and I configured ffmpeg with ./configure --enable-gpl
> --enable-shared --extra-ldflags="-ldl"  --enable-libx264. Yet the
> config.log tells me fatal error: 'x264' file not found, and I get the error
> libx264 not found.

Configure x264 with --enable-shared and then make install.  Then
reconfigure ffmpeg and see if it finds it.

Bob


From soho123.2012 at gmail.com  Tue Jul 30 10:32:27 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Tue, 30 Jul 2013 16:32:27 +0800
Subject: [FFmpeg-user] current timestamp question in ffserver
Message-ID: <CAFd2cZUeUiUpa9q73042T1y1VfT9JbSOGLp-AirMWcaAja3qAw@mail.gmail.com>

Hi ,


Does anyone have idea about timestamp used by ffserver?
In ffserver.c, function http_server(), I can see the code :
cur_time = av_gettime() / 1000;


As my understanding, the return value of av_gettime() is "us"(microsecond),
but why ffserver set
cur_time=av_gettime()/1000? it is in "ns"(nanosecond), right?

From nomiya at galaxy.dti.ne.jp  Tue Jul 30 10:53:18 2013
From: nomiya at galaxy.dti.ne.jp (Masaru Nomiya)
Date: Tue, 30 Jul 2013 17:53:18 +0900
Subject: [FFmpeg-user] libx264 not found
In-Reply-To: <CAJtDrUduV1pFpU9VeVkmGVUmyY0GU6t43M6SkOJ3C6x1+hRa5g@mail.gmail.com>
References: <CAJtDrUduV1pFpU9VeVkmGVUmyY0GU6t43M6SkOJ3C6x1+hRa5g@mail.gmail.com>
Message-ID: <87r4egfmyp.wl%nomiya@galaxy.dti.ne.jp>

Hello,

In the Message; 

  Subject    : [FFmpeg-user] libx264 not found
  Message-ID : <CAJtDrUduV1pFpU9VeVkmGVUmyY0GU6t43M6SkOJ3C6x1+hRa5g at mail.gmail.com>
  Date & Time: Mon, 29 Jul 2013 18:21:01 -0700

Rose Perrone <rose at cs.stanford.edu> has written:

> I'm trying to use the encoder h264 with ffmpeg. I installed x264 from
> source and with few, and I configured ffmpeg with ./configure --enable-gpl
> --enable-shared --extra-ldflags="-ldl"  --enable-libx264. Yet the
> config.log tells me fatal error: 'x264' file not found, and I get the error
> libx264 not found.

Have you compiled x264 with --enable-shared option?

Regards,

---
????? Masaru Nomiya             mail-to: nomiya @ galaxy.dti.ne.jp
????
???? "Bill! You married with Computers.
          Not with Me!"
         "No..., with money."

From jiachielee at live.com  Tue Jul 30 13:13:28 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 04:13:28 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <51F6FB51.2080707@gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
Message-ID: <1375182808947-4660318.post@n4.nabble.com>

Disclaimer: I do not in any way endorse the contents of this slideshow. I?m
just demoing how I use ffmpeg to create slideshow with slides of varying
length.http://sdrv.ms/13rDjUJThe ?job.7z? and ?job-2.7z? contain text files,
which can be made the batch file I used by changing its filename extension.
?job? is used for creating slideshow. ?job-2? is used for extracting the
frames. From the slides, I created files of ?1.mkv, 2.mkv, 3.mkv ? 12.mkv?.
Then I use mkvtoolnix to merge them together into a single. One thing I like
about mkvtoolnix is that it has a GUI front end.In this slideshow, I makes
the first slide lasts for 10 seconds; second slide lasts for 11 seconds;
third slide lasts for 12 seconds; fourth slide lasts for 10 seconds; fifth
slide lasts for 11 seconds; sixth slide lasts for 12 seconds; seventh slide
lasts for 10 seconds; eighth slide lasts for 11 seconds; ninth slide lasts
for 12 seconds; tenth slide lasts for 10 seconds; eleventh slide lasts for
11 seconds; twelfth slide lasts for 12 seconds;I don?t plan to keep this
demo for long. As such, I shall remove it after August 7, 2013.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277p4660318.html
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From yugant at gmail.com  Tue Jul 30 13:32:39 2013
From: yugant at gmail.com (Yugant Khanna)
Date: Tue, 30 Jul 2013 17:02:39 +0530
Subject: [FFmpeg-user] Capture and RTMP
Message-ID: <CAFnvMkuaKkSVfiBY3bgr6NRZtyJ2hMuPPJdBd3JpmwKUeLQV1g@mail.gmail.com>

Hi , currently i am capturing Video in segments with my capture Card using
the following ffmpeg command line :

ffmpeg  -f dshow -i video="Video Capture":audio="Audio Capture" -an  -c:v
libx264 -b:v 128k   -flags -global_header -map 0 -f segment -segment_time 4
 -segment_list test.m3u8 -segment_format mpegts stream%05d.ts



I need to also add user preview to my capture application. Using the
following command line i could through output to the AMS5 server :

ffmpeg -rtbufsize 702000k -f dshow -r 25 -i video="Video
Capture":audio="Audio Capture" -crf 30 -preset ultrafast -acodec aac
-strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -r 25 -b:v
500k -f flv rtmp://192.168.36.251/live/streamName

Can i combine the above 2 commands to Capture and stream simultaneously ?

regards

From smarrocco at ringsidecreative.com  Tue Jul 30 14:37:35 2013
From: smarrocco at ringsidecreative.com (Sam Marrocco)
Date: Tue, 30 Jul 2013 08:37:35 -0400
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <8CDA830CFEB43147A86E9A144320F7C605CD27E5@NHNZDNEX01.dn.nhnz.tv>
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>
	<51E7F54D.2050708@ringsidecreative.com>
	<CAFHbX1KC3Z-dK3GG8U8_2=KL7uyMsdPTQzHbc6Tjk3N0Zqjwcw@mail.gmail.com>
	<51E804EB.4060805@ringsidecreative.com>
	<8CDA830CFEB43147A86E9A144320F7C605CD27E5@NHNZDNEX01.dn.nhnz.tv>
Message-ID: <51F7B38F.6030100@ringsidecreative.com>


On 7/29/2013 6:50 PM, Wayne Poll wrote:

Sorry this is so late as a response (I've been away) but the fix is easy Sam; you can exclude the ninth track like this....

./ffmpeg -i input.mxf -map 0 -map -0:9:0 -c copy -c:v prores -profile:v 3 output.mov

The "-map 0 -map -0:9:0" will tell ffmpeg to copy all streams EXCEPT stream 0:9 - which in my case is a metadata track (possibly TC - not sure).

HTH someone...





Wayne,
Definitely another useful piece of information, thanks.
Unfortunately I *do* need the ninth track (TC) so the tracks must be specified explicitly.

sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>

From andycivil at gmail.com  Tue Jul 30 16:53:31 2013
From: andycivil at gmail.com (Andy Civil)
Date: Tue, 30 Jul 2013 10:53:31 -0400
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375168858849-4660312.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>	<1375099174769-4660296.post@n4.nabble.com>	<51F6FB51.2080707@gmail.com>
	<1375168858849-4660312.post@n4.nabble.com>
Message-ID: <51F7D36B.6080703@gmail.com>

On 2013-07-30 3:20 AM, thljcl wrote:
> I?m sorry. I have not read your previous post. I do not which method you have
> tried. Yes, I can create a slideshow with slides of different length. I can
> even create a sample and share it over SkyDrive with you temporarily before
> I remove it. What I would like to know is how you create your slideshow with
> slides of different length. Would you mind to show me the link that leads to
> your previous post? Or, would you mind to re-write your experience? Your
> efforts are truly appreciated. Thank you.

I describe it in this post:
<http://ffmpeg-users.933282.n4.nabble.com/Create-video-with-multiple-images-tp2524565p2527808.html>
That post references a google site where I describe what I did (which is a good 
thing because it was three years ago and I've forgotten). The links to posts on 
that site no longer work, as the archive must have moved. Also, at the bottom of 
the google site there is a link that you can download my example - it is a 
computer voice reading the question from Francois which I was answering 
(including the bad english!). The audio bitrate is only 16k and the video 
resolution is small, but the whole file is only 268 KB.

However, I think today your method is probably better. Your example slideshow is 
1.1 MB and it has no sound, but the resolution is much bigger and it's also longer.

I have downloaded your examples, I'm too busy to look right now, but I think I 
will learn from them. Thank you!

--
Andy

From edward.chopuryan at gmail.com  Tue Jul 30 10:25:51 2013
From: edward.chopuryan at gmail.com (edward20)
Date: Tue, 30 Jul 2013 01:25:51 -0700 (PDT)
Subject: [FFmpeg-user] Decode H264 and save to yuv file
In-Reply-To: <1375079673569-4660291.post@n4.nabble.com>
References: <1374840030366-4660204.post@n4.nabble.com>
	<tencent_05C78B781D6F668440CD3187@qq.com>
	<1375079673569-4660291.post@n4.nabble.com>
Message-ID: <1375172751997-4660315.post@n4.nabble.com>

Anyone? Please I just need part of C code which
explains how to write decoded YUV data to file.
Is there any example of it?

Here is what I have tried so far



void write_frame_to_file(AVFrame* frame, int width, int height, int iframe)
{
        FILE* outfile;
        char filename[32];
        int y;

        // Open file
        sprintf(filename, "frame%d.yuv", iframe);
        outfile = fopen(filename, "wb");
        if (outfile == NULL) {
                return;
        }

        // Write pixel data
        for (y = 0; y < height; ++y) {
                fwrite(frame->data[0]+y*frame->linesize[0], 1, width*3,
outfile);
        }

        // Close file
        fclose(outfile);
        printf("file is closed\n");
}




--
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From rajivrp at gmail.com  Tue Jul 30 07:43:50 2013
From: rajivrp at gmail.com (Rajiv Pandya)
Date: Tue, 30 Jul 2013 11:13:50 +0530
Subject: [FFmpeg-user] Video bit rate missing from FFProbe output
In-Reply-To: <CA+XaXC=za6GKxgvfjX2tRTbmU4BZ7nGoAUKvOvnxypNbK56Tmw@mail.gmail.com>
References: <CA+XaXC=J7BMNZTF3orYpr_pNrDbmYdVmUycwJeWMa9b7NHGh5w@mail.gmail.com>
	<loom.20130723T220119-727@post.gmane.org>
	<CA+XaXCnhFAYePWHi12_zA_GDqXcqLsa8y9CdMokb1qrrLnh5CQ@mail.gmail.com>
	<loom.20130725T000737-643@post.gmane.org>
	<CA+XaXC=za6GKxgvfjX2tRTbmU4BZ7nGoAUKvOvnxypNbK56Tmw@mail.gmail.com>
Message-ID: <CADCjQEJ5B0BeeuuS0o2y2p+wN02U7Pa4GuDN0qE6KtyJAuME8g@mail.gmail.com>

Hi Carl,

S
orry for top-posting in the last response ... but I didn't fully get what
you said about the patch (to get max bit-rate if no bit-rate is defined) in
your last response.

- Can you pls let me know where can I get that patch from ?
 - Has this behavior changed intentionally in latest FFProbe (as it used to
show the bit-rate in old versions) ?
 - Is this fix going to be a part of major release in near future ?

Thanks a lot

From jpboard2 at yahoo.com  Tue Jul 30 18:53:02 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 30 Jul 2013 09:53:02 -0700 (PDT)
Subject: [FFmpeg-user] Specify log file for 2-pass encoding
Message-ID: <1375203182.495.YahooMailNeo@web164703.mail.gq1.yahoo.com>

I'm using ffmpeg to do 2-pass mpeg encoding with '-ovc lavc' and the
mpeg4 codec on Linux.? This
generates a divx2pass.log file in the directory where the files are located.
I'm trying to encode multiple files at the same time on different CPU cores
and the log files are clobbering each
other.? How can I specify which name
I'd like the log file to have?? The ffmpeg man page says the -passlogfile can
be used to specify
the logfile for pass 1.? But how do I tell ffmpeg to use that
log file in pass 2?

The man page and the HTML documentation doesn't mention divx2pass.log.
Thanks

From pkoshevoy at gmail.com  Tue Jul 30 18:54:30 2013
From: pkoshevoy at gmail.com (Pavel Koshevoy)
Date: Tue, 30 Jul 2013 10:54:30 -0600
Subject: [FFmpeg-user] Decode H264 and save to yuv file
In-Reply-To: <1375172751997-4660315.post@n4.nabble.com>
References: <1374840030366-4660204.post@n4.nabble.com>
	<tencent_05C78B781D6F668440CD3187@qq.com>
	<1375079673569-4660291.post@n4.nabble.com>
	<1375172751997-4660315.post@n4.nabble.com>
Message-ID: <51F7EFC6.1070808@gmail.com>

On 7/30/2013 2:25 AM, edward20 wrote:
> Anyone? Please I just need part of C code which
> explains how to write decoded YUV data to file.


You are asking API questions on the wrong mailing list.  API questions 
are answered on https://lists.ffmpeg.org/mailman/listinfo/libav-user/


> Is there any example of it?

http://ffmpeg.org/doxygen/trunk/examples.html

In particular -- 
http://ffmpeg.org/doxygen/trunk/doc_2examples_2decoding_encoding_8c-example.html


> Here is what I have tried so far
>
>
>
> void write_frame_to_file(AVFrame* frame, int width, int height, int iframe)
> {
>          FILE* outfile;
>          char filename[32];
>          int y;
>
>          // Open file
>          sprintf(filename, "frame%d.yuv", iframe);
>          outfile = fopen(filename, "wb");
>          if (outfile == NULL) {
>                  return;
>          }
>
>          // Write pixel data
>          for (y = 0; y < height; ++y) {
>                  fwrite(frame->data[0]+y*frame->linesize[0], 1, width*3,
> outfile);

not width * 3, it should be frame->linesize[0] (if you really want to 
save it line by line)

And unless the frame pixel format is packed YUV (it's usually planar) 
you wouldn't be saving the UV part of the image.  For planar formats 
you'll probably want to save all the frame->data planes.

Also, there is no guarantee that the frame data is YUV at all (it could 
be BGRA), so you'll probably want to convert the frame to YUV pixel 
format.  Take a look at 
http://ffmpeg.org/doxygen/trunk/doc_2examples_2filtering_video_8c-example.html 
(or 
http://ffmpeg.org/doxygen/trunk/doc_2examples_2scaling_video_8c-example.html 
-- it's probably all you really need).


From jiachielee at live.com  Tue Jul 30 18:54:33 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 09:54:33 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <51F7D36B.6080703@gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375168858849-4660312.post@n4.nabble.com>
	<51F7D36B.6080703@gmail.com>
Message-ID: <1375203273449-4660324.post@n4.nabble.com>

If you file size is your concern, you can re-encode it using lossy
compression after you?ve done with the editing process. I specifically
choose lossless compression and no chroma subsampling to preserve the video
quality. It?s playable by VLC, Media Player Classic Home Cinema, or Windows
Media Player (with K-Lite Codec Pack Mega).
With regards to compatibility, I know that setting of YUV444P and H.264 High
4:4:4 Predictive Profile (Hi444PP) is not exactly widely compatible with
most media players or commercially available media players. It?s meant for
keeping the master copy for future editing.
Again, if compatibility is your concern,

ffmpeg -i "input.mkv" -vcodec "libx264" -crf "18" -pix_fmt "yuv420p"
"output.mkv"

This is a tradeoff of image quality for compatibility and file size. -crf
?18? may be considered a good choice for preserving image quality when it
comes to lossy compression. While it?s generally unpredictable to know the
average bit rate for a given constant rate factor, I do have a theory behind
the algorithms to determine the average bit rate from constant rate factor.
If you are interested, I can post it here as well. For myself, I generally
choose different constant rate factor value for different videos. You may
have wondered why I did not do lossy compression on the slideshows I
created.
The short answer is for local computer playback it is simply unnecessary for
slideshows.
Slideshows are really different from animation in a number of aspects. First
of all, there are many frame duplicates because of the slide of a certain
length. In H.264 inter-frame picture prediction lossless compression, we can
reduce the file size by reducing the redundancy across the frames. Of
course, based on my own tests, H.264 inter-frame picture prediction is not
perfect or ideal in any sense, although considered excellent in comparison
with other codecs. The low bit rate is due to the repetition of the same
image; rather than the reduction of image quality. This is what inter-frame
compression does. This is somewhat similar to repeating the same characters
in text file and compressing it using 7-zip.
Also, when the bit rate is low, it will take less resources for a CPU to
decode it. For live video or animations, however, things are entirely
different. While I may still keep the master copy, it?s not used for local
playback because of the lack of computing resources to decode it quickly
enough during the playback. If I can?t play it even on my PC, why I still
keep it? It is master copy, folks. Lossy compression is required for local
playback. The bit rate for the lossless version is usually very high. Unless
for the purpose of distribution or sharing, I generally don?t do chroma
subsampling even if I do lossy compression.
As far as I know, YouTube cannot decode YUV444p correctly. So you still need
to re-encode it before you want to upload it to YouTube.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277p4660324.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jiachielee at live.com  Tue Jul 30 19:21:21 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 10:21:21 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
Message-ID: <1375204881793-4660326.post@n4.nabble.com>

Oh yes, with regards to why I specify ?yuv444p?, do you happen to know that
the chroma subsampling done by different encoders is not necessarily using
the same algorithms? In other words, while the main idea behind chroma
subsampling is same, each encoder may drop different bits from image during
the encoding the process. To be more specific, when we open the image file
for viewing or editing, generally the picture viewer or media player would
decode the image to bitmap and then present it to us on screen. It?s similar
to word document, which is actually a zip file. Each time we open it,
Microsoft word would extract it. In fact, you can even change the filename
extension of docx to zip if you want. The same applies to Excel, PowerPoint,
etc.
Do you realize that it?s generally a bad idea to save the edits directly to
jpg even when the source is the picture you take using your camera? The best
practice would be to convert it to lossless format such as bmp or PNG. Do
your editing there. You can then save it back to jpg when you have done with
editing. You make many edits and save them directly to jpg, you will suffer
what we call ?generation loss?. Lossless editing in jpg file is still
possible but only for limited instances, such as ?gray scaling?, ?rotating
and flipping?, ?cropping?, and ?rescaling?. Generally, if you draw directly
from jpg in an image editor and save it to jpg, what you will get is ?lossy
editing?.
Granted, if the source is encoded to a ?color space?, you then use the same
encoder to encode the source in the same color space; it?s really not
necessarily to choose ?yuv444p? in this case. But jpeg and H.264 are two
different codecs. Ffmpeg still needs to decode the image before it can
re-encode it. Likewise, you can do frame rate conversion using a different
set of tools, you cannot expect ffmpeg to recover the original lower frame
rate with all the original frames; the algorithms being used might not be
the same.
If the source is PNG image, I don?t even want to specify ?YUV444p?; it will
be chosen automatically if no pixel format is specified.




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From andycivil at gmail.com  Tue Jul 30 19:48:00 2013
From: andycivil at gmail.com (Andy Civil)
Date: Tue, 30 Jul 2013 13:48:00 -0400
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375203273449-4660324.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>	<1375099174769-4660296.post@n4.nabble.com>	<51F6FB51.2080707@gmail.com>	<1375168858849-4660312.post@n4.nabble.com>	<51F7D36B.6080703@gmail.com>
	<1375203273449-4660324.post@n4.nabble.com>
Message-ID: <51F7FC50.3020502@gmail.com>

On 2013-07-30 12:54 PM, thljcl wrote:
> If you file size is your concern,  ...

My interest was really intellectual/academic; I had seen a tutorial on Java 
where the video seemed out of sync with the voiceover, and when I analysed it I 
found that the thing had simply been produced with a frame rate of 1 fps. I 
thought it should be possible to do this much more efficiently. (I'm a fan of 
efficiency, sometimes as an end in itself!)

At first, I thought your method wasn't working well because the photography 
slideshow was over 100MB, but the 'relativity' thing you posted was very good, 
and I'm convinced that your process is useful.

I think it would be a good way to distribute tutorials (slideshow with 
voiceover) especially to places where data rate is an issue. Being able to vary 
the slide delay (to make the slide appear at the right point in the narrative) 
is the feature that is missing from other "make a slideshow with FFmpeg" 
suggestions.

--
Andy

From jpboard2 at yahoo.com  Tue Jul 30 20:06:06 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 30 Jul 2013 11:06:06 -0700 (PDT)
Subject: [FFmpeg-user] Turn Off Multi-Threading
Message-ID: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>

I'm using ffmpeg to edit a large file with the Select filter (-vf select) on a Linux
machine.? When I run 'top' this process is consuming almost 500%
of my CPU (according to 'top').? Is ffmpeg multi-threading this
edit?? Or is the Select filter multi-threaded?? If so, how can I
tell ffmpeg to use a single-thread?

Thank you.

From h.reindl at thelounge.net  Tue Jul 30 20:15:37 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Tue, 30 Jul 2013 20:15:37 +0200
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <51F802C9.2010401@thelounge.net>



Am 30.07.2013 20:06, schrieb James Board:
> I'm using ffmpeg to edit a large file with the Select filter (-vf select) on a Linux
> machine.  When I run 'top' this process is consuming almost 500%
> of my CPU (according to 'top').  Is ffmpeg multi-threading this
> edit?  Or is the Select filter multi-threaded?  If so, how can I
> tell ffmpeg to use a single-thread?

add "-threads 1" as param

ffmpeg is designed to be as fast as possible and so taking
all available ressources which is good in 99% of all cases
beause 100% CPU load does not mean you can't do anything
else in this time on modern hardware

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From jiachielee at live.com  Tue Jul 30 20:13:32 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 11:13:32 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <51F7FC50.3020502@gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375168858849-4660312.post@n4.nabble.com>
	<51F7D36B.6080703@gmail.com>
	<1375203273449-4660324.post@n4.nabble.com>
	<51F7FC50.3020502@gmail.com>
Message-ID: <1375208012491-4660329.post@n4.nabble.com>

Andy, the wallpaper slideshow I created lasts for 22 minutes 30 seconds. It?s
much longer than the ?relativity? slideshow. You did not download it, did
you? Well, I admit that it?s mainly due to my own laziness to specify
varying length for different slide. When each slide has the same duration,
it requires less efforts. By the way, even though it?s lossless compression,
its average bit rate is at 765 kbps. Well, if it?s animation or live
recording, I would say that the bit rate would be around 10 Mbps or larger.
Yes, it?s unbelievably large for lossless format. That also shows how much
data we lose during lossy compression. Lossy compression is a good choice
for distribution, local playback. But we try to do editing, we would try
keep as much fidelity as possible.
I do study special relativity quite a lot. But the slides are written by
someone else using PowerPoint.  I just ?borrowed? it without permission.
Well, it?s publicly available anyway. Ha ha? Special Relativity is so
complex that I don?t even know where to begin with. There are so much to
talk about. It?s not about ?special relativity?. So I just ?borrowed? the
material.
Well, for animation or live recording, using CRF value of 18+- can achieve
the average bit rate around 3000 Kbps. I usually do the calculation first
before I decide which CRF value to be used; sometimes smaller; sometimes
larger; it depends on the complexity of the source.




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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277p4660329.html
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From onemda at gmail.com  Tue Jul 30 20:32:34 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 30 Jul 2013 18:32:34 +0000
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>

On 7/30/13, James Board <jpboard2 at yahoo.com> wrote:
> I'm using ffmpeg to edit a large file with the Select filter (-vf select) on
> a Linux
> machine.  When I run 'top' this process is consuming almost 500%
> of my CPU (according to 'top').  Is ffmpeg multi-threading this
> edit?  Or is the Select filter multi-threaded?  If so, how can I
> tell ffmpeg to use a single-thread?

select filter use single thread only.

Main reason why select filter use so much CPU is probably
decoding process of video itself.

From jpboard2 at yahoo.com  Tue Jul 30 21:28:28 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 30 Jul 2013 12:28:28 -0700 (PDT)
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
Message-ID: <1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>

>select filter use single thread only.

>
>Main reason why select filter use so much CPU is probably
>decoding process of video itself.

How can it use 500 percent of the CPU if it's single thread only?

From onemda at gmail.com  Tue Jul 30 21:30:16 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 30 Jul 2013 19:30:16 +0000
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>

On 7/30/13, James Board <jpboard2 at yahoo.com> wrote:
>>select filter use single thread only.
>
>>
>>Main reason why select filter use so much CPU is probably
>>decoding process of video itself.
>
> How can it use 500 percent of the CPU if it's single thread only?

I said that filter itself is single thread only, read what I wrote above
once more.

From jiachielee at live.com  Tue Jul 30 21:29:58 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 12:29:58 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375208012491-4660329.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375168858849-4660312.post@n4.nabble.com>
	<51F7D36B.6080703@gmail.com>
	<1375203273449-4660324.post@n4.nabble.com>
	<51F7FC50.3020502@gmail.com>
	<1375208012491-4660329.post@n4.nabble.com>
Message-ID: <1375212598592-4660333.post@n4.nabble.com>

Well, let me explain why there is a big difference of average bit rate
between ?Relativity? and ?Wallpapers?. I said previously that ?Wallpapers?
is considerably longer, thus having much larger file size. But you may have
argued that there is a big difference in average bit rate as well. The
average bit rate of ?Relativity? is around 70 kbps; but the average bit rate
of ?wallpapers? is around 765 kbps. It?s more than ten times larger!
The story to be told here is actually about how lossless compression for
image/video works. I actually wrote much about it at my web site at
http://bit.ly/17RMRLW and http://bit.ly/18LfN9F. Granted, my web site is not
exactly top ten web site in any rank. I?m pretty lazy to update it regularly
as well.
That said, I would still tell the theory here, if you are interested. If
not, just ignore this post.
The general idea is that ?lossless compression? relies on the ability to
find redundancy of data. For example, how much will the file size for an
image of 1280 ? 720 pixels? Is it possible to tell the file size by knowing
the resolution? The answer is yes, for uncompressed image file, if we know
the other factor that affects the file size, which is bit depth. The bit
depth reported by Media Player Classic Home Cinema should be more accurately
called ?bit per channel?. A 24-bit image has three channels; the bit per
channel is 8 bit. For an uncompressed image file, the file size can be
calculated:
Resolution ? bit depth ? 1/8
In the case for a 24-bit image file of 1280 ? 720 pixels,
1280?720?24?1/8 Bytes = 2,764,800 Bytes ? 2.64 MiB
Well, you don?t have to take my words for granted. Just open Microsoft Paint
and create a BMP file. The BMP file has contain other information, such as
header and so on; the actual file size is actually slightly bigger.
As such a one-minute uncompressed video of 1280 ? 720 pixels and 24 FPS
would have the file size of 3,981,312,000 Bytes or approximately 3.71 GiB.
By default, when you create a new image file using Microsoft Paint, it only
has one color, which is white. Bit depth specify the number of possible
colors that a pixel can represent. In an image file, a pixel represent a
single color. 24 bit would mean 16,777,216 possible colors. But since a
pixel can only represent a color, an image file of 1280 ? 720 pixels can at
most has 921,600 colors. Human eyes can only discriminate up to 10 million
colors; thus 24 bit is more than enough in most cases. If you save that
?empty? image file to PNG file, the file size is reduced to only 4.16 KiB.
PNG is a lossless file format. That is possible because of the redundancy;
meaning that all other pixels sharing the same colors. If your image file
has 921,600 colors, no matter which lossless file format you choose, you
should get the file size of very close, if not the same.
Of course, as I explained previously, when it comes to lossless encoding of
videos, there is also inter-frame compression, meaning that if redundancy of
data across the frames is spotted, the encoder can reduce the file size as
well. That would explain the great difference of average bit rate between
?Wallpapers? and ?Relativity?. ?Wallpapers? are made from photographs; they
do have more colors compared with ?Relativity?, which contains more texts
(single-color) and single-color background.
In fact, you can create a slideshow video directly from PowerPoint. The
downside is that you cannot specify the length of each slide, the
compression level, frame rate, or basically every encoding setting.
PowerPoint allows us to save the slides to PNG files, which I used to create
the slideshow using ffmpeg.
Also, I don?t keep PNG files. I can always extract them from lossless H.264
video file when I need them for editing. The reason is to save storage
space.




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From jpboard2 at yahoo.com  Tue Jul 30 22:41:17 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 30 Jul 2013 13:41:17 -0700 (PDT)
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
Message-ID: <1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>

> I said that filter itself is single thread only, read what I wrote above

> once more.

I read it.? It doesn't explain why the CPU is at 500 pecent.? Is it because
the rest of ffmpeg is multi-threaded, and that's why it's at 500 percent?
Please forgive me if I don't undersdtan the internal architecture of ffmpeg.
Thanks.

From onemda at gmail.com  Tue Jul 30 22:44:19 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 30 Jul 2013 20:44:19 +0000
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>

On 7/30/13, James Board <jpboard2 at yahoo.com> wrote:
>> I said that filter itself is single thread only, read what I wrote above
>
>> once more.
>
> I read it.  It doesn't explain why the CPU is at 500 pecent.  Is it because
> the rest of ffmpeg is multi-threaded, and that's why it's at 500 percent?
> Please forgive me if I don't undersdtan the internal architecture of
> ffmpeg.

Everything else except select filter that you use (but you did not mentioned it)
can be multithreaded: decoders, encoders, and other filters (but not
select one).


'ffmpeg -filters' shows which filters can use multiple threads.

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

From h.reindl at thelounge.net  Tue Jul 30 22:58:17 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Tue, 30 Jul 2013 22:58:17 +0200
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
Message-ID: <51F828E9.9070600@thelounge.net>



Am 30.07.2013 22:41, schrieb James Board:
>> I said that filter itself is single thread only, read what I wrote above
> 
>> once more.
> 
> I read it.  It doesn't explain why the CPU is at 500 pecent.  Is it because
> the rest of ffmpeg is multi-threaded, and that's why it's at 500 percent?
> Please forgive me if I don't undersdtan the internal architecture of ffmpeg

it has nothing to do with ffmpeg or not ffmpeg

if a process is using more than one thread it
can use more than one CPU and 500% means simply
5 cores with a load of 100% - not more and not less

the same for any other software with more than one thread


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From WPoll at nhnz.tv  Tue Jul 30 23:05:29 2013
From: WPoll at nhnz.tv (Wayne Poll)
Date: Tue, 30 Jul 2013 21:05:29 +0000
Subject: [FFmpeg-user] Is mixing of audio channels happening by default
 in ffmpeg?
In-Reply-To: <51F7B38F.6030100@ringsidecreative.com>
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>
	<51E7F54D.2050708@ringsidecreative.com>
	<CAFHbX1KC3Z-dK3GG8U8_2=KL7uyMsdPTQzHbc6Tjk3N0Zqjwcw@mail.gmail.com>
	<51E804EB.4060805@ringsidecreative.com>
	<8CDA830CFEB43147A86E9A144320F7C605CD27E5@NHNZDNEX01.dn.nhnz.tv>
	<51F7B38F.6030100@ringsidecreative.com>
Message-ID: <8CDA830CFEB43147A86E9A144320F7C605CD5CDA@NHNZDNEX01.dn.nhnz.tv>

When I use this process to convert Sony F55 XAVC 4k footage to Apple ProRes 422 (HQ) quick time files, the time code in the resultant quick time matches the original XAVC MXF time code perfectly, even when excluding the metadata track. 

Wayne Poll?|?Technical & IT Systems Manager?|
NHNZ | 5 Melville Street, Dunedin 9016, New Zealand |


-----Original Message-----
From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Sam Marrocco
Sent: Wednesday, 31 July 2013 0:38
To: FFmpeg user questions
Subject: Re: [FFmpeg-user] Is mixing of audio channels happening by default in ffmpeg?


On 7/29/2013 6:50 PM, Wayne Poll wrote:

Sorry this is so late as a response (I've been away) but the fix is easy Sam; you can exclude the ninth track like this....

./ffmpeg -i input.mxf -map 0 -map -0:9:0 -c copy -c:v prores -profile:v 3 output.mov

The "-map 0 -map -0:9:0" will tell ffmpeg to copy all streams EXCEPT stream 0:9 - which in my case is a metadata track (possibly TC - not sure).

HTH someone...





Wayne,
Definitely another useful piece of information, thanks.
Unfortunately I *do* need the ninth track (TC) so the tracks must be specified explicitly.

sam marrocco | chief technical officer
ringside.cutters.picnic.moonlink

248 548 2500 w
248 910 3344 c

ringsidecreative.com

<http://ringsidecreative.com/>
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
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From bored_to_death85 at yahoo.com  Tue Jul 30 23:11:29 2013
From: bored_to_death85 at yahoo.com (M. V.)
Date: Tue, 30 Jul 2013 14:11:29 -0700 (PDT)
Subject: [FFmpeg-user] problem with converting rtp packets to an audio
	format
In-Reply-To: <1375131311.43048.YahooMailNeo@web165006.mail.bf1.yahoo.com>
References: <1375131311.43048.YahooMailNeo@web165006.mail.bf1.yahoo.com>
Message-ID: <1375218689.55796.YahooMailNeo@web165003.mail.bf1.yahoo.com>

I searched more and more but couldn't find what am I doing wrong. so I think I should simply ask this:

- can ffmpeg read rtp (+sip/sdp) packets from a file and converts them to an output audio file? if yes, how?

I know it's a basic question but I really couldn't find the correct answer to that with document reading and googling.

I would be glad if someone gives me a hint or suggestion.
thank you.





>________________________________
> From: M. V. <bored_to_death85 at yahoo.com>
>To: "ffmpeg-user at ffmpeg.org" <ffmpeg-user at ffmpeg.org> 
>Sent: Tuesday, July 30, 2013 1:25 AM
>Subject: [FFmpeg-user] problem with converting rtp packets to an audio format
> 
>
>hi guys,
>
>I'm new to ffmpeg. I'm developing a program which receives and recognizes rtp (+sip/sdp) packets from interface and writes rtp content (just rtp part, not udp and ip header) to a file. now I wanna use "ffmpeg"? to convert this file to an audio format. for this I use:
>??? #ffmpeg -f rtp -i /locaton/of/my/rtp/file output.au
>and I receive this error:
>
>-----------------------------------------------------------------------------
>[rtp @ 0x2d62880] Guessing on RTP content - if not received properly you need an SDP file describing it
>Guessed Channel Layout for? Input Stream #0.0 : mono
>Input #0, rtp, from '/location/to/my/rtp/file':
>? Duration: N/A, bitrate: 64 kb/s
>??? Stream #0:0: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
>Output #0, au, to '/output.au':
>? Metadata:
>??? encoder???????? : Lavf55.12.100
>??? Stream #0:0: Audio: pcm_s16be ([3][0][0][0] / 0x0003), 8000 Hz, mono, s16, 128 kb/s
>Stream mapping:
>? Stream #0:0 -> #0:0 (pcm_alaw -> pcm_s16be)/location/to/my/rtp/file: Connection timed out
>size=?????? 0kB time=-577014:-32:-22.-77 bitrate=N/A??? 
>size=?????? 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)
>-----------------------------------------------------------------------------
>
>
>so what am I doing wrong?
>- should I write the whole rtp packets (with ethernet/ip/udp headers) to file or just rtp-section of packets are enough?
>- do I also need to write all related sdp (and sip) packets to a separate file?
>- if so, how can I introduce this file to ffmpeg?
>
>I googled a lot and checked documentation, but didn't find an answer. (most of what I found was about playing rtp from ffmpeg)
>
>
>any help would be much appreciated.
>
>thank you.
>_______________________________________________
>ffmpeg-user mailing list
>ffmpeg-user at ffmpeg.org
>http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

From jpboard2 at yahoo.com  Tue Jul 30 23:47:16 2013
From: jpboard2 at yahoo.com (James Board)
Date: Tue, 30 Jul 2013 14:47:16 -0700 (PDT)
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>
Message-ID: <1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>

>Everything else except select filter that you use (but you did not mentioned it)

>can be multithreaded: decoders, encoders, and other filters (but not
>select one).

I'm not using any other filters.? Anyway, does anyone know
how to turn off multi-threading?? I tried "-threads 1" and
I see see CPU usage at almost 200 percent for ffmpeg.

From onemda at gmail.com  Tue Jul 30 23:53:49 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Tue, 30 Jul 2013 21:53:49 +0000
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>
	<1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P4Spn5MC3OPAkpx0kVfx5mArf4wSC520KB6uM7t9DRrWQ@mail.gmail.com>

On 7/30/13, James Board <jpboard2 at yahoo.com> wrote:
>>Everything else except select filter that you use (but you did not
>> mentioned it)
>
>>can be multithreaded: decoders, encoders, and other filters (but not
>>select one).
>
> I'm not using any other filters.  Anyway, does anyone know
> how to turn off multi-threading?  I tried "-threads 1" and
> I see see CPU usage at almost 200 percent for ffmpeg.

Position in command line also matters.

From barsnick at gmx.net  Wed Jul 31 00:17:50 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Wed, 31 Jul 2013 00:17:50 +0200
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <51F828E9.9070600@thelounge.net>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<51F828E9.9070600@thelounge.net>
Message-ID: <20130730221750.GC20885@sunshine.barsnick.net>

On Tue, Jul 30, 2013 at 22:58:17 +0200, Reindl Harald wrote:
> it has nothing to do with ffmpeg or not ffmpeg
> 
> if a process is using more than one thread it
> can use more than one CPU and 500% means simply
> 5 cores with a load of 100% - not more and not less

Which still doesn't answer the question for the poor guy! (And I
believe he realized very well that that is exactly what is happening.)

The question being:
How - if at all - can ffmpeg be restricted to one thread?

Actually, the question is "to one CPU", but everyone seems to realize
that the use of multiple CPUs is due to multithreading. :-)

Since no-one is answering that (except: "position does matter", which
may lead in the right direction), let me offer a different solution
outside of ffmpeg: taskset

Using taskset (on Linux), you can explicitly restrict a process (and I
assume its children and its threads) to a single CPU. This doesn't
avoid multithreading, but it would allow ffmpeg to "clobber" only one
CPU, leaving the others alone.

See also
http://unix.stackexchange.com/questions/23106/limit-process-to-one-cpu-core

Moritz

From h.reindl at thelounge.net  Wed Jul 31 00:23:22 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Wed, 31 Jul 2013 00:23:22 +0200
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <20130730221750.GC20885@sunshine.barsnick.net>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<51F828E9.9070600@thelounge.net>
	<20130730221750.GC20885@sunshine.barsnick.net>
Message-ID: <51F83CDA.3090101@thelounge.net>



Am 31.07.2013 00:17, schrieb Moritz Barsnick:
> On Tue, Jul 30, 2013 at 22:58:17 +0200, Reindl Harald wrote:
>> it has nothing to do with ffmpeg or not ffmpeg
>>
>> if a process is using more than one thread it
>> can use more than one CPU and 500% means simply
>> 5 cores with a load of 100% - not more and not less
> 
> Which still doesn't answer the question for the poor guy! (And I
> believe he realized very well that that is exactly what is happening.)
> 
> The question being:
> How - if at all - can ffmpeg be restricted to one thread?
i answered this *hours* ago

-threads 1



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From h.reindl at thelounge.net  Wed Jul 31 00:30:51 2013
From: h.reindl at thelounge.net (Reindl Harald)
Date: Wed, 31 Jul 2013 00:30:51 +0200
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>
	<1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <51F83E9B.1090301@thelounge.net>



Am 30.07.2013 23:47, schrieb James Board:
>> Everything else except select filter that you use (but you did not mentioned it)
> 
>> can be multithreaded: decoders, encoders, and other filters (but not
>> select one).
> 
> I'm not using any other filters.  Anyway, does anyone know
> how to turn off multi-threading?  I tried "-threads 1" and
> I see see CPU usage at almost 200 percent for ffmpeg

please post complete command line if you
want help instead "i did this and that"

the params for the encoding are *before* the output file

ffmpeg -i "test.m4v" -y -f 'mp4' -vcodec 'libx264' -vb '1024k' -maxrate '4048k' -bufsize '30M' -acodec 'aac'
-strict '-2' -ab '128k' -profile 'main' -threads 1 "out.mp4"

is clearly *one* CPU, in this case H264 main with internal AAC encoder
ffmpeg-2.0 here and "-threads" works since years (0.6 or so)



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From cehoyos at ag.or.at  Wed Jul 31 01:48:40 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 30 Jul 2013 23:48:40 +0000 (UTC)
Subject: [FFmpeg-user] Turn Off Multi-Threading
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<51F802C9.2010401@thelounge.net>
Message-ID: <loom.20130731T014342-606@post.gmane.org>

Reindl Harald <h.reindl <at> thelounge.net> writes:

> ffmpeg is designed to be as fast as possible and so taking
> all available ressources which is good in 99% of all cases

I think it should be added that (especially when talking 
about "resources") it is not at all clear that the decision 
to use as many threads as available when decoding with 
FFmpeg was a very good one, using -threads 1 may be a 
good choice for many use-cases. (The overhead when using 
more than one thread is typically huge.)
This is of course mostly unrelated to the original 
question.

Carl Eugen


From jiachielee at live.com  Wed Jul 31 01:58:19 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 16:58:19 -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: <1375228699675-4660346.post@n4.nabble.com>

In doing scientific research, we learn the laws of Physics based on
experimental data and logical understanding and deduction; not because of
someone says so. That said, I would never claim that I?ll never be wrong; in
fact, I make mistakes as frequent as any human being does. I?m also told by
Buddha not to take others? words for granted for any other reasons other
than my own reasoning and understanding; no matter who they are.
I said that yuv444p can fully represent every single color in 24-bit RGB
color space is based on the algorithms used to encoding RGB color into
yuv444p. We know that lossy compression is irreversible. That is, there is
no way to recover the original information from a file of lossy compression.
The way to determine whether yuv444p can fully represent every single color
in 24-bit RGB color space is to determine whether the algorithm or function
to encode 24-bit RGB color space is reversible or if the function itself has
inverse function. Let me give you an example.
We are taught the existence of inverse trigonometric function. But
trigonometric functions themselves do not have inverse functions. Sounds
strange? Trigonometric functions are inverse functions of inverse
trigonometric function; but vice versa is not true. A function is a
one-to-one or many-to-one relationship. Inverse trigonometric function is
one-to-one relationship but trigonometric function is many-to-one
relationship. That is why inverse trigonometric function, strictly speaking,
is not the inverse function for trigonometric function.
In lossy compression, the algorithm is similar to many-to-one relationship.
That is, for a given encoder?s settings, we have an output file of certain
amount of information. Since some information is discarded during the
encoding process, we cannot recover the discarded information from the
output file. Such a function does not have an inverse. Or, we say that lossy
compression is irreversible.
How does YUV444p relate to RGB24? It depends on the standards being used.
But either way the function does have an inverse. Let?s take a look at ITU-R
BT.601 conversion at 

<iframe width="640" height="390"
src="http://www.youtube.com/embed/FyLN46hGZzU" frameborder="0"
allowfullscreen></iframe>

It?s a slideshow I specifically created to show the derivation. While I
derive the inverse transformation matrix myself; the conversion matrix for
RGB24 to YUV444P I got from http://bit.ly/14gWnv0.
Basically, if you?ve learned linear algebra before, you may understand right
away why YUV444p can represent all colors from R?G?B?. Before you denounce
my answer, please watch the video first to see if there is anything wrong
with my math.
The reason for the loss of color information due to chroma subsampling
scheme such as YUV420p or YUV422p is due to the sharing of color information
between pixels, rather than having each individual pixel has its own color
information in all three channels, which are Y?, Cb, and Cr. There is no
denying that the reason for such transformation is for the chroma
subsampling scheme, i.e. reduction of color information; but it?s not due to
the non-invertible transformation matrix.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Does-converting-to-yuv444p-by-default-make-sense-tp4660219p4660346.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From jam at tigger.ws  Wed Jul 31 02:40:42 2013
From: jam at tigger.ws (James Linder)
Date: Wed, 31 Jul 2013 08:40:42 +0800
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375182808947-4660318.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375182808947-4660318.post@n4.nabble.com>
Message-ID: <BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>


On 30/07/2013, at 7:13 PM, thljcl wrote:

> Disclaimer: I do not in any way endorse the contents of this slideshow. I?m
> just demoing how I use ffmpeg to create slideshow with slides of varying
> length.http://sdrv.ms/13rDjUJThe ?job.7z? and ?job-2.7z? contain text files,
> which can be made the batch file I used by changing its filename extension.
> ?job? is used for creating slideshow. ?job-2? is used for extracting the
> frames. From the slides, I created files of ?1.mkv, 2.mkv, 3.mkv ? 12.mkv?.
> Then I use mkvtoolnix to merge them together into a single. One thing I like
> about mkvtoolnix is that it has a GUI front end.In this slideshow, I makes
> the first slide lasts for 10 seconds; second slide lasts for 11 seconds;
> third slide lasts for 12 seconds; fourth slide lasts for 10 seconds; fifth
> slide lasts for 11 seconds; sixth slide lasts for 12 seconds; seventh slide
> lasts for 10 seconds; eighth slide lasts for 11 seconds; ninth slide lasts
> for 12 seconds; tenth slide lasts for 10 seconds; eleventh slide lasts for
> 11 seconds; twelfth slide lasts for 12 seconds;I don?t plan to keep this
> demo for long. As such, I shall remove it after August 7, 2013.

dvdslideshow is your friend

James

From fernando.brown81 at yahoo.com  Tue Jul 30 22:41:25 2013
From: fernando.brown81 at yahoo.com (Fernando Brown)
Date: Tue, 30 Jul 2013 13:41:25 -0700 (PDT)
Subject: [FFmpeg-user] Question About FFmpeg License
Message-ID: <1375216885.88874.YahooMailNeo@web162801.mail.bf1.yahoo.com>

Hello,

My name is Fernando and I'm having a software company, recently we are developing a software application (Windows platform) and using the FFmpeg.exe (without any modification) just to convert a portion of video files and concatenate them. This program is not a free and is a paid version.
My question is: If we mention the FFmpeg name in our program and also a text file in our installation package to include direct link to FFmpeg website, and also include it in our EULA and again mention that we use FFmpeg under LGPLv2.1; is it sufficient for licensing and legal points?
If NOT, please assist me how to do that.

Looking forward to hear from you.

Regards,
Fernando Brown
fernando.brown81 at yahoo.com

From herbert-rudolf at gmx.at  Tue Jul 30 18:49:16 2013
From: herbert-rudolf at gmx.at (Herbert Rudolf)
Date: Tue, 30 Jul 2013 18:49:16 +0200
Subject: [FFmpeg-user] codec frame size is not set
Message-ID: <51F7EE8C.60302@gmx.at>

Hi..

I have already read much about the "codec frame size is not set"- bug.
But I saw some posts, which were written 2 years ago --> "it will be 
fixed soon"..

But know i have the version from 20130729 
(ffmpeg-20130729-git-c96b3ae-win64-static), and the bug is still there.

I convert a mkv-file to a mp4-file using this command: ffmpeg.exe -i 
"test.mkv" -vcodec copy -acodec copy "test_parsed.mp4"

I think the bug comes because of the german ac3-audio-stream..
The output-log is attached.

Should i use other paramters to fix this issue, or is it still a bug in 
ffmpeg?



Thank you
Herbert
-------------- next part --------------

C:\Users\Niki\Desktop\tmp\ffmpeg-20130729-git-c96b3ae-win64-static\bin>ffmpeg.exe -i "test

ffmpeg version N-55066-gc96b3ae Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 29 2013 18:05:45 with gcc 4.7.3 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --e
enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-
ble-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb -
tmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enabl
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-lib
  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, matroska,webm, from 'test.mkv':
  Metadata:
    title           : Friends - Dunkle Leidenschaft (S01E07)
    creation_time   : 2012-11-07 17:13:04
  Duration: 00:22:35.39, start: 0.000000, bitrate: 4949 kb/s
    Chapter #0.0: start 0.000000, end 96.513000
    Metadata:
      title           : 00:00:00.000
    Chapter #0.1: start 96.513000, end 797.922000
    Metadata:
      title           : 00:01:36.513
    Chapter #0.2: start 797.922000, end 1297.922000
    Metadata:
      title           : 00:13:17.922
    Chapter #0.3: start 1297.922000, end 1355.392000
    Metadata:
      title           : 00:21:37.922
    Stream #0:0: Video: h264 (High), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 23.98 fps, 23.9
    Metadata:
      title           : 720p
    Stream #0:1(ger): Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s (default)
    Metadata:
      title           : AC3
    Stream #0:2(eng): Audio: ac3, 48000 Hz, 5.1(side), fltp, 640 kb/s
    Metadata:
      title           : AC3
[mp4 @ 00000000046000a0] track 1: codec frame size is not set
Output #0, mp4, to 'test_parsed.mp4':
  Metadata:
    title           : Friends - Dunkle Leidenschaft (S01E07)
    encoder         : Lavf55.12.102
    Chapter #0.0: start 0.000000, end 96.513000
    Metadata:
      title           : 00:00:00.000
    Chapter #0.1: start 96.513000, end 797.922000
    Metadata:
      title           : 00:01:36.513
    Chapter #0.2: start 797.922000, end 1297.922000
    Metadata:
      title           : 00:13:17.922
    Chapter #0.3: start 1297.922000, end 1355.392000
    Metadata:
      title           : 00:21:37.922
    Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9
    Metadata:
      title           : 720p
    Stream #0:1(eng): Audio: ac3 ([165][0][0][0] / 0x00A5), 48000 Hz, 5.1(side), 640 kb/s
    Metadata:
      title           : AC3
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:2 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=32474 fps=3548 q=-1.0 Lsize=  787745kB time=00:22:35.39 bitrate=4761.1kbits/s
video:680879kB audio:105890kB subtitle:0 global headers:0kB muxing overhead 0.124005%

From jhbell50 at hotmail.com  Tue Jul 30 21:35:16 2013
From: jhbell50 at hotmail.com (John Bell)
Date: Tue, 30 Jul 2013 15:35:16 -0400
Subject: [FFmpeg-user] Unable to add valid http headers from windows command
	line
Message-ID: <BLU178-DS165A9D28A2C9537E6B6FC9CF560@phx.gbl>

http headers must be terminated by a CRLF, but when invoking ffmpeg from the command line for an HLS stream that requires a header, the \r\n are removed and http gives an error message reporting no CRLF. 
eg:
ffplay -headers "Cookie: eurospirt1keys=goXwiG_clitxCyNihl0ErA; eurospirt1keyt=1375113106\r\n" "http://46.165.210.22/hqstream/eurospirt1/playlist.m3u8?keys=goXwiG_clitxCyNihl0ErA&keyt=1375113106" 
"[http @ 043ac6a0] No trailing CRLF found in HTTP header."

Seems to me that ffmpeg should add any required crlf?s just like CURL does. I?ve requested approval to post this in tracker but I?ve received no email back.

From jhbell50 at hotmail.com  Tue Jul 30 21:40:52 2013
From: jhbell50 at hotmail.com (john Bell)
Date: Tue, 30 Jul 2013 15:40:52 -0400
Subject: [FFmpeg-user] hls stream requiring cookies fails to play
Message-ID: <BLU178-W11DFBE7F7FCB96FFB6FEABCF560@phx.gbl>

When attempting to play hls streams with a variant playlist that require a cookie to be set, ffplay fails  because the subsequent gets do not include the cookie returned by the initial request.  I'd like to post the details in the tracker but have had no response to my enrollment request. 		 	   		  

From cehoyos at ag.or.at  Wed Jul 31 03:06:35 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 31 Jul 2013 01:06:35 +0000 (UTC)
Subject: [FFmpeg-user] Question About FFmpeg License
References: <1375216885.88874.YahooMailNeo@web162801.mail.bf1.yahoo.com>
Message-ID: <loom.20130731T030458-445@post.gmane.org>

Fernando Brown <fernando.brown81 <at> yahoo.com> writes:

> My question is: If we mention the FFmpeg name in our 
> program and also a text file in our installation package
> to include direct link to FFmpeg website, and also 
> include it in our EULA and again mention that we use
> FFmpeg under LGPLv2.1; is it sufficient for licensing 
> and legal points?

You are at least missing providing the source code you 
used to compile FFmpeg.

I am not a lawyer and I am not claiming this is the only 
issue.

Carl Eugen


From cehoyos at ag.or.at  Wed Jul 31 03:09:03 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 31 Jul 2013 01:09:03 +0000 (UTC)
Subject: [FFmpeg-user] codec frame size is not set
References: <51F7EE8C.60302@gmx.at>
Message-ID: <loom.20130731T030758-512@post.gmane.org>

Herbert Rudolf <herbert-rudolf <at> gmx.at> writes:

> I convert a mkv-file to a mp4-file using this command: 
> ffmpeg.exe -i "test.mkv" -vcodec copy -acodec copy 
> "test_parsed.mp4"

> I have already read much about the "codec frame size 
> is not set"- bug.

The issue does not immediately sound familiar 
to me, please provide a sample file.

Carl Eugen


From lou at lrcd.com  Wed Jul 31 03:34:40 2013
From: lou at lrcd.com (Lou)
Date: Tue, 30 Jul 2013 17:34:40 -0800
Subject: [FFmpeg-user] FFMPEG Installation error in Centos
In-Reply-To: <2a3d593bd2c20ee7981da6793c54a756.squirrel@hioxindia.com>
References: <2a3d593bd2c20ee7981da6793c54a756.squirrel@hioxindia.com>
Message-ID: <20130730173440.38c05868@lrcd.com>

On Mon, 29 Jul 2013 16:46:34 +0530
support at hioxindia.com wrote:

> Unable to create and execute files in /tmp.  Set the TMPDIR environment
> variable to another directory and make sure that it is not mounted noexec.
> Sanity test failed.

Run:

$ mount | grep noexec

If it shows noexec for your /tmp then, as the message says, you'll be
unable to create and execute files in /tmp unless you do something
about it such as set the TMPDIR environment variable to another
directory and make sure that it is not also mounted with noexec.

From jiachielee at live.com  Wed Jul 31 06:20:23 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 21:20:23 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375182808947-4660318.post@n4.nabble.com>
	<BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>
Message-ID: <1375244423628-4660355.post@n4.nabble.com>

Is this tool from http://bit.ly/159uJ2e what you are referring to? Thanks for
telling a commercially available tool that costs USD 50. By the way I also
have another free tool that is called ?Movie Maker?, which is free for
Windows users. I think that you have missed the point why I learned to use
ffmpeg; it?s not merely for academic reasons, although that is a big part of
it.
As I previously mentioned, I can do a lot of customizations when it comes to
encoding settings, such as frame rate, quality preservation (Hey, I can do
lossless encoding), file size control, and so on. Ffmpeg is free to use but
it?s not exactly the most user-friendly tool as a command-line tool. But
it?s certainly very powerful. At the end, it?s up to the users to choose
whether or not they only need simple tools for simple job; or they need
something more advanced, but more difficult to be used.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277p4660355.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From andycivil at gmail.com  Wed Jul 31 07:06:00 2013
From: andycivil at gmail.com (Andy Civil)
Date: Wed, 31 Jul 2013 01:06:00 -0400
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375244423628-4660355.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>	<1375099174769-4660296.post@n4.nabble.com>	<51F6FB51.2080707@gmail.com>	<1375182808947-4660318.post@n4.nabble.com>	<BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>
	<1375244423628-4660355.post@n4.nabble.com>
Message-ID: <51F89B38.2020109@gmail.com>

On 2013-07-31 12:20 AM, thljcl wrote:
> Is this tool from http://bit.ly/159uJ2e what you are referring to?

I very much doubt it; I think James was trying to impress us with a 'short' 
answer, and wasn't specific enough with his recommendation!

I'm guessing that the one he was referring to is this one:
http://dvd-slideshow.sourceforge.net/wiki/Main_Page

It is indeed free, and looks interesting. It's also Linux only; I can't complain 
about that because I approve of trying to migrate to Linux - but to be honest 
all my photos and stuff are on a Windows computer so I'd prefer a windows tool, 
at least for now.

Also, I note that the dvd-slideshow can do things like the Ken Burns effect. 
This is very nice for presentation, but obviously kills any idea of saving data 
by having a whole string of "repeat last image" tokens (or whatever you call them).

The bottom line is, I'm interested in your method - I just have to update my 7z 
so that I can uncompress your job files - LOL - but honestly, most people will 
be less infatuated with data rate efficiency, and will enjoy that dvd-slideshow 
program.

P.S. I don't know why you use bit.ly so much, it's OK but I prefer to know where 
I'm going... I discovered that you can PREVIEW a bit.ly link by adding a simple 
"+" sign at the end; I'm sharing this gem for others' interest.

--
Andy

From jiachielee at live.com  Wed Jul 31 07:40:29 2013
From: jiachielee at live.com (thljcl)
Date: Tue, 30 Jul 2013 22:40:29 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <51F89B38.2020109@gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375182808947-4660318.post@n4.nabble.com>
	<BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>
	<1375244423628-4660355.post@n4.nabble.com>
	<51F89B38.2020109@gmail.com>
Message-ID: <1375249229256-4660357.post@n4.nabble.com>

Thanks for clarifying. Linux tool? Eh? It does not matter. Based on what I
saw from the documentation you show me on
http://dvd-slideshow.sourceforge.net/wiki/Main_Page, it does not seem to
offer what I can do with ffmpeg. Well, as you said, it probably did not
matter for average users. They won?t be so picky about encoders? settings as
long as they got something done. Lossless encoding? That probably never
crossed the mind for average users. 
Hm? On the other hand, I seriously doubt the big producers would forget to
have multiple backup of master copies. In the old days, that would mean film
master, i.e. the actual film stocks. In digital age, that of course means
the copies without any lossy compression. After all, they would like to
continue to make money from their cash cows.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277p4660357.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From soho123.2012 at gmail.com  Wed Jul 31 09:43:31 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Wed, 31 Jul 2013 15:43:31 +0800
Subject: [FFmpeg-user] what is "fltp" for audio when using ffmpeg
Message-ID: <CAFd2cZU=eK8-fVVK9rWJBdU9L5Z_Zuch+a++_rpvAeMmhsYTPg@mail.gmail.com>

Hi All,


I got the problem when I try to capture audio + video from USB web cam by
ffmepg.
the problem is audio can not play when I use VLC to play the stream that
ffserver output.
the question is :
I use ffmepg the encode audio to "wmav1" but, I can see the log that is
output by audio filter.
the audio format is "fltp".
What is "fltp"? Does anyone have idea?

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 wmav1
http://l
ocalhost:8090/feed1.ffm
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
  built on Jul 31 2013 14:26:37 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 --dis
able-ffplay --disable-postproc --disable-runtime-cpudetect
--disable-swscale-alpha --disable-mipsdspr1 --d
isable-mipsdspr2 --disable-mipsfpu --enable-small
--prefix=/ffmpeg-2.0/romfs --bindir=/ffmpeg-2.0/romfs --disable-bsfs
--disable-filters --enable-gpl --enab
le-libfaac --enable-nonfree --enable-libx264
--extra-cflags=-I/ShareLib_Install/bin/../include
--extra-libs='-L/ShareLib_Install/bin/../lib -lx264' --enable
-filter='aformat,aresample,copy,format,fps,framestep,resample' --disa
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 a
rgument '30'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or
abbreviation)) with argument '640x48
0'.
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 a
rgument 'copy'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec
('copy' to copy stream)) with a
rgument 'wmav1'.
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 @ 0x71ed30] fd:3 capabilities:4000001
[video4linux2,v4l2 @ 0x71ed30] Current input_channel: 0, input_name: Camera
1
[video4linux2,v4l2 @ 0x71ed30] Setting time per frame to 1/30
[h264 @ 0x71f990] Frame num gap 9 7
[video4linux2,v4l2 @ 0x71ed30] All info found
Input #0, video4linux2,v4l2, from '/dev/video1':
  Duration: N/A, start: 147.060000, bitrate: N/A
    Stream #0:0, 8, 1/1000000: Video: h264, yuv420p, 640x480, 1/2000000, -5
kb/s, 30 fps, 30 tbr, 1000k tb
n, 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: 1375284807.118875, 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://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 wmav1.
Successfully parsed a group of options.
Opening an output file: http://localhost:8090/feed1.ffm.
[ffm @ 0x7f2ff0] Format ffm probed with size=2048 and score=101
[AVIOContext @ 0x7f7590] Statistics: 4096 bytes read, 0 seeks
Wed Jul 31 15:33:27 2013 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 200 4175
Successfully opened the file.
detected 1 logical cores
[graph 0 input from stream 1:0 @ 0x7f3440] Setting 'time_base' to value
'1/48000'
[graph 0 input from stream 1:0 @ 0x7f3440] Setting 'sample_rate' to value
'48000'
[graph 0 input from stream 1:0 @ 0x7f3440] Setting 'sample_fmt' to value
's16'
[graph 0 input from stream 1:0 @ 0x7f3440] Setting 'channel_layout' to
value '0x3'
[graph 0 input from stream 1:0 @ 0x7f3440] tb:1/48000 samplefmt:s16
samplerate:48000 chlayout:0x3
[audio format for output stream 0:0 @ 0x79c700] Setting 'sample_fmts' to
value 'fltp'
[audio format for output stream 0:0 @ 0x79c700] Setting 'sample_rates' to
value '48000'
[audio format for output stream 0:0 @ 0x79c700] Setting 'channel_layouts'
to value '0x3'
[audio format for output stream 0:0 @ 0x79c700] auto-inserting filter
'auto-inserted resampler 0' between
the filter 'Parsed_anull_0' and the filter 'audio format for output stream
0:0'
[AVFilterGraph @ 0x812790] query_formats: 4 queried, 9 merged, 3 already
done, 0 delayed
[auto-inserted resampler 0 @ 0x811210] ch:2 chl:stereo fmt:s16 r:48000Hz ->
ch:2 chl:stereo fmt:fltp r:480
00Hz
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
  Metadata:
    creation_time   : now
    encoder         : Lavf55.12.100
    Stream #0:0, 0, 1/1000000: Audio: wmav1, 48000 Hz, stereo, fltp, 128
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 (pcm_s16le -> wmav1)
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
[alsa @ 0x8174e0] ALSA buffer xrun.
    Last message repeated 1 times    Last message repeated 2 times    Last
message repeated 3 times

From jam at tigger.ws  Wed Jul 31 10:02:11 2013
From: jam at tigger.ws (James Linder)
Date: Wed, 31 Jul 2013 16:02:11 +0800
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <1375244423628-4660355.post@n4.nabble.com>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375182808947-4660318.post@n4.nabble.com>
	<BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>
	<1375244423628-4660355.post@n4.nabble.com>
Message-ID: <8077B185-AEEC-4E0C-96FD-4C7589F8C058@tigger.ws>


On 31/07/2013, at 12:20 PM, thljcl wrote:

> Is this tool from http://bit.ly/159uJ2e what you are referring to? Thanks for
> telling a commercially available tool that costs USD 50. By the way I also
> have another free tool that is called ?Movie Maker?, which is free for
> Windows users. I think that you have missed the point why I learned to use
> ffmpeg; it?s not merely for academic reasons, although that is a big part of
> it.
> As I previously mentioned, I can do a lot of customizations when it comes to
> encoding settings, such as frame rate, quality preservation (Hey, I can do
> lossless encoding), file size control, and so on. Ffmpeg is free to use but
> it?s not exactly the most user-friendly tool as a command-line tool. But
> it?s certainly very powerful. At the end, it?s up to the users to choose
> whether or not they only need simple tools for simple job; or they need
> something more advanced, but more difficult to be used.

Sorry I thought it would be trivial to find
http://dvd-slideshow.sourceforge.net/wiki/Main_Page

It uses ffmpeg to do the heavy lifting
It is very easy to use
It is command line stuff so you may tweek as you see fit
The result is pleasing

James

From jam at tigger.ws  Wed Jul 31 10:07:37 2013
From: jam at tigger.ws (James Linder)
Date: Wed, 31 Jul 2013 16:07:37 +0800
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <51F89B38.2020109@gmail.com>
References: <1375041354497-4660277.post@n4.nabble.com>	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>	<1375099174769-4660296.post@n4.nabble.com>	<51F6FB51.2080707@gmail.com>	<1375182808947-4660318.post@n4.nabble.com>	<BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>
	<1375244423628-4660355.post@n4.nabble.com>
	<51F89B38.2020109@gmail.com>
Message-ID: <A5E0BA25-D43A-41C0-B7DA-734B0D92F9BA@tigger.ws>


On 31/07/2013, at 1:06 PM, Andy Civil wrote:

> On 2013-07-31 12:20 AM, thljcl wrote:
>> Is this tool from http://bit.ly/159uJ2e what you are referring to?
> 
> I very much doubt it; I think James was trying to impress us with a 'short' answer, and wasn't specific enough with his recommendation!
> 
> I'm guessing that the one he was referring to is this one:
> http://dvd-slideshow.sourceforge.net/wiki/Main_Page
> 
> It is indeed free, and looks interesting. It's also Linux only; I can't complain about that because I approve of trying to migrate to Linux - but to be honest all my photos and stuff are on a Windows computer so I'd prefer a windows tool, at least for now.
> 
> Also, I note that the dvd-slideshow can do things like the Ken Burns effect. This is very nice for presentation, but obviously kills any idea of saving data by having a whole string of "repeat last image" tokens (or whatever you call them).
> 
> The bottom line is, I'm interested in your method - I just have to update my 7z so that I can uncompress your job files - LOL - but honestly, most people will be less infatuated with data rate efficiency, and will enjoy that dvd-slideshow program.
> 
> P.S. I don't know why you use bit.ly so much, it's OK but I prefer to know where I'm going... I discovered that you can PREVIEW a bit.ly link by adding a simple "+" sign at the end; I'm sharing this gem for others' interest.

Indeed it is (the source forge link)
My wife uses it unchanged on her mac. I venture to suggest that one could run it on winders too.
Since it is command line ffmpeg you may tweek as you see fit, what I was saying is this is a very nice *outta box" solution to a generic problem/need.
James

From francois.visagie at gmail.com  Wed Jul 31 10:09:37 2013
From: francois.visagie at gmail.com (Francois Visagie)
Date: Wed, 31 Jul 2013 10:09:37 +0200
Subject: [FFmpeg-user] Specify log file for 2-pass encoding
In-Reply-To: <1375203182.495.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1375203182.495.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <00ae01ce8dc5$4f4aa9d0$eddffd70$@gmail.com>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of James Board
> Sent: 30 July 2013 18:53
> To: ffmpeg-user at ffmpeg.org
> Subject: [FFmpeg-user] Specify log file for 2-pass encoding
> 
> I'm using ffmpeg to do 2-pass mpeg encoding with '-ovc lavc' and the
> mpeg4 codec on Linux.? This
> generates a divx2pass.log file in the directory where the files are
located.
> I'm trying to encode multiple files at the same time on different CPU
cores
> and the log files are clobbering each other.? How can I specify which name
I'd
> like the log file to have?? The ffmpeg man page says the -passlogfile can
be
> used to specify the logfile for pass 1.? But how do I tell ffmpeg to use
that log
> file in pass 2?

IIRC you specify the same option for both passes. The documentation on the
'-pass' option:

	?-pass[:stream_specifier] n (output,per-stream)?

	    Select the pass number (1 or 2). It is used to do two-pass video
encoding. The statistics of the video are recorded in the first pass into a
log file (see also the option -passlogfile), and in the second pass _that_ 
	log file is used to generate the video at the exact requested
bitrate.

> 
> The man page and the HTML documentation doesn't mention divx2pass.log.
> Thanks
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From gavr.mail at gmail.com  Wed Jul 31 10:12:18 2013
From: gavr.mail at gmail.com (Kirill Gavrilov)
Date: Wed, 31 Jul 2013 12:12:18 +0400
Subject: [FFmpeg-user] what is "fltp" for audio when using ffmpeg
In-Reply-To: <CAFd2cZU=eK8-fVVK9rWJBdU9L5Z_Zuch+a++_rpvAeMmhsYTPg@mail.gmail.com>
References: <CAFd2cZU=eK8-fVVK9rWJBdU9L5Z_Zuch+a++_rpvAeMmhsYTPg@mail.gmail.com>
Message-ID: <CAJQwZb0sUjLWzVKGv5jmjrh8_MnPvdr69C7xOAN3NKD=why5Bw@mail.gmail.com>

Hello,

What is "fltp"? Does anyone have idea?
>
fltp is Planar Floating point format. Planar format were added relatively
recently to FFmpeg.
The main target is to eliminate extra processing in audio decoders and move
out it to dedicated re-sampling library.

Sorry I can not help with the problem itself - I suppose you should either
try to update your VLC or try adding extra flags to ffserver to perform
re-sampling to compatible format explicitly.
-----------------------------------------------
Kirill Gavrilov,
Software designer.
<kirill at sview.ru>

From soho123.2012 at gmail.com  Wed Jul 31 10:18:53 2013
From: soho123.2012 at gmail.com (Soho Soho123)
Date: Wed, 31 Jul 2013 16:18:53 +0800
Subject: [FFmpeg-user] what is "fltp" for audio when using ffmpeg
In-Reply-To: <CAJQwZb0sUjLWzVKGv5jmjrh8_MnPvdr69C7xOAN3NKD=why5Bw@mail.gmail.com>
References: <CAFd2cZU=eK8-fVVK9rWJBdU9L5Z_Zuch+a++_rpvAeMmhsYTPg@mail.gmail.com>
	<CAJQwZb0sUjLWzVKGv5jmjrh8_MnPvdr69C7xOAN3NKD=why5Bw@mail.gmail.com>
Message-ID: <CAFd2cZV6TYs0AL_UO6MYM-jBT+BME1AcyoKCh_Z_k+hdfT-g8A@mail.gmail.com>

2013/7/31 Kirill Gavrilov <gavr.mail at gmail.com>

> Hello,
>
> What is "fltp"? Does anyone have idea?
> >
> fltp is Planar Floating point format. Planar format were added relatively
> recently to FFmpeg.
> The main target is to eliminate extra processing in audio decoders and move
> out it to dedicated re-sampling library.
>
> Sorry I can not help with the problem itself - I suppose you should either
> try to update your VLC or try adding extra flags to ffserver to perform
> re-sampling to compatible format explicitly.
> -----------------------------------------------
> Kirill Gavrilov,
> Software designer.
> <kirill at sview.ru>
> _______________________________________________
>
> Hi , the setting used by ffserver is :
ffserver output asf stream, and I add global_header already,
But VLC can not play the audio part of asf stream that audio + video output
by ffserver.
Do you have idea about the extra flags?


<Stream test.asf>
Feed feed1.ffm
Format asf

AVOptionVideo flags +global_header
VideoSize 640x480
VideoFrameRate 30
VideoCodec libx264

AVOptionAudio flags +global_header
AudioCodec wmav1
AudioChannels 2
AudioSampleRate 48000
</Stream>

From jiachielee at live.com  Wed Jul 31 10:08:37 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 31 Jul 2013 01:08:37 -0700 (PDT)
Subject: [FFmpeg-user] Create slideshow with ffmpeg from still images
In-Reply-To: <8077B185-AEEC-4E0C-96FD-4C7589F8C058@tigger.ws>
References: <1375041354497-4660277.post@n4.nabble.com>
	<CAPYw7P4X2DYn8ZuVr44wsA+tw=yEdadwmGgg+s4d2KxbADOB3Q@mail.gmail.com>
	<1375099174769-4660296.post@n4.nabble.com>
	<51F6FB51.2080707@gmail.com>
	<1375182808947-4660318.post@n4.nabble.com>
	<BC073029-D4D1-48A7-8676-C88FD11FAFE1@tigger.ws>
	<1375244423628-4660355.post@n4.nabble.com>
	<8077B185-AEEC-4E0C-96FD-4C7589F8C058@tigger.ws>
Message-ID: <1375258117210-4660361.post@n4.nabble.com>

Does it currently support H.264 lossless or FFV1 encoding process?



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Create-slideshow-with-ffmpeg-from-still-images-tp4660277p4660361.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From yugant at gmail.com  Wed Jul 31 11:07:46 2013
From: yugant at gmail.com (Yugant Khanna)
Date: Wed, 31 Jul 2013 14:37:46 +0530
Subject: [FFmpeg-user] Capture and RTMP
In-Reply-To: <CAFnvMkuaKkSVfiBY3bgr6NRZtyJ2hMuPPJdBd3JpmwKUeLQV1g@mail.gmail.com>
References: <CAFnvMkuaKkSVfiBY3bgr6NRZtyJ2hMuPPJdBd3JpmwKUeLQV1g@mail.gmail.com>
Message-ID: <CAFnvMkuiZ5eXMOBiiFYmJ+skT2De=sZ6Kn=cfieZZC+k6wo2+g@mail.gmail.com>

Hi  , i tried  to concat both the above commands to Encode and Stream RTMP
at the same same but in vein .

ffmpeg   -f dshow -r 25 -i video="Decklink Video Capture":audio="Decklink
Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M  -acodec pcm_s24le
-f tee -map 0:v -map 0:a
"[f=segment:segment_time=10:segment_list=test.m3u8:segment_format=mpegts]stream%05d.ts
|[f=flv]rtmp://127.0.0.1/LIVE/STREAMNAME"


ffmpeg version N-51433-g551f683 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Mar 28 2013 21:52:57 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. 22.101 / 52. 22.101
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  0.100 / 55.  0.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 48.105 /  3. 48.105
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[dshow @ 0037bde0] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, dshow, from 'video=Decklink Video Capture:audio=Decklink Audio
Capture
':
  Duration: N/A, start: 0.054090, bitrate: 1536 kb/s
    Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x576, 25
tbr,
10000k tbn, 25 tbc
    Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
[segment @ 03e6cf80] Codec for stream 0 does not use global headers but
containe
r format requires global headers
[segment @ 03e6cf80] Codec for stream 1 does not use global headers but
containe
r format requires global headers
[flv @ 03e83a40] Codec for stream 0 does not use global headers but
container fo
rmat requires global headers
[flv @ 03e83a40] Codec for stream 1 does not use global headers but
container fo
rmat requires global headers
[flv @ 03e83a40] Video codec 'dvvideo' for stream 0 is not compatible with
FLV
[tee @ 03e1c280] Slave '[f=flv]rtmp://192.168.36.251/LIVE/STREAMNAME':
error wri
ting header: Error number -22 occurred
Output #0, tee, to
'[f=segment:segment_time=10:segment_list=test.m3u8:segment_fo
rmat=mpegts]stream%05d.ts |[f=flv]rtmp://192.168.36.251/LIVE/STREAMNAME':
  Metadata:
    encoder         : Lavf55.0.100
    Stream #0:0: Video: dvvideo, yuv411p, 720x576, q=2-31, 2000 kb/s, 90k
tbn, 2
5 tbc
    Stream #0:1: Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> dvvideo)
  Stream #0:1 -> #0:1 (pcm_s16le -> pcm_s24le)
Could not write header for output file #0 (incorrect codec parameters ?):
Invali
d argument

From jiachielee at live.com  Wed Jul 31 11:02:56 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 31 Jul 2013 02:02:56 -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: <1375261376666-4660365.post@n4.nabble.com>

Earlier, I uploaded a slideshow I created to YouTube to demonstrate the way
to derive the inverse transformation matrix for YCbCr to RGB conversion from
RGB to YCbCr transformation matrix. While it?s fair to say that the method I
used is largely unchanged; apparently, I did make some mistakes during the
calculation process, resulting the wrong values for the transformation
matrix for YCbCr to RGB conversion. I?ve removed the earlier post and the
earlier version of slideshow video I uploaded to YouTube. Upon finishing the
revision to make the corrected version, I re-upload the slideshow video
again to YouTube.

<iframe width="640" height="390"
src="http://www.youtube.com/embed/PDVk-fRPzxU" frameborder="0"
allowfullscreen></iframe>


The reason that I made the derivation is to show that based on our
understanding of linear algebra, we know that how YCbCr can be used to fully
encode RGB color information. To be specific, the transformation being used
is actually affine transformation. Please watch the slideshow video and make
your own judgment. I don?t claim to be a know-it-all expert. I even admitted
that I made mistakes during my earlier derivation process. Still, it?s a
careless mistakes, rather than a misunderstanding of the algorithms behind
RGB to YCbCr conversion. There are different industry standards for defining
YCbCr color space model. I only used one to prove a basic concept. 
If you spotted any mistakes, please inform me. Thank you.




--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Does-converting-to-yuv444p-by-default-make-sense-tp4660219p4660365.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From cehoyos at ag.or.at  Wed Jul 31 11:32:52 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 31 Jul 2013 09:32:52 +0000 (UTC)
Subject: [FFmpeg-user] Capture and RTMP
References: <CAFnvMkuaKkSVfiBY3bgr6NRZtyJ2hMuPPJdBd3JpmwKUeLQV1g@mail.gmail.com>
	<CAFnvMkuiZ5eXMOBiiFYmJ+skT2De=sZ6Kn=cfieZZC+k6wo2+g@mail.gmail.com>
Message-ID: <loom.20130731T112907-313@post.gmane.org>

Yugant Khanna <yugant <at> gmail.com> writes:

> ffmpeg ... -vcodec dvvideo 

> [flv  <at>  03e83a40] Video codec 'dvvideo' for 
> stream 0 is not compatible with FLV

flv only supports the following video codecs:
flv1 (Sorenson H263)
h263 (iso)
mpeg4 (asp)
flashsv* (Flash screen video)
vp6* (no encoder within FFmpeg, remuxing only)
h264

You cannot put dvvideo into flv, this is not a 
limitation of FFmpeg.

Carl Eugen


From stefasab at gmail.com  Wed Jul 31 11:38:36 2013
From: stefasab at gmail.com (Stefano Sabatini)
Date: Wed, 31 Jul 2013 11:38:36 +0200
Subject: [FFmpeg-user] Capture and RTMP
In-Reply-To: <CAFnvMkuiZ5eXMOBiiFYmJ+skT2De=sZ6Kn=cfieZZC+k6wo2+g@mail.gmail.com>
References: <CAFnvMkuaKkSVfiBY3bgr6NRZtyJ2hMuPPJdBd3JpmwKUeLQV1g@mail.gmail.com>
	<CAFnvMkuiZ5eXMOBiiFYmJ+skT2De=sZ6Kn=cfieZZC+k6wo2+g@mail.gmail.com>
Message-ID: <20130731093836.GA5975@barisone>

On date Wednesday 2013-07-31 14:37:46 +0530, Yugant Khanna wrote:
> Hi  , i tried  to concat both the above commands to Encode and Stream RTMP
> at the same same but in vein .
> 
> ffmpeg   -f dshow -r 25 -i video="Decklink Video Capture":audio="Decklink
> Audio Capture" -vcodec dvvideo -pix_fmt yuv411p -vb 2M  -acodec pcm_s24le
> -f tee -map 0:v -map 0:a
> "[f=segment:segment_time=10:segment_list=test.m3u8:segment_format=mpegts]stream%05d.ts
> |[f=flv]rtmp://127.0.0.1/LIVE/STREAMNAME"
> 
> 
> ffmpeg version N-51433-g551f683 Copyright (c) 2000-2013 the FFmpeg
> developers
>   built on Mar 28 2013 21:52:57 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. 22.101 / 52. 22.101
>   libavcodec     55.  2.100 / 55.  2.100
>   libavformat    55.  0.100 / 55.  0.100
>   libavdevice    55.  0.100 / 55.  0.100
>   libavfilter     3. 48.105 /  3. 48.105
>   libswscale      2.  2.100 /  2.  2.100
>   libswresample   0. 17.102 /  0. 17.102
>   libpostproc    52.  2.100 / 52.  2.100
> [dshow @ 0037bde0] Estimating duration from bitrate, this may be inaccurate
> Guessed Channel Layout for  Input Stream #0.1 : stereo
> Input #0, dshow, from 'video=Decklink Video Capture:audio=Decklink Audio
> Capture
> ':
>   Duration: N/A, start: 0.054090, bitrate: 1536 kb/s
>     Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x576, 25
> tbr,
> 10000k tbn, 25 tbc
>     Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
> [segment @ 03e6cf80] Codec for stream 0 does not use global headers but
> containe
> r format requires global headers
> [segment @ 03e6cf80] Codec for stream 1 does not use global headers but
> containe
> r format requires global headers
> [flv @ 03e83a40] Codec for stream 0 does not use global headers but
> container fo
> rmat requires global headers
> [flv @ 03e83a40] Codec for stream 1 does not use global headers but
> container fo
> rmat requires global headers
> [flv @ 03e83a40] Video codec 'dvvideo' for stream 0 is not compatible with
> FLV

There are several issues here.
1. FLV does not support DVVIDEO, you should use H.264 or a compatible
codec.

2. you need to specify -flags +global_header for FLV. This is a bit
tricky since you're also using tee. One possible solution would be to
use the bitstream filter dump_extra, but you need a patch to apply
bitstream filters to tee outputs, which is currently work in progress
in -devel.

The short answer is that you can't do that in one pass at the moment,
unless you patch your ffmpeg, or wait a few weeks for the feature to
be integrated.

From nicolas.george at normalesup.org  Wed Jul 31 12:02:54 2013
From: nicolas.george at normalesup.org (Nicolas George)
Date: Wed, 31 Jul 2013 12:02:54 +0200
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <1375228699675-4660346.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>
	<1375228699675-4660346.post@n4.nabble.com>
Message-ID: <20130731100254.GA2485@phare.normalesup.org>

Le duodi 12 thermidor, an CCXXI, thljcl a ?crit?:
> Basically, if you've learned linear algebra before, you may understand right
> away why YUV444p can represent all colors from R'G'B'. Before you denounce
> my answer, please watch the video first to see if there is anything wrong
> with my math.

There is something wrong with your math: you forget that the pixel formats
do not correspond to vector spaces, but to only discrete parallelepipeds.

There are only a few affine bijections from a parallelepipeds to another,
and the RGB <-> YUV conversion is not one of them: at least one of the
parallelepipeds overflows the other.

As it happens, the YUV parallelepiped is wider than the RGB cube, so your
claim that RGB -> YUV is lossless stands on that regard; only you did not
provide the full proof, so you do not get the full mark. But the opposite
direction does not work: for example, full chroma with no luma maps to pixel
values of 278, which is not possible.

And there is the discrete issue. Since the YUV parallelepiped is larger than
the RGB cube, if they hold the same number of discrete points, the YUV
discrete points are further apart, and several RGB points will fall on the
same YUV point. The matrix has a determinant of about 6.4, that is 2.7 extra
bits, so if you want perfect RGB -> YUV conversion from 8-bits depth, you
need at least 9-bits depth on the YUV side. It may become apparent in
gradients.

Regards,

-- 
  Nicolas George
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From barsnick at gmx.net  Wed Jul 31 15:29:03 2013
From: barsnick at gmx.net (Moritz Barsnick)
Date: Wed, 31 Jul 2013 15:29:03 +0200
Subject: [FFmpeg-user] Specify log file for 2-pass encoding
In-Reply-To: <1375203182.495.YahooMailNeo@web164703.mail.gq1.yahoo.com>
References: <1375203182.495.YahooMailNeo@web164703.mail.gq1.yahoo.com>
Message-ID: <20130731132903.GD16993@sunshine.barsnick.net>

On Tue, Jul 30, 2013 at 09:53:02 -0700, James Board wrote:
> I'm using ffmpeg to do 2-pass mpeg encoding with '-ovc lavc' and the
                                                    ^^^^^^^^^

Why do I get the impression that this is mencoder and _not_ ffmpeg?

Moritz

From jpboard2 at yahoo.com  Wed Jul 31 15:34:30 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 31 Jul 2013 06:34:30 -0700 (PDT)
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <51F83E9B.1090301@thelounge.net>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>	<CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>	<1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<51F83E9B.1090301@thelounge.net>
Message-ID: <1375277670.90194.YahooMailNeo@web164702.mail.gq1.yahoo.com>

>> I'm not using any other filters.? Anyway, does anyone know
>> how to turn off multi-threading?? I tried "-threads 1" and
>> I see see CPU usage at almost 200 percent for ffmpeg

>please post complete command line if you
>want help instead "i did this and that"

Here is an example command
??? ffmpeg -i IN.avi -threads 1 -vf select='-between(n, 1, 1000)',setpts=N/FRAME_RATE/TB -c:v ffvhuff -af aselect='-between(n, 1, 1000)',asetpts=N/SR/TB -c:a wmav2 OUT.avi
There are several more between(n, x, y) segments.? I'm thinking that the select filter is using a single thread and
the ffvhuff is using a thread, but that's only a guess.? Also, I typed that command into this email (not cut and paste)
so it's possible I'm off by one or two characters.
?
Why do I want to do this?? My machine has 6 cores and I have dozens of segments to encode.? I tried
setting -threads 6 for the mpeg4 encoding and it only improved by 2x the overall run time.? However, if I
encode 6 segments at a time on 6 different cores, I think I can get closer to 6x speedup.? But the current 
problem is that's it's spawning so many threads, the process slows to a halt, runs out of memory and that's
it.
?
Thank you for helping.

From onemda at gmail.com  Wed Jul 31 15:51:15 2013
From: onemda at gmail.com (Paul B Mahol)
Date: Wed, 31 Jul 2013 13:51:15 +0000
Subject: [FFmpeg-user] Turn Off Multi-Threading
In-Reply-To: <1375277670.90194.YahooMailNeo@web164702.mail.gq1.yahoo.com>
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>
	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>
	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>
	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>
	<CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>
	<1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<51F83E9B.1090301@thelounge.net>
	<1375277670.90194.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <CAPYw7P67gqCGtNK868=5rkd0-+1UTJMGc4H2b9t8MCO-p=mu-A@mail.gmail.com>

On 7/31/13, James Board <jpboard2 at yahoo.com> wrote:
>>> I'm not using any other filters.  Anyway, does anyone know
>>> how to turn off multi-threading?  I tried "-threads 1" and
>>> I see see CPU usage at almost 200 percent for ffmpeg
>
>>please post complete command line if you
>>want help instead "i did this and that"
>
> Here is an example command
>     ffmpeg -i IN.avi -threads 1 -vf select='-between(n, 1, 1000)',setpts=N/FRAME_RATE/TB -c:v ffvhuff -af aselect='-between(n, 1, 1000)',asetpts=N/SR/TB -c:a wmav2 OUT.avi


Perhaps something like this will improve situation:

ffmpeg -threads 1 -i IN.avi -vf select='-between(n, 1,
1000)',setpts=N/FRAME_RATE/TB -threads 1 -c:v ffvhuff -af
aselect='-between(n, 1, 1000)',asetpts=N/SR/TB -threads 1 -c:a wmav2
OUT.avi

> There are several more between(n, x, y) segments.  I'm thinking that the
> select filter is using a single thread and
> the ffvhuff is using a thread, but that's only a guess.  Also, I typed that
> command into this email (not cut and paste)
> so it's possible I'm off by one or two characters.
>
> Why do I want to do this?  My machine has 6 cores and I have dozens of
> segments to encode.  I tried
> setting -threads 6 for the mpeg4 encoding and it only improved by 2x the
> overall run time.  However, if I
> encode 6 segments at a time on 6 different cores, I think I can get closer
> to 6x speedup.  But the current
> problem is that's it's spawning so many threads, the process slows to a
> halt, runs out of memory and that's
> it.
>
> Thank you for helping.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From cehoyos at ag.or.at  Wed Jul 31 15:55:21 2013
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 31 Jul 2013 13:55:21 +0000 (UTC)
Subject: [FFmpeg-user] Turn Off Multi-Threading
References: <1375207566.37214.YahooMailNeo@web164703.mail.gq1.yahoo.com>	<CAPYw7P4eR-tK1JHqMk3E=Atxef_-LpoCcj8B7MxiXTzoBoE-Xg@mail.gmail.com>	<1375212508.91472.YahooMailNeo@web164704.mail.gq1.yahoo.com>	<CAPYw7P6omemaqeeiqpXDPtoWhGm3pWpKhLeKyvwe5vScJRpnGg@mail.gmail.com>	<1375216877.81542.YahooMailNeo@web164706.mail.gq1.yahoo.com>	<CAPYw7P451hKwxh42rW7ARXw63jj_jahMpv8bUATyP7Tg8UfcKA@mail.gmail.com>	<1375220836.4041.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<51F83E9B.1090301@thelounge.net>
	<1375277670.90194.YahooMailNeo@web164702.mail.gq1.yahoo.com>
Message-ID: <loom.20130731T155423-579@post.gmane.org>

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

> Why do I want to do this??
> My machine has 6 cores and I have dozens of 
> segments to encode.

As said, if "segments" means input files then 
please use -threads 1.

Carl Eugen


From jpboard2 at yahoo.com  Wed Jul 31 16:18:47 2013
From: jpboard2 at yahoo.com (James Board)
Date: Wed, 31 Jul 2013 07:18:47 -0700 (PDT)
Subject: [FFmpeg-user] Specify log file for 2-pass encoding
In-Reply-To: <20130731132903.GD16993@sunshine.barsnick.net>
References: <1375203182.495.YahooMailNeo@web164703.mail.gq1.yahoo.com>
	<20130731132903.GD16993@sunshine.barsnick.net>
Message-ID: <1375280327.79154.YahooMailNeo@web164705.mail.gq1.yahoo.com>

>Why do I get the impression that this is mencoder and _not_ ffmpeg?


Because you are pretty smart.? Yes, I was using mencoder not ffmpeg for 
the mpeg encoding and I sent to the wrong list.? Sorry. 

From jiachielee at live.com  Wed Jul 31 16:47:05 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 31 Jul 2013 07:47:05 -0700 (PDT)
Subject: [FFmpeg-user] Does converting to yuv444p by default make sense?
In-Reply-To: <20130731100254.GA2485@phare.normalesup.org>
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>
	<20130731100254.GA2485@phare.normalesup.org>
Message-ID: <1375282025519-4660374.post@n4.nabble.com>

I only attempt to calculate the original value for R (0-255), G (0-255), and
B (0-255). As far as I understand, RGB color space is NOT vector space and I
did not see it as such.
I realized that RGB color space is not a vector space from the very
beginning. The reason is quite simple. For a 3-channels 24-bit RGB color
space, it is a set with only limited number of elements, which is 16,777,216
number of colors. In a vector space, without regards to the actual inner
product space structure, we generally do not have a boundary for a given
field of number. But each color is represented by three unique numbers; in
this case, it is similar to a coordinate system used for describing
Euclidean Space. But as you pointed out, only non-negative integer value is
allowed for R, G, and B; the number is between 0-255, at least in the case
for sRGB color space.
But if I understood correctly, YCbCr is NOT an absolute color space. What is
an absolute color space? It is a color space in which colors are
unambiguous, that is, where the interpretation of colors in the space are
colorimetrically defined without reference to external factors. There are
different ways to define absolute color space. Two examples are CIEXZY and
sRGB. In other words, colors cannot be defined unambiguously without
absolute color space. If YCbCr is not an absolute color space, without
knowing how the absolute color space that it is mapped to being defined, we
cannot have unambiguously defined colors either.
As I said before there are many linear map standards being used, I only use
one example to clarify how they re-encode color information in YCbCr. We DO
NOT and ARE NOT able to define colors without absolute color space, thus
cannot define colors with YCbCr alone. Say we are able to define all
16,777,216 number of colors in sRGB color space; the representation of color
in YCbCr is a merely an affine map from the absolute color space; not the
other way around.
Put it in another way, I merely treated R?, G?, and B? as three unknowns in
a system of three independent linear equations, which are being used to
define color in an absolute color space; but in the example of the video I
showed, is it possible to find unique solution with a known given affine map
and three known values for Y, Cb, and Cr? The answer is yes. If only 0-255
is allowed for the definition of color in each channel of R?, G?, and B?,
there is no way the inverse affine map will yield a negative value or any
other values. In fact, if the affine map is used, there will be exactly
16,777,216 number of possible values of Y, Cb, and Cr since they are defined
from an affine transformation from a RGB color space.
That said, as you can see from the video, due to rounding, it?s still need
to make appropriate rounding or quantization to fulfill the complete
one-to-one relationship. Granted, if YCbCr is treated as an absolute color
space, and other values other than the 16,777,216 number of values are
allowed, solving the equations may lead us to negative values or
non-integers as you suggested.
The fact to the matter is, different device or software or operating system
may use different absolute color space. But YCbCr is not one of them.
So, why we specifically define an affine map to yield YCbCr representations,
a simple way to say is that we separate the color information to three
different channels, instead of Red, Green, Blue, we choose Luma, Chroma Red
Difference, and Chroma Blue difference. That?s not strictly true however.
One would point out that if there is no chroma information, it would be
grayscale or black and white. But the truth is since YCbCr is defined as
affine map to the RGB color space, different levels of ?gray? or ?black? and
?white? can actually still be represented by three numbers, R, G, and B. In
particular, Black is represented by ?0,0,0? in sRGB. Every single color
requires three numbers in sRGB. Even if there is only 256 colors in gray
scale, the colors are still represented by three numbers.
256 ? 1 ? 1= 256
In other words, in gray scale, in YCbCr representations, we only needs a
single unique but maybe different number for both Cb and Cr channels;
whereas in Y channels, 256 numbers are allowed. Unlike sRGB, these 256
numbers in Y channels are not necessarily integers. In the original sRGB
color space, there are also 256 numbers required, but each time the number
for each channel, namely, R, G, and B would change accordingly. That?s why
re-encoding to the relative color space YCbCr makes sense as human eyes are
less sensitive to the change in chroma information compared with luma
information. This re-encoding provides the foundation for chroma
subsampling, which is to reduce the number of pixels to store chroma
information for a given sample. But makes no mistake. The colors are still
defined by sRGB representation in Windows PC or other RGB color space in
other systems. YCbCr is merely a representation given by affine map. That?s
all.




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From jiachielee at live.com  Wed Jul 31 21:13:10 2013
From: jiachielee at live.com (thljcl)
Date: Wed, 31 Jul 2013 12:13:10 -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: <1375297990307-4660376.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

While Charles Poynton may not show me how he derives the inverse of
transformation matrix, he apparently arrives at the same conclusion as I do
with regards to the inverse of transformation matrix. Note that our
equations may seem different to each other as I specifically rewritten it in
different forms in the final slides. But if you pay close attention, they
are basically the same.
Take a look at
http://www.poynton.com/notes/colour_and_gamma/ColorFAQ.html#RTFToC29 and
http://www.youtube.com/watch?v=PDVk-fRPzxU. It?s a good thing to know that
the expert you recommend actually shares the same idea as I do. 




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From nlewis at crawford.com  Wed Jul 31 23:38:06 2013
From: nlewis at crawford.com (Nathan Lewis)
Date: Wed, 31 Jul 2013 17:38:06 -0400
Subject: [FFmpeg-user] Offlist,
 Is mixing of audio channels happening by default in ffmpeg?
In-Reply-To: <loom.20130719T120527-55@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>
Message-ID: <CACegdbJ3EsCjRZZGnPa0VXFdwjagM2wX7fAAtPLzNVCfa6RFyg@mail.gmail.com>

On Fri, Jul 19, 2013 at 6:06 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Bouke (VideoToolShed <bouke <at> videotoolshed.com> writes:
>
> > > I was hoping somebody could test if writing 0xf for
> > > all audio tracks as asked in this mail really helps:
> > > http://thread.gmane.org/gmane.comp.video.ffmpeg.user/45089/focus=45134
> >
> > What do you want me to do? I'm an end user, not a C coder...
>
> No C coder is needed, just somebody who can compile
> FFmpeg. Apparently, this is such an unimportant
> problem that I will probably not work on it.
>
> Thanks, Carl Eugen
>
>
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);  ?