From s1445051 at mail.zih.tu-dresden.de  Tue Apr  1 00:36:47 2014
From: s1445051 at mail.zih.tu-dresden.de (Frank Tetzel)
Date: Tue, 1 Apr 2014 00:36:47 +0200
Subject: [FFmpeg-user] Minimum configuration settings for ffprobe-only
In-Reply-To: <CAFrh=nc92sye_YdJoJxdTFK7KsLLBYH9xYz6kamZ-VMedVD1ug@mail.gmail.com>
References: <CAFrh=nfHpOYR73HG=dFBDBt8Of1A6z9ybw1oOjvVfNvZQ1mxSg@mail.gmail.com>
	<1396210284302.ebde3a7c@Nodemailer>
	<CAFrh=nc92sye_YdJoJxdTFK7KsLLBYH9xYz6kamZ-VMedVD1ug@mail.gmail.com>
Message-ID: <20140401003647.06425198@archthink.fritz.box>

Am Sun, 30 Mar 2014 15:15:09 -0500
schrieb Eugene Gekhter <egekhter at pixcel.com>:

> On Sun, Mar 30, 2014 at 3:11 PM, Nathan Lewis
> <nlewis at mail.crawford.com>wrote:
> 
> > On Sun, Mar 30, 2014 at 3:46 PM, Eugene Gekhter <egekhter at pixcel.com
> > ="mailto:egekhter at pixcel.com">> wrote:
> >
> >
> > What would be the recommended configuration settings if all that was
> >
> > required was to build ffprobe for the sole purpose of determining
> > if ffmpeg
> >
> > can decode the input files?
> >
> > The ultimate goal is to compile this with Emscripten into
> > javascript to
> >
> > detect on client-side if the file a user wants to upload is even
> > worth
> >
> > uploading.
> >
> >
> >
> >
> >
> >
> > Wouldn't running "ffmpeg -i" against the incoming file and
> > returning no errors give you what you want?  If so, just running
> > make against source should yield the results you're after.
> 
> True, I am looking for something as light-weight as possible

Maybe you should limit your supported container formats drastically and
then just check against a whitelist of known video and audio codecs.
That way you could just use small container libraries.

Here's an example for mp4 files using libmp4v2:
http://tetzank.github.io/extractbitrate-js/site.html
(in Firefox Ctrl+Shift+K, in Chromium Ctrl+Shift+J for more info)
https://github.com/tetzank/extractbitrate-js

This was just a toy project of mine to try out emscripten. It just
tries to extract the bitrate.

Using ffprobe or libavformat might be too big.

From michael at redmule.com  Tue Apr  1 00:07:22 2014
From: michael at redmule.com (Michael Hipp)
Date: Mon, 31 Mar 2014 17:07:22 -0500
Subject: [FFmpeg-user] -f segment does not seem to be working
In-Reply-To: <20140331133753.7fae104a@lrcd.com>
References: <5339DE02.9060401@redmule.com> <20140331133753.7fae104a@lrcd.com>
Message-ID: <5339E71A.4080305@redmule.com>

On 3/31/2014 4:37 PM, Lou wrote:
> On Mon, 31 Mar 2014 16:28:34 -0500
> Michael Hipp <michael at redmule.com> wrote:
>
>> I'm attempting to segment an incoming mp3 audio stream:
>>
>> ffmpeg -i udp://239.8.8.8:5000 -acodec copy -f segment -segment_time 5 out%03d.mp3
>>
>> The audio records fine but it never segments. The one file (out000.mp3) just
>> grows and grows.
>>
>> What am I doing wrong?
>>
>> I'm using ffpmeg as ships with ubuntu 12.04 (version
>> 0.8.10-4:0.8.10-0ubuntu0.12.04.1).
> As of 13.10 Ubuntu does not include ffmpeg from FFmpeg. The version you
> are using is from a fork of FFmpeg. This mailing list only supports
> stuff from FFmpeg.
>
> You can try using a recent build of ffmpeg:
> http://ffmpeg.org/download.html#LinuxBuilds
>

Thank you. I'm afraid the version numbers are a bit confusing.

But anyways, I downloaded a static build from 
http://ffmpeg.gusari.org/static/32bit/ as linked on that page and now it 
segments as expected. I only had to add a -map 0 option:

  ./ffmpeg -i udp://239.8.8.8:5000 -acodec copy -map 0 -f segment -segment_time 
60 out%03d.mp3

Thanks,
Michael Hipp

From adh38 at cornell.edu  Tue Apr  1 00:14:33 2014
From: adh38 at cornell.edu (Adam Herbst)
Date: Mon, 31 Mar 2014 15:14:33 -0700
Subject: [FFmpeg-user] sporadic rtmp live stream crashes
In-Reply-To: <D8A180EE-1A4D-499E-B7D3-1E71AE0FB821@dubstep.fm>
References: <CA+uSmNMsLd_o0GAGkBDiBdNA_mXO-SUa+dxNihEVFdmLdBi4uw@mail.gmail.com>
	<D8A180EE-1A4D-499E-B7D3-1E71AE0FB821@dubstep.fm>
Message-ID: <CA+uSmNPOO=hK9a1hqo3oHULOU4JeXB=ge2Z5XDJuQZSiOcaMrw@mail.gmail.com>

You are correct - I forgot to note that the incoming rtmp is Sorensen Spark
video and Speex audio, whereas Wowza only creates HLS from an H264/AAC
stream.  In fact our original goal with ffmpeg was just to transcode
between these two specs, until we found out ffmpeg does straight HLS.

Let me take this opportunity to add that I've messed with the ffmpeg
parameters within my very limited understanding of what they do.  Command
line options I've added that didn't fix it:

-re (read input stream at native frame rate)
-profile:v baseline (not so clear on what this does)
-v:b 400k (video bitrate - tried a few different values, along with
-maxrate 400k -bufsize 800k, because someone said -maxrate and -bufsize
should be v:b and 2*v:b)
-probesize 50k (how much data to scan to determine stream info - instead of
default 500k - someone said the "failed to read packet" might be a timeout
while reading this part)
-dropped -r (output frame rate) and -g (key frame interval) to 20 and 40
-most recently, added timeout=10 and buffer=3600000 to the input rtmp
string - still testing the effects



On Mon, Mar 31, 2014 at 2:21 PM, DopeLabs <dopelabs at dubstep.fm> wrote:

> from what i understand wowza should support HLS out of the box...
>
> using 'cupertinostreamingpacketizer'
>
> http://www.wowza.com/forums/content.php?217#cupertinostreaming
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From dopelabs at dubstep.fm  Tue Apr  1 00:50:38 2014
From: dopelabs at dubstep.fm (DopeLabs)
Date: Mon, 31 Mar 2014 15:50:38 -0700
Subject: [FFmpeg-user] sporadic rtmp live stream crashes
In-Reply-To: <CA+uSmNPOO=hK9a1hqo3oHULOU4JeXB=ge2Z5XDJuQZSiOcaMrw@mail.gmail.com>
References: <CA+uSmNMsLd_o0GAGkBDiBdNA_mXO-SUa+dxNihEVFdmLdBi4uw@mail.gmail.com>
	<D8A180EE-1A4D-499E-B7D3-1E71AE0FB821@dubstep.fm>
	<CA+uSmNPOO=hK9a1hqo3oHULOU4JeXB=ge2Z5XDJuQZSiOcaMrw@mail.gmail.com>
Message-ID: <08ED0FEF-0ADC-40D8-8E10-43F07D9F1A1D@dubstep.fm>

im not entirely sure then.. 

im not sure how you have things setup but you can always just do the transcode and send it right back to wowza in a native format such as flv... this way you have wowza doing all the fun stuff...

but just off the top of my head...


ffmpeg -i rtmp://wowza/private/stream -c:a libfdk_aac -c:v h264 -g 50 -f flv rtmp://wowza/live/stream

then you can just use the native wowza HLS url

http://wowza:1935/live/stream/playlist.m3u8


on first run i noticed lots of dropped frames so i removed -r 25 so ffmpeg will use the input framerate.

i tested this against a flv h264 aac stream on my own wowza setup (as i really dont know much about spark/speex).... so im not 'entirely' sure the you will have the same results.. but this might help =]


here is my command output

$ ffmpeg -i rtmp://wowza/private/stream -c:a libfdk_aac -c:v h264 -g 50 -f flv rtmp://wowza/live/stream
ffmpeg version N-61744-g0dd4eff Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar 23 2014 19:59:06 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  configuration: --prefix=/usr/local --bindir=/usr/local/bin --datadir=/usr/local/share/ffmpeg --docdir=/usr/local/share/doc/ffmpeg --libdir=/usr/local/lib --shlibdir=/usr/local/lib --incdir=/usr/local/include --mandir=/usr/local/share/man --enable-version3 --enable-x11grab --enable-opengl --enable-opencl --enable-openal --enable-libxvid --enable-libx264 --enable-libwavpack --enable-libvpx --enable-libvorbis --enable-libvo-aacenc --enable-libtheora --enable-libssh --enable-librtmp --enable-libopenjpeg --enable-libmp3lame --enable-libfdk-aac --enable-libfaac --enable-libass --enable-vda --enable-avresample --enable-nonfree --enable-gpl
  libavutil      52. 67.100 / 52. 67.100
  libavcodec     55. 52.103 / 55. 52.103
  libavformat    55. 34.101 / 55. 34.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libavresample   1.  2.  0 /  1.  2.  0
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Metadata:
  duration              0.00
  width                 1280.00
  height                720.00
  videodatarate         3023.44
  framerate             34.45
  videocodecid          7.00
  audiodatarate         187.50
  audiosamplerate       44100.00
  audiosamplesize       16.00
  stereo                TRUE
  audiocodecid          10.00
  encoder               Lavf55.25.100
  filesize              0.00
Input #0, flv, from 'rtmp://wowza/private/stream':
  Metadata:
    StreamTitle     : ARCHIVE
    encoder         : Lavf55.25.100
  Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 3096 kb/s, 34.50 fps, 34.42 tbr, 1k tbn, 68.91 tbc
    Stream #0:1: Audio: aac, 44100 Hz, stereo, fltp, 192 kb/s
[libx264 @ 0x20571a0] using SAR=1/1
[libx264 @ 0x20571a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
[libx264 @ 0x20571a0] profile High, level 3.2
[libx264 @ 0x20571a0] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 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=50 keyint_min=5 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, flv, to 'rtmp://wowza/live/stream':
  Metadata:
    StreamTitle     : ARCHIVE
    encoder         : Lavf55.34.101
    Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 1k tbn, 34.42 tbc
    Stream #0:1: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, s16, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (aac -> libfdk_aac)
Press [q] to stop, [?] for help
[flv @ 0x20567c0] Failed to update header with correct duration.ate=1703.4kbits/s    
[flv @ 0x20567c0] Failed to update header with correct filesize.
frame=  316 fps=106 q=-1.0 Lsize=    2515kB time=00:00:09.38 bitrate=2196.0kbits/s    
video:2353kB audio:148kB subtitle:0 data:0 global headers:0kB muxing overhead 0.535417%
[libx264 @ 0x20571a0] frame I:11    Avg QP:22.45  size:  9607
[libx264 @ 0x20571a0] frame P:305   Avg QP:30.15  size:  7552
[libx264 @ 0x20571a0] mb I  I16..4: 73.3% 17.5%  9.1%
[libx264 @ 0x20571a0] mb P  I16..4: 16.5%  0.8%  9.1%  P16..4:  1.6%  0.3%  0.1%  0.0%  0.0%    skip:71.6%
[libx264 @ 0x20571a0] 8x8 transform intra:4.6% inter:9.3%
[libx264 @ 0x20571a0] coded y,uvDC,uvAC intra: 31.0% 0.0% 0.0% inter: 0.9% 0.0% 0.0%
[libx264 @ 0x20571a0] i16 v,h,dc,p: 77% 16%  4%  2%
[libx264 @ 0x20571a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 57%  7% 34%  1%  0%  0%  0%  0%  1%
[libx264 @ 0x20571a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 73%  9% 14%  1%  0%  1%  0%  1%  0%
[libx264 @ 0x20571a0] i8c dc,h,v,p: 100%  0%  0%  0%
[libx264 @ 0x20571a0] Weighted P-Frames: Y:0.7% UV:0.0%
[libx264 @ 0x20571a0] ref P L0: 53.4%  9.1% 24.1% 13.2%  0.2%
[libx264 @ 0x20571a0] kb/s:2099.07



From brian.quandt at gmail.com  Tue Apr  1 01:20:33 2014
From: brian.quandt at gmail.com (Brian Quandt)
Date: Mon, 31 Mar 2014 16:20:33 -0700
Subject: [FFmpeg-user] http input issues against S3
Message-ID: <CAOLkBAkiytOjAQLAvh-H=GKNxhueaWS-RW1HLrr3pA04V8p_ag@mail.gmail.com>

When using http as an input "-i" it seems that error handling (or retrying)
seems weak.

Are there any options to the command line to improve error handling /
retry's on http connections (when used as input).

For instance, if I happen to try to use ffmpeg to transcode from an S3
source, which happens to be on a very heavily used connection, I often get
HTTP 500 type errors (ie 502, 503, etc).   These are nearly never fatal,
but simply indicate a retry is necessary.     ffmpeg using default
parameters, results in messages like:

[https @ 0x1bb2a00] HTTP error 502 Bad Gateway2kbits/s
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x1bb2220] stream 1, offset 0x2072adc0: partial
file
https://api-content.dropbox.com/1/files/auto//of_god_and_dogs_1280x720.mp4:
Invalid data found when processing input

And data is then lost (ie seems like the connection is terminated).

Is there a patch to this behavior or command lines options to set to
improve retry's?

Here's my example ffmpeg command line I'm currently using:

ffmpeg -y -user-agent autodcp/1.0 -headers Authorization: Bearer
nta1234123444123Pr3tp82_S-9wQTdI1?  -i https://somewhere/o.mp4 -f wav
/tmp/out.wav

From adh38 at cornell.edu  Tue Apr  1 01:40:49 2014
From: adh38 at cornell.edu (Adam Herbst)
Date: Mon, 31 Mar 2014 16:40:49 -0700
Subject: [FFmpeg-user] sporadic rtmp live stream crashes
In-Reply-To: <08ED0FEF-0ADC-40D8-8E10-43F07D9F1A1D@dubstep.fm>
References: <CA+uSmNMsLd_o0GAGkBDiBdNA_mXO-SUa+dxNihEVFdmLdBi4uw@mail.gmail.com>
	<D8A180EE-1A4D-499E-B7D3-1E71AE0FB821@dubstep.fm>
	<CA+uSmNPOO=hK9a1hqo3oHULOU4JeXB=ge2Z5XDJuQZSiOcaMrw@mail.gmail.com>
	<08ED0FEF-0ADC-40D8-8E10-43F07D9F1A1D@dubstep.fm>
Message-ID: <CA+uSmNMrSQau4rKsHB+CGqrXuTEq=0JDErmkUYueVLv=O8Q8dw@mail.gmail.com>

Same deal, it's giving me the RTMPSockBuf_Fill error in the early going.
 It must be something about the way ffmpeg is trying to pull the incoming
rtmp stream, regardless of the output format.  Thanks for the suggestions,
much appreciated.  One thing that seems to be helping after a few tests
just now is setting buffer=3600000 in the rtmp string.  I used that number
because I thought I read somewhere that the default was 36000000 (10x
higher) and I wanted to decrease it with the idea that ffmpeg wouldn't be
trying to fill a big buffer when it didn't have any data to work with.  But
now I see the default is 30000 so I'm actually increasing the buffer size.
 So maybe librtmp just needed more room to store incoming packets?  I
welcome further advice because this is still very hazy.


On Mon, Mar 31, 2014 at 3:50 PM, DopeLabs <dopelabs at dubstep.fm> wrote:

> im not entirely sure then..
>
> im not sure how you have things setup but you can always just do the
> transcode and send it right back to wowza in a native format such as flv...
> this way you have wowza doing all the fun stuff...
>
> but just off the top of my head...
>
>
> ffmpeg -i rtmp://wowza/private/stream -c:a libfdk_aac -c:v h264 -g 50 -f
> flv rtmp://wowza/live/stream
>
> then you can just use the native wowza HLS url
>
> http://wowza:1935/live/stream/playlist.m3u8
>
>
> on first run i noticed lots of dropped frames so i removed -r 25 so ffmpeg
> will use the input framerate.
>
> i tested this against a flv h264 aac stream on my own wowza setup (as i
> really dont know much about spark/speex).... so im not 'entirely' sure the
> you will have the same results.. but this might help =]
>
>
> here is my command output
>
> $ ffmpeg -i rtmp://wowza/private/stream -c:a libfdk_aac -c:v h264 -g 50 -f
> flv rtmp://wowza/live/stream
> ffmpeg version N-61744-g0dd4eff Copyright (c) 2000-2014 the FFmpeg
> developers
>   built on Mar 23 2014 19:59:06 with gcc 4.8 (Ubuntu/Linaro
> 4.8.1-10ubuntu9)
>   configuration: --prefix=/usr/local --bindir=/usr/local/bin
> --datadir=/usr/local/share/ffmpeg --docdir=/usr/local/share/doc/ffmpeg
> --libdir=/usr/local/lib --shlibdir=/usr/local/lib
> --incdir=/usr/local/include --mandir=/usr/local/share/man --enable-version3
> --enable-x11grab --enable-opengl --enable-opencl --enable-openal
> --enable-libxvid --enable-libx264 --enable-libwavpack --enable-libvpx
> --enable-libvorbis --enable-libvo-aacenc --enable-libtheora --enable-libssh
> --enable-librtmp --enable-libopenjpeg --enable-libmp3lame
> --enable-libfdk-aac --enable-libfaac --enable-libass --enable-vda
> --enable-avresample --enable-nonfree --enable-gpl
>   libavutil      52. 67.100 / 52. 67.100
>   libavcodec     55. 52.103 / 55. 52.103
>   libavformat    55. 34.101 / 55. 34.101
>   libavdevice    55. 11.100 / 55. 11.100
>   libavfilter     4.  3.100 /  4.  3.100
>   libavresample   1.  2.  0 /  1.  2.  0
>   libswscale      2.  5.102 /  2.  5.102
>   libswresample   0. 18.100 /  0. 18.100
>   libpostproc    52.  3.100 / 52.  3.100
> Metadata:
>   duration              0.00
>   width                 1280.00
>   height                720.00
>   videodatarate         3023.44
>   framerate             34.45
>   videocodecid          7.00
>   audiodatarate         187.50
>   audiosamplerate       44100.00
>   audiosamplesize       16.00
>   stereo                TRUE
>   audiocodecid          10.00
>   encoder               Lavf55.25.100
>   filesize              0.00
> Input #0, flv, from 'rtmp://wowza/private/stream':
>   Metadata:
>     StreamTitle     : ARCHIVE
>     encoder         : Lavf55.25.100
>   Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
>     Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 1280x720
> [SAR 1:1 DAR 16:9], 3096 kb/s, 34.50 fps, 34.42 tbr, 1k tbn, 68.91 tbc
>     Stream #0:1: Audio: aac, 44100 Hz, stereo, fltp, 192 kb/s
> [libx264 @ 0x20571a0] using SAR=1/1
> [libx264 @ 0x20571a0] using cpu capabilities: MMX2 SSE2Fast SSSE3
> FastShuffle SSE4.2 AVX
> [libx264 @ 0x20571a0] profile High, level 3.2
> [libx264 @ 0x20571a0] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC
> codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options:
> cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1
> psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1
> cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12
> 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=50 keyint_min=5 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, flv, to 'rtmp://wowza/live/stream':
>   Metadata:
>     StreamTitle     : ARCHIVE
>     encoder         : Lavf55.34.101
>     Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 1k tbn, 34.42 tbc
>     Stream #0:1: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 44100
> Hz, stereo, s16, 128 kb/s
> Stream mapping:
>   Stream #0:0 -> #0:0 (h264 -> libx264)
>   Stream #0:1 -> #0:1 (aac -> libfdk_aac)
> Press [q] to stop, [?] for help
> [flv @ 0x20567c0] Failed to update header with correct
> duration.ate=1703.4kbits/s
> [flv @ 0x20567c0] Failed to update header with correct filesize.
> frame=  316 fps=106 q=-1.0 Lsize=    2515kB time=00:00:09.38
> bitrate=2196.0kbits/s
> video:2353kB audio:148kB subtitle:0 data:0 global headers:0kB muxing
> overhead 0.535417%
> [libx264 @ 0x20571a0] frame I:11    Avg QP:22.45  size:  9607
> [libx264 @ 0x20571a0] frame P:305   Avg QP:30.15  size:  7552
> [libx264 @ 0x20571a0] mb I  I16..4: 73.3% 17.5%  9.1%
> [libx264 @ 0x20571a0] mb P  I16..4: 16.5%  0.8%  9.1%  P16..4:  1.6%  0.3%
>  0.1%  0.0%  0.0%    skip:71.6%
> [libx264 @ 0x20571a0] 8x8 transform intra:4.6% inter:9.3%
> [libx264 @ 0x20571a0] coded y,uvDC,uvAC intra: 31.0% 0.0% 0.0% inter: 0.9%
> 0.0% 0.0%
> [libx264 @ 0x20571a0] i16 v,h,dc,p: 77% 16%  4%  2%
> [libx264 @ 0x20571a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 57%  7% 34%  1%  0%
>  0%  0%  0%  1%
> [libx264 @ 0x20571a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 73%  9% 14%  1%  0%
>  1%  0%  1%  0%
> [libx264 @ 0x20571a0] i8c dc,h,v,p: 100%  0%  0%  0%
> [libx264 @ 0x20571a0] Weighted P-Frames: Y:0.7% UV:0.0%
> [libx264 @ 0x20571a0] ref P L0: 53.4%  9.1% 24.1% 13.2%  0.2%
> [libx264 @ 0x20571a0] kb/s:2099.07
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From dopelabs at dubstep.fm  Tue Apr  1 01:50:59 2014
From: dopelabs at dubstep.fm (DopeLabs)
Date: Mon, 31 Mar 2014 16:50:59 -0700
Subject: [FFmpeg-user] sporadic rtmp live stream crashes
In-Reply-To: <CA+uSmNMrSQau4rKsHB+CGqrXuTEq=0JDErmkUYueVLv=O8Q8dw@mail.gmail.com>
References: <CA+uSmNMsLd_o0GAGkBDiBdNA_mXO-SUa+dxNihEVFdmLdBi4uw@mail.gmail.com>
	<D8A180EE-1A4D-499E-B7D3-1E71AE0FB821@dubstep.fm>
	<CA+uSmNPOO=hK9a1hqo3oHULOU4JeXB=ge2Z5XDJuQZSiOcaMrw@mail.gmail.com>
	<08ED0FEF-0ADC-40D8-8E10-43F07D9F1A1D@dubstep.fm>
	<CA+uSmNMrSQau4rKsHB+CGqrXuTEq=0JDErmkUYueVLv=O8Q8dw@mail.gmail.com>
Message-ID: <294624F0-A623-4DB3-B21C-73DE6E541752@dubstep.fm>

well thats all ive got... 

but i ponder...  video sourcing from an iphone? is the input stream stable? wowza logs? same issues with an input stream thats source from a hardline?


hope you get it sorted out =]

cheers


From lifeamit at gmail.com  Tue Apr  1 07:09:20 2014
From: lifeamit at gmail.com (Amit Agarwal)
Date: Tue, 1 Apr 2014 10:39:20 +0530
Subject: [FFmpeg-user] How to stitch images to create video
Message-ID: <CAPMi8WYEe3Kd2OP88rAZ4WaAM6RkeibBFojzgp1_o8-oQ=V7og@mail.gmail.com>

Hi,

I am not able to stitch images to create a video out of them. Here is the
command I am using to stitch images together.

ffmpeg -t 10.993 -f image2 -r 29.97 -threads 0 -i
images/i-29x97fps-1280x720-%05d.png -vcodec libx264 -acodec libfaac -ar 48k
-ab 224k -aq 100 -vol 384 -async 1 -y video.mp4

This command it creating video.mp4 but that seems to be corrupt file. No
player is able to play that file and VLC crashes when trying to play this
file.

Can someone please help me fix this command in order to generate a healthy
mp4 video file ?

I am using ffmpeg version 2.1.4 and 2.2. Both are giving same output.



*Here are the logs in case they help:*

ffmpeg -t 10.993 -f image2 -r 29.97 -threads 0 -i
images/i-29x97fps-1280x720-%05d.png -vcodec libx264 -acodec libfaac -ar 48k
-ab 224k -aq 100 -vol 384 -async 1 -y video.mp4
ffmpeg version 2.2-tessus Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar 24 2014 22:25:11 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
  configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm
--extra-version=tessus --disable-shared --enable-static --disable-ffplay
--enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame
--enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid
--enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libxavs --enable-version3
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx
--enable-libgsm --enable-libopus --enable-libmodplug --enable-fontconfig
--enable-libfreetype --enable-libass --enable-libbluray --enable-filters
--enable-runtime-cpudetect
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, image2, from 'images/i-29x97fps-1280x720-%05d.png':
  Duration: 00:00:10.94, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: png, rgba, 1024x769 [SAR 2835:2835 DAR 1024:769],
29.97 fps, 29.97 tbr, 29.97 tbn, 29.97 tbc
Codec AVOption ab (set bitrate (in bits/s)) specified for output file #0
(video.mp4) 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.
No pixel format specified, yuv444p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x102043000] using SAR=1/1
[libx264 @ 0x102043000] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
AVX
[libx264 @ 0x102043000] profile High 4:4:4 Predictive, level 3.1, 4:4:4
8-bit
[libx264 @ 0x102043000] 264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft
2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3
deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11
fast_pskip=1 chroma_qp_offset=4 threads=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=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 'video.mp4':
  Metadata:
    encoder         : Lavf55.33.100
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv444p,
1024x769 [SAR 1:1 DAR 1024:769], q=-1--1, 11988 tbn, 29.97 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (png -> libx264)
Press [q] to stop, [?] for help
frame=  328 fps=100 q=-1.0 Lsize=     387kB time=00:00:10.87 bitrate=
291.2kbits/s
video:382kB audio:0kB subtitle:0 data:0 global headers:0kB muxing overhead
1.209907%
[libx264 @ 0x102043000] frame I:4     Avg QP:12.84  size: 24742
[libx264 @ 0x102043000] frame P:84    Avg QP:20.80  size:  1849
[libx264 @ 0x102043000] frame B:240   Avg QP:24.71  size:   567
[libx264 @ 0x102043000] consecutive B-frames:  2.4%  0.0%  0.0% 97.6%
[libx264 @ 0x102043000] mb I  I16..4:  9.0% 69.5% 21.5%
[libx264 @ 0x102043000] mb P  I16..4:  1.5%  1.5%  0.4%  P16..4: 10.1%
 2.3%  1.0%  0.0%  0.0%    skip:83.2%
[libx264 @ 0x102043000] mb B  I16..4:  0.3%  0.1%  0.0%  B16..8: 11.1%
 0.7%  0.0%  direct: 0.0%  skip:87.8%  L0:50.1% L1:47.3% BI: 2.5%
[libx264 @ 0x102043000] 8x8 transform intra:54.4% inter:62.2%
[libx264 @ 0x102043000] coded y,u,v intra: 30.3% 0.0% 0.0% inter: 0.6% 0.0%
0.0%
[libx264 @ 0x102043000] i16 v,h,dc,p: 54% 39%  3%  3%
[libx264 @ 0x102043000] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 32% 16% 40%  3%  3%
 2%  2%  2%  2%
[libx264 @ 0x102043000] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 46% 26% 16%  3%  3%
 2%  2%  1%  2%
[libx264 @ 0x102043000] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x102043000] ref P L0: 49.9%  4.1% 36.5%  9.6%
[libx264 @ 0x102043000] ref B L0: 59.0% 25.3% 15.7%
[libx264 @ 0x102043000] ref B L1: 82.3% 17.7%
[libx264 @ 0x102043000] kb/s:285.43

From cehoyos at ag.or.at  Tue Apr  1 09:27:35 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 07:27:35 +0000 (UTC)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>
Message-ID: <loom.20140401T092529-671@post.gmane.org>

Nicholas Robbins <nickrobbins <at> yahoo.com> writes:

> I have video, that, by visual inspection, is mixed telecined 
> (2:3) & interlaced.

Could you provide a sample?

Carl Eugen


From cehoyos at ag.or.at  Tue Apr  1 09:30:25 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 07:30:25 +0000 (UTC)
Subject: [FFmpeg-user] Memory leaks in ffmpeg 2.2
References: <1395920879338-4664505.post@n4.nabble.com>
	<1396009615715-4664518.post@n4.nabble.com>
	<CAMCjN1-L2bi0Lj6vP59iupqU5uZNCVronuXoSa7-f=aV0kSO2g@mail.gmail.com>
Message-ID: <loom.20140401T092800-856@post.gmane.org>

Naveen Rudrappa <naveen.mr <at> gmail.com> writes:

>  <at> Carl: were you inferring this behaviour ?

No.
Testing with sdl/ffplay is usually not a good idea, 
you find many bugs in sdl;-(
(Or at least memleaks that have nothing to do with 
FFmpeg.)

This is about a global lock that libavcodec needs 
to avoid race conditions, releasing the lock would 
introduce another race condition (only for dynamic 
libraries iirc).

Please do not top-post here, Carl Eugen


From darren.j.breeze at gmail.com  Tue Apr  1 09:49:01 2014
From: darren.j.breeze at gmail.com (Darren Breeze)
Date: Tue, 01 Apr 2014 15:49:01 +0800
Subject: [FFmpeg-user] compand filter issues
Message-ID: <533A6F6D.1070107@gmail.com>

Hi

I am trying to use the compand filter to compress audio on some files 
using the following build

ffmpeg version git-2014-02-10-03911c4 Copyright (c) 2000-2014 the FFmpeg 
developers
   built on Feb 10 2014 14:12:24 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
   configuration: --prefix=/root/ffmpeg_build 
--extra-cflags=-I/root/ffmpeg_build/include 
--extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/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-libfreetype
   libavutil      52. 63.101 / 52. 63.101
   libavcodec     55. 50.100 / 55. 50.100
   libavformat    55. 31.100 / 55. 31.100
   libavdevice    55.  8.100 / 55.  8.100
   libavfilter     4.  1.102 /  4.  1.102
   libswscale      2.  5.101 /  2.  5.101
   libswresample   0. 17.104 /  0. 17.104
   libpostproc    52.  3.100 / 52.  3.100

with the following command line


/root/bin/ffmpeg -y -threads 0 -i /mnt/datastore/workflow/transcode/inbox/pop01/source_1080p.mp4 -aspect 16:9 -pix_fmt yuv420p -b:v 1800k -minrate 1800k -maxrate 1800k -bufsize 1835k -preset slow -x264opts keyint=33 -vcodec libx264 -acodec libfdk_aac -strict experimental -profile:v main -f mp4 -ac 2 -ab 192k -af compand=.3|.3:.8|.8:-90/-52|-52/-36|-36/-28|-20/-20:6:0:-20:0.8;volume=volume=-7.5dB /tmp/transcode/target_1080p.mp4

Whilst the processing works fine on channel 1 of the output file, channel 2 has high level to the point of clipping out.

I have set up this to a WAV file output with the same results.

the source files have one audio stream of 2 channels

The volume filter is applying an overall level correction based on a previous loudness analysis, the issue remains if this is omitted from the filter.

Not sure what's going on

Darren Breeze





From porparek at gmail.com  Tue Apr  1 11:28:14 2014
From: porparek at gmail.com (Pawel Por)
Date: Tue, 1 Apr 2014 11:28:14 +0200
Subject: [FFmpeg-user] cut at keyframes
Message-ID: <CAJmOuEyk=vbSaJnS8KCxNG5XP3c=pmA1QmRrBRVuRYNcEZbxjQ@mail.gmail.com>

Hi
I'm a great fan of ffmpeg. I've been using the version 1.0.7 of ffmpeg and
it was cutting my mp4 movies at closest keyframes.
It was exactly what I wanted.
It was very useful to me. I was using the following command:
ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 -acodec copy -vcodec copy
-y out.mp4

I've upgraded to the newer version of ffmpeg 2.2 and I found that now when
I use the above command it cuts my movies not at closest keyframes.
I tried "-noaccurate_seek" option (as follows) but it still cuts my movies
not at a closest keyframes.
ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 -acodec copy -vcodec copy
-y out.mp4

I believe that there must be some ffmpeg option to accomplish it (I cannot
find it in "man page").

Please, write me how I can do that.

thanks in advance

From cehoyos at ag.or.at  Tue Apr  1 11:35:12 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 09:35:12 +0000 (UTC)
Subject: [FFmpeg-user] cut at keyframes
References: <CAJmOuEyk=vbSaJnS8KCxNG5XP3c=pmA1QmRrBRVuRYNcEZbxjQ@mail.gmail.com>
Message-ID: <loom.20140401T113438-653@post.gmane.org>

Pawel Por <porparek <at> gmail.com> writes:

> ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 
> -acodec copy -vcodec copy -y out.mp4
> 
> I've upgraded to the newer version of ffmpeg 2.2 
> and I found that now when I use the above command 
> it cuts my movies not at closest keyframes.

Please provide an input sample.

Carl Eugen


From battistel at gmail.com  Tue Apr  1 11:52:05 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Tue, 1 Apr 2014 11:52:05 +0200
Subject: [FFmpeg-user]  Possible issue with asf muxer
Message-ID: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>

Hello,
you can download sample from here:

http://www.datafilehost.com/d/c466f4ad

This file plays fine with ffplay.

Now I try to convert it to raw asf:

ffmpeg -i intermission.wmv  -vcodec rawvideo -pix_fmt yuv420p -acodec
pcm_s16le -ac 2 -ar 48000 -f  asf intermission2.asf

ffmpeg version N-61570-gaa86ccc Copyright (c) 2000-2014 the FFmpeg
developers
  built on Mar 17 2014 22:04:13 with gcc 4.8.2 (GCC)
  configuration: --disable-static --enable-shared --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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 67.100 / 52. 67.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 34.101 / 55. 34.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, asf, from 'D:\Video\media\intermission.wmv':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: MP at LL
    WM/WMADRCPeakReference: 32767
    WM/WMADRCAverageReference: 6698
    WMFSDKVersion   : 9.00.00.2925
    IsVBR           : 0
  Duration: 00:00:40.00, start: 0.000000, bitrate: 229 kb/s
    Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz, stereo,
fltp, 64 kb/s
    Stream #0:1(eng): Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p,
320x240, 300 kb/s, 30 tbr, 1k tbn, 1k tbc
Output #0, asf, to 'intermission2.asf':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: MP at LL
    WM/WMADRCPeakReference: 32767
    WM/WMADRCAverageReference: 6698
    WMFSDKVersion   : 9.00.00.2925
    IsVBR           : 0
    WM/EncodingSettings: Lavf55.34.101
    Stream #0:0(eng): Video: rawvideo (I420 / 0x30323449), yuv420p,
320x240, q=2-31, 200 kb/s, 1k tbn, 30 tbc
    Stream #0:1(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo, s16, 1536 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 (wmv3 -> rawvideo)
  Stream #0:0 -> #0:1 (wmav2 -> pcm_s16le)
Press [q] to stop, [?] for help
frame= 1008 fps=0.0 q=0.0 size=   83504kB time=00:00:33.60
bitrate=20359.1kbits/s dup=1 drop=0
[wmav2 @ 0240e940] nb_frames is 0
Error while decoding stream #0:0: Invalid data found when processing input
frame= 1200 fps=0.0 q=0.0 Lsize=  144086kB time=00:00:40.01
bitrate=29496.2kbits/s dup=1 drop=0
video:135000kB audio:7512kB subtitle:0 data:0 global headers:0kB muxing
overhead 1.104240%


The output file has lots of issue with ffplay: a/v sync, bad audio, ...



Now I try to convert it to raw avi (only muxer is changed from previous cmd
line):

ffmpeg -i intermission.wmv  -vcodec rawvideo -pix_fmt yuv420p -acodec
pcm_s16le -ac 2 -ar 48000 -f  avi intermission2.avi

ffmpeg version N-61570-gaa86ccc Copyright (c) 2000-2014 the FFmpeg
developers
  built on Mar 17 2014 22:04:13 with gcc 4.8.2 (GCC)
  configuration: --disable-static --enable-shared --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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 67.100 / 52. 67.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 34.101 / 55. 34.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, asf, from 'D:\Video\media\intermission.wmv':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: MP at LL
    WM/WMADRCPeakReference: 32767
    WM/WMADRCAverageReference: 6698
    WMFSDKVersion   : 9.00.00.2925
    IsVBR           : 0
  Duration: 00:00:40.00, start: 0.000000, bitrate: 229 kb/s
    Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz, stereo,
fltp, 64 kb/s
    Stream #0:1(eng): Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p,
320x240, 300 kb/s, 30 tbr, 1k tbn, 1k tbc
Output #0, avi, to 'intermission2.avi':
  Metadata:
    WMFSDKNeeded    : 0.0.0.0000
    DeviceConformanceTemplate: MP at LL
    WM/WMADRCPeakReference: 32767
    WM/WMADRCAverageReference: 6698
    WMFSDKVersion   : 9.00.00.2925
    IsVBR           : 0
    ISFT            : Lavf55.34.101
    Stream #0:0(eng): Video: rawvideo (I420 / 0x30323449), yuv420p,
320x240, q=2-31, 200 kb/s, 30 tbn, 30 tbc
    Stream #0:1(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo, s16, 1536 kb/s
Stream mapping:
  Stream #0:1 -> #0:0 (wmv3 -> rawvideo)
  Stream #0:0 -> #0:1 (wmav2 -> pcm_s16le)
Press [q] to stop, [?] for help
frame= 1189 fps=0.0 q=0.0 size=  103870kB time=00:00:39.66
bitrate=21451.4kbits/s
[wmav2 @ 0252e940] nb_frames is 0
Error while decoding stream #0:0: Invalid data found when processing input
frame= 1199 fps=0.0 q=0.0 Lsize=  142443kB time=00:00:40.01
bitrate=29160.2kbits/s
video:134888kB audio:7512kB subtitle:0 data:0 global headers:0kB muxing
overhead 0.030529%


Everything is fine.

I suppose something is wrong with asf muxer.



Thanks,
MB

From cehoyos at ag.or.at  Tue Apr  1 12:09:42 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 10:09:42 +0000 (UTC)
Subject: [FFmpeg-user] Possible issue with asf muxer
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
Message-ID: <loom.20140401T120815-600@post.gmane.org>

Massimo Battistel <battistel <at> gmail.com> writes:

> ffmpeg -i intermission.wmv  -vcodec rawvideo -pix_fmt yuv420p 
> -acodec pcm_s16le -ac 2 -ar 48000 -f  asf intermission2.asf

Do you have any indication that rawvideo and / or 
rawaudio are supported in asf?
Do you have a sample or did you test the output 
with WMP?

Carl Eugen


From nickrobbins at yahoo.com  Tue Apr  1 13:01:07 2014
From: nickrobbins at yahoo.com (Nicholas Robbins)
Date: Tue, 1 Apr 2014 04:01:07 -0700 (PDT)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
In-Reply-To: <loom.20140401T092529-671@post.gmane.org>
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>
	<loom.20140401T092529-671@post.gmane.org>
Message-ID: <1396350067.93176.YahooMailNeo@web160804.mail.bf1.yahoo.com>

> On Tuesday, April 1, 2014 3:28 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> > Nicholas Robbins <nickrobbins <at> yahoo.com> writes:
> 
>>  I have video, that, by visual inspection, is mixed telecined 
>>  (2:3) & interlaced.
> 
> Could you provide a sample?
> 
> Carl Eugen
> 

I'm happy with the results of my filter chain, do you still want me to upload a sample? 

From nickrobbins at yahoo.com  Tue Apr  1 13:18:00 2014
From: nickrobbins at yahoo.com (Nicholas Robbins)
Date: Tue, 1 Apr 2014 04:18:00 -0700 (PDT)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
In-Reply-To: <1396350067.93176.YahooMailNeo@web160804.mail.bf1.yahoo.com>
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>	<loom.20140401T092529-671@post.gmane.org>
	<1396350067.93176.YahooMailNeo@web160804.mail.bf1.yahoo.com>
Message-ID: <1396351080.23052.YahooMailNeo@web160803.mail.bf1.yahoo.com>



> On Tuesday, April 1, 2014 7:01 AM, Nicholas Robbins <nickrobbins at yahoo.com> wrote:

> >>  On Tuesday, April 1, 2014 3:28 AM, Carl Eugen Hoyos 
> <cehoyos at ag.or.at> wrote:
> 
>>  > Nicholas Robbins <nickrobbins <at> yahoo.com> writes:
>> 
>>> ? I have video, that, by visual inspection, is mixed telecined 
>>> ? (2:3) & interlaced.
>> 
>>  Could you provide a sample?
>> 
>>  Carl Eugen
>> 
> 
> I'm happy with the results of my filter chain, do you still want me to 
> upload a sample? 


for posterity, uploaded robbins-short.mkv to uploads.ffmpeg.org/incoming

From cehoyos at ag.or.at  Tue Apr  1 13:46:53 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 11:46:53 +0000 (UTC)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>
	<1396302123.84118.YahooMailNeo@web160803.mail.bf1.yahoo.com>
Message-ID: <loom.20140401T134544-689@post.gmane.org>

Nicholas Robbins <nickrobbins <at> yahoo.com> writes:

> I tried " -vf pullup,dejudder,idet,yadif=deint=interlaced,fps=60000/1001"

This is either missing "mode=1" or do you actually want 
30fps output?
As is, this produces just many duplicated frames because 
yadif by default does not duplicate the frame rate.

Carl Eugen


From sonia.p.alves at hotmail.com  Tue Apr  1 13:57:24 2014
From: sonia.p.alves at hotmail.com (Sonia Alves)
Date: Tue, 1 Apr 2014 12:57:24 +0100
Subject: [FFmpeg-user] problem when trying to convert a video for a copy
 of it self
In-Reply-To: <20140331183458.GA27376@sunshine.barsnick.net>
References: <DUB111-W630831FB900DE919C77045A8630@phx.gbl>,
	<loom.20140331T172513-87@post.gmane.org>,
	<DUB111-W63BC9967F1FA252AE72C50A8630@phx.gbl>,
	<20140331183458.GA27376@sunshine.barsnick.net>
Message-ID: <DUB111-W1200FF4DEA5EB4C499CEBBDA8620@phx.gbl>

Sure.
Here it is:
ffmpeg -i Oscars.mp4 -f mp4 -vcodec copy -acodec copy Oscars2.mp4
ffmpeg version N-61886-g64b7914 Copyright (c) 2000-2014 the FFmpeg developers  built on Mar 27 2014 22:09:30 with gcc 4.8.2 (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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib  libavutil      52. 69.100 / 52. 69.100  libavcodec     55. 55.101 / 55. 55.101  libavformat    55. 35.101 / 55. 35.101  libavdevice    55. 11.100 / 55. 11.100  libavfilter     4.  3.100 /  4.  3.100  libswscale      2.  5.102 /  2.  5.102  libswresample   0. 18.100 /  0. 18.100  libpostproc    52.  3.100 / 52.  3.100Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Oscars.mp4':  Metadata:    major_brand     : mp42    minor_version   : 0    compatible_brands: isommp42    creation_time   : 2014-03-08 17:35:47  Duration: 00:01:24.52, start: 0.000000, bitrate: 570 kb/s    Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 472 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)    Metadata:      handler_name    : VideoHandler    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 96 kb/s (default)    Metadata:      creation_time   : 2014-03-08 17:35:47      handler_name    : IsoMedia File Produced by Google, 5-11-2011File 'Oscars2.mp4' already exists. Overwrite ? [y/N] yOutput #0, mp4, to 'Oscars2.mp4':  Metadata:    major_brand     : mp42    minor_version   : 0    compatible_brands: isommp42    encoder         : Lavf55.35.101    Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 640x360 [SAR 1:1 DAR 16:9], q=2-31, 472 kb/s, 29.97 fps, 30k tbn, 30k tbc (default)    Metadata:      handler_name    : VideoHandler    Stream #0:1(und): Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, 96kb/s (default)    Metadata:      creation_time   : 2014-03-08 17:35:47      handler_name    : IsoMedia File Produced by Google, 5-11-2011Stream mapping:  Stream #0:0 -> #0:0 (copy)  Stream #0:1 -> #0:1 (copy)Press [q] to stop, [?] for helpframe= 2533 fps=0.0 q=-1.0 Lsize=    5934kB time=00:01:24.52 bitrate= 575.2kbits/svideo:4872kB audio:991kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.215870%


> Date: Mon, 31 Mar 2014 20:34:58 +0200
> From: barsnick at gmx.net
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] problem when trying to convert a video for a copy of it self
> 
> On Mon, Mar 31, 2014 at 16:53:10 +0100, Sonia Alves wrote:
> > I'm sending you the output of the command line above in attachment.
> 
> Could you please copy the text into the email instead of attaching a
> screenshot? It makes the mail smaller, easier to view, and easier to
> point out issues.
> 
> Moritz
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
 		 	   		  

From cehoyos at ag.or.at  Tue Apr  1 15:04:21 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 13:04:21 +0000 (UTC)
Subject: [FFmpeg-user] Minimum configuration settings for ffprobe-only
References: <CAFrh=nfHpOYR73HG=dFBDBt8Of1A6z9ybw1oOjvVfNvZQ1mxSg@mail.gmail.com>
	<loom.20140331T103425-710@post.gmane.org>
	<CAFrh=ndskYO2L79=hGWo0MKAD_GfueF7fvF3nskVPBDWO+NUaQ@mail.gmail.com>
	<loom.20140331T172740-218@post.gmane.org>
	<CAFrh=nefE3oqMq4bh5U_14iD-Cb5=MC9aWPUYJ3tDfY8ESEVhg@mail.gmail.com>
	<5339B14C.5010206@gmail.com>
	<CAFrh=nc3Ay6+T1ShPb=X9TwnnXGZAVN2Gm=5q-EUuusrTbZFTg@mail.gmail.com>
Message-ID: <loom.20140401T150246-272@post.gmane.org>

Eugene Gekhter <egekhter <at> pixcel.com> writes:

> 98% of the time ffprobe would be dealing with normal 
> consumer-generated files. Your example above is the 
> exception, not the norm.

My original suspicion was (and still is) that in 99% 
of all cases, users will upload media files and not 
files that FFmpeg cannot read.

And please note that random data is typically recognized 
as mp3 or aac.

Carl Eugen


From battistel at gmail.com  Tue Apr  1 15:15:08 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Tue, 1 Apr 2014 15:15:08 +0200
Subject: [FFmpeg-user] Possible issue with asf muxer
In-Reply-To: <loom.20140401T120815-600@post.gmane.org>
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
	<loom.20140401T120815-600@post.gmane.org>
Message-ID: <CAHss8oGUkg_T=1DWk=UhU9_pjujkdh=zLkBEhSBwAH7MpwbfYg@mail.gmail.com>

>
>
> Do you have any indication that rawvideo and / or
> rawaudio are supported in asf?
>

Don't know if there is official doc about this, but I often do that with no
problems. Only the file I provided gives me issues.


> Do you have a sample or did you test the output
> with WMP?
>

The sample can be downloaded from here:
http://www.datafilehost.com/d/c466f4ad

Running the script I provided, you can generate the output file.

I've tested the output with ffplay and vlc 2.1.2. Both have problems with
asf and both are fine with avi, but with this sample only.


Another interesting information:
If you just remux to asf the output generated by script, you fix the issue:

ffmpeg -i intermission2.asf -c copy intermission3.asf

Output file (intermission3.asf) plays fine both on ffplay and vlc 2.1.2.


thanks,
MB

From nickrobbins at yahoo.com  Tue Apr  1 16:21:32 2014
From: nickrobbins at yahoo.com (Nicholas Robbins)
Date: Tue, 1 Apr 2014 07:21:32 -0700 (PDT)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
In-Reply-To: <loom.20140401T134544-689@post.gmane.org>
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>	<1396302123.84118.YahooMailNeo@web160803.mail.bf1.yahoo.com>
	<loom.20140401T134544-689@post.gmane.org>
Message-ID: <1396362092.74307.YahooMailNeo@web160806.mail.bf1.yahoo.com>

> On Tuesday, April 1, 2014 7:47 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> > Nicholas Robbins <nickrobbins <at> yahoo.com> writes:
> 
>>  I tried " -vf 
> pullup,dejudder,idet,yadif=deint=interlaced,fps=60000/1001"
> 
> This is either missing "mode=1" or do you actually want 
> 30fps output?
> As is, this produces just many duplicated frames because 
> yadif by default does not duplicate the frame rate.
> 
> Carl Eugen

60fps output would be better. So 
"-vf pullup,dejudder,idet,yadif=mode=1:deint=interlaced,fps=60000/1001"
is the way to go?


From cehoyos at ag.or.at  Tue Apr  1 16:32:04 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 14:32:04 +0000 (UTC)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>	<1396302123.84118.YahooMailNeo@web160803.mail.bf1.yahoo.com>
	<loom.20140401T134544-689@post.gmane.org>
	<1396362092.74307.YahooMailNeo@web160806.mail.bf1.yahoo.com>
Message-ID: <loom.20140401T163058-100@post.gmane.org>

Nicholas Robbins <nickrobbins <at> yahoo.com> writes:

> So "-vf pullup,dejudder,idet,yadif=mode=1:deint=interlaced,fps=60000/1001"
> is the way to go?

I had the feeling that idet is simply not made for this 
task (I think you can see where it incorrectly invokes 
the deinterlacer). Did you try fieldmatch?

Carl Eugen


From nickrobbins at yahoo.com  Tue Apr  1 16:43:07 2014
From: nickrobbins at yahoo.com (Nicholas Robbins)
Date: Tue, 1 Apr 2014 07:43:07 -0700 (PDT)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
In-Reply-To: <loom.20140401T163058-100@post.gmane.org>
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>	<1396302123.84118.YahooMailNeo@web160803.mail.bf1.yahoo.com>	<loom.20140401T134544-689@post.gmane.org>	<1396362092.74307.YahooMailNeo@web160806.mail.bf1.yahoo.com>
	<loom.20140401T163058-100@post.gmane.org>
Message-ID: <1396363387.70498.YahooMailNeo@web160806.mail.bf1.yahoo.com>

> On Tuesday, April 1, 2014 10:33 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> > Nicholas Robbins <nickrobbins <at> yahoo.com> writes:
> 
>>  So "-vf 
> pullup,dejudder,idet,yadif=mode=1:deint=interlaced,fps=60000/1001"
>>  is the way to go?
> 
> I had the feeling that idet is simply not made for this 
> task (I think you can see where it incorrectly invokes 
> the deinterlacer). Did you try fieldmatch?
> 
> Carl Eugen

IIRC fieldmatch isn't useful for partially telecined material, since it requires decimate which doesn't know to kill only the fieldmatch-produced extra frame. You either end up with extra frames in the telecined sections or drop useful frames in the non-telecined sections. What filter chain are you suggesting?

-Nick

From Alexander.Achleitner at zf.com  Tue Apr  1 17:25:21 2014
From: Alexander.Achleitner at zf.com (Alexander.Achleitner at zf.com)
Date: Tue, 1 Apr 2014 15:25:21 +0000
Subject: [FFmpeg-user] Problem Compiling ffmpeg
Message-ID: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>

Hi,

i want to compile the ffmpeg sources for an ARM  TI AM3359 with linux OS. I tried some
config arguments, but it doesn't work.
My gcc Cross-Compiler Toolchain on an Ubuntu Live Image works correcty,
as I can build qt-Applications which run on the ARM Board.
One Error message is
"...
AR   libavdevice/libavdevice.a
Sorry  -  this program has been built wihout plugin support "

Configuration was,
./configure -enable-cross-compile -cc=arm-cortexa8-linux-gnueabihf-gcc  --cxx=arm-cortexa8-linux-gnueabihf-gcc
--as=arm-cortexa8-linux-gnueabihf-gcc  --ar=arm-cortexa8-linux-gnueabihf-gcc  --arch=armel  --target-os=linux


Can anyone help me please.



Mit freundlichen Gr??en/Kind regards

Dr. Alexander Achleitner




From tevans.uk at googlemail.com  Tue Apr  1 17:29:37 2014
From: tevans.uk at googlemail.com (Tom Evans)
Date: Tue, 1 Apr 2014 16:29:37 +0100
Subject: [FFmpeg-user] How to stitch images to create video
In-Reply-To: <CAPMi8WYEe3Kd2OP88rAZ4WaAM6RkeibBFojzgp1_o8-oQ=V7og@mail.gmail.com>
References: <CAPMi8WYEe3Kd2OP88rAZ4WaAM6RkeibBFojzgp1_o8-oQ=V7og@mail.gmail.com>
Message-ID: <CAFHbX1JxscmrzGaBOBjBt0wizrULoxLhHbWOBZf=+nf-znzqWA@mail.gmail.com>

On Tue, Apr 1, 2014 at 6:09 AM, Amit Agarwal <lifeamit at gmail.com> wrote:
> Hi,
>
> I am not able to stitch images to create a video out of them. Here is the
> command I am using to stitch images together.
>
> ffmpeg -t 10.993 -f image2 -r 29.97 -threads 0 -i
> images/i-29x97fps-1280x720-%05d.png -vcodec libx264 -acodec libfaac -ar 48k
> -ab 224k -aq 100 -vol 384 -async 1 -y video.mp4
>
> This command it creating video.mp4 but that seems to be corrupt file. No
> player is able to play that file and VLC crashes when trying to play this
> file.
>
> Can someone please help me fix this command in order to generate a healthy
> mp4 video file ?
>
> I am using ffmpeg version 2.1.4 and 2.2. Both are giving same output.
>
>
>
> *Here are the logs in case they help:*
>
> ffmpeg -t 10.993 -f image2 -r 29.97 -threads 0 -i
> images/i-29x97fps-1280x720-%05d.png -vcodec libx264 -acodec libfaac -ar 48k
> -ab 224k -aq 100 -vol 384 -async 1 -y video.mp4
> ffmpeg version 2.2-tessus Copyright (c) 2000-2014 the FFmpeg developers
>   built on Mar 24 2014 22:25:11 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
>   configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm
> --extra-version=tessus --disable-shared --enable-static --disable-ffplay
> --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame
> --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid
> --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libxavs --enable-version3
> --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx
> --enable-libgsm --enable-libopus --enable-libmodplug --enable-fontconfig
> --enable-libfreetype --enable-libass --enable-libbluray --enable-filters
> --enable-runtime-cpudetect
>   libavutil      52. 66.100 / 52. 66.100
>   libavcodec     55. 52.102 / 55. 52.102
>   libavformat    55. 33.100 / 55. 33.100
>   libavdevice    55. 10.100 / 55. 10.100
>   libavfilter     4.  2.100 /  4.  2.100
>   libswscale      2.  5.102 /  2.  5.102
>   libswresample   0. 18.100 /  0. 18.100
>   libpostproc    52.  3.100 / 52.  3.100
> Input #0, image2, from 'images/i-29x97fps-1280x720-%05d.png':
>   Duration: 00:00:10.94, start: 0.000000, bitrate: N/A
>     Stream #0:0: Video: png, rgba, 1024x769 [SAR 2835:2835 DAR 1024:769],
> 29.97 fps, 29.97 tbr, 29.97 tbn, 29.97 tbc
> Codec AVOption ab (set bitrate (in bits/s)) specified for output file #0
> (video.mp4) 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.
> No pixel format specified, yuv444p for H.264 encoding chosen.
> Use -pix_fmt yuv420p for compatibility with outdated media players.

This wasn't clear enough? I'll take it out of the quoting to make it
even clearer.

No pixel format specified, yuv444p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.

Cheers

Tom

From egekhter at pixcel.com  Tue Apr  1 18:02:34 2014
From: egekhter at pixcel.com (Eugene Gekhter)
Date: Tue, 1 Apr 2014 11:02:34 -0500
Subject: [FFmpeg-user] Minimum configuration settings for ffprobe-only
In-Reply-To: <loom.20140401T150246-272@post.gmane.org>
References: <CAFrh=nfHpOYR73HG=dFBDBt8Of1A6z9ybw1oOjvVfNvZQ1mxSg@mail.gmail.com>
	<loom.20140331T103425-710@post.gmane.org>
	<CAFrh=ndskYO2L79=hGWo0MKAD_GfueF7fvF3nskVPBDWO+NUaQ@mail.gmail.com>
	<loom.20140331T172740-218@post.gmane.org>
	<CAFrh=nefE3oqMq4bh5U_14iD-Cb5=MC9aWPUYJ3tDfY8ESEVhg@mail.gmail.com>
	<5339B14C.5010206@gmail.com>
	<CAFrh=nc3Ay6+T1ShPb=X9TwnnXGZAVN2Gm=5q-EUuusrTbZFTg@mail.gmail.com>
	<loom.20140401T150246-272@post.gmane.org>
Message-ID: <CAFrh=ne7GOz0dgB710YELcLgJozJnXKVxmQhckxbiwKrjb5_mg@mail.gmail.com>

On Tue, Apr 1, 2014 at 8:04 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Eugene Gekhter <egekhter <at> pixcel.com> writes:
>
> > 98% of the time ffprobe would be dealing with normal
> > consumer-generated files. Your example above is the
> > exception, not the norm.
>
> My original suspicion was (and still is) that in 99%
> of all cases, users will upload media files and not
> files that FFmpeg cannot read.
>
> And please note that random data is typically recognized
> as mp3 or aac.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


I have a proof of concept.
I've been able to get two different error messages:
1. Unsupported codec
2. Invalid data found when processing input

Without digging deeper into the source code, are there other cases that
would result in different error messages?

From joao_m_s_m at hotmail.com  Tue Apr  1 18:19:14 2014
From: joao_m_s_m at hotmail.com (=?iso-8859-1?B?Sm/jbyBNYWNoYWRv?=)
Date: Tue, 1 Apr 2014 16:19:14 +0000
Subject: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
Message-ID: <DUB130-W17C4A888B6363BD773124EE8620@phx.gbl>

Hi a
 		 	   		  

From joao_m_s_m at hotmail.com  Tue Apr  1 18:28:57 2014
From: joao_m_s_m at hotmail.com (=?iso-8859-1?B?Sm/jbyBNYWNoYWRv?=)
Date: Tue, 1 Apr 2014 16:28:57 +0000
Subject: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
In-Reply-To: <DUB130-W17C4A888B6363BD773124EE8620@phx.gbl>
References: <DUB130-W17C4A888B6363BD773124EE8620@phx.gbl>
Message-ID: <DUB130-W80C9C295EE04FA99F54850E8620@phx.gbl>

Hi all, 

I am trying to make some modifications on ffplay.c using eclipse.

I have managed to successfully compile the ffmpeg using the following link:

https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide (thanks to whoever did it).

Now I am trying to compile the ffplay.c using the eclipse, so I can debud, change and compile it.

Does anyone knows how to do this?

I tried this link (http://www3.ntu.edu.sg/home/ehchua/programming/howto/eclipsecpp_howto.html) but I think I didn't compile it with MingW and MSYS. 

I also found this link which I'll try after https://trac.ffmpeg.org/wiki/Setup_Eclipse_IDE_for_FFmpeg_%28Linux%29.

Any suggestion?

Thanks in advance,
Jo?o Machado.
 		 	   		   		 	   		  

From peter at peterphi.com  Tue Apr  1 20:19:16 2014
From: peter at peterphi.com (Peter Wright)
Date: Tue, 1 Apr 2014 19:19:16 +0100
Subject: [FFmpeg-user] AAC in MOV 1 frame off
Message-ID: <etPan.533b0324.6b8b4567.11409@Peters-Retina-Mac.local>

Hi,

I am having a problem with AV sync being off by 1 frame when using AAC and MP3 (tried all 3 AAC encoders, internal, libfaac and libfdk_aac) in MOV.
I've tried with 2.2 and with git master (d506deaeaa98013505241d8149d82327efea0379)

I have used lavfi to generate SMPTE bars (with burned-in timecode) and a sine-wave with a beep every second. The beep happens on frame 00.

To generate the samples (10 second clips are generated):

ffmpeg -f lavfi -i "smptebars=size=1920x1080:rate=25" -f lavfi -i ?"sine=b=2:r=48000" -vf "drawtext=timecode='00\:00\:00\:00':r=25:x=(w-tw)/2:y=h-(2*lh):fontcolor=white:box=1:boxcolor=0x00000000 at 1:fontsize=100" -vcodec mpeg4 -t 10 -y -strict experimental -acodec aac aac.mov

ffmpeg -f lavfi -i "smptebars=size=1920x1080:rate=25" -f lavfi -i ?"sine=b=2:r=48000" -vf "drawtext=timecode='00\:00\:00\:00':r=25:x=(w-tw)/2:y=h-(2*lh):fontcolor=white:box=1:boxcolor=0x00000000 at 1:fontsize=100" -vcodec mpeg4 -t 10 -y -acodec libmp3lame mp3.mov

ffmpeg -f lavfi -i "smptebars=size=1920x1080:rate=25" -f lavfi -i ?"sine=b=2:r=48000" -vf "drawtext=timecode='00\:00\:00\:00':r=25:x=(w-tw)/2:y=h-(2*lh):fontcolor=white:box=1:boxcolor=0x00000000 at 1:fontsize=100" -vcodec mpeg4 -t 10 -y -acodec pcm_s16le pcm.mov

I'm playing back in Final Cut Pro X and Quicktime (latest version for Mavericks).

When I play back "aac.mov", the first beep is not audible and the next beep happens primarily on frame 00:00:00:24 (with a short pop in 00:00:00:23)
When I play back "mp3.mov", the first beep is audible and the next beep happens primarily on frame 00:00:01:00 (with a short pop on 00:00:00:24)
When I play back "pcm.mov", the first beep is audible and the next beep happens entirely on frame 00:00:01:00

I've noticed that for the MP3 and AAC sample there are packets generated with a negative PTS value, I'm not sure if this is part of the problem? See ffprobe results below

For aac.mov:
ffprobe -show_packets -of xml aac.mov
ffprobe version 2.2 Copyright (c) 2007-2014 the FFmpeg developers
? built on Mar 29 2014 02:29:54 with Apple LLVM version 5.1 (clang-503.0.38) (based on LLVM 3.4svn)
? configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-fontconfig --enable-libfreetype --disable-indev=jack --disable-outdev=xv --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --enable-vda --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-nonfree --enable-libfdk-aac --enable-libfaac
? libavutil ? ? ?52. 66.100 / 52. 66.100
? libavcodec ? ? 55. 52.102 / 55. 52.102
? libavformat ? ?55. 33.100 / 55. 33.100
? libavdevice ? ?55. 10.100 / 55. 10.100
? libavfilter ? ? 4. ?2.100 / ?4. ?2.100
? libavresample ? 1. ?2. ?0 / ?1. ?2. ?0
? libswscale ? ? ?2. ?5.102 / ?2. ?5.102
? libswresample ? 0. 18.100 / ?0. 18.100
? libpostproc ? ?52. ?3.100 / 52. ?3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'aac.mov':
? Metadata:
? ? major_brand ? ? : qt
? ? minor_version ? : 512
? ? compatible_brands: qt
? ? encoder ? ? ? ? : Lavf55.33.100
? Duration: 00:00:10.02, start: 0.021333, bitrate: 2018 kb/s
? ? Stream #0:0(eng): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 1953 kb/s, 25 fps, 25 tbr, 12800 tbn, 25 tbc (default)
? ? Metadata:
? ? ? handler_name ? ?: DataHandler
? ? Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 63 kb/s (default)
? ? Metadata:
? ? ? handler_name ? ?: DataHandler
<packet codec_type="audio" stream_index="1" pts="-1024" pts_time="-0.021333" dts="-1024" dts_time="-0.021333" duration="1024" duration_time="0.021333" size="325" pos="36" flags="K"/>
<packet codec_type="video" stream_index="0" pts="0" pts_time="0.000000" dts="0" dts_time="0.000000" duration="512" duration_time="0.040000" size="42424" pos="361" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="0" pts_time="0.000000" dts="0" dts_time="0.000000" duration="1024" duration_time="0.021333" size="344" pos="42785" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="1024" pts_time="0.021333" dts="1024" dts_time="0.021333" duration="1024" duration_time="0.021333" size="381" pos="43129" flags="K"/>
<packet codec_type="video" stream_index="0" pts="512" pts_time="0.040000" dts="512" dts_time="0.040000" duration="512" duration_time="0.040000" size="24350" pos="43510" flags="_"/>
<packet codec_type="audio" stream_index="1" pts="2048" pts_time="0.042667" dts="2048" dts_time="0.042667" duration="1024" duration_time="0.021333" size="277" pos="67860" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="3072" pts_time="0.064000" dts="3072" dts_time="0.064000" duration="1024" duration_time="0.021333" size="131" pos="68137" flags="K"/>
...


For mp3.mov:
ffprobe -show_packets -of xml mp3.mov
ffprobe version 2.2 Copyright (c) 2007-2014 the FFmpeg developers
? built on Mar 29 2014 02:29:54 with Apple LLVM version 5.1 (clang-503.0.38) (based on LLVM 3.4svn)
? configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-fontconfig --enable-libfreetype --disable-indev=jack --disable-outdev=xv --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --enable-vda --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-nonfree --enable-libfdk-aac --enable-libfaac
? libavutil ? ? ?52. 66.100 / 52. 66.100
? libavcodec ? ? 55. 52.102 / 55. 52.102
? libavformat ? ?55. 33.100 / 55. 33.100
? libavdevice ? ?55. 10.100 / 55. 10.100
? libavfilter ? ? 4. ?2.100 / ?4. ?2.100
? libavresample ? 1. ?2. ?0 / ?1. ?2. ?0
? libswscale ? ? ?2. ?5.102 / ?2. ?5.102
? libswresample ? 0. 18.100 / ?0. 18.100
? libpostproc ? ?52. ?3.100 / 52. ?3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'mp3.mov':
? Metadata:
? ? major_brand ? ? : qt
? ? minor_version ? : 512
? ? compatible_brands: qt
? ? encoder ? ? ? ? : Lavf55.33.100
? Duration: 00:00:10.02, start: 0.023021, bitrate: 2018 kb/s
? ? Stream #0:0(eng): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 1953 kb/s, 25 fps, 25 tbr, 12800 tbn, 25 tbc (default)
? ? Metadata:
? ? ? handler_name ? ?: DataHandler
? ? Stream #0:1(eng): Audio: mp3 (.mp3 / 0x33706D2E), 48000 Hz, mono, s16p, 64 kb/s (default)
? ? Metadata:
? ? ? handler_name ? ?: DataHandler
<packet codec_type="audio" stream_index="1" pts="-1105" pts_time="-0.023021" dts="-1105" dts_time="-0.023021" duration="1152" duration_time="0.024000" size="192" pos="36" flags="K"/>
<packet codec_type="video" stream_index="0" pts="0" pts_time="0.000000" dts="0" dts_time="0.000000" duration="512" duration_time="0.040000" size="42424" pos="228" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="47" pts_time="0.000979" dts="47" dts_time="0.000979" duration="1152" duration_time="0.024000" size="192" pos="42652" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="1199" pts_time="0.024979" dts="1199" dts_time="0.024979" duration="1152" duration_time="0.024000" size="192" pos="42844" flags="K"/>
<packet codec_type="video" stream_index="0" pts="512" pts_time="0.040000" dts="512" dts_time="0.040000" duration="512" duration_time="0.040000" size="24350" pos="43036" flags="_"/>
<packet codec_type="audio" stream_index="1" pts="2351" pts_time="0.048979" dts="2351" dts_time="0.048979" duration="1152" duration_time="0.024000" size="192" pos="67386" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="3503" pts_time="0.072979" dts="3503" dts_time="0.072979" duration="1152" duration_time="0.024000" size="192" pos="67578" flags="K"/>
...


for pcm.mov:
ffprobe -show_packets -of xml pcm.mov
ffprobe version 2.2 Copyright (c) 2007-2014 the FFmpeg developers
? built on Mar 29 2014 02:29:54 with Apple LLVM version 5.1 (clang-503.0.38) (based on LLVM 3.4svn)
? configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-fontconfig --enable-libfreetype --disable-indev=jack --disable-outdev=xv --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --enable-vda --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-nonfree --enable-libfdk-aac --enable-libfaac
? libavutil ? ? ?52. 66.100 / 52. 66.100
? libavcodec ? ? 55. 52.102 / 55. 52.102
? libavformat ? ?55. 33.100 / 55. 33.100
? libavdevice ? ?55. 10.100 / 55. 10.100
? libavfilter ? ? 4. ?2.100 / ?4. ?2.100
? libavresample ? 1. ?2. ?0 / ?1. ?2. ?0
? libswscale ? ? ?2. ?5.102 / ?2. ?5.102
? libswresample ? 0. 18.100 / ?0. 18.100
? libpostproc ? ?52. ?3.100 / 52. ?3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pcm.mov':
? Metadata:
? ? major_brand ? ? : qt
? ? minor_version ? : 512
? ? compatible_brands: qt
? ? encoder ? ? ? ? : Lavf55.33.100
? Duration: 00:00:10.00, start: 0.000000, bitrate: 2725 kb/s
? ? Stream #0:0(eng): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 1953 kb/s, 25 fps, 25 tbr, 12800 tbn, 25 tbc (default)
? ? Metadata:
? ? ? handler_name ? ?: DataHandler
? ? Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, s16, 768 kb/s (default)
? ? Metadata:
? ? ? handler_name ? ?: DataHandler
<packet codec_type="video" stream_index="0" pts="0" pts_time="0.000000" dts="0" dts_time="0.000000" duration="512" duration_time="0.040000" size="42424" pos="36" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="0" pts_time="0.000000" dts="0" dts_time="0.000000" duration="1024" duration_time="0.021333" size="2048" pos="42460" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="1024" pts_time="0.021333" dts="1024" dts_time="0.021333" duration="1024" duration_time="0.021333" size="2048" pos="44508" flags="K"/>
<packet codec_type="video" stream_index="0" pts="512" pts_time="0.040000" dts="512" dts_time="0.040000" duration="512" duration_time="0.040000" size="24350" pos="46556" flags="_"/>
<packet codec_type="audio" stream_index="1" pts="2048" pts_time="0.042667" dts="2048" dts_time="0.042667" duration="1024" duration_time="0.021333" size="2048" pos="70906" flags="K"/>
<packet codec_type="audio" stream_index="1" pts="3072" pts_time="0.064000" dts="3072" dts_time="0.064000" duration="1024" duration_time="0.021333" size="2048" pos="72954" flags="K"/>
<packet codec_type="video" stream_index="0" pts="1024" pts_time="0.080000" dts="1024" dts_time="0.080000" duration="512" duration_time="0.040000" size="4281" pos="75002" flags="_"/>
...


Any ideas? I?ve tried -copyts and -vsync to no avail


From barsnick at gmx.net  Tue Apr  1 22:16:48 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Tue, 1 Apr 2014 22:16:48 +0200
Subject: [FFmpeg-user] How to stitch images to create video
In-Reply-To: <CAFHbX1JxscmrzGaBOBjBt0wizrULoxLhHbWOBZf=+nf-znzqWA@mail.gmail.com>
References: <CAPMi8WYEe3Kd2OP88rAZ4WaAM6RkeibBFojzgp1_o8-oQ=V7og@mail.gmail.com>
	<CAFHbX1JxscmrzGaBOBjBt0wizrULoxLhHbWOBZf=+nf-znzqWA@mail.gmail.com>
Message-ID: <20140401201648.GC14296@sunshine.barsnick.net>

On Tue, Apr 01, 2014 at 16:29:37 +0100, Tom Evans wrote:
> > Use -pix_fmt yuv420p for compatibility with outdated media players.
> 
> This wasn't clear enough? I'll take it out of the quoting to make it
> even clearer.

I would have spontaneously answered the same, but does VLC count as an
outdated media player? That said, thinking about it: since it crashed,
it might be that VLC _could have_ supported this video, but indeed had
a different problem.

Moritz

From barsnick at gmx.net  Tue Apr  1 22:25:02 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Tue, 1 Apr 2014 22:25:02 +0200
Subject: [FFmpeg-user] Problem Compiling ffmpeg
In-Reply-To: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
Message-ID: <20140401202502.GD14296@sunshine.barsnick.net>

On Tue, Apr 01, 2014 at 15:25:21 +0000, Alexander.Achleitner at zf.com wrote:
> i want to compile the ffmpeg sources for an ARM  TI AM3359 with linux OS. I tried some
> config arguments, but it doesn't work.

Small hint: Did you read the line of output which says:

> Include the log file "config.log" produced by configure as this will
> help solving the problem.

??

Please do include that file.

Thanks,
Moritz

From cehoyos at ag.or.at  Tue Apr  1 22:56:52 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 20:56:52 +0000 (UTC)
Subject: [FFmpeg-user] AAC in MOV 1 frame off
References: <etPan.533b0324.6b8b4567.11409@Peters-Retina-Mac.local>
Message-ID: <loom.20140401T225539-103@post.gmane.org>

Peter Wright <peter <at> peterphi.com> writes:

> I am having a problem with AV sync being off by 1 
> frame when using AAC and MP3

Iirc, this was reported several times, but I 
believe that neither myself nor another FFmpeg 
developer understood the issue:
How can you know that AV-sync is off by 0.04 seconds?

Carl Eugen


From cehoyos at ag.or.at  Tue Apr  1 22:58:30 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 20:58:30 +0000 (UTC)
Subject: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
References: <DUB130-W17C4A888B6363BD773124EE8620@phx.gbl>
	<DUB130-W80C9C295EE04FA99F54850E8620@phx.gbl>
Message-ID: <loom.20140401T225744-805@post.gmane.org>

Jo?o Machado <joao_m_s_m <at> hotmail.com> writes:

> Now I am trying to compile the ffplay.c using the eclipse, 
> so I can debud, change and compile it.

> I also found this link which I'll try after 
> https://trac.ffmpeg.org/wiki/Setup_Eclipse_IDE_for_FFmpeg_%28Linux%29.

I suggest you ask here if anything does not work as 
explained in the wiki.

Carl Eugen


From naveen.mr at gmail.com  Tue Apr  1 23:10:30 2014
From: naveen.mr at gmail.com (Naveen Rudrappa)
Date: Tue, 1 Apr 2014 22:10:30 +0100
Subject: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
In-Reply-To: <loom.20140401T225744-805@post.gmane.org>
References: <DUB130-W17C4A888B6363BD773124EE8620@phx.gbl>
	<DUB130-W80C9C295EE04FA99F54850E8620@phx.gbl>
	<loom.20140401T225744-805@post.gmane.org>
Message-ID: <CAMCjN1-zB=2+rqXOHKP9KJDC1RP_8Ezt3Vxan_OGvXB=uBwxZw@mail.gmail.com>

Joao,
if there is no way out - suggest you to look at the command line compiler
and linker options for ffplay.c
use verbose method to see these options:

make V=s

It should give fair bit of info required to populate "project -> properties
-> c/c++ build -> settings" in eclipse
I did manage to compile this way sometime ago




On Tue, Apr 1, 2014 at 9:58 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Jo?o Machado <joao_m_s_m <at> hotmail.com> writes:
>
> > Now I am trying to compile the ffplay.c using the eclipse,
> > so I can debud, change and compile it.
>
> > I also found this link which I'll try after
> > https://trac.ffmpeg.org/wiki/Setup_Eclipse_IDE_for_FFmpeg_%28Linux%29.
>
> I suggest you ask here if anything does not work as
> explained in the wiki.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>



-- 
,
Naveen Rudrappa

From cehoyos at ag.or.at  Tue Apr  1 23:27:14 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 21:27:14 +0000 (UTC)
Subject: [FFmpeg-user] Problem Compiling ffmpeg
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
Message-ID: <loom.20140401T232247-362@post.gmane.org>

 <Alexander.Achleitner <at> zf.com> writes:

> One Error message is
> "...
> AR   libavdevice/libavdevice.a
> Sorry  -  this program has been built wihout plugin support "

For build problems, please run "make V=1" twice and 
post the complete, uncut output of the second run here.
But see below first.

> Configuration was,
> ./configure -enable-cross-compile 

(Your configure line looks as if it would not work, 
are you sure that you posted it correctly?)

> -cc=arm-cortexa8-linux-gnueabihf-gcc 

> --cxx=arm-cortexa8-linux-gnueabihf-gcc

Is this needed?

> --as=arm-cortexa8-linux-gnueabihf-gcc 

This should not be needed.

> --ar=arm-cortexa8-linux-gnueabihf-gcc 

This looks like the cause for your problem above.

> --arch=armel 

"arm" should be enough.

> --target-os=linux

Isn't --cross-prefix what you are looking for?

Carl Eugen


From cehoyos at ag.or.at  Tue Apr  1 23:34:42 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 21:34:42 +0000 (UTC)
Subject: [FFmpeg-user] How to stitch images to create video
References: <CAPMi8WYEe3Kd2OP88rAZ4WaAM6RkeibBFojzgp1_o8-oQ=V7og@mail.gmail.com>
Message-ID: <loom.20140401T233229-702@post.gmane.org>

Amit Agarwal <lifeamit <at> gmail.com> writes:

> This command it creating video.mp4 but that seems to 
> be corrupt file.

Does the output file play with current ffplay or current 
MPlayer?

> No player is able to play that file and VLC crashes 
> when trying to play this file.

Which version of vlc did you test?

Please understand that if FFmpeg writes invalid files 
that cannot be played with any player, this would be 
an important issue that we'd like to know about.
If old versions of media players do not like files 
produced with current FFmpeg, I don't think there 
is anything we can do.

Carl Eugen


From cehoyos at ag.or.at  Tue Apr  1 23:32:05 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 21:32:05 +0000 (UTC)
Subject: [FFmpeg-user] Mixed Telecine & Interlaced
References: <1396298646.52178.YahooMailNeo@web160804.mail.bf1.yahoo.com>	<1396302123.84118.YahooMailNeo@web160803.mail.bf1.yahoo.com>	<loom.20140401T134544-689@post.gmane.org>	<1396362092.74307.YahooMailNeo@web160806.mail.bf1.yahoo.com>
	<loom.20140401T163058-100@post.gmane.org>
	<1396363387.70498.YahooMailNeo@web160806.mail.bf1.yahoo.com>
Message-ID: <loom.20140401T232943-558@post.gmane.org>

Nicholas Robbins <nickrobbins <at> yahoo.com> writes:

> What filter chain are you suggesting?

I simply wanted to repeat that idet (although it does to 
some degree work as expected contrary to what I claimed 
on this mailing list before) does not work perfectly for 
the changes between interlaced and telecined / 
progressive and therefore should probably be improved.

Carl Eugen


From cehoyos at ag.or.at  Tue Apr  1 23:36:24 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 1 Apr 2014 21:36:24 +0000 (UTC)
Subject: [FFmpeg-user] -f segment does not seem to be working
References: <5339DE02.9060401@redmule.com> <20140331133753.7fae104a@lrcd.com>
	<5339E71A.4080305@redmule.com>
Message-ID: <loom.20140401T233526-17@post.gmane.org>

Michael Hipp <michael <at> redmule.com> writes:

> Thank you. I'm afraid the version numbers are a bit 
> confusing.

See here for more information on why some people want 
you to believe you originally tested FFmpeg:
http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html

Carl Eugen


From michael at redmule.com  Wed Apr  2 00:03:57 2014
From: michael at redmule.com (Michael Hipp)
Date: Tue, 01 Apr 2014 17:03:57 -0500
Subject: [FFmpeg-user] -f segment does not seem to be working
In-Reply-To: <loom.20140401T233526-17@post.gmane.org>
References: <5339DE02.9060401@redmule.com>
	<20140331133753.7fae104a@lrcd.com>	<5339E71A.4080305@redmule.com>
	<loom.20140401T233526-17@post.gmane.org>
Message-ID: <533B37CD.3070204@redmule.com>

On 4/1/2014 4:36 PM, Carl Eugen Hoyos wrote:
> Michael Hipp <michael <at> redmule.com> writes:
>
>> Thank you. I'm afraid the version numbers are a bit
>> confusing.
> See here for more information on why some people want
> you to believe you originally tested FFmpeg:
> http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html
>
> Carl Eugen
>

Thanks. A sad read. A fork is one thing, but it turns out Ubuntu ships with 
ffmpeg that isn't ffmpeg.

Anyways, my stuff is working fine now.

Thanks,
Michael

From peter at peterphi.com  Wed Apr  2 01:45:44 2014
From: peter at peterphi.com (Peter Wright)
Date: Wed, 2 Apr 2014 00:45:44 +0100
Subject: [FFmpeg-user] AAC in MOV 1 frame off
In-Reply-To: <loom.20140401T225539-103@post.gmane.org>
References: <etPan.533b0324.6b8b4567.11409@Peters-Retina-Mac.local>
	<loom.20140401T225539-103@post.gmane.org>
Message-ID: <etPan.533b4fa9.2ae8944a.4829@Peters-Retina-Mac.local>

On 1 April 2014 at 21:57:20, Carl Eugen Hoyos (cehoyos at ag.or.at) wrote:
Peter Wright <peter <at> peterphi.com> writes:?

> I am having a problem with AV sync being off by 1?
> frame when using AAC and MP3?

Iirc, this was reported several times, but I?
believe that neither myself nor another FFmpeg?
developer understood the issue:?
How can you know that AV-sync is off by 0.04 seconds??

Carl Eugen?
Hi Carl,

Firstly, I?m assuming that quicktime / final cut pro do the correct thing (I?m trying to write files that will be used by end-users in FCP and quicktime so from my perspective I want the files to work in quicktime) and using them to play back a frame of video and its associated audio one at a time. However, there does seem to be something odd going on anyway (with the first audio packet having a PTS value that puts it before the first video frame - see below)

The sample I?m generating (see previous e-mail for command) using lavfilter is emitting 48k samples/second (sine wave with a beep once a second). This should mean that the first 1920 audio samples in the stream are a ?beep?, with the next?46080 being a constant tone. When I play it back with ffplay the first thing I can hear is the beep, then the tone. When I play back in Quicktime the initial beep is not audible. This problem also manifests for an audio-only MOV file.

On playback the tone isn?t off by precisely 0.04 seconds, but it?s roughly that: when I play only frame 00:00:00:23 I can hear a small click which I think is part of the 1920 sample beep (which is clearly audible when 00:00:00:24 is played). I imported the waveform from the .MOV into Audacity - the beep starts at sample 45,888 which is 2112 samples too early (it?s also 1920+192 - I?m not sure if 192 appearing is a coincidence)

When I looked at the file with ffprobe -show_packets I see that the first audio packet has a PTS Time of?-0.021333 (and a duration time of 0.021333), whereas the first video packet has a PTS Time of 0.0000. So if I understand things correctly that entire audio packet should be played back before the first frame of video is displayed (and while 1024 audio samples doesn?t completely explain why the sample is ~1920+192 samples early it does make me think that?s something odd is happening)

So, to recap: the problem goes away when emitting PCM (and both codecs which exhibit this problem, AAC and MP3, have a negative PTS on the first packet)

I?d like to help figure out what?s going on here if I can, do you have any pointers for where I should look to get you more detail? I have some familiarity with the code but not a great deal with the specifics of the encoders and the mov muxer (one experiment I?d like to try is to find out who?s assigning that negative PTS value and see what happens if it?s changed to zero instead)





From francesco at bltitalia.com  Tue Apr  1 16:35:49 2014
From: francesco at bltitalia.com (francesco at bltitalia.com)
Date: Tue, 1 Apr 2014 16:35:49 +0200 (added by postmaster@tin.it)
Subject: [FFmpeg-user] Decoding XDCAM 4:2:2 using libavcodec-5.5 : error
 0xbebbb1b7 (AVERROR_INVALIDDATA)
Message-ID: <522D2C8B0AEB75D7@vsmtp2.tin.it> (added by postmaster@tin.it)

I post this message on libav list too, but no one response.
Hi to all 
I have written a code for decoding IMX-D10 at 30, 40 and 50 M and works
fine.  Attempting to decode XDCAM 4:2:2 at 1920x1080 it seems that only
first frame (I-Frame) is decoded.  After this I receive the error
0xbebbbb1b7 from avcodec_decode_video2 routine.  Here is the code:

 pCodec = NULL;   pFrame = NULL;  pCodecCtx = NULL;
  av_init_packet(&packet);

  pCodec=avcodec_find_decoder(CODEC_ID_MPEG2VIDEO);
  if( pCodec==NULL)
  {   str1.printf("Error at frame %ld: Unable to find CODEC_ID_MPEG2VIDEO
exited",frmcnt,err);smsmemo->Memo1->Lines->Add(str1);
     errDS10 = -3; goto end_convDS10;
   }
  pCodecCtx = avcodec_alloc_context3(pCodec);
  // Allocate video frame
  pFrame=avcodec_alloc_frame();
 // Open codec
  if(avcodec_open2(pCodecCtx, pCodec,NULL)<0) { 
smsmemo->Memo1->Lines->Add("Unable to open codec!"); errDS10 = -5; goto
end_convDS10;}

  packet.size = bfsz; // This is the size passed to the routine
  packet.data = fbuf; // This is the buffer passed to the routine

  frameFinished = 1;    err = 0;
  do
  {
  // Decode video frame
   err = avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished,&packet);
   if(err<0)
   {  str1.printf("Error (%lx) at frame %ld returned from
avcodec_decode_video2, exited",err,frmcnt); smsmemo->Memo1->Lines->Add(str1);
     errDS10 = -7; goto end_convDS10;
   }
   Application->ProcessMessages();

  }
  while ((!frameFinished)&&(packet.size>0))  ;

The first frame is decoded correctly; is an I-Frame and these are the first
bytes of header:

00 00 01 B3 78 04 38 33 7A 12 22 02 10 20 20 26
20 26 2C 2C 2C 2C 2C 2C 34 30 34 36 36 36 34 34
34 34 36 36 36 3A 3A 3A 44 44 44 3A 3A 3A 36 36
3A 3A 40 40 44 44 4A 4C 4A 46 46 44 46 4C 4C 50
50 50 60 60 5C 5C 70 70 74 8A 8A A7 10 11 11 12
12 12 13 13 13 13 14 14 14 14 14 15 15 15 15 15
15 16 16 16 16 16 16 16 17 17 17 17 17 17 17 17
18 18 18 19 18 18 18 19 1A 1A 1A 1A 19 1B 1B 1B
1B 1B 1C 1C 1C 1C 1E 1E 1E 1F 1F 21 00 00 01 B5
18 24 00 01 01 00 00 00 01 B5 21 01 01 01 1E 02
21 C0 00 00 01 B8 00 08 00 40 00 00 01 00 00 8B
22 10 00 00 00 01 B5 ...

as you can see from first pachet header (00 00 00 01 00) is an I-Frame (byte
5 is 0x8B that is 10001011 and insulating bits 3,4 and 5 we have 1 =>
I-Frame). Second packet is an B frame, here is header:

00 00 01 00 00 1A D7 83 B8 00 00 01 B5 84 45 4B
9C 00 00 00 01 B5 38 40 80 80 98 80 98 B0 B0 B0
B0 B0 B0 D0 C0 D0 D8 D8 D8 D0 D0 D0 D0 D8 D8 D8
E8 E8 E9 11 11 10 E8 E8 E8 D8 D8 E8 E9 01 01 11
11 29 31 29 19 19 11 19 31

we have 0x1a that is 00011010 and now bits 3,4and 5 are 011 that identify a
B-Frame.  I receive error till next I frame.  There is something I'm missing
or it's a bug (or limit ) of this version of libavcodec ?
Regards


From praks411 at gmail.com  Tue Apr  1 10:22:15 2014
From: praks411 at gmail.com (praks411)
Date: Tue, 1 Apr 2014 01:22:15 -0700 (PDT)
Subject: [FFmpeg-user] Memory leaks in ffmpeg 2.2
In-Reply-To: <loom.20140401T092800-856@post.gmane.org>
References: <1395920879338-4664505.post@n4.nabble.com>
	<1396009615715-4664518.post@n4.nabble.com>
	<CAMCjN1-L2bi0Lj6vP59iupqU5uZNCVronuXoSa7-f=aV0kSO2g@mail.gmail.com>
	<loom.20140401T092800-856@post.gmane.org>
Message-ID: <1396340535174-4664575.post@n4.nabble.com>

Thanks for the suggestions. Currently my whole setup is on windows so it is
difficult to get valgrind running from me. The lock condition which you have
mentioned is true, as I was able to verify it by explicitly releasing the
lock, the memory leaks were gone but there were some other issues, the code
was crashing possibly due to another race conditions.

Pradeep



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Memory-leaks-in-ffmpeg-2-2-tp4664505p4664575.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From lifeamit at gmail.com  Wed Apr  2 06:05:23 2014
From: lifeamit at gmail.com (Amit Agarwal)
Date: Wed, 2 Apr 2014 09:35:23 +0530
Subject: [FFmpeg-user] How to stitch images to create video
In-Reply-To: <loom.20140401T233229-702@post.gmane.org>
References: <CAPMi8WYEe3Kd2OP88rAZ4WaAM6RkeibBFojzgp1_o8-oQ=V7og@mail.gmail.com>
	<loom.20140401T233229-702@post.gmane.org>
Message-ID: <CAPMi8WbAnCWQYa106x9i7AEfEWD2dDct7uvwg2Ykxa99tbitKg@mail.gmail.com>

I am using VLC version 2.0.7 on Mac 10.9.

Yes, using '-pix_fmt yuv420p' helped solved the problem.

I did not try to play corrupt file with ffplay or mplayer.


On Wed, Apr 2, 2014 at 3:04 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> Amit Agarwal <lifeamit <at> gmail.com> writes:
>
> > This command it creating video.mp4 but that seems to
> > be corrupt file.
>
> Does the output file play with current ffplay or current
> MPlayer?
>
> > No player is able to play that file and VLC crashes
> > when trying to play this file.
>
> Which version of vlc did you test?
>
> Please understand that if FFmpeg writes invalid files
> that cannot be played with any player, this would be
> an important issue that we'd like to know about.
> If old versions of media players do not like files
> produced with current FFmpeg, I don't think there
> is anything we can do.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From porparek at gmail.com  Wed Apr  2 09:16:43 2014
From: porparek at gmail.com (Pawel Por)
Date: Wed, 2 Apr 2014 09:16:43 +0200
Subject: [FFmpeg-user] cut at keyframes
In-Reply-To: <CAJmOuEyk=vbSaJnS8KCxNG5XP3c=pmA1QmRrBRVuRYNcEZbxjQ@mail.gmail.com>
References: <CAJmOuEyk=vbSaJnS8KCxNG5XP3c=pmA1QmRrBRVuRYNcEZbxjQ@mail.gmail.com>
Message-ID: <CAJmOuEx=CW-P7ut1JYHy-BKN_xW666q_U7p=2+NxQo2-dchfTw@mail.gmail.com>

Thaks for reply. I've uploaded test.mp4 (10MB) file into the following
location:
http://www.sendspace.pl/en/file/3e79185144b8a84429197cd


On Tue, Apr 1, 2014 at 11:28 AM, Pawel Por <porparek at gmail.com> wrote:

> Hi
> I'm a great fan of ffmpeg. I've been using the version 1.0.7 of ffmpeg and
> it was cutting my mp4 movies at closest keyframes.
> It was exactly what I wanted.
> It was very useful to me. I was using the following command:
> ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 -acodec copy -vcodec copy
> -y out.mp4
>
> I've upgraded to the newer version of ffmpeg 2.2 and I found that now when
> I use the above command it cuts my movies not at closest keyframes.
> I tried "-noaccurate_seek" option (as follows) but it still cuts my movies
> not at a closest keyframes.
> ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 -acodec copy -vcodec copy
> -y out.mp4
>
> I believe that there must be some ffmpeg option to accomplish it (I cannot
> find it in "man page").
>
> Please, write me how I can do that.
>
> thanks in advance
>
>

From cus at passwd.hu  Wed Apr  2 09:54:08 2014
From: cus at passwd.hu (Marton Balint)
Date: Wed, 2 Apr 2014 09:54:08 +0200 (CEST)
Subject: [FFmpeg-user] Decoding XDCAM 4:2:2 using libavcodec-5.5 : error
 0xbebbb1b7 (AVERROR_INVALIDDATA)
In-Reply-To: <522D2C8B0AEB75D7@vsmtp2.tin.it> (added by postmaster@tin.it)
References: <522D2C8B0AEB75D7@vsmtp2.tin.it> (added by postmaster@tin.it)
Message-ID: <alpine.LNX.2.00.1404020951500.6616@iq.passwd.hu>


On Tue, 1 Apr 2014, francesco at bltitalia.com wrote:

> I post this message on libav list too, but no one response.
> Hi to all
> I have written a code for decoding IMX-D10 at 30, 40 and 50 M and works
> fine.  Attempting to decode XDCAM 4:2:2 at 1920x1080 it seems that only
> first frame (I-Frame) is decoded.  After this I receive the error
> 0xbebbbb1b7 from avcodec_decode_video2 routine.  Here is the code:
>
> pCodec = NULL;   pFrame = NULL;  pCodecCtx = NULL;
>  av_init_packet(&packet);
>
>  pCodec=avcodec_find_decoder(CODEC_ID_MPEG2VIDEO);
>  if( pCodec==NULL)
>  {   str1.printf("Error at frame %ld: Unable to find CODEC_ID_MPEG2VIDEO
> exited",frmcnt,err);smsmemo->Memo1->Lines->Add(str1);
>     errDS10 = -3; goto end_convDS10;
>   }
>  pCodecCtx = avcodec_alloc_context3(pCodec);
>  // Allocate video frame
>  pFrame=avcodec_alloc_frame();
> // Open codec
>  if(avcodec_open2(pCodecCtx, pCodec,NULL)<0) {
> smsmemo->Memo1->Lines->Add("Unable to open codec!"); errDS10 = -5; goto
> end_convDS10;}
>
>  packet.size = bfsz; // This is the size passed to the routine
>  packet.data = fbuf; // This is the buffer passed to the routine
>
>  frameFinished = 1;    err = 0;
>  do
>  {
>  // Decode video frame
>   err = avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished,&packet);
>   if(err<0)
>   {  str1.printf("Error (%lx) at frame %ld returned from
> avcodec_decode_video2, exited",err,frmcnt); smsmemo->Memo1->Lines->Add(str1);
>     errDS10 = -7; goto end_convDS10;
>   }
>   Application->ProcessMessages();
>
>  }
>  while ((!frameFinished)&&(packet.size>0))  ;
>
> The first frame is decoded correctly; is an I-Frame and these are the first
> bytes of header:
>
> 00 00 01 B3 78 04 38 33 7A 12 22 02 10 20 20 26
> 20 26 2C 2C 2C 2C 2C 2C 34 30 34 36 36 36 34 34
> 34 34 36 36 36 3A 3A 3A 44 44 44 3A 3A 3A 36 36
> 3A 3A 40 40 44 44 4A 4C 4A 46 46 44 46 4C 4C 50
> 50 50 60 60 5C 5C 70 70 74 8A 8A A7 10 11 11 12
> 12 12 13 13 13 13 14 14 14 14 14 15 15 15 15 15
> 15 16 16 16 16 16 16 16 17 17 17 17 17 17 17 17
> 18 18 18 19 18 18 18 19 1A 1A 1A 1A 19 1B 1B 1B
> 1B 1B 1C 1C 1C 1C 1E 1E 1E 1F 1F 21 00 00 01 B5
> 18 24 00 01 01 00 00 00 01 B5 21 01 01 01 1E 02
> 21 C0 00 00 01 B8 00 08 00 40 00 00 01 00 00 8B
> 22 10 00 00 00 01 B5 ...
>
> as you can see from first pachet header (00 00 00 01 00) is an I-Frame (byte
> 5 is 0x8B that is 10001011 and insulating bits 3,4 and 5 we have 1 =>
> I-Frame). Second packet is an B frame, here is header:
>
> 00 00 01 00 00 1A D7 83 B8 00 00 01 B5 84 45 4B
> 9C 00 00 00 01 B5 38 40 80 80 98 80 98 B0 B0 B0
> B0 B0 B0 D0 C0 D0 D8 D8 D8 D0 D0 D0 D0 D8 D8 D8
> E8 E8 E9 11 11 10 E8 E8 E8 D8 D8 E8 E9 01 01 11
> 11 29 31 29 19 19 11 19 31
>
> we have 0x1a that is 00011010 and now bits 3,4and 5 are 011 that identify a
> B-Frame.  I receive error till next I frame.  There is something I'm missing
> or it's a bug (or limit ) of this version of libavcodec ?

Have you tried it with ffmpeg git head? Does ffplay play all frames? If 
you can, please upload a sample somewhere.

Regards,
Marton

From elliottbalsley at gmail.com  Wed Apr  2 10:03:30 2014
From: elliottbalsley at gmail.com (Elliott Balsley)
Date: Wed, 2 Apr 2014 01:03:30 -0700
Subject: [FFmpeg-user] frame rate change is too slow
Message-ID: <CADzzJYmv8aaAejQX4WKP5sHf+QdK0j4Rni1vMgMRQBGfgKiVSg@mail.gmail.com>

I have a video shot at 96fps with a 23.98 timebase (so it's in slow
motion).  I want to make it play at normal speed, by dropping 3 out of
every 4 frames.  I would expect this to encode 4x faster, but it
actually takes about twice as long as a normal encode.  I'm using the
-r flag twice; is there a better way?

$ ffmpeg -ss 60 -t 10 -i
/Volumes/RAID/GILT/Raw_Footage/Day07/A045R1NA/A045C001_140402_R1NA.mov
-pix_fmt yuv420p -r 24000/1001 test.mov
ffmpeg version 2.1.git-ffb7d71 Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar  7 2014 16:54:41 with Apple LLVM version 5.0
(clang-500.2.79) (based on LLVM 3.3svn)
  configuration: --prefix=/opt/local --enable-swscale
--enable-avfilter --enable-avresample --enable-libmp3lame
--enable-libvorbis --enable-libopus --enable-libtheora
--enable-libschroedinger --enable-libopenjpeg --enable-libmodplug
--enable-libvpx --enable-libspeex --enable-libass --enable-libbluray
--enable-gnutls --enable-fontconfig --enable-libfreetype
--disable-indev=jack --disable-outdev=xv --mandir=/opt/local/share/man
--enable-shared --enable-pthreads --cc=/usr/bin/cc --arch=x86_64
--enable-yasm --enable-gpl --enable-postproc --enable-libx264
--enable-libxvid --enable-nonfree --enable-libfdk-aac --enable-libfaac
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libavresample   1.  2.  0 /  1.  2.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Volumes/RAID/GILT/Raw_Footage/Day07/A045R1NA/A045C001_140402_R1NA.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt  ARRI
    creation_time   : 2014-04-02 04:27:44
    timecode        : 00:01:33:13
  Duration: 00:06:13.71, start: 0.000000, bitrate: 190892 kb/s
    Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le,
1920x1080, 183719 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 23976
tbn, 23976 tbc (default)
    Metadata:
      creation_time   : 2014-04-02 04:27:44
      handler_name    : Apple Alias Data Handler
    Stream #0:1(eng): Data: none (tmcd / 0x64636D74) (default)
    Metadata:
      creation_time   : 2014-04-02 04:27:44
      handler_name    : Apple Alias Data Handler
      timecode        : 00:01:33:13
File 'test.mov' already exists. Overwrite ? [y/N] y
[libx264 @ 0x7fb913852c00] using SAR=1/1
[libx264 @ 0x7fb913852c00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x7fb913852c00] profile High, level 4.0
[libx264 @ 0x7fb913852c00] 264 - core 142 - H.264/MPEG-4 AVC codec -
Copyleft 2003-2014 - 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=36 lookahead_threads=6 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=23 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, mov, to 'test.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt  ARRI
    timecode        : 00:01:33:13
    encoder         : Lavf55.33.100
    Stream #0:0(eng): Video: h264 (libx264) (avc1 / 0x31637661),
yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 24k tbn, 23.98 tbc
(default)
    Metadata:
      creation_time   : 2014-04-02 04:27:44
      handler_name    : Apple Alias Data Handler
Stream mapping:
  Stream #0:0 -> #0:0 (prores -> libx264)
Press [q] to stop, [?] for help
frame=  240 fps= 35 q=-1.0 Lsize=    4459kB time=00:00:09.92
bitrate=3680.1kbits/s
video:4455kB audio:0kB subtitle:0 data:0 global headers:0kB muxing
overhead 0.092370%
[libx264 @ 0x7fb913852c00] frame I:2     Avg QP:21.98  size:102144
[libx264 @ 0x7fb913852c00] frame P:100   Avg QP:24.07  size: 27494
[libx264 @ 0x7fb913852c00] frame B:138   Avg QP:25.37  size: 11650
[libx264 @ 0x7fb913852c00] consecutive B-frames: 11.2% 31.7% 13.8% 43.3%
[libx264 @ 0x7fb913852c00] mb I  I16..4:  9.3% 71.6% 19.1%
[libx264 @ 0x7fb913852c00] mb P  I16..4:  1.0%  6.6%  1.0%  P16..4:
51.9%  7.9%  5.7%  0.0%  0.0%    skip:25.8%
[libx264 @ 0x7fb913852c00] mb B  I16..4:  0.4%  2.3%  0.5%  B16..8:
38.3%  1.7%  0.2%  direct: 4.2%  skip:52.4%  L0:46.9% L1:51.7% BI:
1.4%
[libx264 @ 0x7fb913852c00] 8x8 transform intra:74.2% inter:84.8%
[libx264 @ 0x7fb913852c00] coded y,uvDC,uvAC intra: 60.9% 55.0% 19.4%
inter: 16.9% 25.1% 0.6%
[libx264 @ 0x7fb913852c00] i16 v,h,dc,p: 20% 21%  6% 52%
[libx264 @ 0x7fb913852c00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 15% 15%
7%  8% 10%  9%  8%  8%
[libx264 @ 0x7fb913852c00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 29% 17% 13%
7%  8% 10%  7%  6%  4%
[libx264 @ 0x7fb913852c00] i8c dc,h,v,p: 60% 17% 19%  4%
[libx264 @ 0x7fb913852c00] Weighted P-Frames: Y:23.0% UV:17.0%
[libx264 @ 0x7fb913852c00] ref P L0: 43.6% 14.8% 26.0% 12.3%  3.3%
[libx264 @ 0x7fb913852c00] ref B L0: 74.4% 21.5%  4.2%
[libx264 @ 0x7fb913852c00] ref B L1: 92.2%  7.8%
[libx264 @ 0x7fb913852c00] kb/s:3645.46

$ ffmpeg -r 96000/1001 -ss 60 -t 10 -i
/Volumes/RAID/GILT/Raw_Footage/Day07/A045R1NA/A045C001_140402_R1NA.mov
-pix_fmt yuv420p -r 24000/1001 test.mov
ffmpeg version 2.1.git-ffb7d71 Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar  7 2014 16:54:41 with Apple LLVM version 5.0
(clang-500.2.79) (based on LLVM 3.3svn)
  configuration: --prefix=/opt/local --enable-swscale
--enable-avfilter --enable-avresample --enable-libmp3lame
--enable-libvorbis --enable-libopus --enable-libtheora
--enable-libschroedinger --enable-libopenjpeg --enable-libmodplug
--enable-libvpx --enable-libspeex --enable-libass --enable-libbluray
--enable-gnutls --enable-fontconfig --enable-libfreetype
--disable-indev=jack --disable-outdev=xv --mandir=/opt/local/share/man
--enable-shared --enable-pthreads --cc=/usr/bin/cc --arch=x86_64
--enable-yasm --enable-gpl --enable-postproc --enable-libx264
--enable-libxvid --enable-nonfree --enable-libfdk-aac --enable-libfaac
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libavresample   1.  2.  0 /  1.  2.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Volumes/RAID/GILT/Raw_Footage/Day07/A045R1NA/A045C001_140402_R1NA.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt  ARRI
    creation_time   : 2014-04-02 04:27:44
    timecode        : 00:01:33:13
  Duration: 00:06:13.71, start: 0.000000, bitrate: 190892 kb/s
    Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le,
1920x1080, 183719 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 23976
tbn, 23976 tbc (default)
    Metadata:
      creation_time   : 2014-04-02 04:27:44
      handler_name    : Apple Alias Data Handler
    Stream #0:1(eng): Data: none (tmcd / 0x64636D74) (default)
    Metadata:
      creation_time   : 2014-04-02 04:27:44
      handler_name    : Apple Alias Data Handler
      timecode        : 00:01:33:13
[libx264 @ 0x7fa2d0852c00] using SAR=1/1
[libx264 @ 0x7fa2d0852c00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x7fa2d0852c00] profile High, level 4.0
[libx264 @ 0x7fa2d0852c00] 264 - core 142 - H.264/MPEG-4 AVC codec -
Copyleft 2003-2014 - 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=36 lookahead_threads=6 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=23 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, mov, to 'test.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt  ARRI
    timecode        : 00:01:33:13
    encoder         : Lavf55.33.100
    Stream #0:0(eng): Video: h264 (libx264) (avc1 / 0x31637661),
yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 24k tbn, 23.98 tbc
(default)
    Metadata:
      creation_time   : 2014-04-02 04:27:44
      handler_name    : Apple Alias Data Handler
Stream mapping:
  Stream #0:0 -> #0:0 (prores -> libx264)
Press [q] to stop, [?] for help
frame=  243 fps= 16 q=-1.0 Lsize=    6151kB time=00:00:10.05
bitrate=5013.1kbits/s dup=0 drop=716
video:6147kB audio:0kB subtitle:0 data:0 global headers:0kB muxing
overhead 0.067901%
[libx264 @ 0x7fa2d0852c00] frame I:2     Avg QP:22.64  size: 86343
[libx264 @ 0x7fa2d0852c00] frame P:126   Avg QP:24.43  size: 31866
[libx264 @ 0x7fa2d0852c00] frame B:115   Avg QP:25.70  size: 18313
[libx264 @ 0x7fa2d0852c00] consecutive B-frames: 12.3% 70.0% 11.1%  6.6%
[libx264 @ 0x7fa2d0852c00] mb I  I16..4:  9.0% 72.2% 18.7%
[libx264 @ 0x7fa2d0852c00] mb P  I16..4:  1.2%  7.4%  1.1%  P16..4:
54.8%  9.5%  5.8%  0.0%  0.0%    skip:20.2%
[libx264 @ 0x7fa2d0852c00] mb B  I16..4:  0.9%  5.0%  0.8%  B16..8:
37.9%  3.1%  0.5%  direct: 8.1%  skip:43.6%  L0:46.5% L1:50.8% BI:
2.8%
[libx264 @ 0x7fa2d0852c00] 8x8 transform intra:75.4% inter:82.9%
[libx264 @ 0x7fa2d0852c00] coded y,uvDC,uvAC intra: 60.5% 55.6% 19.7%
inter: 22.0% 36.6% 0.8%
[libx264 @ 0x7fa2d0852c00] i16 v,h,dc,p: 16% 22%  5% 56%
[libx264 @ 0x7fa2d0852c00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 18% 16% 15%
7%  8% 10%  9%  8%  9%
[libx264 @ 0x7fa2d0852c00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 17% 14%
7%  8% 10%  7%  6%  4%
[libx264 @ 0x7fa2d0852c00] i8c dc,h,v,p: 61% 17% 19%  4%
[libx264 @ 0x7fa2d0852c00] Weighted P-Frames: Y:24.6% UV:23.0%
[libx264 @ 0x7fa2d0852c00] ref P L0: 43.6% 13.9% 21.6% 16.0%  4.9%
[libx264 @ 0x7fa2d0852c00] ref B L0: 67.7% 30.7%  1.6%
[libx264 @ 0x7fa2d0852c00] ref B L1: 96.9%  3.1%
[libx264 @ 0x7fa2d0852c00] kb/s:4967.86

From Alexander.Achleitner at zf.com  Wed Apr  2 10:55:18 2014
From: Alexander.Achleitner at zf.com (Alexander.Achleitner at zf.com)
Date: Wed, 2 Apr 2014 08:55:18 +0000
Subject: [FFmpeg-user] Problem Compiling ffmpeg
In-Reply-To: <loom.20140401T232247-362@post.gmane.org>
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
	<loom.20140401T232247-362@post.gmane.org>
Message-ID: <7B7B8356F2BD81429073BEF692648A1313A79D34@FRDS01162.emea.zf-world.com>

Hi Carl,

thank you for your reply. Setting to arm is much better. I tried the other, as I didn't know
the correct one.

Now the make stops at another step. Can you help me again please.
Linux is not my top development environment.


./configure --enable-cross-compile --cc=arm-cortexa8-linux-gnueabihf-gcc --cxx=arm-cortexa8-linux-gnueabihf-gcc  --as=arm-cortexa8-linux-gnueabihf-gcc  --arch=arm --target-os=linux



 make V=1
arm-cortexa8-linux-gnueabihf-gcc -I. -I./ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -march=armv7-a -std=c99 -fomit-frame-pointer -marm -pthread -g -Wdeclaration-after-statement -Wall -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -Wwrite-strings -Wtype-limits -Wundef -Wmissing-prototypes -Wno-pointer-to-int-cast -Wstrict-prototypes -Wno-parentheses -Wno-switch -Wno-format-zero-length -Wno-pointer-sign -O3 -fno-math-errno -fno-signed-zeros -fno-tree-vectorize -Werror=implicit-function-declaration -Werror=missing-prototypes -Werror=return-type -Werror=vla -Wno-maybe-uninitialized   -MMD -MF ffprobe.d -MT ffprobe.o -c -o ffprobe.o ffprobe.c
arm-cortexa8-linux-gnueabihf-gcc -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavresample -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -Wl,--as-needed -Wl,--warn-common -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample  -o ffprobe_g cmdutils.o ffprobe.o   -lavdevice -lavfilter -lavformat -lavcodec -lswresample -lswscale -lavutil -lasound -lm -lz -pthread -lrt  
cp -p ffprobe_g ffprobe
strip ffprobe
strip: Unable to recognise the format of the input file `ffprobe'
make: *** [ffprobe] Error 1

phycore at phycore-VirtualBox:~/Desktop/zfas/EnvironmentSet/ffmpeg-2.2$ make V=1
arm-cortexa8-linux-gnueabihf-gcc -I. -I./ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -march=armv7-a -std=c99 -fomit-frame-pointer -marm -pthread -g -Wdeclaration-after-statement -Wall -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -Wwrite-strings -Wtype-limits -Wundef -Wmissing-prototypes -Wno-pointer-to-int-cast -Wstrict-prototypes -Wno-parentheses -Wno-switch -Wno-format-zero-length -Wno-pointer-sign -O3 -fno-math-errno -fno-signed-zeros -fno-tree-vectorize -Werror=implicit-function-declaration -Werror=missing-prototypes -Werror=return-type -Werror=vla -Wno-maybe-uninitialized   -MMD -MF ffserver.d -MT ffserver.o -c -o ffserver.o ffserver.c
arm-cortexa8-linux-gnueabihf-gcc -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavresample -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -Wl,--as-needed -Wl,--warn-common -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample  -o ffserver_g cmdutils.o ffserver.o   -lavdevice -lavfilter -lavformat -lavcodec -lswresample -lswscale -lavutil -lasound -lm -lz -pthread -lrt  
cp -p ffserver_g ffserver
strip ffserver
strip: Unable to recognise the format of the input file `ffserver'
make: *** [ffserver] Error 1
phycore at phycore-VirtualBox:~/Desktop/zfas/EnvironmentSet/ffmpeg-2.2$ 



Yours sincerely,


Alexander Achleitner



-----Urspr?ngliche Nachricht-----
Von: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] Im Auftrag von Carl Eugen Hoyos
Gesendet: Dienstag, 1. April 2014 23:27
An: ffmpeg-user at ffmpeg.org
Betreff: Re: [FFmpeg-user] Problem Compiling ffmpeg

 <Alexander.Achleitner <at> zf.com> writes:

> One Error message is
> "...
> AR   libavdevice/libavdevice.a
> Sorry  -  this program has been built wihout plugin support "

For build problems, please run "make V=1" twice and post the complete, uncut output of the second run here.
But see below first.

> Configuration was,
> ./configure -enable-cross-compile

(Your configure line looks as if it would not work, are you sure that you posted it correctly?)

> -cc=arm-cortexa8-linux-gnueabihf-gcc

> --cxx=arm-cortexa8-linux-gnueabihf-gcc

Is this needed?

> --as=arm-cortexa8-linux-gnueabihf-gcc

This should not be needed.

> --ar=arm-cortexa8-linux-gnueabihf-gcc

This looks like the cause for your problem above.

> --arch=armel

"arm" should be enough.

> --target-os=linux

Isn't --cross-prefix what you are looking for?

Carl Eugen

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

From rainer at dcs.co.at  Wed Apr  2 13:48:40 2014
From: rainer at dcs.co.at (Rainer Fritz)
Date: Wed, 2 Apr 2014 13:48:40 +0200
Subject: [FFmpeg-user] Filter lut3d with GPU Accelaration?
Message-ID: <A41B36AF-FDFC-4F81-AF11-516724B417F6@dcs.co.at>

Dear All !

Thx for such a great tool ffmpeg is. I use the lut3d filter often and getting currently
on a MacbbokPro i7 about 12fps when transcoding from Prores4444 to DNxHD for example.

Is there a possibility to accelarate this filter by the GPU (openCL or openGL)? How much work will it be to implement
this? Who can be contacted regarding this?

THX

Rainer Fritz



From u at pkh.me  Wed Apr  2 14:03:31 2014
From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=)
Date: Wed, 2 Apr 2014 14:03:31 +0200
Subject: [FFmpeg-user] Filter lut3d with GPU Accelaration?
In-Reply-To: <A41B36AF-FDFC-4F81-AF11-516724B417F6@dcs.co.at>
References: <A41B36AF-FDFC-4F81-AF11-516724B417F6@dcs.co.at>
Message-ID: <20140402120331.GH30770@leki.pkh.me>

On Wed, Apr 02, 2014 at 01:48:40PM +0200, Rainer Fritz wrote:
> Dear All !
> 
> Thx for such a great tool ffmpeg is. I use the lut3d filter often and getting currently
> on a MacbbokPro i7 about 12fps when transcoding from Prores4444 to DNxHD for example.
> 

What version of FFmpeg are you using? I've added the threading a while
ago, make sure 'S' is present when you run:

? ffmpeg -v 0 -filters|grep lut3d
 TS. lut3d            V->V       Adjust colors using a 3D LUT.

> Is there a possibility to accelarate this filter by the GPU (openCL or openGL)? How much work will it be to implement this?

Patch welcome I guess, I don't plan to work on this.

> Who can be contacted regarding this?
> 

You could eventually open a ticket in the trac (http://trac.ffmpeg.org)

> THX
> 
> Rainer Fritz
> 

-- 
Cl?ment B.
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From joao_m_s_m at hotmail.com  Wed Apr  2 16:07:12 2014
From: joao_m_s_m at hotmail.com (=?iso-8859-1?B?Sm/jbyBNYWNoYWRv?=)
Date: Wed, 2 Apr 2014 14:07:12 +0000
Subject: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
In-Reply-To: <CAMCjN1-zB=2+rqXOHKP9KJDC1RP_8Ezt3Vxan_OGvXB=uBwxZw@mail.gmail.com>
References: <DUB130-W17C4A888B6363BD773124EE8620@phx.gbl>,
	<DUB130-W80C9C295EE04FA99F54850E8620@phx.gbl>,
	<loom.20140401T225744-805@post.gmane.org>,
	<CAMCjN1-zB=2+rqXOHKP9KJDC1RP_8Ezt3Vxan_OGvXB=uBwxZw@mail.gmail.com>
Message-ID: <DUB130-W6DDE8EDE93794AC4FFEC2E86D0@phx.gbl>

Hi again,

I have this huge command line:

gcc -I. -I./ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -I/home/joao/ffmpeg_build/include -std=c99 -fomit-frame-pointer -pthread -I/usr/include/fribidi -I/usr/include/freetype2 -I/home/joao/ffmpeg_build/include/opus -D_GNU_SOURCE=1 -D_REENTRANT -I/usr/include/SDL -g -Wdeclaration-after-statement -Wall -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -Wwrite-strings -Wtype-limits -Wundef -Wmissing-prototypes -Wno-pointer-to-int-cast -Wstrict-prototypes -Wno-parentheses -Wno-switch -Wno-format-zero-length -Wno-pointer-sign -O3 -fno-math-errno -fno-signed-zeros -fno-tree-vectorize -Werror=implicit-function-declaration -Werror=missing-prototypes -Werror=return-type -Werror=vla -Wno-maybe-uninitialized  -D_GNU_SOURCE=1 -D_REENTRANT -I/usr/include/SDL -MMD -MF ffplay.d -MT ffplay.o -c -o ffplay.o ffplay.c

Is everything mandatory?

Can you tell me what I need to add to GCC C++-> Compiler and  GCC C++-> Linker?

Thanks in advance,
JM.


> Date: Tue, 1 Apr 2014 22:10:30 +0100
> From: naveen.mr at gmail.com
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
> 
> Joao,
> if there is no way out - suggest you to look at the command line compiler
> and linker options for ffplay.c
> use verbose method to see these options:
> 
> make V=s
> 
> It should give fair bit of info required to populate "project -> properties
> -> c/c++ build -> settings" in eclipse
> I did manage to compile this way sometime ago
> 
> 
> 
> 
> On Tue, Apr 1, 2014 at 9:58 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> 
> > Jo?o Machado <joao_m_s_m <at> hotmail.com> writes:
> >
> > > Now I am trying to compile the ffplay.c using the eclipse,
> > > so I can debud, change and compile it.
> >
> > > I also found this link which I'll try after
> > > https://trac.ffmpeg.org/wiki/Setup_Eclipse_IDE_for_FFmpeg_%28Linux%29.
> >
> > I suggest you ask here if anything does not work as
> > explained in the wiki.
> >
> > Carl Eugen
> >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> 
> 
> 
> -- 
> ,
> Naveen Rudrappa
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
 		 	   		  

From peter at peterphi.com  Wed Apr  2 18:00:01 2014
From: peter at peterphi.com (Peter Wright)
Date: Wed, 2 Apr 2014 17:00:01 +0100
Subject: [FFmpeg-user] AAC in MOV 1 frame off
In-Reply-To: <etPan.533b4fa9.2ae8944a.4829@Peters-Retina-Mac.local>
References: <etPan.533b0324.6b8b4567.11409@Peters-Retina-Mac.local>
	<loom.20140401T225539-103@post.gmane.org>
	<etPan.533b4fa9.2ae8944a.4829@Peters-Retina-Mac.local>
Message-ID: <etPan.533c3401.6b8b4567.4c36@Peters-Retina-Mac.local>

On 2 April 2014 at 00:45:45, Peter Wright (peter at peterphi.com) wrote:
On 1 April 2014 at 21:57:20, Carl Eugen Hoyos (cehoyos at ag.or.at) wrote:
Peter Wright <peter <at> peterphi.com> writes:?

> I am having a problem with AV sync being off by 1?
> frame when using AAC and MP3?

Iirc, this was reported several times, but I?
believe that neither myself nor another FFmpeg?
developer understood the issue:?
How can you know that AV-sync is off by 0.04 seconds??

Carl Eugen?
Hi Carl,

Firstly, I?m assuming that quicktime / final cut pro do the correct thing (I?m trying to write files that will be used by end-users in FCP and quicktime so from my perspective I want the files to work in quicktime) and using them to play back a frame of video and its associated audio one at a time. However, there does seem to be something odd going on anyway (with the first audio packet having a PTS value that puts it before the first video frame - see below)

The sample I?m generating (see previous e-mail for command) using lavfilter is emitting 48k samples/second (sine wave with a beep once a second). This should mean that the first 1920 audio samples in the stream are a ?beep?, with the next?46080 being a constant tone. When I play it back with ffplay the first thing I can hear is the beep, then the tone. When I play back in Quicktime the initial beep is not audible. This problem also manifests for an audio-only MOV file.

On playback the tone isn?t off by precisely 0.04 seconds, but it?s roughly that: when I play only frame 00:00:00:23 I can hear a small click which I think is part of the 1920 sample beep (which is clearly audible when 00:00:00:24 is played). I imported the waveform from the .MOV into Audacity - the beep starts at sample 45,888 which is 2112 samples too early (it?s also 1920+192 - I?m not sure if 192 appearing is a coincidence)

When I looked at the file with ffprobe -show_packets I see that the first audio packet has a PTS Time of?-0.021333 (and a duration time of 0.021333), whereas the first video packet has a PTS Time of 0.0000. So if I understand things correctly that entire audio packet should be played back before the first frame of video is displayed (and while 1024 audio samples doesn?t completely explain why the sample is ~1920+192 samples early it does make me think that?s something odd is happening)

So, to recap: the problem goes away when emitting PCM (and both codecs which exhibit this problem, AAC and MP3, have a negative PTS on the first packet)

I?d like to help figure out what?s going on here if I can, do you have any pointers for where I should look to get you more detail? I have some familiarity with the code but not a great deal with the specifics of the encoders and the mov muxer (one experiment I?d like to try is to find out who?s assigning that negative PTS value and see what happens if it?s changed to zero instead)



I think I know what?s happening here - and, specifically, why the second beep starts 2112 samples early (I had originally thought 2112 was 1920+192 but it seems to be a magic number from historical AAC encoders). It also explains why the problem occurs for MP3.

Appendix G of the Quicktime Specification (?https://developer.apple.com/library/mac/documentation/quicktime/qtff/QTFFAppenG/QTFFAppenG.html?) describes how AAC is positioned relative to video, by using the Edit List atom and the Sample Group Structures atom. Crucially,??In the absence of the sample group structures, the classic solution of expecting an implicit encoding delay of 2112 samples and the edit list to start at the beginning of encoder delay will be assumed as described in the previous section?. FFmpeg doesn?t write an spgd atom, so it triggers this behaviour in Apple?s code.

That indicates that it?s necessary to write sgpd and sbgp atoms for the audio tracks (to confirm that the MOV is being encoded with explicit delay information). I tried writing some initial code for this (patch attached for movenc.c) but don?t see a difference, so I assume the fallback behaviour of Apple?s code is still being triggered.

Interestingly, the files generated by Final Cut Pro X export to MOV don?t contain the sgdp/sbgp atoms





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From georg+ffmpeg at schaathun.net  Wed Apr  2 19:36:08 2014
From: georg+ffmpeg at schaathun.net (Hans Georg Schaathun)
Date: Wed, 2 Apr 2014 18:36:08 +0100
Subject: [FFmpeg-user] How to get compatible encoding when converting
	various sources?
Message-ID: <20140402173608.GA18183@golay.schaathun.net>

Hi,

I try to create videos by recording small snippets, with different
methods, and then splice them together to a longer video.  Problem
is, even using the same output codec options, the snippets appear
not to be compatible.  The voice in some of the snippets are in
a deeper pitch in the spliced video compared to the constituent 
file.

Does anyone have any advice to offer?

I am not an expert in multimedia coding, least of all sound.
I can guess that there is probably some incompatibility in the
sound encoding, but I cannot imagine what options are missing
to force it all into one format.

The source files are generated with the following two ffmpeg
calls:
1.  From a high quality AVI file, recorded by ffmpeg from screencast:
ffmpeg -i in.avi -acodec libfdk_aac -ab 128k -ac 2 \
	-vcodec libx264 -s 640x480 -pix_fmt yuv420p -movflags \
	-threads 0 -y out.mp4
2.  From a PNG still and a FLAC file recorded by audacity:
ffmpeg -loop 1 -i in.png -i in.flac \
	-acodec libfdk_aac -ab 128k -ac 2 \
	-vcodec libx264 -s 640x480 -pix_fmt yuv420p -movflags \
        -shortest -threads 0 -y out.mp4

The results are spliced with the following command
        ffmpeg -f concat -i in.list -c copy out.mp4

In the cases where all the snippets are from source 1, there is
no problem.  When the first snippet is from source 2, all the snippets 
from source 1 get the deranged audio pitch.  (I have not tried 
combining a starting source 1 with subsequent source 2-s.)

I am also doing the same exercise using OGG/Theora/Vorbis
instead of mpeg4/h264/aac, but the result is even worse,
with sound clipping.

I shall be grateful for any pointers.

If it matters, it runs on Ubuntu 13.04 with the following version 
information:
% ffmpeg -version
ffmpeg version git-2013-08-28-70106e7
built on Aug 28 2013 11:27:12 with gcc 4.7 (Ubuntu/Linaro
4.7.3-1ubuntu1)
configuration: --prefix=/local/home/georg/prg/ffmpeg
--extra-cflags=-I/local/home/georg/prg/ffmpeg/include
--extra-ldflags=-L/local/home/georg/prg/ffmpeg/lib --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
libavutil      52. 42.100 / 52. 42.100
libavcodec     55. 29.100 / 55. 29.100
libavformat    55. 14.102 / 55. 14.102
libavdevice    55.  3.100 / 55.  3.100
libavfilter     3. 82.102 /  3. 82.102
libswscale      2.  5.100 /  2.  5.100
libswresample   0. 17.103 /  0. 17.103
libpostproc    52.  3.100 / 52.  3.100


-- 
:-- Hans Georg

From asdfg1945 at gmail.com  Wed Apr  2 10:02:43 2014
From: asdfg1945 at gmail.com (Arachnia)
Date: Wed, 2 Apr 2014 01:02:43 -0700 (PDT)
Subject: [FFmpeg-user] segment options not work in combining audio and video
 the same time?
Message-ID: <1396425763948-4664610.post@n4.nabble.com>

I want to convert a rtsp stream to ts and m3u8. My command is 

ffmpeg -i 'rtsp://localhost:5544/rtsptest' -vcodec libx264 -crf 24 -b:v 128k
-s 352x288 -profile:v baseline -pix_fmt yuv420p -flags -global_header -map 0
-f segment -segment_time 10 -segment_list_size 2 -segment_list test.m3u8
-segment_list_type m3u8 -segment_format mpegts 240p%08d.ts

It works fine. And then I try to add a empty audio track, command is

ffmpeg -f s16le -i /dev/zero -f rtsp -i 'rtsp://localhost:5544/rtsptest'
-vcodec libx264 -crf 24 -b:v 128k -s 352x288 -profile:v baseline -pix_fmt
yuv420p -c:a libfaac -ar 22050 -async 1 -ac 2 -b:a 1k -flags -global_header
-map 0 -f segment -segment_time 10 -segment_list_size 2 -segment_list
test.m3u8 -segment_list_type m3u8 -segment_format mpegts 240p%08d.ts

The ts file I get only have audio stream!

if I encode stream to a flv file, or a mp4 file, the commands all OK!

ffmpeg -f s16le -i /dev/zero -f rtsp -i 'rtsp://localhost:5544/rtsptest'
-vcodec libx264 -crf 24 -b:v 128k -s 352x288 -profile:v baseline -pix_fmt
yuv420p -c:a libfaac -ar 22050 -async 1 -ac 2 -b:a 1k -f flv test.flv

And I found that if I change position of input devices in command like this

 ffmpeg -f rtsp -i 'rtsp://localhost:5544/rtsptest' -f s16le -i /dev/zero
-vcodec libx264 -crf 24 -b:v 128k -s 352x288 -profile:v baseline -pix_fmt
yuv420p -c:a libfaac -ar 22050 -async 1 -ac 2 -b:a 1k -flags -global_header
-map 0 -f segment -segment_time 10 -segment_list_size 2 -segment_list
test.m3u8 -segment_list_type m3u8 -segment_format mpegts 240p%08d.ts

Then ts files only have video stream.

Something wrong with me?




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From ghofmann at sofa.de  Wed Apr  2 10:18:34 2014
From: ghofmann at sofa.de (Blaukasten)
Date: Wed, 2 Apr 2014 01:18:34 -0700 (PDT)
Subject: [FFmpeg-user] vidstabdetect
Message-ID: <1396426714549-4664612.post@n4.nabble.com>

Hello I am probably doing a beginner mistake. I am trying to use the filter
vidstabdetect and getting the following error message:

C:\Program Files\WinFF>ffmpeg -i "E:\somename.MOV" -vf
vidstabdetect=result="E:\dummy.trf" "E:\output.avi"
ffmpeg version N-60120-g52d2bcc Copyright (c) 2000-2014 the FFmpeg
developers
  built on Jan 23 2014 22:06:41 with gcc 4.8.2 (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-libvidstab
--enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
--enable-libwavp
ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 63.100 / 52. 63.100
  libavcodec     55. 49.100 / 55. 49.100
  libavformat    55. 25.101 / 55. 25.101
  libavdevice    55.  5.102 / 55.  5.102
  libavfilter     4.  1.100 /  4.  1.100
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100

...

[vidstabdetect @ 0000000004ff0940] [Eval @ 000000000022a930] Undefined
constant
or missing '(' in 'dummy.trf'
[vidstabdetect @ 0000000004ff0940] Unable to parse option value "dummy.trf"
[vidstabdetect @ 0000000004ff0940] [Eval @ 000000000022a9a0] Undefined
constant
or missing '(' in 'dummy.trf'
[vidstabdetect @ 0000000004ff0940] Unable to parse option value "dummy.trf"
[vidstabdetect @ 0000000004ff0940] Error setting option shakiness to value
dummy.trf.
[Parsed_vidstabdetect_0 @ 0000000002809560] Error applying options to the
filter.
[AVFilterGraph @ 0000000002863840] Error initializing filter 'vidstabdetect'
with args 'result=E:dummy.trf'
Error opening filters!

I was expecting that the filter would create and write the file dummy.rtf,
but it looks like it wants to read this file. Any idea?
Thanks
Gerhard



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From barsnick at gmx.net  Wed Apr  2 21:12:09 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Wed, 2 Apr 2014 21:12:09 +0200
Subject: [FFmpeg-user] vidstabdetect
In-Reply-To: <1396426714549-4664612.post@n4.nabble.com>
References: <1396426714549-4664612.post@n4.nabble.com>
Message-ID: <20140402191209.GA9915@sunshine.barsnick.net>

Hi,

On Wed, Apr 02, 2014 at 01:18:34 -0700, Blaukasten wrote:
> C:\Program Files\WinFF>ffmpeg -i "E:\somename.MOV" -vf
> vidstabdetect=result="E:\dummy.trf" "E:\output.avi"

":" is the default separator for options. If you want to use it
literally, you need to escape it. I just forced myself to use Windows,
and by trial and error figured out that it works like this:
  vidstabdetect=result="E\\:\\\\dummy.trf"

My guess was that this would work, but it doesn't:
  vidstabdetect=result="E\:\dummy.trf"

It's something in the way Windows and libav handle escapeing, I think
it was discussed here recently.

Moritz

From barsnick at gmx.net  Wed Apr  2 21:19:28 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Wed, 2 Apr 2014 21:19:28 +0200
Subject: [FFmpeg-user] segment options not work in combining audio and
 video the same time?
In-Reply-To: <1396425763948-4664610.post@n4.nabble.com>
References: <1396425763948-4664610.post@n4.nabble.com>
Message-ID: <20140402191928.GB9915@sunshine.barsnick.net>

Hi,

On Wed, Apr 02, 2014 at 01:02:43 -0700, Arachnia wrote:

> Something wrong with me?

Yes! You didn't show us the complete output of your ffmpeg commands,
which would make it much more obvious how ffmpeg is interpreting stuff.
Honestly, ffmpeg's output is meant for explaining what it's doing, why
do you hide it from us? ;-)

> if I encode stream to a flv file, or a mp4 file, the commands all OK!

Do you realize you are using totally different commands for encoding to
segment vs. encoding to file?

Most notably, your segment commands use this option:
> -map 0
without any other map options, thereby telling ffmpeg to use only the
first input file/stream. This explains all your observations.

Moritz

From barsnick at gmx.net  Wed Apr  2 21:25:37 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Wed, 2 Apr 2014 21:25:37 +0200
Subject: [FFmpeg-user] vidstabdetect
In-Reply-To: <20140402191209.GA9915@sunshine.barsnick.net>
References: <1396426714549-4664612.post@n4.nabble.com>
	<20140402191209.GA9915@sunshine.barsnick.net>
Message-ID: <20140402192537.GC9915@sunshine.barsnick.net>

Addendum:

On Wed, Apr 02, 2014 at 21:12:09 +0200, Moritz Barsnick wrote:
> It's something in the way Windows and libav handle escapeing, I think
> it was discussed here recently.

This is the post I was referring to:
http://ffmpeg.org/pipermail/ffmpeg-user/2014-January/019505.html

Moritz

From barsnick at gmx.net  Wed Apr  2 21:33:03 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Wed, 2 Apr 2014 21:33:03 +0200
Subject: [FFmpeg-user] compand filter issues
In-Reply-To: <533A6F6D.1070107@gmail.com>
References: <533A6F6D.1070107@gmail.com>
Message-ID: <20140402193303.GD9915@sunshine.barsnick.net>

Hi,
since nobody has tackled this issue yet:

On Tue, Apr 01, 2014 at 15:49:01 +0800, Darren Breeze wrote:

> /root/bin/ffmpeg -y -threads 0 -i /mnt/datastore/workflow/transcode/inbox/pop01/source_1080p.mp4 -aspect 16:9 -pix_fmt yuv420p -b:v 1800k -minrate 1800k -maxrate 1800k -bufsize 1835k -preset slow -x264opts keyint=33 -vcodec libx264 -acodec libfdk_aac -strict experimental -profile:v main -f mp4 -ac 2 -ab 192k -af compand=.3|.3:.8|.8:-90/-52|-52/-36|-36/-28|-20/-20:6:0:-20:0.8;volume=volume=-7.5dB /tmp/transcode/target_1080p.mp4
> 
> Whilst the processing works fine on channel 1 of the output file, channel 2 has high level to the point of clipping out.

The complete, uncut output from your ffmpeg command would help.
Possibly with the added argument "-loglevel debug".

My guess is that possibly your volume filter isn't being applied at
all. (The command output would show us, if that were the case.) I'm not
a total expert and may be mistaken here, but you are using ';' as a
separator:
> -af compand=.3|.3:.8|.8:-90/-52|-52/-36|-36/-28|-20/-20:6:0:-20:0.8;volume=volume=-7.5dB
but that is meant for separating filter chains, not filters. You
probably should be using ','. [*]

Moritz

[*] "Filters in the same linear chain are separated by commas, and
    distinct linear chains of filters are separated by semicolons."
    http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction

From mediastream at gmail.com  Wed Apr  2 21:54:17 2014
From: mediastream at gmail.com (Dennis)
Date: Wed, 2 Apr 2014 15:54:17 -0400
Subject: [FFmpeg-user] reusing same passlog file.
Message-ID: <CAPCJWRoPmU_m_VQF4iDh5=LjYjNaHkp4H5VR2NGvOX0_U7-xdw@mail.gmail.com>

Trying to figure out way to minimize NFS reads of passlog and mbtree files
on a diskless server.

pass 1:
ffmpeg -i /nfs/source.m2v -pass 1 output.mp4
this creates: ffmpeg2pass-0.log.mbtree and ffmpeg2pass-0.log.

now to create two pass2 files, there has to be a `ln` or `copy` command to
create a separate log for each pass2 video output track:
ln ffmpeg2pass-0.log.mbtree ffmpeg2pass-1.log.mbtree
ln ffmpeg2pass-0.log ffmpeg2pass-1.log

pass2 (two outputs):
ffmpeg -i /nfs/source.m2v  -pass 2 -s 640x480 vga.mp4   -pass 2 -s 320x240
qvga.mp4

Can FFmpeg uses exactly same set of passlog files in both pass2's. Can it
read the same passlog and mbtree file and provide it  to multiple video
output processes, as long as "passlogfilename" is specified?

Otherwise once this scales up, it results in heavy traffic of mbtree files.

From richard.heade at gmail.com  Thu Apr  3 00:19:42 2014
From: richard.heade at gmail.com (richard heade)
Date: Wed, 02 Apr 2014 15:19:42 -0700
Subject: [FFmpeg-user] ffmpeg -ss start-time issue
In-Reply-To: <20140327103111.4bd20fcb@tomh>
References: <533430A3.7010904@gmail.com> <20140327103111.4bd20fcb@tomh>
Message-ID: <533C8CFE.5060507@gmail.com>

I have tried a number of variations on generating timestamps on the
video, like

ffmpeg -i HiDef.mpg -vf
drawtext="fontfile=/usr/share/fonts/truetype/DejaVuSans.ttf: \
                                text='\%T %D': x=10: y=10: fontsize=24:
fontcolor=white" \
                                                        -vcodec copy
-acodec copy xgen_white.mp4

(I also tried "fontcolor=black")

and

ffmpeg -i HiDef.mpg -vf
"drawtext=fontsize=18:fontfile=/usr/share/fonts/truetype/DejaVuSansMono-Bold.ttf:
\
    box=1:text='%T %D':x=(w-text_w-5):y=(h-text_h-5)" -vcodec copy
-acodec copy timestamp.avi

and I haven't been able to detect any timestamps on the video.


On 03/27/2014 07:31 AM, Tom Horsley wrote:
> On Thu, 27 Mar 2014 07:07:31 -0700
> richard heade wrote:
>
>> If
>> they are incompatible is there another way, compatible with ffmpeg, to
>> get the frame count, elapsed time, and frame type from a video?
> I've never found any two video editors that agreed on timestamps :-(.
>
> I use the big hammer of generating a small version of the video
> with the timestamps printed on each frame by ffmpeg so I'm
> sure the same version of ffmpeg will believe the same timestamp:
>
> http://home.comcast.net/~tomhorsley/game/ffmpeg.html#generatetimestamps
>
> I can then play that video and pause to pick timestamps off
> the image.
>


From horsley1953 at gmail.com  Thu Apr  3 00:48:14 2014
From: horsley1953 at gmail.com (Tom Horsley)
Date: Wed, 2 Apr 2014 18:48:14 -0400
Subject: [FFmpeg-user] ffmpeg -ss start-time issue
In-Reply-To: <533C8CFE.5060507@gmail.com>
References: <533430A3.7010904@gmail.com> <20140327103111.4bd20fcb@tomh>
	<533C8CFE.5060507@gmail.com>
Message-ID: <20140402184814.114ab837@zooty>

On Wed, 02 Apr 2014 15:19:42 -0700
richard heade wrote:

> -vcodec copy

I doubt seriously you can modify a video to add anything
and also use -vcodec copy.

From rainer at dcs.co.at  Thu Apr  3 00:53:31 2014
From: rainer at dcs.co.at (Rainer Fritz)
Date: Thu, 3 Apr 2014 00:53:31 +0200
Subject: [FFmpeg-user] Filter lut3d with GPU Accelaration?
In-Reply-To: <20140402120331.GH30770@leki.pkh.me>
References: <A41B36AF-FDFC-4F81-AF11-516724B417F6@dcs.co.at>
	<20140402120331.GH30770@leki.pkh.me>
Message-ID: <92DF188A-9AFD-4D21-B3EE-8C562B507EC5@dcs.co.at>

Hi Clement !

I'm using an actual version:
configuration: --prefix=/usr/local --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libass --enable-libcelt --enable-libfaac --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-openssl --enable-libtheora --enable-libvo-aacenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-libx265 --enable-opencl --enable-opengl
  libavutil      52. 66.101 / 52. 66.101
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 34.101 / 55. 34.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100

My cmd line looks like that:
ffmpeg -i /Users/k-effects/Desktop/J001C034_140110_R6MS.mov -vf lut3d="/Users/k-effects/Desktop/Alexa_LogC2Video_Rec709.cube" -s 1920x1080 -c:v dnxhd -pix_fmt yuv422p -b:v 120M /Users/k-effects/Desktop/Test_dnxhd.mxf

Which works awesome! But thought LUT conversion could be made by GPU to get something like realtime? would love to use ffplay in realtime with the lut3d filter?.

Keep up the awesome work !

best,

Rainer




Am 02.04.2014 um 14:03 schrieb Cl?ment B?sch <u at pkh.me>:

> On Wed, Apr 02, 2014 at 01:48:40PM +0200, Rainer Fritz wrote:
>> Dear All !
>> 
>> Thx for such a great tool ffmpeg is. I use the lut3d filter often and getting currently
>> on a MacbbokPro i7 about 12fps when transcoding from Prores4444 to DNxHD for example.
>> 
> 
> What version of FFmpeg are you using? I've added the threading a while
> ago, make sure 'S' is present when you run:
> 
> ? ffmpeg -v 0 -filters|grep lut3d
> TS. lut3d            V->V       Adjust colors using a 3D LUT.
> 
>> Is there a possibility to accelarate this filter by the GPU (openCL or openGL)? How much work will it be to implement this?
> 
> Patch welcome I guess, I don't plan to work on this.
> 
>> Who can be contacted regarding this?
>> 
> 
> You could eventually open a ticket in the trac (http://trac.ffmpeg.org)
> 
>> THX
>> 
>> Rainer Fritz
>> 
> 
> -- 
> Cl?ment B.
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From asdfg1945 at gmail.com  Thu Apr  3 05:03:01 2014
From: asdfg1945 at gmail.com (Arachnia)
Date: Wed, 2 Apr 2014 20:03:01 -0700 (PDT)
Subject: [FFmpeg-user] segment options not work in combining audio and
 video the same time?
In-Reply-To: <20140402191928.GB9915@sunshine.barsnick.net>
References: <1396425763948-4664610.post@n4.nabble.com>
	<20140402191928.GB9915@sunshine.barsnick.net>
Message-ID: <1396494181183-4664627.post@n4.nabble.com>

Yes, map option is the key!

options of ffmpeg are too many and often make me confused.

Thanks a lot Moritz.

P.S.
I know it is different commands for encoding to segment vs. encoding to
file, but I thought that commands format all like 

ffmpeg [input format] [input path] [encode options] [output format] [output
path]

just a test.  ;-)






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From joao_m_s_m at hotmail.com  Thu Apr  3 14:04:45 2014
From: joao_m_s_m at hotmail.com (=?iso-8859-1?B?Sm/jbyBNYWNoYWRv?=)
Date: Thu, 3 Apr 2014 12:04:45 +0000
Subject: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
In-Reply-To: <DUB130-W6DDE8EDE93794AC4FFEC2E86D0@phx.gbl>
References: <DUB130-W17C4A888B6363BD773124EE8620@phx.gbl>, ,
	<DUB130-W80C9C295EE04FA99F54850E8620@phx.gbl>, ,
	<loom.20140401T225744-805@post.gmane.org>, ,
	<CAMCjN1-zB=2+rqXOHKP9KJDC1RP_8Ezt3Vxan_OGvXB=uBwxZw@mail.gmail.com>,
	<DUB130-W6DDE8EDE93794AC4FFEC2E86D0@phx.gbl>
Message-ID: <DUB130-W56F6D6D6A1BED41E5B143DE86C0@phx.gbl>

Hi all,

I did manage to compile it with the following commands:

gcc -I. -I./ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -I/absolutePathTo/ffmpeg_build/include -std=c99 -fomit-frame-pointer -pthread -I/usr/include/fribidi -I/usr/include/freetype2 -I/absolutePathTo/ffmpeg_build/include/opus -D_GNU_SOURCE=1 -D_REENTRANT -I/usr/include/SDL -g -Wdeclaration-after-statement -Wall -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -Wwrite-strings -Wtype-limits -Wundef -Wmissing-prototypes -Wno-pointer-to-int-cast -Wstrict-prototypes -Wno-parentheses -Wno-switch -Wno-format-zero-length -Wno-pointer-sign -O3 -fno-math-errno -fno-signed-zeros -fno-tree-vectorize -Werror=implicit-function-declaration -Werror=missing-prototypes -Werror=return-type -Werror=vla -Wno-maybe-uninitialized  -D_GNU_SOURCE=1 -D_REENTRANT -I/usr/include/SDL -MMD -MF ffplay.d -MT ffplay.o -c -o ffplay.o ffplay.c

gcc -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavresample -Llibavutil -Llibpostproc -Llibswscale -Llibswresample -L/absolutePathTo/ffmpeg_build/lib  -Wl,--as-needed -Wl,--warn-common -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample  -o ffplay_g cmdutils.o ffplay.o   -lavdevice -lavfilter -lavformat -lavcodec -lpostproc -lswresample -lswscale -lavutil -lva -lXfixes -lXext -lX11 -lasound -lSDL -lx264 -lvpx -lvpx -lvpx -lvpx -lvorbisenc -lvorbis -logg -ltheoraenc -ltheoradec -logg -L/home/joao/ffmpeg_build/lib -lopus -lmp3lame -lfdk-aac -lass -lm -lz -pthread -ldl  -lSDL

cp -p ffplay_g ffplay

strip ffplay

Cheers,

Jo?o M.
> From: joao_m_s_m at hotmail.com
> To: ffmpeg-user at ffmpeg.org
> Date: Wed, 2 Apr 2014 14:07:12 +0000
> Subject: Re: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
> 
> Hi again,
> 
> I have this huge command line:
> 
> gcc -I. -I./ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -I/home/joao/ffmpeg_build/include -std=c99 -fomit-frame-pointer -pthread -I/usr/include/fribidi -I/usr/include/freetype2 -I/home/joao/ffmpeg_build/include/opus -D_GNU_SOURCE=1 -D_REENTRANT -I/usr/include/SDL -g -Wdeclaration-after-statement -Wall -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -Wwrite-strings -Wtype-limits -Wundef -Wmissing-prototypes -Wno-pointer-to-int-cast -Wstrict-prototypes -Wno-parentheses -Wno-switch -Wno-format-zero-length -Wno-pointer-sign -O3 -fno-math-errno -fno-signed-zeros -fno-tree-vectorize -Werror=implicit-function-declaration -Werror=missing-prototypes -Werror=return-type -Werror=vla -Wno-maybe-uninitialized  -D_GNU_SOURCE=1 -D_REENTRANT -I/usr/include/SDL -MMD -MF ffplay.d -MT ffplay.o -c -o ffplay.o ffplay.c
> 
> Is everything mandatory?
> 
> Can you tell me what I need to add to GCC C++-> Compiler and  GCC C++-> Linker?
> 
> Thanks in advance,
> JM.
> 
> 
> > Date: Tue, 1 Apr 2014 22:10:30 +0100
> > From: naveen.mr at gmail.com
> > To: ffmpeg-user at ffmpeg.org
> > Subject: Re: [FFmpeg-user] Compile FFpaly on Ubuntu 13.10 and Eclipse
> > 
> > Joao,
> > if there is no way out - suggest you to look at the command line compiler
> > and linker options for ffplay.c
> > use verbose method to see these options:
> > 
> > make V=s
> > 
> > It should give fair bit of info required to populate "project -> properties
> > -> c/c++ build -> settings" in eclipse
> > I did manage to compile this way sometime ago
> > 
> > 
> > 
> > 
> > On Tue, Apr 1, 2014 at 9:58 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> > 
> > > Jo?o Machado <joao_m_s_m <at> hotmail.com> writes:
> > >
> > > > Now I am trying to compile the ffplay.c using the eclipse,
> > > > so I can debud, change and compile it.
> > >
> > > > I also found this link which I'll try after
> > > > https://trac.ffmpeg.org/wiki/Setup_Eclipse_IDE_for_FFmpeg_%28Linux%29.
> > >
> > > I suggest you ask here if anything does not work as
> > > explained in the wiki.
> > >
> > > Carl Eugen
> > >
> > > _______________________________________________
> > > ffmpeg-user mailing list
> > > ffmpeg-user at ffmpeg.org
> > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > 
> > 
> > 
> > -- 
> > ,
> > Naveen Rudrappa
> > _______________________________________________
> > 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  Wed Apr  2 11:49:08 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 2 Apr 2014 09:49:08 +0000 (UTC)
Subject: [FFmpeg-user] Possible issue with asf muxer
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
Message-ID: <loom.20140402T114402-194@post.gmane.org>

Massimo Battistel <battistel <at> gmail.com> writes:

> ffmpeg -i intermission.wmv  -vcodec rawvideo -pix_fmt yuv420p 
> -acodec pcm_s16le -ac 2 -ar 48000 -f  asf intermission2.asf

I tested this command line (without -ac 2 -ar 48000 -f asf) 
with your sample and the output file plays fine with WMP 
(and vlc) afaict.

How does converting the output file with ffmpeg work?

Carl Eugen


From cehoyos at ag.or.at  Wed Apr  2 11:12:32 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 2 Apr 2014 09:12:32 +0000 (UTC)
Subject: [FFmpeg-user] cut at keyframes
References: <CAJmOuEyk=vbSaJnS8KCxNG5XP3c=pmA1QmRrBRVuRYNcEZbxjQ@mail.gmail.com>
Message-ID: <loom.20140402T110825-747@post.gmane.org>

Pawel Por <porparek <at> gmail.com> writes:

> ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 
> -acodec copy -vcodec copy -y out.mp4
> 
> I've upgraded to the newer version of ffmpeg 2.2 and I 
> found that now when I use the above command it cuts my 
> movies not at closest keyframes.

I tested the following with the sample you uploaded 
with FFmpeg 1.0 and current git head:
$ ffmpeg -ss 1 -i test.mp4 -acodec copy -vcodec copy -t 3 out.mp4
Both versions produce the same output frames, what do I 
miss?

Please don't top-post here, it is considered rude.

Carl Eugen


From phpdev at ehrhardt.nl  Wed Apr  2 07:06:01 2014
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Wed, 02 Apr 2014 07:06:01 +0200
Subject: [FFmpeg-user] countless dropping frema warnings
Message-ID: <ab5nj9t6d8ke9o1f2qgm51mjqm24d2pkee@4ax.com>

This commit is leading to countless warnings:
http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=41e7e46cac507e993af6d656c8d3672ba3f9e304

The warnings might be correct, because I am changing the framerate from
50fps to '-r 25'. But I never got those warnings before and they are
cluttering the console output.

After removing this from ffmpeg.c, everything seems OK:

    if (nb_frames == 0) {
        nb_frames_drop++;
        av_log(NULL, AV_LOG_WARNING,
               "*** dropping frame %d from stream %d at ts %"PRId64"\n",
               ost->frame_number, ost->st->index, in_picture->pts);
        return;
    } else 

Commandline and output after the sig separator. Take note of the summary
at the end:

frame= 6028 fps=123 q=-1.0 Lsize=   12849kB time=00:04:01.04
	bitrate= 436.7kbits/s dup=0 drop=6022    

Jan
-- 
ffmpeg.exe -i "P:\DCIM\101VBPBH\S1010001.MP4" -s 384x288 -aspect 4:3 \
	-sws_flags bicubic -vcodec libx264 -b:v 320k -r 25 -strict -2 \
	-acodec aac -ab 96k -ar 48000 -ac 2 -y output.mp4

ffmpeg version N-62039-gc00f368 Copyright (c) 2000-2014 the FFmpeg
developers
  built on Apr  1 2014 23:38:04 with gcc 4.8.2 (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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 71.100 / 52. 71.100
  libavcodec     55. 56.107 / 55. 56.107
  libavformat    55. 36.100 / 55. 36.100
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'P:\DCIM\101VBPBH\S1010001.MP4':
  Metadata:
    major_brand     : mp42
    minor_version   : 1
    compatible_brands: mp42avc1
    creation_time   : 2012-09-09 16:31:46
  Duration: 00:04:01.00, start: 0.000000, bitrate: 6598 kb/s
    Stream #0:0(eng): Video: h264 (Constrained Baseline) (avc1 /
0x31637661), yuv420p(tv, bt470bg), 640x480 [SAR 1:1 DAR 4:3], 6063 kb/s,
50 fps, 50 tbr, 60k tbn, 100 tbc (default)
    Metadata:
      creation_time   : 2012-09-09 16:31:46
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo,
fltp, 255 kb/s (default)
    Metadata:
      creation_time   : 2012-09-09 16:31:46
[libx264 @ 0288d980] using SAR=1/1
[libx264 @ 0288d980] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
AVX
[libx264 @ 0288d980] profile High, level 2.1
[libx264 @ 0288d980] 264 - core 142 r2409 d6b4e63 - H.264/MPEG-4 AVC
codec - Copyleft 2003-2014 - 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=abr
mbtree=1 bitrate=320 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4
ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'output.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 1
    compatible_brands: mp42avc1
    encoder         : Lavf55.36.100
    Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021),
yuv420p, 384x288 [SAR 1:1 DAR 4:3], q=-1--1, 320 kb/s, 12800 tbn, 25 tbc
(default)
    Metadata:
      creation_time   : 2012-09-09 16:31:46
    Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz,
stereo, fltp, 96 kb/s (default)
    Metadata:
      creation_time   : 2012-09-09 16:31:46
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (aac -> aac)
Press [q] to stop, [?] for help
*** dropping frame 6 from stream 0 at ts 3
*** dropping frame 7 from stream 0 at ts 4
*** dropping frame 8 from stream 0 at ts 5
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*** dropping frame 81 from stream 0 at ts 78
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*** dropping frame 84 from stream 0 at ts 81
*** dropping frame 85 from stream 0 at ts 82
frame=   86 fps=0.0 q=29.0 size=      63kB time=00:00:02.94 bitrate=
174.6kbits/s dup=0 drop=80    
*** dropping frame 86 from stream 0 at ts 83
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*** dropping frame 144 from stream 0 at ts 141
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*** dropping frame 146 from stream 0 at ts 143
*** dropping frame 147 from stream 0 at ts 144
frame=  148 fps=148 q=28.0 size=     191kB time=00:00:05.44 bitrate=
287.3kbits/s dup=0 drop=142    
*** dropping frame 148 from stream 0 at ts 145
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*** dropping frame 189 from stream 0 at ts 186
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*** dropping frame 191 from stream 0 at ts 188

et cetera et cetera

*** dropping frame 5960 from stream 0 at ts 5957
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*** dropping frame 5962 from stream 0 at ts 5959
*** dropping frame 5963 from stream 0 at ts 5960
frame= 5963 fps=123 q=31.0 size=   12506kB time=00:03:57.99 bitrate=
430.5kbits/s dup=0 drop=5958    
*** dropping frame 5964 from stream 0 at ts 5961
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*** dropping frame 6024 from stream 0 at ts 6021
*** dropping frame 6025 from stream 0 at ts 6022
frame= 6026 fps=123 q=31.0 size=   12596kB time=00:04:00.93 bitrate=
428.3kbits/s dup=0 drop=6020    
*** dropping frame 6026 from stream 0 at ts 6023
*** dropping frame 6027 from stream 0 at ts 6024
frame= 6028 fps=123 q=-1.0 Lsize=   12849kB time=00:04:01.04 bitrate=
436.7kbits/s dup=0 drop=6022    

video:9838kB audio:2834kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 1.404214%
[libx264 @ 0288d980] frame I:27    Avg QP:19.57  size: 14911
[libx264 @ 0288d980] frame P:2501  Avg QP:22.94  size:  3257
[libx264 @ 0288d980] frame B:3500  Avg QP:25.28  size:   436
[libx264 @ 0288d980] consecutive B-frames: 12.2% 23.9% 21.6% 42.3%
[libx264 @ 0288d980] mb I  I16..4:  7.8% 45.6% 46.6%
[libx264 @ 0288d980] mb P  I16..4:  1.2%  2.6%  0.8%  P16..4: 41.8%
15.7% 10.2%  0.0%  0.0%    skip:27.7%
[libx264 @ 0288d980] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8: 33.4% 3.2%
0.9%  direct: 1.5%  skip:60.9%  L0:41.7% L1:49.9% BI: 8.5%
[libx264 @ 0288d980] final ratefactor: 22.67
[libx264 @ 0288d980] 8x8 transform intra:53.9% inter:55.0%
[libx264 @ 0288d980] coded y,uvDC,uvAC intra: 51.3% 54.3% 20.0% inter:
14.4% 9.8% 0.5%
[libx264 @ 0288d980] i16 v,h,dc,p: 32% 27% 10% 31%
[libx264 @ 0288d980] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 18% 28%  3%  5%
7%  6%  3%  6%
[libx264 @ 0288d980] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 26% 29% 15%  3%  5%
6%  7%  3%  5%
[libx264 @ 0288d980] i8c dc,h,v,p: 56% 20% 20%  4%
[libx264 @ 0288d980] Weighted P-Frames: Y:13.5% UV:2.5%
[libx264 @ 0288d980] ref P L0: 60.7% 13.1% 16.9%  9.0%  0.3%
[libx264 @ 0288d980] ref B L0: 75.8% 18.9%  5.3%
[libx264 @ 0288d980] ref B L1: 89.5% 10.5%
[libx264 @ 0288d980] kb/s:334.20


From cehoyos at ag.or.at  Wed Apr  2 11:40:41 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 2 Apr 2014 09:40:41 +0000 (UTC)
Subject: [FFmpeg-user] Problem Compiling ffmpeg
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
	<loom.20140401T232247-362@post.gmane.org>
Message-ID: <loom.20140402T114020-927@post.gmane.org>

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

> Isn't --cross-prefix what you are looking for?

Any reason why you ignored this?

Carl Eugen


From georgk66 at gmail.com  Thu Apr  3 17:06:15 2014
From: georgk66 at gmail.com (Georg Korger)
Date: Thu, 3 Apr 2014 17:06:15 +0200
Subject: [FFmpeg-user] wrap to mxf - stream order
Message-ID: <CABxxoaXyUjy4j4BHa37xwZzjiLWq=YjMku_1F7oip1QsveaK4Q@mail.gmail.com>

Hi everybody!

I try to wrap an mxf File with 8 mono audio channels to one
with the same video content and a stereo audio track from the first two
mono tracks:

ffmpeg -y -i input.mxf -c:v copy -filter_complex amerge c:\temp\test.mxf

The syntax seems to be o.k. but i get the following output:

******* START
ffmpeg version N-59396-g4156df5 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Dec 27 2013 22:01:59 with gcc 4.8.2 (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-libvidstab
--enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
--enable-libwavp
ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 59.100 / 52. 59.100
  libavcodec     55. 46.100 / 55. 46.100
  libavformat    55. 22.100 / 55. 22.100
  libavdevice    55.  5.102 / 55.  5.102
  libavfilter     4.  0.103 /  4.  0.103
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  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 'input.mxf':
  Metadata:
    uid             : 0003475a-c6f9-4ca0-931f-8259f117faf6
    generation_uid  : 0000475a-c6f9-4ca0-931f-8259f117faf6
    company_name    : Hamburg Pro Media GmbH
    product_name    : MXF4mac GC Export FCP
    product_version : 1.5.1
    product_uid     : 00000000-0000-0000-0000-000000000000
    modification_date: 2014-03-31 14:07:23
    timecode        : 10:00:00:00
  Duration: 00:03:59.84, start: 0.000000, bitrate: 59993 kb/s
    Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709), 1920x1080
[SAR 1
:1 DAR 16:9], max. 50000 kb/s, 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 @ 033383a0] No channel layout for input 1
[Parsed_amerge_0 @ 033383a0] Input channel layouts overlap: output layout
will b
e determined by the number of distinct input channels
[mxf @ 0340d360] there must be exactly one video stream and it must be the
first
 one
Output #0, mxf, to 'c:\temp\test.mxf':
  Metadata:
    uid             : 0003475a-c6f9-4ca0-931f-8259f117faf6
    generation_uid  : 0000475a-c6f9-4ca0-931f-8259f117faf6
    company_name    : Hamburg Pro Media GmbH
    product_name    : MXF4mac GC Export FCP
    product_version : 1.5.1
    product_uid     : 00000000-0000-0000-0000-000000000000
    modification_date: 2014-03-31 14:07:23
    timecode        : 10:00:00:00
    encoder         : Lavf55.22.100
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
(default)
    Stream #0:1: Video: mpeg2video, yuv422p, 1920x1080 [SAR 1:1 DAR 16:9],
q=2-3
1, max. 50000 kb/s, 25 fps, 90k tbn, 25 tbc
Stream mapping:
  Stream #0:1 (pcm_s24le) -> amerge:in0
  Stream #0:2 (pcm_s24le) -> amerge:in1
  amerge -> Stream #0:0 (pcm_s16le)
  Stream #0:0 -> #0:1 (copy)
Could not write header for output file #0 (incorrect codec parameters ?):
Error
number -1 occurred
******* END


If i do not use the amerge Filter, it works and i get an output file with
one mono audio channel:

ffmpeg -y -i input.mxf -c:v copy c:\temp\test.mxf

******* START
ffmpeg version N-59396-g4156df5 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Dec 27 2013 22:01:59 with gcc 4.8.2 (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-libvidstab
--enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
--enable-libwavp
ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 59.100 / 52. 59.100
  libavcodec     55. 46.100 / 55. 46.100
  libavformat    55. 22.100 / 55. 22.100
  libavdevice    55.  5.102 / 55.  5.102
  libavfilter     4.  0.103 /  4.  0.103
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  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 'input.mxf':
  Metadata:
    uid             : 0003475a-c6f9-4ca0-931f-8259f117faf6
    generation_uid  : 0000475a-c6f9-4ca0-931f-8259f117faf6
    company_name    : Hamburg Pro Media GmbH
    product_name    : MXF4mac GC Export FCP
    product_version : 1.5.1
    product_uid     : 00000000-0000-0000-0000-000000000000
    modification_date: 2014-03-31 14:07:23
    timecode        : 10:00:00:00
  Duration: 00:03:59.84, start: 0.000000, bitrate: 59993 kb/s
    Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709), 1920x1080
[SAR 1
:1 DAR 16:9], max. 50000 kb/s, 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
Output #0, mxf, to 'c:\temp\test.mxf':
  Metadata:
    uid             : 0003475a-c6f9-4ca0-931f-8259f117faf6
    generation_uid  : 0000475a-c6f9-4ca0-931f-8259f117faf6
    company_name    : Hamburg Pro Media GmbH
    product_name    : MXF4mac GC Export FCP
    product_version : 1.5.1
    product_uid     : 00000000-0000-0000-0000-000000000000
    modification_date: 2014-03-31 14:07:23
    timecode        : 10:00:00:00
    encoder         : Lavf55.22.100
    Stream #0:0: Video: mpeg2video, yuv422p, 1920x1080 [SAR 1:1 DAR 16:9],
q=2-3
1, max. 50000 kb/s, 25 fps, 25 tbn, 25 tbc
    Stream #0:1: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (pcm_s24le -> pcm_s16le)
Press [q] to stop, [?] for help
frame=   80 fps=0.0 q=-1.0 size=   19729kB time=00:00:03.20
bitrate=50505.7kbits
******* END

The difference is, that in the first case, the output of amerge will become
the
Stream #0:0 to the mxf wrapper, which seems not to be allowed!
In the second case video will become the Stream #0:0 and everything
seems to work.
I didn't find a way to change the streams in the mxf wrapper.
Can you help me in that case?

Thank you, Georg Korger

From battistel at gmail.com  Thu Apr  3 18:34:06 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Thu, 3 Apr 2014 18:34:06 +0200
Subject: [FFmpeg-user] Possible issue with asf muxer
In-Reply-To: <loom.20140402T114402-194@post.gmane.org>
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
	<loom.20140402T114402-194@post.gmane.org>
Message-ID: <CAHss8oES8SmM3f9U57_ArOu4HHTVqX8UER8Aq87Xz1UPU1+dhw@mail.gmail.com>

>
> I tested this command line (without -ac 2 -ar 48000 -f asf)
> with your sample and the output file plays fine with WMP
> (and vlc) afaict.
>

I've tested the same command line (the one without -ac 2 -ar 48000 -f asf)
with latest ffmpeg Zeranoe builds and vlc and I still see the issue (both
ffplay and vlc 2.1.3, wmp is fine).
Video freezes for the first 8 seconds.
My platform is Windows 7 64bit.


How does converting the output file with ffmpeg work?


I don't understand. Can you elaborate?


Thanks,
MB

From ghofmann at sofa.de  Thu Apr  3 09:27:26 2014
From: ghofmann at sofa.de (Blaukasten)
Date: Thu, 3 Apr 2014 00:27:26 -0700 (PDT)
Subject: [FFmpeg-user] vidstabdetect
In-Reply-To: <20140402192537.GC9915@sunshine.barsnick.net>
References: <1396426714549-4664612.post@n4.nabble.com>
	<20140402191209.GA9915@sunshine.barsnick.net>
	<20140402192537.GC9915@sunshine.barsnick.net>
Message-ID: <1396510046923-4664628.post@n4.nabble.com>

Thank you, Moritz, this did the job! I thought quoting ("...") would make
escaping unnecessary, didn't think the ":" had such a high priority.
Gerhard



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From info at scenator.de  Thu Apr  3 23:33:55 2014
From: info at scenator.de (scenator)
Date: Thu, 3 Apr 2014 14:33:55 -0700 (PDT)
Subject: [FFmpeg-user] bad snapshot-quality from avchd-videofile
In-Reply-To: <1396559792909-4664636.post@n4.nabble.com>
References: <1396559792909-4664636.post@n4.nabble.com>
Message-ID: <1396560835594-4664638.post@n4.nabble.com>

Unfortunately, the topic was bad formated:

call:
C:\ProgramData\ffmpeg.exe -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss
00:00:01.20 -vframes 1 C:\Users\Frank\AppData\Roaming\Snapshot_1.jpg

There is no difference in quality by calling ...\Snapshot_1.bmp (6MB) or
...\Snapshot_1.jpg (157 kb)

video-details:
ffmpeg version N-54499-g63d7684 Copyright (c) 2000-2013 the FFmpeg
developers built on Jul 6 2013 10:30:06 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. 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. 78.102 / 3. 78.102
libswscale 2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100 Input #0, mpegts, from
'd:\Temp\Stop!1.HZ!00.32.49.80!bigfile!bigfile!.m2ts': Duration:
00:16:22.28, start: 1.000033, bitrate: 17304 kb/s Program 1 Stream
#0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080
[SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:1[0x1100]:
Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream
#0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090) 

Best regards in advantage,
  Frank 



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From peter_vince at yahoo.co.uk  Fri Apr  4 00:38:08 2014
From: peter_vince at yahoo.co.uk (Peter Vince)
Date: Thu, 3 Apr 2014 23:38:08 +0100 (BST)
Subject: [FFmpeg-user] Format of YUV files
Message-ID: <1396564688.23788.YahooMailNeo@web172304.mail.ir2.yahoo.com>

Hello,

???? I am very new to ffMPEG, but have been able to create MPEG-2 files from a collection of still images.? I see from the documentation (section 7.4 on http://ffmpeg.org/ffmpeg.html) that it is possible to use individual YUV files as well.? It says they are "raw" files, so I tried, but got a pixel format error.? I had the U and V files at half horizontal resolution (as specified in the above), and had the data sitting on 128, with plus and minus values about that.? I wonder if they should sit on zero with positive and negative numbers ("signed 8-bit"?).? Are there any sample files anywhere?

???? Thank you,

????????? Peter Vince


From cehoyos at ag.or.at  Fri Apr  4 01:19:27 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 3 Apr 2014 23:19:27 +0000 (UTC)
Subject: [FFmpeg-user] Format of YUV files
References: <1396564688.23788.YahooMailNeo@web172304.mail.ir2.yahoo.com>
Message-ID: <loom.20140404T011645-836@post.gmane.org>

Peter Vince <peter_vince <at> yahoo.co.uk> writes:

> Are there any sample files anywhere?

$ ffmpeg -i singleimage -f image2 -pix_fmt yuv420p out.Y
$ ffmpeg -i videostream -f image2 -pix_fmt yuv420p out%4d.Y

You can use other pix_fmts like yuv422p.

Carl Eugen


From cehoyos at ag.or.at  Fri Apr  4 01:21:29 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 3 Apr 2014 23:21:29 +0000 (UTC)
Subject: [FFmpeg-user] bad snapshot-quality from avchd-videofile
References: <1396559792909-4664636.post@n4.nabble.com>
	<1396560835594-4664638.post@n4.nabble.com>
Message-ID: <loom.20140404T012014-601@post.gmane.org>

scenator <info <at> scenator.de> writes:

> C:\ProgramData\ffmpeg.exe -i "d:\Temp\STOP!1~1.M2T" -y 
> -s 1920x1080 -ss 00:00:01.20 -vframes 1 
> C:\Users\Frank\AppData\Roaming\Snapshot_1.jpg

This is missing a quantiser like -qscale 2 (or a higher 
value) or a bitrate.

Why are you specifying the output frame size (-s) if 
you want it to be like the input?

Carl Eugen


From cehoyos at ag.or.at  Fri Apr  4 01:26:27 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 3 Apr 2014 23:26:27 +0000 (UTC)
Subject: [FFmpeg-user] wrap to mxf - stream order
References: <CABxxoaXyUjy4j4BHa37xwZzjiLWq=YjMku_1F7oip1QsveaK4Q@mail.gmail.com>
Message-ID: <loom.20140404T012530-570@post.gmane.org>

Georg Korger <georgk66 <at> gmail.com> writes:

> ffmpeg -y -i input.mxf -c:v copy -filter_complex amerge 
> c:\temp\test.mxf

ffmpeg -i input.mxf -vcodec copy -filter_complex amerge[out] 
-map 0:0 -map [out] test.mxf

Carl Eugen


From cehoyos at ag.or.at  Fri Apr  4 01:29:59 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 3 Apr 2014 23:29:59 +0000 (UTC)
Subject: [FFmpeg-user] frame rate change is too slow
References: <CADzzJYmv8aaAejQX4WKP5sHf+QdK0j4Rni1vMgMRQBGfgKiVSg@mail.gmail.com>
Message-ID: <loom.20140404T012916-466@post.gmane.org>

Elliott Balsley <elliottbalsley <at> gmail.com> writes:

> I want to make it play at normal speed, by dropping 3 out of
> every 4 frames.  I would expect this to encode 4x faster, 
> but it actually takes about twice as long as a normal encode.

Did you test to encode a whole file with both 
methods (without "-t")?

Carl Eugen


From info at scenator.de  Thu Apr  3 23:26:00 2014
From: info at scenator.de (scenator)
Date: Thu, 3 Apr 2014 14:26:00 -0700 (PDT)
Subject: [FFmpeg-user] snapshot from videfile: incorrect position
Message-ID: <1396560360295-4664637.post@n4.nabble.com>

Hello,

I hope, it is quite simple for you...

I want to take a snapshot from sec. 437,960212.

Therefore, I have to go fastforward to 435.96 (2Sec. before snapshot),
then play 2.00 Seconds and take my snapshot

So I call:
C:\ProgramData\ffmpeg.exe -ss 435.96 -i "d:\Temp\STOP!1~1.M2T" -y -s
1920x1080 -ss 00:00:02.00 -vframes 1
C:\Users\Frank\AppData\Roaming\Snapshot.jpg

result: the extracted frame ist a frame round about half a second before
437,96.

So I tried some combinations like:
 -ss 435.96 -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss 00:00:01.75... ->
got closer to 437,96..
 -ss 435.96 -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss 00:00:01.70... ->
exactely the wished frame 437,96.

But another target-position eg. 525,16:
-ss 00:00:01.80 is exactely the wished frame, etc.

So could you please explain, how to find out the exact target frame?

Best regards in advantage,
  Frank
  



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From info at scenator.de  Thu Apr  3 23:16:32 2014
From: info at scenator.de (scenator)
Date: Thu, 3 Apr 2014 14:16:32 -0700 (PDT)
Subject: [FFmpeg-user] bad snapshot-quality from avchd-videofile
Message-ID: <1396559792909-4664636.post@n4.nabble.com>

Hello,I need your help again...The Quality of /all/ extracted snapshot (from
a HD-videofile) isn't acceptable, please watch the result:
<http://ffmpeg-users.933282.n4.nabble.com/file/n4664636/Snapshot_1.jpg>
C:\ProgramData\ffmpeg.exe -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss
00:00:01.20 -vframes 1 C:\Users\Frank\AppData\Roaming\Snapshot_1.jpgThere is
no difference in quality by calling ...\Snapshot_1.bmp (6MB) or
...\Snapshot_1.jpg (157 kb)ffmpeg version N-54499-g63d7684 Copyright (c)
2000-2013 the FFmpeg developers  built on Jul  6 2013 10:30:06 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. 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. 78.102 /  3. 78.102  libswscale     
2.  3.100 /  2.  3.100  libswresample   0. 17.102 /  0. 17.102  libpostproc   
52.  3.100 / 52.  3.100Input #0, mpegts, from
'd:\Temp\Stop!1.HZ!00.32.49.80!bigfile!bigfile!.m2ts':  Duration:
00:16:22.28, start: 1.000033, bitrate: 17304 kb/s  Program 1     Stream
#0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080
[SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc    Stream #0:1[0x1100]:
Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s   
Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] /
0x0090)Best regards in advantage,  Frank



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From me at renecalles.de  Fri Apr  4 07:03:11 2014
From: me at renecalles.de (=?utf-8?Q?Ren=C3=A9_Calles?=)
Date: Fri, 4 Apr 2014 07:03:11 +0200
Subject: [FFmpeg-user] bad snapshot-quality from avchd-videofile
In-Reply-To: <1396559792909-4664636.post@n4.nabble.com>
References: <1396559792909-4664636.post@n4.nabble.com>
Message-ID: <etPan.533e3d0f.6b8b4567.171ef@amaconatable.fritz.box>

Hi,

as Carl already mentioned you should set the -qscale argument with e.g. -qscale 2.

Cheers,
Ren?
Am 4. April 2014 bei 05:35:20, scenator (info at scenator.de) schrieb:

Hello,I need your help again...The Quality of /all/ extracted snapshot (from  
a HD-videofile) isn't acceptable, please watch the result:  
<http://ffmpeg-users.933282.n4.nabble.com/file/n4664636/Snapshot_1.jpg>  
C:\ProgramData\ffmpeg.exe -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss  
00:00:01.20 -vframes 1 C:\Users\Frank\AppData\Roaming\Snapshot_1.jpgThere is  
no difference in quality by calling ...\Snapshot_1.bmp (6MB) or  
...\Snapshot_1.jpg (157 kb)ffmpeg version N-54499-g63d7684 Copyright (c)  
2000-2013 the FFmpeg developers built on Jul 6 2013 10:30:06 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. 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. 78.102 / 3. 78.102 libswscale  
2. 3.100 / 2. 3.100 libswresample 0. 17.102 / 0. 17.102 libpostproc  
52. 3.100 / 52. 3.100Input #0, mpegts, from  
'd:\Temp\Stop!1.HZ!00.32.49.80!bigfile!bigfile!.m2ts': Duration:  
00:16:22.28, start: 1.000033, bitrate: 17304 kb/s Program 1 Stream  
#0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080  
[SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:1[0x1100]:  
Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s  
Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] /  
0x0090)Best regards in advantage, Frank  



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/bad-snapshot-quality-from-avchd-videofile-tp4664636.html  
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From me at renecalles.de  Fri Apr  4 07:08:02 2014
From: me at renecalles.de (=?utf-8?Q?Ren=C3=A9_Calles?=)
Date: Fri, 4 Apr 2014 07:08:02 +0200
Subject: [FFmpeg-user] snapshot from videfile: incorrect position
In-Reply-To: <1396560360295-4664637.post@n4.nabble.com>
References: <1396560360295-4664637.post@n4.nabble.com>
Message-ID: <etPan.533e3e32.74b0dc51.171ef@amaconatable.fritz.box>

Hi,

as far as i know using -ss **before** the -i input, FFmpeg will fast forward without decoding the frames. When it reaches the Timestamp you are seeking to it will choose the next Keyframe (I-Frame) by a principle called nearest neighbor.

It seems the frame at 435.96 is no Keyframe while the frame at 525,16 is a keyframe.

So, for more accurate seeking it is preferred to use -ss argument **after** -i input.

Bests,
Ren?
Am 4. April 2014 bei 05:34:56, scenator (info at scenator.de) schrieb:

Hello,  

I hope, it is quite simple for you...  

I want to take a snapshot from sec. 437,960212.  

Therefore, I have to go fastforward to 435.96 (2Sec. before snapshot),  
then play 2.00 Seconds and take my snapshot  

So I call:  
C:\ProgramData\ffmpeg.exe -ss 435.96 -i "d:\Temp\STOP!1~1.M2T" -y -s  
1920x1080 -ss 00:00:02.00 -vframes 1  
C:\Users\Frank\AppData\Roaming\Snapshot.jpg  

result: the extracted frame ist a frame round about half a second before  
437,96.  

So I tried some combinations like:  
-ss 435.96 -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss 00:00:01.75... ->  
got closer to 437,96..  
-ss 435.96 -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss 00:00:01.70... ->  
exactely the wished frame 437,96.  

But another target-position eg. 525,16:  
-ss 00:00:01.80 is exactely the wished frame, etc.  

So could you please explain, how to find out the exact target frame?  

Best regards in advantage,  
Frank  




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From u at pkh.me  Fri Apr  4 07:59:47 2014
From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=)
Date: Fri, 4 Apr 2014 07:59:47 +0200
Subject: [FFmpeg-user] Filter lut3d with GPU Accelaration?
In-Reply-To: <92DF188A-9AFD-4D21-B3EE-8C562B507EC5@dcs.co.at>
References: <A41B36AF-FDFC-4F81-AF11-516724B417F6@dcs.co.at>
	<20140402120331.GH30770@leki.pkh.me>
	<92DF188A-9AFD-4D21-B3EE-8C562B507EC5@dcs.co.at>
Message-ID: <20140404055947.GB13370@leki.pkh.me>

On Thu, Apr 03, 2014 at 12:53:31AM +0200, Rainer Fritz wrote:
> Hi Clement !
> 
> I'm using an actual version:
> configuration: --prefix=/usr/local --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libass --enable-libcelt --enable-libfaac --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-openssl --enable-libtheora --enable-libvo-aacenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-libx265 --enable-opencl --enable-opengl
>   libavutil      52. 66.101 / 52. 66.101
>   libavcodec     55. 52.102 / 55. 52.102
>   libavformat    55. 34.101 / 55. 34.101
>   libavdevice    55. 11.100 / 55. 11.100
>   libavfilter     4.  3.100 /  4.  3.100
>   libswscale      2.  5.101 /  2.  5.101
>   libswresample   0. 18.100 /  0. 18.100
>   libpostproc    52.  3.100 / 52.  3.100
> 
> My cmd line looks like that:
> ffmpeg -i /Users/k-effects/Desktop/J001C034_140110_R6MS.mov -vf lut3d="/Users/k-effects/Desktop/Alexa_LogC2Video_Rec709.cube" -s 1920x1080 -c:v dnxhd -pix_fmt yuv422p -b:v 120M /Users/k-effects/Desktop/Test_dnxhd.mxf
> 

You should check the speed of the filter itself. Compare the FPS between:
    ffmpeg -i /Users/k-effects/Desktop/J001C034_140110_R6MS.mov -vf -f null -
    ffmpeg -i /Users/k-effects/Desktop/J001C034_140110_R6MS.mov -vf lut3d="/Users/k-effects/Desktop/Alexa_LogC2Video_Rec709.cube" -f null -

Also, you should specify the -s as a video filter in the -vf. Depending on
the input size it will affect the speed; for instance if this is some 4k
footage, you probably want to put the scale before lut3d. If it's an
upscale, you probably want to put the scale after lut3d. Like, -vf
"lut3d=bla.cube,scale=1920x1080".

> Which works awesome! But thought LUT conversion could be made by GPU to get something like realtime? would love to use ffplay in realtime with the lut3d filter?.
> 

I don't have the time nor the motivation to work on GPU accel, but we have
an OpenCL integration already, so feel free to contribute or open a ticket
as I suggested.

Note: please do not top post, it is considered rude on this mailing list.

[...]

-- 
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From u at pkh.me  Fri Apr  4 08:00:19 2014
From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=)
Date: Fri, 4 Apr 2014 08:00:19 +0200
Subject: [FFmpeg-user] Filter lut3d with GPU Accelaration?
In-Reply-To: <20140404055947.GB13370@leki.pkh.me>
References: <A41B36AF-FDFC-4F81-AF11-516724B417F6@dcs.co.at>
	<20140402120331.GH30770@leki.pkh.me>
	<92DF188A-9AFD-4D21-B3EE-8C562B507EC5@dcs.co.at>
	<20140404055947.GB13370@leki.pkh.me>
Message-ID: <20140404060019.GC13370@leki.pkh.me>

On Fri, Apr 04, 2014 at 07:59:47AM +0200, Cl?ment B?sch wrote:
> You should check the speed of the filter itself. Compare the FPS between:
>     ffmpeg -i /Users/k-effects/Desktop/J001C034_140110_R6MS.mov -vf -f null -

drop the -vf here ofc.

-- 
Cl?ment B.
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From adf.lists at gmail.com  Fri Apr  4 11:18:17 2014
From: adf.lists at gmail.com (Andy Furniss)
Date: Fri, 04 Apr 2014 10:18:17 +0100
Subject: [FFmpeg-user] bad snapshot-quality from avchd-videofile
In-Reply-To: <1396559792909-4664636.post@n4.nabble.com>
References: <1396559792909-4664636.post@n4.nabble.com>
Message-ID: <533E78D9.7060608@gmail.com>

scenator wrote:
> Hello,I need your help again...The Quality of /all/ extracted snapshot (from
> a HD-videofile) isn't acceptable, please watch the result:
> <http://ffmpeg-users.933282.n4.nabble.com/file/n4664636/Snapshot_1.jpg>
> C:\ProgramData\ffmpeg.exe -i "d:\Temp\STOP!1~1.M2T" -y -s 1920x1080 -ss
> 00:00:01.20 -vframes 1 C:\Users\Frank\AppData\Roaming\Snapshot_1.jpgThere is
> no difference in quality by calling ...\Snapshot_1.bmp (6MB)

The source is interlaced so you are getting a weaved frame.

Try with a deinterlacer eg. -vf yadif=1

With the =1 yadif will produce one frame per field so you will have x2 
frames to choose from.



From peter_vince at yahoo.co.uk  Fri Apr  4 11:29:25 2014
From: peter_vince at yahoo.co.uk (Peter Vince)
Date: Fri, 4 Apr 2014 10:29:25 +0100 (BST)
Subject: [FFmpeg-user] Format of YUV files
In-Reply-To: <loom.20140404T011645-836@post.gmane.org>
References: <1396564688.23788.YahooMailNeo@web172304.mail.ir2.yahoo.com>
	<loom.20140404T011645-836@post.gmane.org>
Message-ID: <1396603765.37821.YahooMailNeo@web172304.mail.ir2.yahoo.com>

Terrific - thanks Carl!

???? Peter




----- Original Message -----
From: Carl Eugen Hoyos <cehoyos at ag.or.at>
To: ffmpeg-user at ffmpeg.org
Cc: 
Sent: Friday, 4 April 2014, 0:19
Subject: Re: [FFmpeg-user] Format of YUV files

Peter Vince <peter_vince <at> yahoo.co.uk> writes:

> Are there any sample files anywhere?

$ ffmpeg -i singleimage -f image2 -pix_fmt yuv420p out.Y
$ ffmpeg -i videostream -f image2 -pix_fmt yuv420p out%4d.Y

You can use other pix_fmts like yuv422p.

Carl Eugen

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


From Alexander.Achleitner at zf.com  Fri Apr  4 11:44:24 2014
From: Alexander.Achleitner at zf.com (Alexander.Achleitner at zf.com)
Date: Fri, 4 Apr 2014 09:44:24 +0000
Subject: [FFmpeg-user] Problem Compiling ffmpeg
In-Reply-To: <loom.20140402T114020-927@post.gmane.org>
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
	<loom.20140401T232247-362@post.gmane.org>
	<loom.20140402T114020-927@post.gmane.org>
Message-ID: <7B7B8356F2BD81429073BEF692648A1313A7A0C1@FRDS01162.emea.zf-world.com>

Hi Carl,

maybe. The problem is that i don't know waht is missing for success. I use an Eclipse IDE
to develop for the arm AM335 controller. To compile the ffmpeg I use the compiler set
in the Eclipse project, but there are some options  (-O0 -g3 -Wall -c -fmessage-length=0).
But I think they are not correlated with my problem. -O is for optimization, -g3  , -Wall is for suppress
warnings, -c  is for only compiling , -fmessage is for diagnostic message   from the gcc options.

All my Eclipse Qt projects are running on the AM335. But I really have problems
to compile the ffmpeg sources in Ubuntu for the AM335 target.
Second problem, what can I do with --croos-prefix ? There is no explanation for
this option.

When i start the make process, a lot of objects are created. That's ok.  But then,
the make stopps at this position, 
 
MAN	doc/libswresample.3
MAN	doc/libavcodec.3
MAN	doc/libavformat.3
MAN	doc/libavdevice.3
MAN	doc/libavfilter.3
TXT	doc/fate.txt
CC	ffmpeg.o
ffmpeg.c: In function 'write_frame':
ffmpeg.c:592:9: warning: 'destruct' is deprecated (declared at ./libavcodec/avcodec.h:1140) [-Wdeprecated-declarations]
ffmpeg.c: In function 'do_streamcopy':
ffmpeg.c:1562:5: warning: passing argument 2 of 'av_copy_packet_side_data' discards 'const' qualifier from pointer target type [enabled by default]
In file included from libavformat/avformat.h:255:0,
                 from ffmpeg.c:44:
./libavcodec/avcodec.h:3593:5: note: expected 'struct AVPacket *' but argument is of type 'const struct AVPacket *'
CC	cmdutils.o
CC	ffmpeg_opt.o
CC	ffmpeg_filter.o
LD	ffmpeg_g
CP	ffmpeg
STRIP	ffmpeg
strip: Unable to recognise the format of the input file `ffmpeg'
make: *** [ffmpeg] Error 1



Regards,


Alexander Achleitner



-----Urspr?ngliche Nachricht-----
Von: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] Im Auftrag von Carl Eugen Hoyos
Gesendet: Mittwoch, 2. April 2014 11:41
An: ffmpeg-user at ffmpeg.org
Betreff: Re: [FFmpeg-user] Problem Compiling ffmpeg

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

> Isn't --cross-prefix what you are looking for?

Any reason why you ignored this?

Carl Eugen

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ffmpeg-user at ffmpeg.org
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From barsnick at gmx.net  Fri Apr  4 12:13:15 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Fri, 4 Apr 2014 12:13:15 +0200
Subject: [FFmpeg-user] Problem Compiling ffmpeg
In-Reply-To: <7B7B8356F2BD81429073BEF692648A1313A7A0C1@FRDS01162.emea.zf-world.com>
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
	<loom.20140401T232247-362@post.gmane.org>
	<loom.20140402T114020-927@post.gmane.org>
	<7B7B8356F2BD81429073BEF692648A1313A7A0C1@FRDS01162.emea.zf-world.com>
Message-ID: <20140404101315.GA12382@sunshine.barsnick.net>

Hi Alexander,

On Fri, Apr 04, 2014 at 09:44:24 +0000, Alexander.Achleitner at zf.com wrote:
> CC	ffmpeg_opt.o
> CC	ffmpeg_filter.o
> LD	ffmpeg_g
> CP	ffmpeg
> STRIP	ffmpeg
> strip: Unable to recognise the format of the input file `ffmpeg'
> make: *** [ffmpeg] Error 1

You are using the (native) program "strip" and not the (cross) program
"arm-cortexa8-linux-gnueabihf-strip".

The former does not handle the cross binaries, and therefore errors
out.

The latter would have been used if you had not ignored Carl's hint at
using "--cross-prefix".

(I personally don't see the point of configuring and using an IDE to
build a project which already has a perfectly working configure/make
flow. But that's just me.)

Moritz

From cehoyos at ag.or.at  Fri Apr  4 13:28:06 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 4 Apr 2014 11:28:06 +0000 (UTC)
Subject: [FFmpeg-user] Problem Compiling ffmpeg
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
	<loom.20140401T232247-362@post.gmane.org>
	<loom.20140402T114020-927@post.gmane.org>
	<7B7B8356F2BD81429073BEF692648A1313A7A0C1@FRDS01162.emea.zf-world.com>
	<20140404101315.GA12382@sunshine.barsnick.net>
Message-ID: <loom.20140404T132431-348@post.gmane.org>

Moritz Barsnick <barsnick <at> gmx.net> writes:

> I personally don't see the point of configuring and 
> using an IDE to build a project which already has a 
> perfectly working configure/make flow. 

> But that's just me.

On the contrary:
We invested a huge amount of work into the configure 
script and I don't remember a request for an additional 
feature atm.
At the same time, people don't stop to ask how not to 
use the script (and ignore both possible solutions and 
the etiquette on this mailing list). This is quite 
annoying.

Is ./configure --help really an unusual way to learn 
about its options?

Carl Eugen


From barsnick at gmx.net  Fri Apr  4 14:38:35 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Fri, 4 Apr 2014 14:38:35 +0200
Subject: [FFmpeg-user] configure features: pkgconfig support
Message-ID: <20140404123809.GA12991@sunshine.barsnick.net>

Was: Re: [FFmpeg-user] Problem Compiling ffmpeg

On Fri, Apr 04, 2014 at 11:28:06 +0000, Carl Eugen Hoyos wrote:
> and I don't remember a request for an additional feature atm.

Okay, here goes:
How about pkgconfig support for libx264 and libfdk_aac? Or is there
some legacy (e.g. old versions without .pc) which this would break?

Thanks,
Moritz

From cehoyos at ag.or.at  Fri Apr  4 14:47:35 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 4 Apr 2014 12:47:35 +0000 (UTC)
Subject: [FFmpeg-user] configure features: pkgconfig support
References: <20140404123809.GA12991@sunshine.barsnick.net>
Message-ID: <loom.20140404T144606-495@post.gmane.org>

Moritz Barsnick <barsnick <at> gmx.net> writes:

> How about pkgconfig support for libx264 and libfdk_aac?

I was so far able to prohibit this...
What does not work for you? Could you 
explain the use-case that currently does 
not work but would work with pkgconfig?

> Or is there some legacy (e.g. old versions without .pc) 
> which this would break?

I wouldn't call OSX and mingw legacy...

Carl Eugen


From porparek at gmail.com  Fri Apr  4 14:52:59 2014
From: porparek at gmail.com (Pawel Por)
Date: Fri, 4 Apr 2014 14:52:59 +0200
Subject: [FFmpeg-user] cut at keyframes
In-Reply-To: <CAJmOuEx=CW-P7ut1JYHy-BKN_xW666q_U7p=2+NxQo2-dchfTw@mail.gmail.com>
References: <CAJmOuEyk=vbSaJnS8KCxNG5XP3c=pmA1QmRrBRVuRYNcEZbxjQ@mail.gmail.com>
	<CAJmOuEx=CW-P7ut1JYHy-BKN_xW666q_U7p=2+NxQo2-dchfTw@mail.gmail.com>
Message-ID: <CAJmOuEwChxb5av+4m7r7w3CK0HreQCrGF=eEdwsUJVSDvx7ePQ@mail.gmail.com>

>>* ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3
*>>* -acodec copy -vcodec copy -y out.mp4
*>> >>* I've upgraded to the newer version of ffmpeg 2.2 and I
*>>* found that now when I use the above command it cuts my
*>>* movies not at closest keyframes.
*
>I tested the following with the sample you uploaded
>with FFmpeg 1.0 and current git head:
>$ ffmpeg -ss 1 -i test.mp4 -acodec copy -vcodec copy -t 3 out.mp4
>Both versions produce the same output frames, what do I
>miss?
>Please don't top-post here, it is considered rude.
>Carl Eugen

Thanks for reply.
My goal is to cut my mp4 movie exactly at keyframe.
Is it possible with the most recent version of ffmpeg ?
I've used -noaccurate_seek option but it didn't help.

Please tell me how I can do that.

When it turns out that it is not possible with the most recent version
of ffmpeg I will downgrade to version 1.0.7 and provide you with a
feedback.

Do I top-post ? It looks like I'm not aware of it. I'm sorry.



On Wed, Apr 2, 2014 at 9:16 AM, Pawel Por <porparek at gmail.com> wrote:

> Thaks for reply. I've uploaded test.mp4 (10MB) file into the following
> location:
> http://www.sendspace.pl/en/file/3e79185144b8a84429197cd
>
>
> On Tue, Apr 1, 2014 at 11:28 AM, Pawel Por <porparek at gmail.com> wrote:
>
>> Hi
>> I'm a great fan of ffmpeg. I've been using the version 1.0.7 of ffmpeg
>> and it was cutting my mp4 movies at closest keyframes.
>> It was exactly what I wanted.
>> It was very useful to me. I was using the following command:
>> ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 -acodec copy -vcodec copy
>> -y out.mp4
>>
>> I've upgraded to the newer version of ffmpeg 2.2 and I found that now
>> when I use the above command it cuts my movies not at closest keyframes.
>> I tried "-noaccurate_seek" option (as follows) but it still cuts my
>> movies not at a closest keyframes.
>> ffmpeg -noaccurate_seek -i input.mp4 -ss 1 -t 3 -acodec copy -vcodec copy
>> -y out.mp4
>>
>> I believe that there must be some ffmpeg option to accomplish it (I
>> cannot find it in "man page").
>>
>> Please, write me how I can do that.
>>
>> thanks in advance
>>
>>
>

From georgk66 at gmail.com  Fri Apr  4 15:54:00 2014
From: georgk66 at gmail.com (Georg Korger)
Date: Fri, 4 Apr 2014 15:54:00 +0200
Subject: [FFmpeg-user] wrap to mxf - stream order
In-Reply-To: <loom.20140404T012530-570@post.gmane.org>
References: <CABxxoaXyUjy4j4BHa37xwZzjiLWq=YjMku_1F7oip1QsveaK4Q@mail.gmail.com>
	<loom.20140404T012530-570@post.gmane.org>
Message-ID: <CABxxoaXKzBxuQnSx45Hehsy5+0H14eBu_xrCTOpZ8AkGzS00FQ@mail.gmail.com>

Thank you, Carl Eugen, works great. I tried a lot with the -map, but
did not understand it really



2014-04-04 1:26 GMT+02:00 Carl Eugen Hoyos <cehoyos at ag.or.at>:

> Georg Korger <georgk66 <at> gmail.com> writes:
>
> > ffmpeg -y -i input.mxf -c:v copy -filter_complex amerge
> > c:\temp\test.mxf
>
> ffmpeg -i input.mxf -vcodec copy -filter_complex amerge[out]
> -map 0:0 -map [out] test.mxf
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From info at scenator.de  Fri Apr  4 20:12:33 2014
From: info at scenator.de (scenator)
Date: Fri, 4 Apr 2014 11:12:33 -0700 (PDT)
Subject: [FFmpeg-user] bad snapshot-quality from avchd-videofile
In-Reply-To: <loom.20140404T012014-601@post.gmane.org>
References: <1396559792909-4664636.post@n4.nabble.com>
	<1396560835594-4664638.post@n4.nabble.com>
	<loom.20140404T012014-601@post.gmane.org>
Message-ID: <1396635153684-4664657.post@n4.nabble.com>

Hello,

thank you very much!

when I use ... -qscale 2 an error occoures: Please use -q:a or -q:v, -qscale
is ambiguous

So I tried -q 2 and everything is fine

In relation to the other comments above I got the best results with:
... -vframes 1 -qscale 2 -vf yadif=1

1000 thanks for your advices. 

Best regards,
  Frank



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/bad-snapshot-quality-from-avchd-videofile-tp4664636p4664657.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From info at scenator.de  Fri Apr  4 20:38:17 2014
From: info at scenator.de (scenator)
Date: Fri, 4 Apr 2014 11:38:17 -0700 (PDT)
Subject: [FFmpeg-user] snapshot from videfile: incorrect position
In-Reply-To: <1396560360295-4664637.post@n4.nabble.com>
References: <1396560360295-4664637.post@n4.nabble.com>
Message-ID: <1396636697397-4664658.post@n4.nabble.com>

Hello again,

yes I know, that -ss **after** -i is more precisious than **before** -i, but
then there is no fastforward and I have to wait this 437.96 seconds (7
minutes!!)- thererfore this is no solution for me.

I thought, when I go fast forward to 435.96 and then the last 1.75 sec
seeking the wanted position with "normal play" would deliver me th accurate
position.

Any other ideas??
Best regards,
  Frank





--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/snapshot-from-videfile-incorrect-position-tp4664637p4664658.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From phpdev at ehrhardt.nl  Fri Apr  4 23:07:32 2014
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Fri, 04 Apr 2014 23:07:32 +0200
Subject: [FFmpeg-user] countless dropping frema warnings
References: <ab5nj9t6d8ke9o1f2qgm51mjqm24d2pkee@4ax.com>
Message-ID: <8o7uj9dp7vvcuqo1d50g8e8rvpvj15nibi@4ax.com>

Jan Ehrhardt in gmane.comp.video.ffmpeg.user (Wed, 02 Apr 2014 07:06:01
+0200):
>This commit is leading to countless warnings:
>http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=41e7e46cac507e993af6d656c8d3672ba3f9e304

I ended up with a patch to make the 'Drop' error VERBOSE, just like the
'Dup' error a few lines later.

The patch is here:
https://github.com/Jan-E/MyMpeg/blob/master/ffmpeg_patches/revert_dropping_frame_error.patch

And at the end of this message

Jan

diff --git a/ffmpeg.c b/ffmpeg.c
index 25001f5..2850746 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -895,7 +895,7 @@ static void do_video_out(AVFormatContext *s,
     nb_frames = FFMIN(nb_frames, ost->max_frames - ost->frame_number);
     if (nb_frames == 0) {
         nb_frames_drop++;
-        av_log(NULL, AV_LOG_WARNING,
+        av_log(NULL, AV_LOG_VERBOSE,
                "*** dropping frame %d from stream %d at ts
%"PRId64"\n",
                ost->frame_number, ost->st->index, in_picture->pts);
         return;


From cehoyos at ag.or.at  Fri Apr  4 23:21:57 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 4 Apr 2014 21:21:57 +0000 (UTC)
Subject: [FFmpeg-user] bad snapshot-quality from avchd-videofile
References: <1396559792909-4664636.post@n4.nabble.com>
	<1396560835594-4664638.post@n4.nabble.com>
	<loom.20140404T012014-601@post.gmane.org>
	<1396635153684-4664657.post@n4.nabble.com>
Message-ID: <loom.20140404T232104-261@post.gmane.org>

scenator <info <at> scenator.de> writes:

> when I use ... -qscale 2 an error occoures: 
> Please use -q:a or -q:v, -qscale is ambiguous

Isn't this (much) more a hint than an error?
(I always use -qscale)

Please do not top-post here, it is considered rude.

Carl Eugen


From cehoyos at ag.or.at  Fri Apr  4 23:47:46 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 4 Apr 2014 21:47:46 +0000 (UTC)
Subject: [FFmpeg-user] countless dropping frema warnings
References: <ab5nj9t6d8ke9o1f2qgm51mjqm24d2pkee@4ax.com>
	<8o7uj9dp7vvcuqo1d50g8e8rvpvj15nibi@4ax.com>
Message-ID: <loom.20140404T234720-134@post.gmane.org>

Jan Ehrhardt <phpdev <at> ehrhardt.nl> writes:

> -        av_log(NULL, AV_LOG_WARNING,
> +        av_log(NULL, AV_LOG_VERBOSE,

Too late;-(

Carl Eugen


From cehoyos at ag.or.at  Sat Apr  5 13:25:38 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 5 Apr 2014 11:25:38 +0000 (UTC)
Subject: [FFmpeg-user] Format of YUV files
References: <1396564688.23788.YahooMailNeo@web172304.mail.ir2.yahoo.com>
	<loom.20140404T011645-836@post.gmane.org>
Message-ID: <loom.20140405T132426-170@post.gmane.org>

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

> $ ffmpeg -i singleimage -f image2 -pix_fmt yuv420p out.Y
> $ ffmpeg -i videostream -f image2 -pix_fmt yuv420p out%4d.Y

The option "-f image2" should not be necessary anymore with 
current FFmpeg git head.
(The option "-pix_fmt" is only necessary if you 
want to change it wrt to the colour space of the 
input file.)

Carl Eugen


From cehoyos at ag.or.at  Sat Apr  5 13:28:02 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 5 Apr 2014 11:28:02 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?Decoding_XDCAM_4=3A2=3A2_using_libavcodec?=
	=?utf-8?q?-5=2E5_=3A_error_0xbebbb1b7_=28AVERROR=5FINVALIDDATA=29?=
References: <18047.0687723302$1396519528@news.gmane.org>
Message-ID: <loom.20140405T132729-692@post.gmane.org>

 <francesco <at> bltitalia.com> writes:

> Attempting to decode XDCAM 4:2:2 at 1920x1080 it 
> seems that only first frame (I-Frame) is decoded.

Does it work with ffmpeg (the application)?

Carl Eugen


From elliottbalsley at gmail.com  Sun Apr  6 04:10:51 2014
From: elliottbalsley at gmail.com (Elliott Balsley)
Date: Sat, 5 Apr 2014 19:10:51 -0700
Subject: [FFmpeg-user] frame rate change is too slow
In-Reply-To: <loom.20140404T012916-466@post.gmane.org>
References: <CADzzJYmv8aaAejQX4WKP5sHf+QdK0j4Rni1vMgMRQBGfgKiVSg@mail.gmail.com>
	<loom.20140404T012916-466@post.gmane.org>
Message-ID: <FC9588B4-DF99-4FDE-8872-7B713DB4119A@gmail.com>

No I haven't. Should that make a difference? 

> On Apr 3, 2014, at 16:29, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> 
> Elliott Balsley <elliottbalsley <at> gmail.com> writes:
> 
>> I want to make it play at normal speed, by dropping 3 out of
>> every 4 frames.  I would expect this to encode 4x faster, 
>> but it actually takes about twice as long as a normal encode.
> 
> Did you test to encode a whole file with both 
> methods (without "-t")?
> 
> Carl Eugen
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From brant at mac.com  Sun Apr  6 07:15:02 2014
From: brant at mac.com (Brant Merryman)
Date: Sun, 06 Apr 2014 00:15:02 -0500
Subject: [FFmpeg-user] Problems configuring AAC decoder.
Message-ID: <02DDAA45-AB71-459E-9BA2-E59F9482E417@mac.com>

Hi. I'm trying configure an AAC codec to decode raw AAC audio frames. I opened the codec using aacDecoder_Open(TT_MP4_RAW, 1); Next, I want to configure it. These audio frames are being sent in a custom (non-standard) transport stream. The transport stream tells me a lot of details about itself including what kind of audio is being sent, the sample rate, the number of channels, and so forth.

I need to call aacDecoder_ConfigRaw to set the codec for decoding these frames. According to the documentation I found, the second argument is a Audio Specific Config (ASC). I was able to find the following two pages about the Audio Specific Config:

http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio
http://thompsonng.blogspot.com/2010/03/aac-configuration.html

According to this, the Audio Specific Config consists of:

5 bits: object type
4 bits: frequency index
4 bits: channel configuration

In the stream I'm testing with right now, it has two channels and a sample rate of 44100.


So, if I set object_type to AAC_Main which is 1, and the frequency index to 4 which is 44100, and channels to 2, I use the following to get a value:
UInt16 audio_specific = (object_type << 11) | (frequency_index << 7)| (number_of_channels << 3);

audio_specific is 0xA10 (2576 decimal).

Then I make the following call to aacDecoder_ConfigRaw:

UCHAR asc[2];
memcpy(&audio_specific, &asc, sizeof(asc));

UCHAR a_s_conf[] = { asc[0], asc[1] };
UCHAR * conf[] = { a_s_conf };
UINT conf_len = sizeof(a_s_conf);                    
AAC_DECODER_ERROR dec_err = aacDecoder_ConfigRaw(m_AAC_decoder, conf, &conf_len );

At this point, dec_err is AAC_DEC_UNSUPPORTED_FORMAT.

What am I doing wrong? Is it the way that I am formatting the Audio Specific Config - or the way that I'm feeding it into the call?

Thanks.



From j.ramson at gmail.com  Sun Apr  6 13:17:09 2014
From: j.ramson at gmail.com (Johannes Ramson)
Date: Sun, 6 Apr 2014 13:17:09 +0200
Subject: [FFmpeg-user] feed ffmpeg with quicktime reference?
Message-ID: <CAO3XUFzK6z2LaP7QoCgW1BHWZabxHU+Ldw6uBZPNM2x2m-pgeQ@mail.gmail.com>

is ffmpeg able to use quicktime reference clips (made with mediacomposer
and pointing to files at networkbase storages) as input?
I did some test with following results:

"C:\Users\TTVavid2\Desktop\XDCAM-01_2014-04-04 Knud
Vetten.Copy.01.Copy.01.Sub.m
ov"

C:\Users\TTVavid2\Desktop>set pfad=C:\Users\TTVavid2\Desktop\

C:\Users\TTVavid2\Desktop>set CURRENTDIR=C:\Users\TTVavid2\Desktop

C:\Users\TTVavid2\Desktop>set FILENAME=XDCAM-01_2014-04-04 Knud
Vetten.Copy.01.C
opy.01.Sub

C:\Users\TTVavid2\Desktop>set
in="C:\Users\TTVavid2\Desktop\XDCAM-01_2014-04-04
Knud Vetten.Copy.01.Copy.01.Sub.mov"

C:\Users\TTVavid2\Desktop>C:\TOOLS\ffmpeg\bin\ffmpeg -i
"C:\Users\TTVavid2\Deskt
op\XDCAM-01_2014-04-04 Knud Vetten.Copy.01.Copy.01.Sub.mov"
ffmpeg version N-57821-g1fb3b49 Copyright (c) 2000-2013 the FFmpeg
developers
  built on Nov  2 2013 18:06:38 with gcc 4.8.2 (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-libvidstab
--enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
--enable-libwavp
ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 49.100 / 52. 49.100
  libavcodec     55. 40.101 / 55. 40.101
  libavformat    55. 21.100 / 55. 21.100
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000000000327040] stream 0, error opening alias:
path
='/ttvunity1/tTVUNITY1/Unity1_Avid_Projekte_und_MediaFiles/Unity1_Avid_Projekte_
und_MediaFiles/FAKT_KOPFSCHMERZ/Avid
MediaFiles/MXF/1/Avid_Mob.0V01.533F0533F06A
D.mxf', dir='1', filename='Avid_Mob.0V01.533F0533F06AD.mxf', volume='*',
nlvl_fr
om=-1, nlvl_to=-1
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000000000327040] stream 1, error opening alias:
path
='/ttvunity1/tTVUNITY1/Unity1_Avid_Projekte_und_MediaFiles/Unity1_Avid_Projekte_
und_MediaFiles/FAKT_KOPFSCHMERZ/Avid
MediaFiles/MXF/1/Avid_Mob.0A01.533F0533F06A
D.mxf', dir='1', filename='Avid_Mob.0A01.533F0533F06AD.mxf', volume='*',
nlvl_fr
om=-1, nlvl_to=-1
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000000000327040] stream 2, error opening alias:
path
='/ttvunity1/tTVUNITY1/Unity1_Avid_Projekte_und_MediaFiles/Unity1_Avid_Projekte_
und_MediaFiles/FAKT_KOPFSCHMERZ/Avid
MediaFiles/MXF/1/Avid_Mob.0A02.533F0533F06A
E.mxf', dir='1', filename='Avid_Mob.0A02.533F0533F06AE.mxf', volume='*',
nlvl_fr
om=-1, nlvl_to=-1
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000000000327040] Could not find codec
parameters for
 stream 0 (Video: mpeg2video (xd5c / 0x63356478), 1920x1080, 50015 kb/s):
unspec
ified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize'
options
Guessed Channel Layout for  Input Stream #0.1 : mono
Guessed Channel Layout for  Input Stream #0.2 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'C:\Users\TTVavid2\Desktop\XDCAM-01_2014
-04-04 Knud Vetten.Copy.01.Copy.01.Sub.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2014-04-04 21:12:50
    comment         : Source ID =
060a2b340101010101010f00-13-00-00-00-{533f11ec
-b25c-02c9-060e2b347f7f2a80}
    comment-eng     : Source ID =
060a2b340101010101010f00-13-00-00-00-{533f11ec
-b25c-02c9-060e2b347f7f2a80}
    timecode        : 10:08:09:23
  Duration: 00:00:20.00, start: 0.000000, bitrate: 4 kb/s
    Stream #0:0(eng): Video: mpeg2video (xd5c / 0x63356478), 1920x1080,
50015 kb
/s, 25 fps, 25 tbr, 2500 tbn, 2500 tbc (default)
    Metadata:
      creation_time   : 2014-04-04 21:12:50
      handler_name    : Apple Alias-Datensteuerung
    Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono,
s16,
 768 kb/s (default)
    Metadata:
      creation_time   : 2014-04-04 21:12:50
      handler_name    : Apple Alias-Datensteuerung
    Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono,
s16,
 768 kb/s (default)
    Metadata:
      creation_time   : 2014-04-04 21:12:50
      handler_name    : Apple Alias-Datensteuerung
    Stream #0:3(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
    Metadata:
      creation_time   : 2014-04-04 21:12:51
      handler_name    : Apple Alias-Datensteuerung
      timecode        : 10:08:09:23
The quicktime-ref is pointing to op-atom mxf-files with avc-intra100 in it.
Unfortunately i dont know how to give you samples of these clips, cause if
i pick them out of my network the quicktime-ref wont find them in your
setting - right?

kind regards

-- 
Johannes Ramson

From james.darnley at gmail.com  Sun Apr  6 13:49:13 2014
From: james.darnley at gmail.com (James Darnley)
Date: Sun, 06 Apr 2014 13:49:13 +0200
Subject: [FFmpeg-user] Problems configuring AAC decoder.
In-Reply-To: <02DDAA45-AB71-459E-9BA2-E59F9482E417@mac.com>
References: <02DDAA45-AB71-459E-9BA2-E59F9482E417@mac.com>
Message-ID: <53413F39.10405@gmail.com>

On 2014-04-06 07:15, Brant Merryman wrote:
> UCHAR asc[2];
> memcpy(&audio_specific, &asc, sizeof(asc));
> 
> UCHAR a_s_conf[] = { asc[0], asc[1] };
> UCHAR * conf[] = { a_s_conf };
> UINT conf_len = sizeof(a_s_conf);                    
> AAC_DECODER_ERROR dec_err = aacDecoder_ConfigRaw(m_AAC_decoder, conf, &conf_len );
> 
> At this point, dec_err is AAC_DEC_UNSUPPORTED_FORMAT.
> 
> What am I doing wrong? Is it the way that I am formatting the Audio Specific Config - or the way that I'm feeding it into the call?

This doesn't look like any of FFmpeg's API, certainly not the public one
anyway.  What exactly are you doing?  What/whose code is this?


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From brant at mac.com  Sun Apr  6 17:00:28 2014
From: brant at mac.com (Brant Merryman)
Date: Sun, 06 Apr 2014 10:00:28 -0500
Subject: [FFmpeg-user] Problems configuring AAC decoder.
In-Reply-To: <53413F39.10405@gmail.com>
References: <02DDAA45-AB71-459E-9BA2-E59F9482E417@mac.com>
	<53413F39.10405@gmail.com>
Message-ID: <FC590765-BB69-4B5A-BB41-7AD29EC4490B@mac.com>

The aacDecoder_ConfigRaw call that I'm trying to use is declared in aacdecoder_lib.h which is part of fdk-aac. Is that not part of FFMPEG?

(That would certainly explain part of my confusion.)


On Apr 6, 2014, at 6:49 AM, James Darnley <james.darnley at gmail.com> wrote:

> On 2014-04-06 07:15, Brant Merryman wrote:
>> UCHAR asc[2];
>> memcpy(&audio_specific, &asc, sizeof(asc));
>> 
>> UCHAR a_s_conf[] = { asc[0], asc[1] };
>> UCHAR * conf[] = { a_s_conf };
>> UINT conf_len = sizeof(a_s_conf);                    
>> AAC_DECODER_ERROR dec_err = aacDecoder_ConfigRaw(m_AAC_decoder, conf, &conf_len );
>> 
>> At this point, dec_err is AAC_DEC_UNSUPPORTED_FORMAT.
>> 
>> What am I doing wrong? Is it the way that I am formatting the Audio Specific Config - or the way that I'm feeding it into the call?
> 
> This doesn't look like any of FFmpeg's API, certainly not the public one
> anyway.  What exactly are you doing?  What/whose code is this?
> 
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From cehoyos at ag.or.at  Sun Apr  6 21:03:01 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 6 Apr 2014 19:03:01 +0000 (UTC)
Subject: [FFmpeg-user] feed ffmpeg with quicktime reference?
References: <CAO3XUFzK6z2LaP7QoCgW1BHWZabxHU+Ldw6uBZPNM2x2m-pgeQ@mail.gmail.com>
Message-ID: <loom.20140406T210159-818@post.gmane.org>

Johannes Ramson <j.ramson <at> gmail.com> writes:

> [mov,mp4,m4a,3gp,3g2,mj2  <at>  0000000000327040] stream 0, 
> error opening alias: path
>
='/ttvunity1/tTVUNITY1/Unity1_Avid_Projekte_und_MediaFiles/Unity1_Avid_Projekte_
> und_MediaFiles/FAKT_KOPFSCHMERZ/Avid
> MediaFiles/MXF/1/Avid_Mob.0V01.533F0533F06A
> D.mxf'

Does this path exist on your system?

Carl Eugen


From cehoyos at ag.or.at  Sun Apr  6 21:04:10 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 6 Apr 2014 19:04:10 +0000 (UTC)
Subject: [FFmpeg-user] Problems configuring AAC decoder.
References: <02DDAA45-AB71-459E-9BA2-E59F9482E417@mac.com>
	<53413F39.10405@gmail.com>
	<FC590765-BB69-4B5A-BB41-7AD29EC4490B@mac.com>
Message-ID: <loom.20140406T210320-354@post.gmane.org>

Brant Merryman <brant <at> mac.com> writes:

> The aacDecoder_ConfigRaw call that I'm trying to use 
> is declared in aacdecoder_lib.h which is part of
> fdk-aac. Is that not part of FFMPEG?

Definitely not.

Unrelated: Don't forget that fdk-aac is assumed to be 
non-free by its developers.

Carl Eugen


From j.ramson at gmail.com  Sun Apr  6 22:09:19 2014
From: j.ramson at gmail.com (Johannes Ramson)
Date: Sun, 6 Apr 2014 22:09:19 +0200
Subject: [FFmpeg-user] feed ffmpeg with quicktime reference?
In-Reply-To: <loom.20140406T210159-818@post.gmane.org>
References: <CAO3XUFzK6z2LaP7QoCgW1BHWZabxHU+Ldw6uBZPNM2x2m-pgeQ@mail.gmail.com>
	<loom.20140406T210159-818@post.gmane.org>
Message-ID: <CAO3XUFx1OVYAFWjORZfCubbBDczRxNN1gmvVdKiPrGGvGJXc0g@mail.gmail.com>

Yes, i checked it twice. I also did a test with qt-ref pointing to files at
internal drives - same result.


2014-04-06 21:03 GMT+02:00 Carl Eugen Hoyos <cehoyos at ag.or.at>:

> Johannes Ramson <j.ramson <at> gmail.com> writes:
>
> > [mov,mp4,m4a,3gp,3g2,mj2  <at>  0000000000327040] stream 0,
> > error opening alias: path
> >
>
> ='/ttvunity1/tTVUNITY1/Unity1_Avid_Projekte_und_MediaFiles/Unity1_Avid_Projekte_
> > und_MediaFiles/FAKT_KOPFSCHMERZ/Avid
> > MediaFiles/MXF/1/Avid_Mob.0V01.533F0533F06A
> > D.mxf'
>
> Does this path exist on your system?
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>



-- 
Johannes Ramson

From cehoyos at ag.or.at  Sun Apr  6 22:31:01 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 6 Apr 2014 20:31:01 +0000 (UTC)
Subject: [FFmpeg-user] feed ffmpeg with quicktime reference?
References: <CAO3XUFzK6z2LaP7QoCgW1BHWZabxHU+Ldw6uBZPNM2x2m-pgeQ@mail.gmail.com>
	<loom.20140406T210159-818@post.gmane.org>
	<CAO3XUFx1OVYAFWjORZfCubbBDczRxNN1gmvVdKiPrGGvGJXc0g@mail.gmail.com>
Message-ID: <loom.20140406T223025-979@post.gmane.org>

Johannes Ramson <j.ramson <at> gmail.com> writes:

> Yes, i checked it twice.

What does ls show for the directory in question?

Please don't top-post here, it is considered rude.

Carl Eugen


From pfribeiro at gmail.com  Sun Apr  6 22:33:32 2014
From: pfribeiro at gmail.com (kathodus)
Date: Sun, 6 Apr 2014 13:33:32 -0700 (PDT)
Subject: [FFmpeg-user] Raspberry Pi to YouTube Live
In-Reply-To: <52E59181.4010101@gmail.com>
References: <52E59181.4010101@gmail.com>
Message-ID: <1396816412582-4664673.post@n4.nabble.com>

Hi Daryll,

I came across your post a few days ago while trying to do exactly the same.
I have now succeeded to live stream to Youtube and would like to share my
solution so that others do not need to spend as much time tinkering with
ffmpeg :-)

1) I disregarded Raspbian's ffmpeg version and just downloaded the source
followed by a custom compile. I used the guide in
https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide and changed the script
in http://chrisandjo.com/redletter/compiling-ffmpeg/ to cope with the
appropriate packages. (Namely I was after adding support for libfdk_aac,
which in the end wasn't necessary).

2) Through numerous attempts I worked out that the bitrate sent with audio
being AAC-encoded was very low! (in fact significantly lower than what
raspivid was producing) and thus Youtube refused to take it. The trick is
with the order of the command line arguments to ffmpeg. I used the following
(where XXXXXXX is the youtube RTMP stream identifier):

raspivid -o - -t 0 -w 1280 -h 720 -fps 25 -b 600000 -g 50 | ./ffmpeg -re -ar
44100 -ac 2 -acodec pcm_s16le -f s16le -ac 2 -i /dev/zero -f h264 -i -
-vcodec copy -acodec aac -ab 128k -g 50 -strict experimental -f flv
rtmp://a.rtmp.youtube.com/live2/XXXXXXX

I note the following:

a) '-re' is the right enabler for getting the appropriate bitrate out of the
audio encoding. It seems the Pi can cope with AAC encoding very well. I have
obviously not tested any real audio input as I have no soundcard for it.
b) '-g 50' there is a minimum requirement for the keyframe interval of less
than or equal to 2 seconds. I have been able to work out that this
corresponds to less than or equal to 2*fps. (Note that I could not find the
unit for the -g parameter anywhere, not even for ffmpeg itself. So this is
guessing really.) See Youtube's streaming requirements
https://support.google.com/youtube/answer/2853702?topic=2853713&hl=en.

Despite Youtube's bandwidth requirements. I could easily stream
1080p at 25fps+AAC with only 600~700Kbps of bandwidth available.

c) The audio stream part of the command seems to need to come before the
video part, because of '-re'.

Hope this is useful to people looking into streaming to Youtube live with
the Pi! :-)



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Raspberry-Pi-to-YouTube-Live-tp4663435p4664673.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From horsley1953 at gmail.com  Sun Apr  6 22:43:16 2014
From: horsley1953 at gmail.com (Tom Horsley)
Date: Sun, 6 Apr 2014 16:43:16 -0400
Subject: [FFmpeg-user] Could segments from one mpg have different audio
	offsets?
Message-ID: <20140406164316.183c64e5@zooty>

Sometimes I like to archive recorded programs from my TiVO,
downloading and tovodecoding the .mpg file, then using -ss
and -t to grab and transcode with libx264 the individual
chunks between the commercials.

Every so often, I'll find each individual chunk plays
flawlessly, but when I use the concat filter to stick
them all together, I wind up with audio sync problems
in the resulting concatenated video.

It just occurred to me that I've never checked to
see if the individual chunks all had the same
offset between video and audio streams.

Could that be what is going on? Might something
about the mpg file cause ffmpeg to generate results
where different offsets show up in separately
encoded chunks from the same source?

I'll have to be sure to check that the next time
I have an example with this problem, but I thought
I should ask to see if it is worth looking at,
or, for that matter, if someone else has a better
idea about how perfectly in sync segments could
wind up out of sync when I join them.

From cehoyos at ag.or.at  Sun Apr  6 23:28:12 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 6 Apr 2014 21:28:12 +0000 (UTC)
Subject: [FFmpeg-user] Could segments from one mpg have different audio
	offsets?
References: <20140406164316.183c64e5@zooty>
Message-ID: <loom.20140406T232729-367@post.gmane.org>

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

> Every so often, I'll find each individual chunk plays
> flawlessly, but when I use the concat filter to stick
> them all together, I wind up with audio sync problems
> in the resulting concatenated video.

How can I reproduce this?
(Two short amples, command line and complete, uncut 
console output.)

Carl Eugen


From horsley1953 at gmail.com  Mon Apr  7 00:11:23 2014
From: horsley1953 at gmail.com (Tom Horsley)
Date: Sun, 6 Apr 2014 18:11:23 -0400
Subject: [FFmpeg-user] Could segments from one mpg have different audio
 offsets?
In-Reply-To: <loom.20140406T232729-367@post.gmane.org>
References: <20140406164316.183c64e5@zooty>
	<loom.20140406T232729-367@post.gmane.org>
Message-ID: <20140406181123.63705de5@zooty>

On Sun, 6 Apr 2014 21:28:12 +0000 (UTC)
Carl Eugen Hoyos wrote:

> > Every so often, I'll find each individual chunk plays
> > flawlessly, but when I use the concat filter to stick
> > them all together, I wind up with audio sync problems
> > in the resulting concatenated video.  
> 
> How can I reproduce this?

It seems to depend entirely on the source material.

I use the same scripts and commands for every show I
archive, but every once in a while I see it happen.
I don't have any examples of it laying around at the
moment, I was just wondering if there were any ideas
(or even known problems) that might cause it.

What does the concat filter do if the files being
concatenated have different offsets?

From Alexander.Achleitner at zf.com  Mon Apr  7 08:20:39 2014
From: Alexander.Achleitner at zf.com (Alexander.Achleitner at zf.com)
Date: Mon, 7 Apr 2014 06:20:39 +0000
Subject: [FFmpeg-user] Problem Compiling ffmpeg
In-Reply-To: <20140404101315.GA12382@sunshine.barsnick.net>
References: <7B7B8356F2BD81429073BEF692648A1313A79C1B@FRDS01162.emea.zf-world.com>
	<loom.20140401T232247-362@post.gmane.org>
	<loom.20140402T114020-927@post.gmane.org>
	<7B7B8356F2BD81429073BEF692648A1313A7A0C1@FRDS01162.emea.zf-world.com>
	<20140404101315.GA12382@sunshine.barsnick.net>
Message-ID: <7B7B8356F2BD81429073BEF692648A1313A7A179@FRDS01162.emea.zf-world.com>

Hi Moritz,

thank you and all others for help. I tested ffmpeg on the AM335. It works. For example I can extract pictures 
from an avi file. 
Great.

I think some of my problems came from the point that I have used only Windows in the last ten years.

The configuration script is good, but at some options I didn't know what I have to set.

I always wanted to compile ffmpeg separately and not in Eclipse. I need the libraries for the Project
in the IDE.


Many thanks for help.


Regards,



Alexander Achleitner


-----Urspr?ngliche Nachricht-----
Von: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] Im Auftrag von Moritz Barsnick
Gesendet: Freitag, 4. April 2014 12:13
An: FFmpeg user discussions
Betreff: Re: [FFmpeg-user] Problem Compiling ffmpeg

Hi Alexander,

On Fri, Apr 04, 2014 at 09:44:24 +0000, Alexander.Achleitner at zf.com wrote:
> CC	ffmpeg_opt.o
> CC	ffmpeg_filter.o
> LD	ffmpeg_g
> CP	ffmpeg
> STRIP	ffmpeg
> strip: Unable to recognise the format of the input file `ffmpeg'
> make: *** [ffmpeg] Error 1

You are using the (native) program "strip" and not the (cross) program "arm-cortexa8-linux-gnueabihf-strip".

The former does not handle the cross binaries, and therefore errors out.

The latter would have been used if you had not ignored Carl's hint at using "--cross-prefix".

(I personally don't see the point of configuring and using an IDE to build a project which already has a perfectly working configure/make flow. But that's just me.)

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

From nichot20 at yahoo.com  Mon Apr  7 09:46:59 2014
From: nichot20 at yahoo.com (Tim Nicholson)
Date: Mon, 07 Apr 2014 08:46:59 +0100
Subject: [FFmpeg-user] configure features: pkgconfig support
In-Reply-To: <loom.20140404T144606-495@post.gmane.org>
References: <20140404123809.GA12991@sunshine.barsnick.net>
	<loom.20140404T144606-495@post.gmane.org>
Message-ID: <534257F3.70205@yahoo.com>

On 04/04/14 13:47, Carl Eugen Hoyos wrote:
> Moritz Barsnick <barsnick <at> gmx.net> writes:
> 
>> How about pkgconfig support for libx264 and libfdk_aac?
> 
> I was so far able to prohibit this...
> What does not work for you? Could you 
> explain the use-case that currently does 
> not work but would work with pkgconfig?
> 
>> Or is there some legacy (e.g. old versions without .pc) 
>> which this would break?
> 
> I wouldn't call OSX and mingw legacy...
> 

Interesting as a recent thread on ffmbc irc seemed to suggest that
switching to pc for libx264 helped with the latest versions of mingw..

> Carl Eugen
> [..]

-- 
Tim

From mark.himsley at gmail.com  Mon Apr  7 10:53:05 2014
From: mark.himsley at gmail.com (Mark Himsley)
Date: Mon, 7 Apr 2014 09:53:05 +0100
Subject: [FFmpeg-user] configure features: pkgconfig support
In-Reply-To: <534257F3.70205@yahoo.com>
References: <20140404123809.GA12991@sunshine.barsnick.net>
	<loom.20140404T144606-495@post.gmane.org>
	<534257F3.70205@yahoo.com>
Message-ID: <CAPAFRRRtZAHk3gi3cQyw4nAQs3b1iBuKUf0_AP9_UpN4Fqi=ow@mail.gmail.com>

On 7 Apr 2014 08:47, "Tim Nicholson" <nichot20 at yahoo.com> wrote:
>
> On 04/04/14 13:47, Carl Eugen Hoyos wrote:
> > Moritz Barsnick <barsnick <at> gmx.net> writes:
> >
> >> How about pkgconfig support for libx264 and libfdk_aac?
> >
> > I was so far able to prohibit this...
> > What does not work for you? Could you
> > explain the use-case that currently does
> > not work but would work with pkgconfig?
> >
> >> Or is there some legacy (e.g. old versions without .pc)
> >> which this would break?
> >
> > I wouldn't call OSX and mingw legacy...
>
> Interesting as a recent thread on ffmbc irc seemed to suggest that
> switching to pc for libx264 helped with the latest versions of mingw..


Although, I was talking about cross-compiling with mingw on Linux, and
there I tend to sym-link the Linux pkg-config to
/usr/bin/i586-mingw32msvc-pkg-config so that the --cross-prefix finds a
pkg-config.

I agree that OS-X doesn't include pkg-config by default, but I always
install one from one of the package suppliers. So many libraries require
pkg-config I believe its vital to have it available when compiling.


-- 

Mark

From cehoyos at ag.or.at  Mon Apr  7 11:27:19 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 7 Apr 2014 09:27:19 +0000 (UTC)
Subject: [FFmpeg-user] configure features: pkgconfig support
References: <20140404123809.GA12991@sunshine.barsnick.net>
	<loom.20140404T144606-495@post.gmane.org>
	<534257F3.70205@yahoo.com>
	<CAPAFRRRtZAHk3gi3cQyw4nAQs3b1iBuKUf0_AP9_UpN4Fqi=ow@mail.gmail.com>
Message-ID: <loom.20140407T112431-995@post.gmane.org>

Mark Himsley <mark.himsley <at> gmail.com> writes:

> So many libraries require pkg-config I believe its 
> vital to have it available when compiling.

Do you mean libraries in general or libraries that 
you use with FFmpeg?

Carl Eugen


From phpdev at ehrhardt.nl  Mon Apr  7 12:04:19 2014
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Mon, 07 Apr 2014 12:04:19 +0200
Subject: [FFmpeg-user] configure features: pkgconfig support
References: <20140404123809.GA12991@sunshine.barsnick.net>
	<loom.20140404T144606-495@post.gmane.org>
	<534257F3.70205@yahoo.com>
	<CAPAFRRRtZAHk3gi3cQyw4nAQs3b1iBuKUf0_AP9_UpN4Fqi=ow@mail.gmail.com>
Message-ID: <ort4k9tatducoo5k11nk23hk8ejv9us7t1@4ax.com>

Mark Himsley in gmane.comp.video.ffmpeg.user (Mon, 7 Apr 2014 09:53:05
+0100):
>Although, I was talking about cross-compiling with mingw on Linux, and
>there I tend to sym-link the Linux pkg-config to
>/usr/bin/i586-mingw32msvc-pkg-config so that the --cross-prefix finds a
>pkg-config.

See
http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=5&t=163&sid=11fa1ca48be4d664dccc5cb1e4035e15&start=10#p556

With Zeranoe's script for installing a cross-compile environment on
Linux, it is best to symlink pkg-config as well.

Jan


From lumiwa at gmail.com  Mon Apr  7 14:02:28 2014
From: lumiwa at gmail.com (Ajtim)
Date: Mon, 07 Apr 2014 08:02:28 -0400
Subject: [FFmpeg-user] webm-mpg
Message-ID: <3216739.gx8Bv37gmr@lumiwa.farms.net>

Hi!

My system is FreeBSD and I have problem to plays converted web to dvd format 
on standalone video player. It is "chocking". How I did conversion:



ffmpeg -i xy.webm -aspect 16:9 -target ntsc-dvd dvd.mpg

ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
built on Mar 14 2014 06:32:04 with FreeBSD clang version 3.3
(tags/RELEASE_33/final 183502) 20130610
configuration: --enable-libaacplus --disable-indev=alsa --disable-
outdev=alsa --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-
libass --enable-libcdio --enable-libcelt --enable-libfaac --enable-libfdk-aac
--enable-ffserver --enable-fontconfig --enable-libfreetype --enable-frei0r --
enable-gnutls --enable-libgsm --enable-iconv --enable-libmp3lame --disable-
libbluray --enable-libv4l2 --enable-libmodplug --enable-openal --enable-
libopencv --enable-libopenjpeg --enable-libopus --disable-libpulse --disable-
indev=pulse --disable-outdev=pulse --enable-librtmp --enable-libschroedinger
--enable-libspeex --enable-libtheora --enable-vaapi --enable-vdpau --enable-
libvorbis --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --
enable-libx264 --enable-libxvid --prefix=/usr/local --mandir=/usr/local/man --
datadir=/usr/local/share/ffmpeg --enable-shared --enable-gpl --enable-postproc
--enable-avfilter --enable-avresample --enable-pthreads --enable-memalign-hack
--disable-libstagefright-h264 --disable-libutvideo --disable-libsoxr --cc=cc
--extra-cflags='-msse -I/usr/local/include/vorbis -I/usr/local/include' --
extra-ldflags='-L/usr/local/lib ' --extra-libs=-pthread --disable-debug --
enable-nonfree --enable-version3
libavutil 52. 48.101 / 52. 48.101
libavcodec 55. 39.101 / 55. 39.101
libavformat 55. 19.104 / 55. 19.104
libavdevice 55. 5.100 / 55. 5.100
libavfilter 3. 90.100 / 3. 90.100
libavresample 1. 1. 0 / 1. 1. 0
libswscale 2. 5.101 / 2. 5.101
libswresample 0. 17.104 / 0. 17.104
libpostproc 52. 3.100 / 52. 3.100


Input #0, matroska,webm, from 'jd.webm':
Duration: 01:28:43.57, start: 0.000000, bitrate: 408 kb/s
Stream #0:0: Video: vp8, yuv420p, 576x320, SAR 1:1 DAR 9:5, 1k fps, 25
tbr, 1k tbn, 1k tbc (default)
Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)
Output #0, dvd, to 'dvd.mpg':
Metadata:
encoder : Lavf55.19.104
Stream #0:0: Video: mpeg2video, yuv420p, 720x480 [SAR 32:27 DAR 16:9],
q=2-31, 6000 kb/s, 90k tbn, 29.97 tbc (default)
Stream #0:1: Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s (default)
Stream mapping:
Stream #0:0 -> #0:0 (vp8 -> mpeg2video)
Stream #0:1 -> #0:1 (vorbis -> ac3)
Press [q] to stop, [?] for help
frame=159547 fps=577 q=2.0 Lsize= 1391468kB time=01:28:43.57
bitrate=2141.2kbits/s dup=26458 drop=0
video:1055105kB audio:291134kB subtitle:0 global headers:0kB muxing
overhead 3.359724%

% of overhead is sometimes higher sometimes 2...%


Than I run:


dvdauthor -o dvd/ dvd.mpg
DVDAuthor::dvdauthor, version 0.7.1.
Build options: gnugetopt imagemagick iconv freetype fribidi fontconfig
Send bug reports to <dvdauthor-users at lists.sourceforge.net>

I

INFO: default video format is NTSC
INFO: dvdauthor creating VTS
STAT: Picking VTS 01

STAT: Processing dvd.mpg...
WARN: Audio pts for channel 0 moves backwards by 225; please remultiplex
input.
WARN: Previous sector: 0.531 - 0.595
WARN: Current sector: 0.592 - 0.624
WARN: Audio pts for channel 0 moves backwards by 32; please remultiplex input.
WARN: Previous sector: 0.656 - 0.688
WARN: Current sector: 0.688 - 0.720
WARN: Audio pts for channel 0 moves backwards by 11; please remultiplex input.
WARN: Previous sector: 0.784 - 0.816
WARN: Current sector: 0.816 - 0.880
STAT: VOBU 8816 at 1358MB, 1 PGCs
INFO: Video pts = 0.533 .. 5324.084
INFO: Audio[0] pts = 0.531 .. 5324.112
STAT: VOBU 8817 at 1358MB, 1 PGCs
CHAPTERS: VTS[1/1] 0.000
INFO: Generating VTS with the following video attributes:
INFO: MPEG version: mpeg2
INFO: TV standard: ntsc
INFO: Aspect ratio: 16:9
INFO: Resolution: 720x480
INFO: Audio ch 0 format: ac3/2ch, 48khz drc

STAT: fixed 8817 VOBUs

Sometimes I have muc more "WARN"

and
dvdauthor -o dvd/ -T

and final burn iso with:

growisofs -Z /dev/cd0 -dvd-video dvd/

And as I wrote above when I play DVD on the standalone player it is chocking 
and some are not usable to watch but if I play on computer it works without 
problems.

Thanks in advance.

-- 
ajtiM
--------
http://www.redbubble.com/people/lumiwa

From cehoyos at ag.or.at  Mon Apr  7 14:10:02 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 7 Apr 2014 12:10:02 +0000 (UTC)
Subject: [FFmpeg-user] webm-mpg
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
Message-ID: <loom.20140407T140932-542@post.gmane.org>

Ajtim <lumiwa <at> gmail.com> writes:

> ffmpeg -i xy.webm -aspect 16:9 -target ntsc-dvd dvd.mpg

Could you confirm that you live in a NTSC-country?

Carl Eugen


From lumiwa at gmail.com  Mon Apr  7 14:25:25 2014
From: lumiwa at gmail.com (Ajtim)
Date: Mon, 07 Apr 2014 08:25:25 -0400
Subject: [FFmpeg-user] webm-mpg
In-Reply-To: <loom.20140407T140932-542@post.gmane.org>
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
	<loom.20140407T140932-542@post.gmane.org>
Message-ID: <2781278.6yQgsxU8qn@lumiwa.farms.net>

On Monday 07 April 2014 12:10:02 Carl Eugen Hoyos wrote:
> Ajtim <lumiwa <at> gmail.com> writes:
> > ffmpeg -i xy.webm -aspect 16:9 -target ntsc-dvd dvd.mpg
> 
> Could you confirm that you live in a NTSC-country?
> 
> Carl Eugen
> 

Yes, USA.

-- 
ajtiM
--------
http://www.redbubble.com/people/lumiwa

From cehoyos at ag.or.at  Mon Apr  7 14:44:40 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 7 Apr 2014 12:44:40 +0000 (UTC)
Subject: [FFmpeg-user] cut at keyframes
References: <CAJmOuEyk=vbSaJnS8KCxNG5XP3c=pmA1QmRrBRVuRYNcEZbxjQ@mail.gmail.com>
	<CAJmOuEx=CW-P7ut1JYHy-BKN_xW666q_U7p=2+NxQo2-dchfTw@mail.gmail.com>
	<CAJmOuEwChxb5av+4m7r7w3CK0HreQCrGF=eEdwsUJVSDvx7ePQ@mail.gmail.com>
Message-ID: <loom.20140407T144338-551@post.gmane.org>

Pawel Por <porparek <at> gmail.com> writes:

> My goal is to cut my mp4 movie exactly at keyframe.
> Is it possible with the most recent version of ffmpeg ?

With the sample you uploaded, I only managed to cut at 
keyframe boundaries.
How did you succeed in cutting somewhere else?

As said, please stop top-posting!

Carl Eugen


From cehoyos at ag.or.at  Mon Apr  7 15:40:55 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 7 Apr 2014 13:40:55 +0000 (UTC)
Subject: [FFmpeg-user] webm-mpg
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
Message-ID: <loom.20140407T153941-66@post.gmane.org>

Ajtim <lumiwa <at> gmail.com> writes:

> ffmpeg -i xy.webm -aspect 16:9 -target ntsc-dvd dvd.mpg

> And as I wrote above when I play DVD on the standalone 
> player it is chocking  and some are not usable to watch

Does this only happen for a particular input file or 
for all 25fps input streams like -f lavfi -i testsrc ?

Carl Eugen


From mark.himsley at gmail.com  Mon Apr  7 15:42:14 2014
From: mark.himsley at gmail.com (Mark Himsley)
Date: Mon, 7 Apr 2014 14:42:14 +0100
Subject: [FFmpeg-user] configure features: pkgconfig support
In-Reply-To: <loom.20140407T112431-995@post.gmane.org>
References: <20140404123809.GA12991@sunshine.barsnick.net>
	<loom.20140404T144606-495@post.gmane.org>
	<534257F3.70205@yahoo.com>
	<CAPAFRRRtZAHk3gi3cQyw4nAQs3b1iBuKUf0_AP9_UpN4Fqi=ow@mail.gmail.com>
	<loom.20140407T112431-995@post.gmane.org>
Message-ID: <CAPAFRRTbBXrRT9GArNOUg+pKGf-dDGJaPZyfNYL=7ofw9XSEmQ@mail.gmail.com>

On 7 April 2014 10:27, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Mark Himsley <mark.himsley <at> gmail.com> writes:
>
>> So many libraries require pkg-config I believe its
>> vital to have it available when compiling.
>
> Do you mean libraries in general or libraries that
> you use with FFmpeg?


Hi Carl,

Compiling libraries that I use with FFmpeg.

-- 
Mark

From lumiwa at gmail.com  Mon Apr  7 16:00:41 2014
From: lumiwa at gmail.com (Ajtim)
Date: Mon, 07 Apr 2014 10:00:41 -0400
Subject: [FFmpeg-user] webm-mpg
In-Reply-To: <loom.20140407T153941-66@post.gmane.org>
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
	<loom.20140407T153941-66@post.gmane.org>
Message-ID: <6944999.DObiVSqYxs@lumiwa.farms.net>

On Monday 07 April 2014 13:40:55 Carl Eugen Hoyos wrote:
> Ajtim <lumiwa <at> gmail.com> writes:
> > ffmpeg -i xy.webm -aspect 16:9 -target ntsc-dvd dvd.mpg
> > 
> > And as I wrote above when I play DVD on the standalone
> > player it is chocking  and some are not usable to watch
> 
> Does this only happen for a particular input file or
> for all 25fps input streams like -f lavfi -i testsrc ?
> 
> Carl Eugen
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

I am doing just as I wrote and each dvd has more or less the same problems.
Yesterday I use:
ffmpeg -i jd.webm -aspect 16:9 -target ntsc-dvd -vcodec mpeg1video dvd.mpg

and has problems too.

-- 
ajtiM
--------
http://www.redbubble.com/people/lumiwa

From mark.himsley at gmail.com  Mon Apr  7 17:14:55 2014
From: mark.himsley at gmail.com (Mark Himsley)
Date: Mon, 7 Apr 2014 16:14:55 +0100
Subject: [FFmpeg-user] Raspberry Pi to YouTube Live
In-Reply-To: <1396816412582-4664673.post@n4.nabble.com>
References: <52E59181.4010101@gmail.com>
	<1396816412582-4664673.post@n4.nabble.com>
Message-ID: <CAPAFRRTx11P46AHawi692SU7NNLQ+QEJeESbHEYdA033o61EjA@mail.gmail.com>

Hi,

On 6 April 2014 21:33, kathodus <pfribeiro at gmail.com> wrote:
> (Note that I could not find the
> unit for the -g parameter anywhere, not even for ffmpeg itself. So this is
> guessing really.)

-g is one of the parameters associated with individual video encoders.
If you have a look at this section of the docs you'll see that many
video CODECs describe the 'g' option as "the size of a GOP", or
similar wording:

https://www.ffmpeg.org/ffmpeg-all.html#Video-Encoders

> -g 50' there is a minimum requirement for the keyframe interval of less
> than or equal to 2 seconds. I have been able to work out that this
> corresponds to less than or equal to 2*fps

The -g option to (some) video encoders, sets - in frames - the maximum
size for a GOP. In the example you've given, 50 frames at 25 FPS will
give you a keyframe at least every 2 seconds.

Hope that helps.

-- 
Mark Himsley

From lumiwa at gmail.com  Mon Apr  7 17:33:03 2014
From: lumiwa at gmail.com (Ajtim)
Date: Mon, 07 Apr 2014 11:33:03 -0400
Subject: [FFmpeg-user] webm-mpg
In-Reply-To: <loom.20140407T153941-66@post.gmane.org>
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
	<loom.20140407T153941-66@post.gmane.org>
Message-ID: <11704832.9LLmcIFEb0@lumiwa.farms.net>

On Monday 07 April 2014 13:40:55 Carl Eugen Hoyos wrote:
> Ajtim <lumiwa <at> gmail.com> writes:
> > ffmpeg -i xy.webm -aspect 16:9 -target ntsc-dvd dvd.mpg
> > 
> > And as I wrote above when I play DVD on the standalone
> > player it is chocking  and some are not usable to watch
> 
> Does this only happen for a particular input file or
> for all 25fps input streams like -f lavfi -i testsrc ?
> 
> Carl Eugen
> 
ffmpeg -f lavfi -i testsrc test.mpg
ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Mar 14 2014 06:32:04 with FreeBSD clang version 3.3 
(tags/RELEASE_33/final 183502) 20130610
  configuration: --enable-libaacplus --disable-indev=alsa --disable-outdev=alsa 
--enable-libopencore-amrnb --enable-libopencore-amrwb --disable-libass --
enable-libcdio --enable-libcelt --enable-libfaac --enable-libfdk-aac --enable-
ffserver --enable-fontconfig --enable-libfreetype --enable-frei0r --enable-
gnutls --enable-libgsm --enable-iconv --enable-libmp3lame --disable-libbluray 
--enable-libv4l2 --enable-libmodplug --enable-openal --enable-libopencv --
enable-libopenjpeg --enable-libopus --disable-libpulse --disable-indev=pulse 
--disable-outdev=pulse --enable-librtmp --enable-libschroedinger --enable-
libspeex --enable-libtheora --enable-vaapi --enable-vdpau --enable-libvorbis 
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libx264 
--enable-libxvid --prefix=/usr/local --mandir=/usr/local/man --
datadir=/usr/local/share/ffmpeg --enable-shared --enable-gpl --enable-postproc 
--enable-avfilter --enable-avresample --enable-pthreads --enable-memalign-hack 
--disable-libstagefright-h264 --disable-libutvideo --disable-libsoxr --cc=cc 
--extra-cflags='-msse -I/usr/local/include/vorbis -I/usr/local/include' --
extra-ldflags='-L/usr/local/lib ' --extra-libs=-pthread --disable-debug --
enable-nonfree --enable-version3
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101                                                                                     
  libavformat    55. 19.104 / 55. 19.104                                                                                     
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
Input #0, lavfi, from 'testsrc':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 320x240 [SAR 
1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc
[mpeg @ 0x80e042a20] VBV buffer size not set, muxing may fail
Output #0, mpeg, to 'test.mpg':
  Metadata:
    encoder         : Lavf55.19.104
    Stream #0:0: Video: mpeg1video, yuv420p, 320x240 [SAR 1:1 DAR 4:3], 
q=2-31, 200 kb/s, 90k tbn, 25 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> mpeg1video)
Press [q] to stop, [?] for help
frame=169548 fps=1029 q=8.4 Lsize=  167326kB time=01:53:01.88 bitrate= 
202.1kbits/s    
video:165850kB audio:0kB subtitle:0 global headers:0kB muxing overhead 
0.889996%


-- 
ajtiM
--------
http://www.redbubble.com/people/lumiwa

From jeff at thefirst.org  Mon Apr  7 17:54:58 2014
From: jeff at thefirst.org (Jeff Thompson)
Date: Mon, 07 Apr 2014 08:54:58 -0700
Subject: [FFmpeg-user] sync audio in a single file
Message-ID: <5342CA52.7030003@thefirst.org>

I have an avi file with duration 1 hour. The audio rate is out of sync 
and ends about 2 minutes before the end of the video. I used the ffmpeg 
-async option:

|ffmpeg -i unsynced.avi -i unsynced.avi -acodec adpcm_ms -vcodec copy -map 0:0 -map 1:1 -async 1000 synced.avi|

But the output audio is still out of sync. Am I using the ffmpeg command 
correctly to sync the audio with the video in the one input file?

Thanks,
- Jeff


From cehoyos at ag.or.at  Mon Apr  7 18:26:42 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 7 Apr 2014 16:26:42 +0000 (UTC)
Subject: [FFmpeg-user] webm-mpg
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
	<loom.20140407T153941-66@post.gmane.org>
	<11704832.9LLmcIFEb0@lumiwa.farms.net>
Message-ID: <loom.20140407T182511-757@post.gmane.org>

Ajtim <lumiwa <at> gmail.com> writes:

> > Does this only happen for a particular input file or
> > for all 25fps input streams like -f lavfi -i testsrc ?
>
> ffmpeg -f lavfi -i testsrc test.mpg

I believe the relevant command line would be:
$ ffmpeg -f lavfi -i testsrc -aspect 16:9 -target ntsc-dvd test.mpg

> ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers

And please test current git head.

Carl Eugen


From jklimek at gmail.com  Mon Apr  7 19:48:21 2014
From: jklimek at gmail.com (John Klimek)
Date: Mon, 7 Apr 2014 13:48:21 -0400
Subject: [FFmpeg-user] How can I transcode a video to 800x480,
	non-square pixels?
Message-ID: <CAOEv1kbeXzOFex_E2VzALE=eAa_QP3jYBhzOWMy1jS=O5FCG+w@mail.gmail.com>

I've purchased a car stereo that lists the following dimensions:

6.95 in diagonal (156.6 mm x 81.6 mm)
800x480, 16:9 aspect ratio

If the pixels were square, that would be 15:9 but the physical dimensions
are definitely 16:9.

If I have a video encoded at 720p or 1080p and I want to convert it to use
the entire 800x480 display and the non-square (?) pixels, how can I do this
with FFmpeg?

From lumiwa at gmail.com  Mon Apr  7 19:50:59 2014
From: lumiwa at gmail.com (Ajtim)
Date: Mon, 07 Apr 2014 13:50:59 -0400
Subject: [FFmpeg-user] webm-mpg
In-Reply-To: <loom.20140407T182511-757@post.gmane.org>
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
	<11704832.9LLmcIFEb0@lumiwa.farms.net>
	<loom.20140407T182511-757@post.gmane.org>
Message-ID: <2388653.fcmLg4bu7A@lumiwa.farms.net>

On Monday 07 April 2014 16:26:42 Carl Eugen Hoyos wrote:
> Ajtim <lumiwa <at> gmail.com> writes:
> > > Does this only happen for a particular input file or
> > > for all 25fps input streams like -f lavfi -i testsrc ?
> > 
> > ffmpeg -f lavfi -i testsrc test.mpg
> 
> I believe the relevant command line would be:
> $ ffmpeg -f lavfi -i testsrc -aspect 16:9 -target ntsc-dvd test.mpg
> 
> > ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
> 
> And please test current git head.
> 
> Carl Eugen

 ffmpeg -f lavfi -i testsrc -aspect 16:9 -target ntsc-dvd test.mpg
ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Mar 14 2014 06:32:04 with FreeBSD clang version 3.3 
(tags/RELEASE_33/final 183502) 20130610
  configuration: --enable-libaacplus --disable-indev=alsa --disable-outdev=alsa 
--enable-libopencore-amrnb --enable-libopencore-amrwb --disable-libass --
enable-libcdio --enable-libcelt --enable-libfaac --enable-libfdk-aac --enable-
ffserver --enable-fontconfig --enable-libfreetype --enable-frei0r --enable-
gnutls --enable-libgsm --enable-iconv --enable-libmp3lame --disable-libbluray 
--enable-libv4l2 --enable-libmodplug --enable-openal --enable-libopencv --
enable-libopenjpeg --enable-libopus --disable-libpulse --disable-indev=pulse 
--disable-outdev=pulse --enable-librtmp --enable-libschroedinger --enable-
libspeex --enable-libtheora --enable-vaapi --enable-vdpau --enable-libvorbis 
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx --enable-libx264 
--enable-libxvid --prefix=/usr/local --mandir=/usr/local/man --
datadir=/usr/local/share/ffmpeg --enable-shared --enable-gpl --enable-postproc 
--enable-avfilter --enable-avresample --enable-pthreads --enable-memalign-hack 
--disable-libstagefright-h264 --disable-libutvideo --disable-libsoxr --cc=cc 
--extra-cflags='-msse -I/usr/local/include/vorbis -I/usr/local/include' --
extra-ldflags='-L/usr/local/lib ' --extra-libs=-pthread --disable-debug --
enable-nonfree --enable-version3
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101
  libavformat    55. 19.104 / 55. 19.104
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
Input #0, lavfi, from 'testsrc':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 320x240 [SAR 
1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc
Output #0, dvd, to 'test.mpg':
  Metadata:
    encoder         : Lavf55.19.104
    Stream #0:0: Video: mpeg2video, yuv420p, 720x480 [SAR 32:27 DAR 16:9], 
q=2-31, 6000 kb/s, 90k tbn, 29.97 tbc
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo -> mpeg2video)
Press [q] to stop, [?] for help
frame=163598 fps=685 q=2.0 Lsize=  644534kB time=01:30:58.68 bitrate= 
967.3kbits/s dup=27130 drop=0    
video:608658kB audio:0kB subtitle:0 global headers:0kB muxing overhead 
5.894228%

Thank you for helping me...

-- 
ajtiM
--------
http://www.redbubble.com/people/lumiwa

From lists.davep at gmail.com  Mon Apr  7 19:56:56 2014
From: lists.davep at gmail.com (Dave P)
Date: Mon, 7 Apr 2014 13:56:56 -0400
Subject: [FFmpeg-user] How can I transcode a video to 800x480,
	non-square pixels?
In-Reply-To: <CAOEv1kbeXzOFex_E2VzALE=eAa_QP3jYBhzOWMy1jS=O5FCG+w@mail.gmail.com>
References: <CAOEv1kbeXzOFex_E2VzALE=eAa_QP3jYBhzOWMy1jS=O5FCG+w@mail.gmail.com>
Message-ID: <CANPFk1dhNxYjF3-boNP=7zAqwE81wfAwU7XU0q2r++9JDBDdAw@mail.gmail.com>

On Mon, Apr 7, 2014 at 1:48 PM, John Klimek <jklimek at gmail.com> wrote:

>
> If I have a video encoded at 720p or 1080p and I want to convert it to use
> the entire 800x480 display and the non-square (?) pixels, how can I do this
> with FFmpeg?


-filter:v scale=800:480,setdar=16/9

From lou at lrcd.com  Mon Apr  7 21:09:16 2014
From: lou at lrcd.com (Lou)
Date: Mon, 7 Apr 2014 11:09:16 -0800
Subject: [FFmpeg-user] Raspberry Pi to YouTube Live
In-Reply-To: <1396816412582-4664673.post@n4.nabble.com>
References: <52E59181.4010101@gmail.com>
	<1396816412582-4664673.post@n4.nabble.com>
Message-ID: <20140407110916.7fe96745@lrcd.com>

On Sun, 6 Apr 2014 13:33:32 -0700 (PDT)
kathodus <pfribeiro at gmail.com> wrote:

> b) '-g 50' there is a minimum requirement for the keyframe interval of less
> than or equal to 2 seconds. I have been able to work out that this
> corresponds to less than or equal to 2*fps.

-g will be ignored when you stream copy the video with -vcodec copy.

From jklimek at gmail.com  Mon Apr  7 21:19:20 2014
From: jklimek at gmail.com (John Klimek)
Date: Mon, 7 Apr 2014 15:19:20 -0400
Subject: [FFmpeg-user] How can I transcode a video to 800x480,
	non-square pixels?
In-Reply-To: <CANPFk1dhNxYjF3-boNP=7zAqwE81wfAwU7XU0q2r++9JDBDdAw@mail.gmail.com>
References: <CAOEv1kbeXzOFex_E2VzALE=eAa_QP3jYBhzOWMy1jS=O5FCG+w@mail.gmail.com>
	<CANPFk1dhNxYjF3-boNP=7zAqwE81wfAwU7XU0q2r++9JDBDdAw@mail.gmail.com>
Message-ID: <CAOEv1ka7GLB3G8KmVKr8UbShucik6eXnogYFxrw_29YMiCgkDw@mail.gmail.com>

Thanks!  Does the input resolution or aspect ratio matter?


On Mon, Apr 7, 2014 at 1:56 PM, Dave P <lists.davep at gmail.com> wrote:

> On Mon, Apr 7, 2014 at 1:48 PM, John Klimek <jklimek at gmail.com> wrote:
>
> >
> > If I have a video encoded at 720p or 1080p and I want to convert it to
> use
> > the entire 800x480 display and the non-square (?) pixels, how can I do
> this
> > with FFmpeg?
>
>
> -filter:v scale=800:480,setdar=16/9
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From 690271929 at qq.com  Tue Apr  8 10:20:14 2014
From: 690271929 at qq.com (=?ISO-8859-1?B?SmFja3k=?=)
Date: Tue, 8 Apr 2014 16:20:14 +0800
Subject: [FFmpeg-user] how can i compile ffmpeg with dshow on windows
In-Reply-To: <CAOEv1ka7GLB3G8KmVKr8UbShucik6eXnogYFxrw_29YMiCgkDw@mail.gmail.com>
References: <CAOEv1kbeXzOFex_E2VzALE=eAa_QP3jYBhzOWMy1jS=O5FCG+w@mail.gmail.com>
	<CANPFk1dhNxYjF3-boNP=7zAqwE81wfAwU7XU0q2r++9JDBDdAw@mail.gmail.com>
	<CAOEv1ka7GLB3G8KmVKr8UbShucik6eXnogYFxrw_29YMiCgkDw@mail.gmail.com>
Message-ID: <tencent_7A78F6C4352BF8B8355B79BB@qq.com>

hi,all:
     i compile ffmpeg with dshow on windows using MinGW.
the config: --configure --enable-indevs, but there is no dshow in the configure result.
How can i compile ffmpeg with dshow??

From werner.robitza at gmail.com  Tue Apr  8 12:07:46 2014
From: werner.robitza at gmail.com (Werner Robitza)
Date: Tue, 8 Apr 2014 12:07:46 +0200
Subject: [FFmpeg-user] Assembling files from several input files
Message-ID: <CAN0LTaM_p3q7=5FdBVzGC3NKDv6jU_SiCR_vcu818OkEro224w@mail.gmail.com>

Let's assume I have four files:

video1.mp4
video2.mp4
audio1.m4a
audio2.m4a

What would a simple command be that could concatenate/mux the files in such
a way that it contains:

video1 from 0 to 20 seconds, video 2 from 20 to 30 seconds
audio1 from 0 to 10 seconds, audio2 from 10 to 30

Basically, I need to freely mux/concatenate parts of different sources. I
know I could cut the video parts individually before, then concat them into
a final video, then do the same with audio, and finally encode and mux
everything, but would it be possible to do that in one filter expression?

From 690271929 at qq.com  Tue Apr  8 12:13:16 2014
From: 690271929 at qq.com (=?ISO-8859-1?B?SmFja3k=?=)
Date: Tue, 8 Apr 2014 18:13:16 +0800
Subject: [FFmpeg-user] Assembling files from several input files
In-Reply-To: <CAN0LTaM_p3q7=5FdBVzGC3NKDv6jU_SiCR_vcu818OkEro224w@mail.gmail.com>
References: <CAN0LTaM_p3q7=5FdBVzGC3NKDv6jU_SiCR_vcu818OkEro224w@mail.gmail.com>
Message-ID: <tencent_4A4CEB671F885143667947A5@qq.com>

there no simple command.
1) concat video,concat audio
2)ffmpeg -i video -i audio -c copy outfile




------------------ Original ------------------
From:  "Werner Robitza";<werner.robitza at gmail.com>;
Date:  Tue, Apr 8, 2014 06:07 PM
To:  "ffmpeg-user"<ffmpeg-user at ffmpeg.org>; 

Subject:  [FFmpeg-user] Assembling files from several input files



Let's assume I have four files:

video1.mp4
video2.mp4
audio1.m4a
audio2.m4a

What would a simple command be that could concatenate/mux the files in such
a way that it contains:

video1 from 0 to 20 seconds, video 2 from 20 to 30 seconds
audio1 from 0 to 10 seconds, audio2 from 10 to 30

Basically, I need to freely mux/concatenate parts of different sources. I
know I could cut the video parts individually before, then concat them into
a final video, then do the same with audio, and finally encode and mux
everything, but would it be possible to do that in one filter expression?
_______________________________________________
ffmpeg-user mailing list
ffmpeg-user at ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user
.

From cehoyos at ag.or.at  Tue Apr  8 17:10:58 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 8 Apr 2014 15:10:58 +0000 (UTC)
Subject: [FFmpeg-user] webm-mpg
References: <3216739.gx8Bv37gmr@lumiwa.farms.net>
	<11704832.9LLmcIFEb0@lumiwa.farms.net>
	<loom.20140407T182511-757@post.gmane.org>
	<2388653.fcmLg4bu7A@lumiwa.farms.net>
Message-ID: <loom.20140408T170902-449@post.gmane.org>

Ajtim <lumiwa <at> gmail.com> writes:

>  ffmpeg -f lavfi -i testsrc -aspect 16:9 -target ntsc-dvd test.mpg

Does the output file show the same problems as 
you reported originally?

> ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers

As said:
This is old and it has very limited usefulness (for you!) 
to test this version: If the problem gets fixed, you will 
have to update and before it gets fixed, you will have to 
confirm that it is still reproducible with current git 
head.

Carl Eugen


From rogerdpack2 at gmail.com  Tue Apr  8 20:03:51 2014
From: rogerdpack2 at gmail.com (Roger Pack)
Date: Tue, 8 Apr 2014 12:03:51 -0600
Subject: [FFmpeg-user] how can i compile ffmpeg with dshow on windows
In-Reply-To: <tencent_7A78F6C4352BF8B8355B79BB@qq.com>
References: <CAOEv1kbeXzOFex_E2VzALE=eAa_QP3jYBhzOWMy1jS=O5FCG+w@mail.gmail.com>
	<CANPFk1dhNxYjF3-boNP=7zAqwE81wfAwU7XU0q2r++9JDBDdAw@mail.gmail.com>
	<CAOEv1ka7GLB3G8KmVKr8UbShucik6eXnogYFxrw_29YMiCgkDw@mail.gmail.com>
	<tencent_7A78F6C4352BF8B8355B79BB@qq.com>
Message-ID: <CAL1QdWdQWntYCEBAdw=3DR-7+P03Ea8S40NZ-p9gLGK1mA3j6g@mail.gmail.com>

It should "just work" if you have the dshow headers:

http://betterlogic.com/roger/2013/09/directx-headers-for-mingw/

On 4/8/14, Jacky <690271929 at qq.com> wrote:
> hi,all:
>      i compile ffmpeg with dshow on windows using MinGW.
> the config: --configure --enable-indevs, but there is no dshow in the
> configure result.
> How can i compile ffmpeg with dshow??
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From 413872259 at qq.com  Tue Apr  8 05:09:36 2014
From: 413872259 at qq.com (=?gb18030?B?wO654tD3?=)
Date: Tue, 8 Apr 2014 11:09:36 +0800
Subject: [FFmpeg-user] config
Message-ID: <tencent_44EAF8B438011D3A2D49C885@qq.com>


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From Dieter.Herzmann at topix.ch  Mon Apr  7 23:04:00 2014
From: Dieter.Herzmann at topix.ch (Dieter Herzmann)
Date: Mon, 7 Apr 2014 23:04:00 +0200
Subject: [FFmpeg-user] Urgent problem with ffmpeg installation on OS X
 10.9.2, Xcode Version 5.1 (5B130a)
Message-ID: <CF68DCCE.21EA4%dieter.herzmann@topix.ch>

Manual for installation
http://sangatpedas.com/20140218/installing-ffmpeg-osx-mavericks/

Slightly changed script for installation
build_ffmpeg.sh

Log file with error
build_ffmpeg.log


I need a very fast solution on this.


Many thanks


Kind regards

Dieter Herzmann



Topix AG  |  Bildstrasse 5  |  CH-9015 St. Gallen

Phone +41 71 313 80 40  |  Fax +41 71 313 80 44  |  Hotline +41 71 313 80 42

dieter.herzmann at topix.ch<mailto:dieter.herzmann at topix.ch>  |  www.topix.ch<http://www.topix.ch>



+++ Topix AG ? IT-Solutions for Media Production +++
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From james.darnley at gmail.com  Tue Apr  8 20:21:36 2014
From: james.darnley at gmail.com (James Darnley)
Date: Tue, 08 Apr 2014 20:21:36 +0200
Subject: [FFmpeg-user] Urgent problem with ffmpeg installation on OS X
 10.9.2, Xcode Version 5.1 (5B130a)
In-Reply-To: <CF68DCCE.21EA4%dieter.herzmann@topix.ch>
References: <CF68DCCE.21EA4%dieter.herzmann@topix.ch>
Message-ID: <53443E30.9080600@gmail.com>

On 2014-04-07 23:04, Dieter Herzmann wrote:
> Manual for installation
> http://sangatpedas.com/20140218/installing-ffmpeg-osx-mavericks/
> 
> Slightly changed script for installation
> build_ffmpeg.sh
> 
> Log file with error
> build_ffmpeg.log

What exactly have you given us?  That script isn't part of FFmpeg and
that log isn't from configure.

> ERROR: libtheora 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.

The log that you provided tells you the "problem" and gives you a
suggests what you should do, which you have completely ignored.

Did you install libtheora?  Did you install the headers if they are
packaged separately?  Do you even need libtheora?  Leave the option out
if you don't.

And just why are you disabling mmx and by extension all later SIMD?

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From wally at voosen.eu  Tue Apr  8 20:35:49 2014
From: wally at voosen.eu (wally)
Date: Tue, 08 Apr 2014 20:35:49 +0200
Subject: [FFmpeg-user] mpg to mp4 request
Message-ID: <53444185.50609@voosen.eu>

Hello,
i want to ask for help on converting a mpg videofile recorded from TV
to a mp4 format my SAT Receiver can play, (mp4).
Below are the format infos of source and target output by ffmpeg -i:
All done on Linux.

Which ffmpeg commandline do this job ?

source:

     Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 720x576 
[SAR 64:45 DAR 16:9], max. 9500 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
     Stream #0:2[0x83]: Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s

target:

     Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), 
yuv420p(tv), 720x406 [SAR 1:1 DAR 360:203], 1894 kb/s, 25 fps, 25 tbr, 
25k tbn, 50 tbc (default)
     Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 
fltp, 93 kb/s (default)


 > ffmpeg -i ./source.mpg
ffmpeg version 2.2 Copyright (c) 2000-2014 the FFmpeg developers
   built on Mar 24 2014 17:15:07 with gcc 4.8 (SUSE Linux)
   configuration: --shlibdir=/usr/lib --prefix=/usr 
--mandir=/usr/share/man --libdir=/usr/lib --enable-shared 
--disable-static --enable-debug --disable-stripping 
--extra-cflags='-fomit-frame-pointer -fmessage-length=0 
-grecord-gcc-switches -fstack-protector -O2 -Wall -D_FORTIFY_SOURCE=2 
-funwind-tables -fasynchronous-unwind-tables -g -D_LARGEFILE_SOURCE 
-D_FILE_OFFSET_BITS=64 -I/usr/include/gsm' --enable-gpl --enable-x11grab 
--enable-version3 --enable-pthreads --enable-avfilter --enable-libpulse 
--enable-libvpx --enable-libopus --enable-libass --enable-libx265 
--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 --enable-gnutls
   libavutil      52. 66.100 / 52. 66.100
   libavcodec     55. 52.102 / 55. 52.102
   libavformat    55. 33.100 / 55. 33.100
   libavdevice    55. 10.100 / 55. 10.100
   libavfilter     4.  2.100 /  4.  2.100
   libavresample   1.  2.  0 /  1.  2.  0
   libswscale      2.  5.102 /  2.  5.102
   libswresample   0. 18.100 /  0. 18.100
   libpostproc    52.  3.100 / 52.  3.100
[NULL @ 0x82baf20] start time is not set in estimate_timings_from_pts
Input #0, mpeg, from './Rossini.mpg':
   Duration: 01:49:08.19, start: 0.252622, bitrate: 4815 kb/s
     Stream #0:0[0x1bf]: Data: dvd_nav_packet
     Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 720x576 
[SAR 64:45 DAR 16:9], max. 9500 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
     Stream #0:2[0x83]: Audio: ac3, 48000 Hz, stereo, fltp, 448 kb/s
At least one output file must be specified
 >



 > ffmpeg -i ./target.mp4
ffmpeg version 2.2 Copyright (c) 2000-2014 the FFmpeg developers
   built on Mar 24 2014 17:15:07 with gcc 4.8 (SUSE Linux)
   configuration: --shlibdir=/usr/lib --prefix=/usr 
--mandir=/usr/share/man --libdir=/usr/lib --enable-shared 
--disable-static --enable-debug --disable-stripping 
--extra-cflags='-fomit-frame-pointer -fmessage-length=0 
-grecord-gcc-switches -fstack-protector -O2 -Wall -D_FORTIFY_SOURCE=2 
-funwind-tables -fasynchronous-unwind-tables -g -D_LARGEFILE_SOURCE 
-D_FILE_OFFSET_BITS=64 -I/usr/include/gsm' --enable-gpl --enable-x11grab 
--enable-version3 --enable-pthreads --enable-avfilter --enable-libpulse 
--enable-libvpx --enable-libopus --enable-libass --enable-libx265 
--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 --enable-gnutls
   libavutil      52. 66.100 / 52. 66.100
   libavcodec     55. 52.102 / 55. 52.102
   libavformat    55. 33.100 / 55. 33.100
   libavdevice    55. 10.100 / 55. 10.100
   libavfilter     4.  2.100 /  4.  2.100
   libavresample   1.  2.  0 /  1.  2.  0
   libswscale      2.  5.102 /  2.  5.102
   libswresample   0. 18.100 /  0. 18.100
   libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './target.mp4':
   Metadata:
     major_brand     : mp42
     minor_version   : 0
     compatible_brands: isommp42
     creation_time   : 2014-03-31 22:03:51
   Duration: 01:20:35.84, start: 0.000000, bitrate: 1992 kb/s
     Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), 
yuv420p(tv), 720x406 [SAR 1:1 DAR 360:203], 1894 kb/s, 25 fps, 25 tbr, 
25k tbn, 50 tbc (default)
     Metadata:
       creation_time   : 2014-03-31 22:03:51
       handler_name    : Mainconcept MP4 Video Media Handler
     Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, 
fltp, 93 kb/s (default)
     Metadata:
       creation_time   : 2014-03-31 22:03:51
       handler_name    : Mainconcept MP4 Sound Media Handler
At least one output file must be specified
 >


From lou at lrcd.com  Tue Apr  8 20:48:21 2014
From: lou at lrcd.com (Lou)
Date: Tue, 8 Apr 2014 10:48:21 -0800
Subject: [FFmpeg-user] Assembling files from several input files
In-Reply-To: <CAN0LTaM_p3q7=5FdBVzGC3NKDv6jU_SiCR_vcu818OkEro224w@mail.gmail.com>
References: <CAN0LTaM_p3q7=5FdBVzGC3NKDv6jU_SiCR_vcu818OkEro224w@mail.gmail.com>
Message-ID: <20140408104821.252f90ea@lrcd.com>

On Tue, 8 Apr 2014 12:07:46 +0200
Werner Robitza <werner.robitza at gmail.com> wrote:

> Let's assume I have four files:
> 
> video1.mp4
> video2.mp4
> audio1.m4a
> audio2.m4a
> 
> What would a simple command be that could concatenate/mux the files in such
> a way that it contains:
> 
> video1 from 0 to 20 seconds, video 2 from 20 to 30 seconds
> audio1 from 0 to 10 seconds, audio2 from 10 to 30
> 
> Basically, I need to freely mux/concatenate parts of different sources. I
> know I could cut the video parts individually before, then concat them into
> a final video, then do the same with audio, and finally encode and mux
> everything, but would it be possible to do that in one filter expression?

Hi Werner,

Maybe not simple, but this can be done with the (a)trim, (a)setpts, and
concat filters:

ffmpeg -i v1.mp4 -i v2.mp4 -i a1.m4a -i a2.m4a -filter_complex \
"[0:v]trim=0:20,setpts=PTS-STARTPTS[v0]; \
 [1:v]trim=20:30,setpts=PTS-STARTPTS[v1]; \
 [2:a]atrim=0:10,asetpts=PTS-STARTPTS[a0]; \
 [3:a]atrim=10:30,asetpts=PTS-STARTPTS[a1]; \
 [v0][v1]concat=n=2:v=1:a=0[vcat]; \
 [a0][a1]concat=n=2:v=0:a=1[acat]" \
-map "[vcat]" -map "[acat]" output.mp4

You may need to add other filters like aformat if the inputs are
too dissimilar.

From mrfun.china at gmail.com  Wed Apr  9 03:43:53 2014
From: mrfun.china at gmail.com (YIRAN LI)
Date: Wed, 9 Apr 2014 11:43:53 +1000
Subject: [FFmpeg-user] Can't set --extra-ldflags=-Wl,--strip-all
Message-ID: <CAN16yyODXAmpRayH_N5JTW+Pkd3RkD-AAfaMiGMRHnmtZubz3A@mail.gmail.com>

Hi,

I tried to set --strip-all option to configure but failed, while other
options like -Wl, -M could be correctly set. (build ffmpeg dlls on MinGW
for Windows)

According to attached config.log
gcc -Wl,--strip-all -o /tmp/ffconf.pVuzGaQU.exe /tmp/ffconf.GDKMVvFe.o
failed.
But I ran same command on MingW and everything is fine.

hxuanyu at hxuanyu-PC /c/sourcecode/components/thecomponents/ffmpeg16
$ gcc -c -o test.o test.c

hxuanyu at hxuanyu-PC /c/sourcecode/components/thecomponents/ffmpeg16
$ gcc -Wl,--strip-all -o test.exe test.o

hxuanyu at hxuanyu-PC /c/sourcecode/components/thecomponents/ffmpeg16
$ which gcc; gcc --version
/mingw/bin/gcc.exe
gcc.exe (GCC) 4.8.1
Copyright (C) 2013 Free Software Foundation, Inc.
This is free software; see the source for copying conditions.  There is NO
warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.

Thanks
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From reuben.m at gmail.com  Wed Apr  9 06:26:51 2014
From: reuben.m at gmail.com (Reuben Martin)
Date: Tue, 8 Apr 2014 23:26:51 -0500
Subject: [FFmpeg-user] post-encode stream mapping
Message-ID: <CAC9da+6gOMN5VqbBKU4TMXx8uX_irTraLcySio4nHPsrEGTEuQ@mail.gmail.com>

Is it possible to map an encoded stream to multiple muxers for the purpose
of encoding once for multiple outputs?

I want to encode 3 separate outputs from the same input. (3 different
resolutions of the same video) However the audio will be encoded the same
for all 3. So it is possible to encode the audio once, and pass it to all 3
of the muxers? I know re-encoding the same audio 3 times for each output is
not all that intensive, but this is for live video, and the system is
almost at limit. Any small reduction in process load is a plus.

(Stream-copy is not an option in this scenario.)

-Reuben

From werner.robitza at gmail.com  Wed Apr  9 09:32:14 2014
From: werner.robitza at gmail.com (Werner Robitza)
Date: Wed, 9 Apr 2014 09:32:14 +0200
Subject: [FFmpeg-user] Assembling files from several input files
In-Reply-To: <20140408104821.252f90ea@lrcd.com>
References: <CAN0LTaM_p3q7=5FdBVzGC3NKDv6jU_SiCR_vcu818OkEro224w@mail.gmail.com>
	<20140408104821.252f90ea@lrcd.com>
Message-ID: <CAN0LTaNbeWQ1=3HL0g7zMfRf1QtYHY186Rb-69L0bfFeWGjXQQ@mail.gmail.com>

On Tue, Apr 8, 2014 at 8:48 PM, Lou <lou at lrcd.com> wrote:
> Maybe not simple, but this can be done with the (a)trim, (a)setpts, and
> concat filters:
>
> ffmpeg -i v1.mp4 -i v2.mp4 -i a1.m4a -i a2.m4a -filter_complex \
> "[0:v]trim=0:20,setpts=PTS-STARTPTS[v0]; \
>  [1:v]trim=20:30,setpts=PTS-STARTPTS[v1]; \
>  [2:a]atrim=0:10,asetpts=PTS-STARTPTS[a0]; \
>  [3:a]atrim=10:30,asetpts=PTS-STARTPTS[a1]; \
>  [v0][v1]concat=n=2:v=1:a=0[vcat]; \
>  [a0][a1]concat=n=2:v=0:a=1[acat]" \
> -map "[vcat]" -map "[acat]" output.mp4

Thank you, Lou. Perhaps I can come up with a way to generate such an
expression procedurally.
If anyone has suggestions on other tools that allow such a form of
concatenation more easily, I'd be happy to hear about that.

> You may need to add other filters like aformat if the inputs are
> too dissimilar.

Yeah, I was planning on decoding everything into raw formats before.

From ulzii.ict at gmail.com  Wed Apr  9 09:37:03 2014
From: ulzii.ict at gmail.com (Ulziisaikhan L)
Date: Wed, 9 Apr 2014 15:37:03 +0800
Subject: [FFmpeg-user] mosaic stream stops
Message-ID: <CABXT5GvMXK-1f-eK_EVTM5oRJz5hG61oW_q0g33spsk73+0khg@mail.gmail.com>

dear all,


I'm streaming mosaic from UDP multicast.
mosaic have 6 and 4 windows but both of them stream normally up to 2 day.
actually 6 streams overloading buffer then I reduced 4 stream but still
terminated. any idea ?


*4 window mosaic stream*
/root/bin/ffmpeg -f mpegts -i 'udp://
238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.15:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.40:8001?fifo_size
=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.45:8001?fifo_size=1000000&overrun_nonfatal=1' -filter_complex
"nullsrc=size=1024x600 [base]; [0:v] setpts=PTS-STARTPTS, scale=512x300
[upperleft
]; [1:v] setpts=PTS-STARTPTS, scale=512x300 [upperright]; [2:v]
setpts=PTS-STARTPTS, scale=512x300 [lowerleft]; [3:v] setpts=PTS-STARTPTS,
scale=512x300 [lowerright]; [base][upperleft] overla
y=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=512 [tmp2];
[tmp2][lowerleft] overlay=shortest=1:y=300 [tmp3]; [tmp3][lowerright]
overlay=shortest=1:x=512:y=300" -strict experime
ntal -vcodec libx264 -preset ultrafast -profile:v main -an -f mpegts udp://
238.1.1.42:8001?pkt_size=1316


*6 windows mosaic stream*
/root/bin/ffmpeg -f mpegts -i 'udp://
238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.27:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.15:8001?fifo_size
=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.124:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.14:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.40:8001?fifo_si
ze=1000000&overrun_nonfatal=1' -filter_complex "nullsrc=size=1024x600
[base]; [0:v] setpts=PTS-STARTPTS, scale=341x300 [upperleft]; [1:v]
setpts=PTS-STARTPTS, scale=341x300 [uppermiddle]; [2:
v] setpts=PTS-STARTPTS, scale=341x300 [upperright]; [3:v]
setpts=PTS-STARTPTS, scale=341x300 [lowerleft]; [4:v] setpts=PTS-STARTPTS,
scale=341x300 [lowermiddle]; [5:v] setpts=PTS-STARTPTS, sc
ale=341x300 [lowerright]; [base][upperleft] overlay=shortest=1 [tmp1];
[tmp1] [uppermiddle] overlay=shortest=1:x=341 [tmp2]; [tmp2] [upperright]
overlay=shortest=1:x=682 [tmp3]; [tmp3] [lower
left] overlay=shortest=1:y=300 [tmp4]; [tmp4] [lowermiddle]
overlay=shortest=1:x=341:y=300 [tmp5]; [tmp5] [lowerright]
overlay=shortest=1:x=682:y=300" -strict experimental -vcodec libx264 -pr
eset ultrafast -profile:v main -an -f mpegts udp://
238.1.1.42:8001?pkt_size=1316


thanks

From lifeamit at gmail.com  Wed Apr  9 09:41:53 2014
From: lifeamit at gmail.com (Amit Agarwal)
Date: Wed, 9 Apr 2014 13:11:53 +0530
Subject: [FFmpeg-user] Buffer overflow problem in ffmpeg
Message-ID: <CAPMi8WYitukWiyS_oARDjphHuTnLmfHPDRWF2CyeKY_=oxGCZg@mail.gmail.com>

I am trying to overlay a video onto another video by using filter_complex.
But it doesn't do overlay properly and says 'Buffer queue overflow'. It
works fine if I remove delay by not using -itsoffset. Please let me know
what can I do to overcome this problem. I am using latest version of ffmpeg.
It is 2.2 N-62296-g4484152.

*Here is the command I am trying. *
ffmpeg -i .Files/1szqpq/hjmuqD/creation/overlaid-video.mp4 -itsoffset
22.952 -i
.Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm
-filter_complex "overlay=x='if(gte(t,34.28),NAN,0)':y=0" -y
.Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4


*And here is the output log.*

ffmpeg version 2.2-tessus Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar 24 2014 22:25:11 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
  configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm
--extra-version=tessus --disable-shared --enable-static --disable-ffplay
--enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame
--enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid
--enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libxavs --enable-version3
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx
--enable-libgsm --enable-libopus --enable-libmodplug --enable-fontconfig
--enable-libfreetype --enable-libass --enable-libbluray --enable-filters
--enable-runtime-cpudetect
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'.Files/1szqpq/hjmuqD/creation/overlaid-video.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf55.33.100
  Duration: 00:00:44.91, start: 0.000000, bitrate: 443 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], 392 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn,
59.94 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
fltp, 128 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Input #1, matroska,webm, from
'.Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm':
  Metadata:
    encoder         : libebml v1.2.0 + libmatroska v1.1.0
    creation_time   : 2014-04-09 06:16:15
  Duration: 00:00:11.33, start: 0.000000, bitrate: 996 kb/s
    Stream #1:0: Video: vp8, yuv420p, 274x486, SAR 1:1 DAR 137:243, 30.05
fps, 30.05 tbr, 1k tbn, 1k tbc (default)
    Stream #1:1: Audio: vorbis, 48000 Hz, stereo, fltp (default)
[libx264 @ 0x104000600] using SAR=1/1
[libx264 @ 0x104000600] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
AVX
[libx264 @ 0x104000600] profile High, level 3.1
[libx264 @ 0x104000600] 264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft
2003-2014 - 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=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=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 '.Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf55.33.100
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 29.97 tbc (default)
    Stream #0:1(und): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
44100 Hz, stereo, s16, 128 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 (h264) -> overlay:main (graph 0)
  Stream #1:0 (vp8) -> overlay:overlay (graph 0)
  overlay (graph 0) -> Stream #0:0 (libx264)
  Stream #0:1 -> #0:1 (aac -> libvo_aacenc)
Press [q] to stop, [?] for help
[Parsed_overlay_0 @ 0x103800000] [framesync @ 0x1038000e8] Buffer queue
overflow, dropping.
    Last message repeated 624 times
frame= 1345 fps=181 q=-1.0 Lsize=    2130kB time=00:00:44.81 bitrate=
389.4kbits/s dup=624 drop=0
video:1851kB audio:251kB subtitle:0 data:0 global headers:0kB muxing
overhead 1.348262%
[libx264 @ 0x104000600] frame I:8     Avg QP:13.93  size: 30628
[libx264 @ 0x104000600] frame P:402   Avg QP:17.93  size:  2802
[libx264 @ 0x104000600] frame B:935   Avg QP:20.63  size:   560
[libx264 @ 0x104000600] consecutive B-frames:  2.4% 14.4%  1.1% 82.1%
[libx264 @ 0x104000600] mb I  I16..4:  9.0% 73.4% 17.6%
[libx264 @ 0x104000600] mb P  I16..4:  1.7%  1.7%  0.3%  P16..4: 10.9%
 2.3%  0.9%  0.0%  0.0%    skip:82.2%
[libx264 @ 0x104000600] mb B  I16..4:  0.1%  0.1%  0.0%  B16..8:  7.6%
 0.4%  0.0%  direct: 0.1%  skip:91.7%  L0:46.5% L1:50.8% BI: 2.8%
[libx264 @ 0x104000600] 8x8 transform intra:53.1% inter:78.0%
[libx264 @ 0x104000600] coded y,uvDC,uvAC intra: 31.4% 36.8% 7.9% inter:
1.1% 2.3% 0.0%
[libx264 @ 0x104000600] i16 v,h,dc,p: 31% 27%  7% 35%
[libx264 @ 0x104000600] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 41% 20% 23%  3%  3%
 3%  3%  3%  3%
[libx264 @ 0x104000600] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 25% 14%  3%  4%
 4%  4%  3%  3%
[libx264 @ 0x104000600] i8c dc,h,v,p: 74% 12% 11%  2%
[libx264 @ 0x104000600] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x104000600] ref P L0: 68.4%  8.7% 17.1%  5.8%
[libx264 @ 0x104000600] ref B L0: 85.5% 13.7%  0.8%
[libx264 @ 0x104000600] ref B L1: 97.5%  2.5%
[libx264 @ 0x104000600] kb/s:337.78

From cehoyos at ag.or.at  Wed Apr  9 10:14:30 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 9 Apr 2014 08:14:30 +0000 (UTC)
Subject: [FFmpeg-user] Can't set --extra-ldflags=-Wl,--strip-all
References: <CAN16yyODXAmpRayH_N5JTW+Pkd3RkD-AAfaMiGMRHnmtZubz3A@mail.gmail.com>
Message-ID: <loom.20140409T100654-790@post.gmane.org>

YIRAN LI <mrfun.china <at> gmail.com> writes:

> I tried to set --strip-all option to configure but failed

Since the build system strips by default, you don't have to 
change anything.
If you don't want stripping, add --disable-stripping, 
if you want stripping, don't add any options.

Please understand that generally, you get no support here 
if you add funny extra-cflags or extra-ldflags. If you 
believe that configure is missing some option to improve 
your build experience, please tell us here on the mailing 
list.
(But I am still against making pkg-config mandatory, 
imo this would open a can of worms.)

Carl Eugen


From cehoyos at ag.or.at  Wed Apr  9 10:20:06 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 9 Apr 2014 08:20:06 +0000 (UTC)
Subject: [FFmpeg-user] post-encode stream mapping
References: <CAC9da+6gOMN5VqbBKU4TMXx8uX_irTraLcySio4nHPsrEGTEuQ@mail.gmail.com>
Message-ID: <loom.20140409T101937-790@post.gmane.org>

Reuben Martin <reuben.m <at> gmail.com> writes:

> Is it possible to map an encoded stream to multiple 
> muxers for the purpose of encoding once for multiple 
> outputs?

Did you test the tee muxer?

Carl Eugen


From ulzii.ict at gmail.com  Wed Apr  9 11:13:50 2014
From: ulzii.ict at gmail.com (Ulziisaikhan L)
Date: Wed, 9 Apr 2014 17:13:50 +0800
Subject: [FFmpeg-user] mosaic stream stops
In-Reply-To: <CABXT5GvMXK-1f-eK_EVTM5oRJz5hG61oW_q0g33spsk73+0khg@mail.gmail.com>
References: <CABXT5GvMXK-1f-eK_EVTM5oRJz5hG61oW_q0g33spsk73+0khg@mail.gmail.com>
Message-ID: <CABXT5GvkaA109jmhpXD+ahn7COdMd+Lce3ezO3dkDnnF4gFE=w@mail.gmail.com>

your kind reply will be highly appreciated

thanks


On Wed, Apr 9, 2014 at 3:37 PM, Ulziisaikhan L <ulzii.ict at gmail.com> wrote:

> dear all,
>
>
> I'm streaming mosaic from UDP multicast.
> mosaic have 6 and 4 windows but both of them stream normally up to 2 day.
> actually 6 streams overloading buffer then I reduced 4 stream but still
> terminated. any idea ?
>
>
> *4 window mosaic stream*
> /root/bin/ffmpeg -f mpegts -i 'udp://
> 238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.15:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.40:8001?fifo_size
> =1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.45:8001?fifo_size=1000000&overrun_nonfatal=1' -filter_complex
> "nullsrc=size=1024x600 [base]; [0:v] setpts=PTS-STARTPTS, scale=512x300
> [upperleft
> ]; [1:v] setpts=PTS-STARTPTS, scale=512x300 [upperright]; [2:v]
> setpts=PTS-STARTPTS, scale=512x300 [lowerleft]; [3:v] setpts=PTS-STARTPTS,
> scale=512x300 [lowerright]; [base][upperleft] overla
> y=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=512 [tmp2];
> [tmp2][lowerleft] overlay=shortest=1:y=300 [tmp3]; [tmp3][lowerright]
> overlay=shortest=1:x=512:y=300" -strict experime
> ntal -vcodec libx264 -preset ultrafast -profile:v main -an -f mpegts udp://
> 238.1.1.42:8001?pkt_size=1316
>
>
> *6 windows mosaic stream*
> /root/bin/ffmpeg -f mpegts -i 'udp://
> 238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.27:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.15:8001?fifo_size
> =1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.124:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.14:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
> 238.1.1.40:8001?fifo_si
> ze=1000000&overrun_nonfatal=1' -filter_complex "nullsrc=size=1024x600
> [base]; [0:v] setpts=PTS-STARTPTS, scale=341x300 [upperleft]; [1:v]
> setpts=PTS-STARTPTS, scale=341x300 [uppermiddle]; [2:
> v] setpts=PTS-STARTPTS, scale=341x300 [upperright]; [3:v]
> setpts=PTS-STARTPTS, scale=341x300 [lowerleft]; [4:v] setpts=PTS-STARTPTS,
> scale=341x300 [lowermiddle]; [5:v] setpts=PTS-STARTPTS, sc
> ale=341x300 [lowerright]; [base][upperleft] overlay=shortest=1 [tmp1];
> [tmp1] [uppermiddle] overlay=shortest=1:x=341 [tmp2]; [tmp2] [upperright]
> overlay=shortest=1:x=682 [tmp3]; [tmp3] [lower
> left] overlay=shortest=1:y=300 [tmp4]; [tmp4] [lowermiddle]
> overlay=shortest=1:x=341:y=300 [tmp5]; [tmp5] [lowerright]
> overlay=shortest=1:x=682:y=300" -strict experimental -vcodec libx264 -pr
> eset ultrafast -profile:v main -an -f mpegts udp://
> 238.1.1.42:8001?pkt_size=1316
>
>
> thanks
>

From ffmpeg-list at dubistmeinheld.de  Wed Apr  9 14:55:01 2014
From: ffmpeg-list at dubistmeinheld.de (Takashi)
Date: Wed, 09 Apr 2014 14:55:01 +0200
Subject: [FFmpeg-user] Is FLAC 8 Bit supported?
Message-ID: <53454325.3070104@dubistmeinheld.de>

I'm trying to create an 8 Bit FLAC audio file using ffmpeg, but without
success. (It's working with -sample_fmt s16 but not with -sample_fmt
u8). What am I doing wrong? Is this a bug or feature request or not
possible? (8 Bit is at least in the specs:
https://xiph.org/flac/format.html)

/usr/local/bin/ffmpeg -y -i test.mp3 -sample_fmt u8 test.flac
ffmpeg version N-62299-g60ef0c6 Copyright (c) 2000-2014 the FFmpeg
developers
  built on Apr  9 2014 14:43:15 with gcc 4.7 (SUSE Linux)
  configuration: --enable-gpl --enable-version3 --enable-libopenjpeg
--enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame
--enable-libopus --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libvorbis --enable-libtheora --enable-libvpx --enable-libspeex
--enable-libxvid --enable-libx264 --disable-network --disable-stripping
--enable-libschroedinger --enable-libgsm --enable-avfilter
--enable-pthreads --disable-ffserver --extra-cflags=-I/usr/local/include
--extra-ldflags=-L/usr/local/lib --disable-protocol=httpproxy
--enable-bzlib --enable-libvpx --enable-zlib
  libavutil      52. 75.100 / 52. 75.100
  libavcodec     55. 58.103 / 55. 58.103
  libavformat    55. 36.102 / 55. 36.102
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mp3, from 'test.mp3':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.33.100
  Duration: 00:00:55.25, start: 0.025057, bitrate: 128 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
[flac @ 0x2ac3300] Specified sample format u8 is invalid or not supported
Output #0, flac, to 'test.flac':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf54.33.100
    Stream #0:0: Audio: flac, 44100 Hz, stereo, u8, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 -> flac)
Error while opening encoder for output stream #0:0 - maybe incorrect
parameters such as bit_rate, rate, width or height

From cehoyos at ag.or.at  Wed Apr  9 15:21:05 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 9 Apr 2014 13:21:05 +0000 (UTC)
Subject: [FFmpeg-user] Is FLAC 8 Bit supported?
References: <53454325.3070104@dubistmeinheld.de>
Message-ID: <loom.20140409T151945-680@post.gmane.org>

Takashi <ffmpeg-list <at> dubistmeinheld.de> writes:

> I'm trying to create an 8 Bit FLAC audio file using ffmpeg

Only 16 and 32 bit are supported.
If you need lower quality, transcode to pcm_u8 or pcm_s8 first.

Carl Eugen


From gcboney at bellsouth.net  Wed Apr  9 16:12:43 2014
From: gcboney at bellsouth.net (George Boney)
Date: Wed, 09 Apr 2014 10:12:43 -0400
Subject: [FFmpeg-user] How to set <StreamVideoSampleAccess> for flv stream
Message-ID: <5345555B.8080107@bellsouth.net>

Hello,

I have an ActionScript Application that connects to FFSERVER (run by a 
different entity) to display an FLV stream. Since I need to 'bitmap' 
operations, I need to set sample access

In Wowza, in Application.xml, I can set:
	  <StreamVideoSampleAccess>*</StreamVideoSampleAccess>

In AMS/FMS, I can set:
	client.videoSampleAccess = "publicdomain";

How do I set this for a flv stream in FFMPEG?

George
Additional Details:

When my application issues a ?bitmapData.draw? in ActionScript, it 
returns an error ?SecurityError: Error #2122? or ?SecurityError: Error 
2123?.

These errors are Adobe Sandbox errors. ActionScript default security 
does not allow you to access the bitmap data just because you can access 
the stream. It is not completely clear from the documentation, but I 
think it occurs because my swf file is loaded from a different domain 
than the video stream.

According to Adobe documentation, this error can be resolved by either:
a) a crossdomain.xml file from the same URL as the video stream, or
b) the publisher setting ?videoSampleAccess? in the stream. (see
http://help.adobe.com/en_US/FlashMediaServer/3.5_SS_ASD/WS5b3ccc516d4fbf351e63e3d11a11afc95e-7ec3.html#WS5b3ccc516d4fbf351e63e3d11a11afc95e-7fcb 
)

The people that administer the FFSERVER are willing to work with me on 
this issue, but after a month or two of searching, we still don?t know 
how to implement either of these solutions. I suspect solution (b) is 
the best method, but I would expect there to be some parameter we set in 
the <stream> configuration to enable videoSampleAccess (either directly 
or through Metadata), and we were unable to find one.

There are some comments on the web that discuss the publisher setting 
?RTSPSampleAccess, True, True?, but this is an FLV stream so I don?t 
think that would apply in this casel.

Any suggestions would be helpful. I was going to look at the source for 
configuring a FLV stream and just see if there were any undocumented 
parameters, but I am not that familiar with the FFSERVER source and have 
not had the time.

Thanks,
George




From porparek at gmail.com  Wed Apr  9 16:40:53 2014
From: porparek at gmail.com (Pawel Por)
Date: Wed, 9 Apr 2014 16:40:53 +0200
Subject: [FFmpeg-user] cut at keyframes
Message-ID: <CAJmOuExOi-SvhcZAtomG1rHKHyaA1PmpLaaB7jjDud+YdrGrSw@mail.gmail.com>

>>* My goal is to cut my mp4 movie exactly at keyframe.
*>>* Is it possible with the most recent version of ffmpeg ?
*
> With the sample you uploaded, I only managed to cut at
> keyframe boundaries.
> How did you succeed in cutting somewhere else?

You are right. I also did not succeed cutting somewhere else than at the
keyframe boundary.
It looks like I was confused by the brightness of the cut point. When I use
eg
-ss 0.5 -t 3
and
-ss 1 -t 3

I got the same beginning cut point which is always the keyframe but for
some reasons one of them is darker. I was confused at it when a movie was
quite static.
Anyway it works fine now for me.

From lifeamit at gmail.com  Wed Apr  9 18:12:14 2014
From: lifeamit at gmail.com (Amit Agarwal)
Date: Wed, 9 Apr 2014 21:42:14 +0530
Subject: [FFmpeg-user] Buffer overflow problem in ffmpeg
In-Reply-To: <CAPMi8WYitukWiyS_oARDjphHuTnLmfHPDRWF2CyeKY_=oxGCZg@mail.gmail.com>
References: <CAPMi8WYitukWiyS_oARDjphHuTnLmfHPDRWF2CyeKY_=oxGCZg@mail.gmail.com>
Message-ID: <CAPMi8WZS8fovKpGTtyC26Qyab0zkTn5YngQj-YBJheuzrDy6Dg@mail.gmail.com>

Hi,

Please some one help me on this. I am really blocked on this.


On Wed, Apr 9, 2014 at 1:11 PM, Amit Agarwal <lifeamit at gmail.com> wrote:

> I am trying to overlay a video onto another video by using filter_complex.
> But it doesn't do overlay properly and says 'Buffer queue overflow'. It
> works fine if I remove delay by not using -itsoffset. Please let me know
> what can I do to overcome this problem. I am using latest version of ffmpeg.
> It is 2.2 N-62296-g4484152.
>
> *Here is the command I am trying. *
> ffmpeg -i .Files/1szqpq/hjmuqD/creation/overlaid-video.mp4 -itsoffset
> 22.952 -i
> .Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm
> -filter_complex "overlay=x='if(gte(t,34.28),NAN,0)':y=0" -y
> .Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4
>
>
> *And here is the output log.*
>
> ffmpeg version 2.2-tessus Copyright (c) 2000-2014 the FFmpeg developers
>   built on Mar 24 2014 22:25:11 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
>   configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm
> --extra-version=tessus --disable-shared --enable-static --disable-ffplay
> --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame
> --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid
> --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libxavs --enable-version3
> --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx
> --enable-libgsm --enable-libopus --enable-libmodplug --enable-fontconfig
> --enable-libfreetype --enable-libass --enable-libbluray --enable-filters
> --enable-runtime-cpudetect
>   libavutil      52. 66.100 / 52. 66.100
>   libavcodec     55. 52.102 / 55. 52.102
>   libavformat    55. 33.100 / 55. 33.100
>   libavdevice    55. 10.100 / 55. 10.100
>   libavfilter     4.  2.100 /  4.  2.100
>   libswscale      2.  5.102 /  2.  5.102
>   libswresample   0. 18.100 /  0. 18.100
>   libpostproc    52.  3.100 / 52.  3.100
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
> '.Files/1szqpq/hjmuqD/creation/overlaid-video.mp4':
>   Metadata:
>     major_brand     : isom
>     minor_version   : 512
>     compatible_brands: isomiso2avc1mp41
>     encoder         : Lavf55.33.100
>   Duration: 00:00:44.91, start: 0.000000, bitrate: 443 kb/s
>     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 392 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn,
> 59.94 tbc (default)
>     Metadata:
>       handler_name    : VideoHandler
>     Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
> fltp, 128 kb/s (default)
>     Metadata:
>       handler_name    : SoundHandler
> Input #1, matroska,webm, from
> '.Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm':
>   Metadata:
>     encoder         : libebml v1.2.0 + libmatroska v1.1.0
>     creation_time   : 2014-04-09 06:16:15
>   Duration: 00:00:11.33, start: 0.000000, bitrate: 996 kb/s
>     Stream #1:0: Video: vp8, yuv420p, 274x486, SAR 1:1 DAR 137:243, 30.05
> fps, 30.05 tbr, 1k tbn, 1k tbc (default)
>     Stream #1:1: Audio: vorbis, 48000 Hz, stereo, fltp (default)
> [libx264 @ 0x104000600] using SAR=1/1
> [libx264 @ 0x104000600] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> AVX
> [libx264 @ 0x104000600] profile High, level 3.1
> [libx264 @ 0x104000600] 264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft
> 2003-2014 - 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=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=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 '.Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4':
>   Metadata:
>     major_brand     : isom
>     minor_version   : 512
>     compatible_brands: isomiso2avc1mp41
>     encoder         : Lavf55.33.100
>     Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 29.97 tbc (default)
>     Stream #0:1(und): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
> 44100 Hz, stereo, s16, 128 kb/s (default)
>     Metadata:
>       handler_name    : SoundHandler
> Stream mapping:
>   Stream #0:0 (h264) -> overlay:main (graph 0)
>   Stream #1:0 (vp8) -> overlay:overlay (graph 0)
>   overlay (graph 0) -> Stream #0:0 (libx264)
>   Stream #0:1 -> #0:1 (aac -> libvo_aacenc)
> Press [q] to stop, [?] for help
> [Parsed_overlay_0 @ 0x103800000] [framesync @ 0x1038000e8] Buffer queue
> overflow, dropping.
>     Last message repeated 624 times
> frame= 1345 fps=181 q=-1.0 Lsize=    2130kB time=00:00:44.81 bitrate=
> 389.4kbits/s dup=624 drop=0
> video:1851kB audio:251kB subtitle:0 data:0 global headers:0kB muxing
> overhead 1.348262%
> [libx264 @ 0x104000600] frame I:8     Avg QP:13.93  size: 30628
> [libx264 @ 0x104000600] frame P:402   Avg QP:17.93  size:  2802
> [libx264 @ 0x104000600] frame B:935   Avg QP:20.63  size:   560
> [libx264 @ 0x104000600] consecutive B-frames:  2.4% 14.4%  1.1% 82.1%
> [libx264 @ 0x104000600] mb I  I16..4:  9.0% 73.4% 17.6%
> [libx264 @ 0x104000600] mb P  I16..4:  1.7%  1.7%  0.3%  P16..4: 10.9%
>  2.3%  0.9%  0.0%  0.0%    skip:82.2%
> [libx264 @ 0x104000600] mb B  I16..4:  0.1%  0.1%  0.0%  B16..8:  7.6%
>  0.4%  0.0%  direct: 0.1%  skip:91.7%  L0:46.5% L1:50.8% BI: 2.8%
> [libx264 @ 0x104000600] 8x8 transform intra:53.1% inter:78.0%
> [libx264 @ 0x104000600] coded y,uvDC,uvAC intra: 31.4% 36.8% 7.9% inter:
> 1.1% 2.3% 0.0%
> [libx264 @ 0x104000600] i16 v,h,dc,p: 31% 27%  7% 35%
> [libx264 @ 0x104000600] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 41% 20% 23%  3%  3%
>  3%  3%  3%  3%
> [libx264 @ 0x104000600] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 25% 14%  3%  4%
>  4%  4%  3%  3%
> [libx264 @ 0x104000600] i8c dc,h,v,p: 74% 12% 11%  2%
> [libx264 @ 0x104000600] Weighted P-Frames: Y:0.0% UV:0.0%
> [libx264 @ 0x104000600] ref P L0: 68.4%  8.7% 17.1%  5.8%
> [libx264 @ 0x104000600] ref B L0: 85.5% 13.7%  0.8%
> [libx264 @ 0x104000600] ref B L1: 97.5%  2.5%
> [libx264 @ 0x104000600] kb/s:337.78
>

From cehoyos at ag.or.at  Wed Apr  9 21:22:57 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 9 Apr 2014 19:22:57 +0000 (UTC)
Subject: [FFmpeg-user] mpg to mp4 request
References: <53444185.50609@voosen.eu>
Message-ID: <loom.20140409T211826-560@post.gmane.org>

wally <wally <at> voosen.eu> writes:

> Which ffmpeg commandline do this job ?

Did you test the obvious?
$ ffmpeg -i source.mpg -vcodec libx264 -vb 1.9M -acodec aac 
-strict -2 -ab 90k out.mp4

>    Metadata:
>      major_brand     : mp42
>      minor_version   : 0
>      compatible_brands: isommp42

FFmpeg only writes isom files, you can overwrite this, 
but I am not sure that makes the output file mp42.

Carl Eugen


From elliottbalsley at gmail.com  Thu Apr 10 04:53:37 2014
From: elliottbalsley at gmail.com (Elliott Balsley)
Date: Wed, 9 Apr 2014 19:53:37 -0700
Subject: [FFmpeg-user] Variables in bash script causes 'Unrecognized option'
Message-ID: <CADzzJYmz1e9yL-+u5XDs_PNiJa=YF9L16Sy=3inQeyjXzWix8A@mail.gmail.com>

I want to use ffmpeg in a bash script with variables, like this:

ffpreopts="-loglevel warning"
ffmpeg "$ffpreopts"

But it fails with error: "Unrecognized option 'loglevel warning'.
This is only an example, I've tried various options besides loglevel
and none work when passed in a variable like this.

The command "ffmpeg -loglevel warning" works fine in a Terminal.  So
what's wrong with my syntax?

Full output from script:

ffmpeg version 2.1.git-ffb7d71 Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar  7 2014 16:54:41 with Apple LLVM version 5.0
(clang-500.2.79) (based on LLVM 3.3svn)
  configuration: --prefix=/opt/local --enable-swscale
--enable-avfilter --enable-avresample --enable-libmp3lame
--enable-libvorbis --enable-libopus --enable-libtheora
--enable-libschroedinger --enable-libopenjpeg --enable-libmodplug
--enable-libvpx --enable-libspeex --enable-libass --enable-libbluray
--enable-gnutls --enable-fontconfig --enable-libfreetype
--disable-indev=jack --disable-outdev=xv --mandir=/opt/local/share/man
--enable-shared --enable-pthreads --cc=/usr/bin/cc --arch=x86_64
--enable-yasm --enable-gpl --enable-postproc --enable-libx264
--enable-libxvid --enable-nonfree --enable-libfdk-aac --enable-libfaac
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libavresample   1.  2.  0 /  1.  2.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Unrecognized option 'loglevel warning'.
Error splitting the argument list: Option not found

From blacktrash at gmx.net  Thu Apr 10 08:01:26 2014
From: blacktrash at gmx.net (Christian Ebert)
Date: Thu, 10 Apr 2014 07:01:26 +0100
Subject: [FFmpeg-user] Variables in bash script causes 'Unrecognized
 option'
In-Reply-To: <CADzzJYmz1e9yL-+u5XDs_PNiJa=YF9L16Sy=3inQeyjXzWix8A@mail.gmail.com>
References: <CADzzJYmz1e9yL-+u5XDs_PNiJa=YF9L16Sy=3inQeyjXzWix8A@mail.gmail.com>
Message-ID: <20140410060126.GB867@krille.blacktrash.org>

* Elliott Balsley on Wednesday, April 09, 2014 at 19:53:37 -0700
> I want to use ffmpeg in a bash script with variables, like this:
> 
> ffpreopts="-loglevel warning"
> ffmpeg "$ffpreopts"

ffmpeg $ffpreopts

Try unquoting the variable in your script.

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

From nichot20 at yahoo.com  Thu Apr 10 08:48:39 2014
From: nichot20 at yahoo.com (Tim Nicholson)
Date: Thu, 10 Apr 2014 07:48:39 +0100
Subject: [FFmpeg-user] Urgent problem with ffmpeg installation on OS X
 10.9.2, Xcode Version 5.1 (5B130a)
In-Reply-To: <CF68DCCE.21EA4%dieter.herzmann@topix.ch>
References: <CF68DCCE.21EA4%dieter.herzmann@topix.ch>
Message-ID: <53463EC7.5080101@yahoo.com>

On 07/04/14 22:04, Dieter Herzmann wrote:
> Manual for installation
> http://sangatpedas.com/20140218/installing-ffmpeg-osx-mavericks/
> 
> Slightly changed script for installation
> build_ffmpeg.sh
> 
> Log file with error
> build_ffmpeg.log
> 
> 
> I need a very fast solution on this.
> 
> 
> Many thanks
> 
> 
> Kind regards
> 
> Dieter Herzmann
> 
> [...]


I suggest you try:-

http://hexeract.wordpress.com/2009/04/12/how-to-compile-ffmpegmplayer-for-macosx/

Noting some of the tweaks you need to make for Maverics and Xcode.
(mostly due to clang 'features')
-- 
Tim

From cehoyos at ag.or.at  Thu Apr 10 09:28:16 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 10 Apr 2014 07:28:16 +0000 (UTC)
Subject: [FFmpeg-user] Urgent problem with ffmpeg installation on OS X
	10.9.2, Xcode Version 5.1 (5B130a)
References: <CF68DCCE.21EA4%dieter.herzmann@topix.ch>
	<53463EC7.5080101@yahoo.com>
Message-ID: <loom.20140410T092713-130@post.gmane.org>

Tim Nicholson <nichot20 <at> yahoo.com> writes:

> I suggest you try:-
> 
>
http://hexeract.wordpress.com/2009/04/12/how-to-compile-ffmpegmplayer-for-macosx/

The script seems to have several problems, what 
specific "tweaks" does it show that seem useful to you?

And does openal really work for? I wasn't successful...

Carl Eugen


From lifeamit at gmail.com  Thu Apr 10 11:49:54 2014
From: lifeamit at gmail.com (Amit Agarwal)
Date: Thu, 10 Apr 2014 15:19:54 +0530
Subject: [FFmpeg-user] Buffer overflow problem in ffmpeg
In-Reply-To: <CAPMi8WYitukWiyS_oARDjphHuTnLmfHPDRWF2CyeKY_=oxGCZg@mail.gmail.com>
References: <CAPMi8WYitukWiyS_oARDjphHuTnLmfHPDRWF2CyeKY_=oxGCZg@mail.gmail.com>
Message-ID: <CAPMi8WbQiXo4LV2aTRPAJjTt2ufOg9Q-OGm5cTTLAQMsLBqBgw@mail.gmail.com>

Hi,

Could some one please give me some hint on how can I fix this ?


On Wed, Apr 9, 2014 at 1:11 PM, Amit Agarwal <lifeamit at gmail.com> wrote:

> I am trying to overlay a video onto another video by using filter_complex.
> But it doesn't do overlay properly and says 'Buffer queue overflow'. It
> works fine if I remove delay by not using -itsoffset. Please let me know
> what can I do to overcome this problem. I am using latest version of ffmpeg.
> It is 2.2 N-62296-g4484152.
>
> *Here is the command I am trying. *
> ffmpeg -i .Files/1szqpq/hjmuqD/creation/overlaid-video.mp4 -itsoffset
> 22.952 -i
> .Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm
> -filter_complex "overlay=x='if(gte(t,34.28),NAN,0)':y=0" -y
> .Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4
>
>
> *And here is the output log.*
>
> ffmpeg version 2.2-tessus Copyright (c) 2000-2014 the FFmpeg developers
>   built on Mar 24 2014 22:25:11 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
>   configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm
> --extra-version=tessus --disable-shared --enable-static --disable-ffplay
> --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame
> --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid
> --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libxavs --enable-version3
> --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx
> --enable-libgsm --enable-libopus --enable-libmodplug --enable-fontconfig
> --enable-libfreetype --enable-libass --enable-libbluray --enable-filters
> --enable-runtime-cpudetect
>   libavutil      52. 66.100 / 52. 66.100
>   libavcodec     55. 52.102 / 55. 52.102
>   libavformat    55. 33.100 / 55. 33.100
>   libavdevice    55. 10.100 / 55. 10.100
>   libavfilter     4.  2.100 /  4.  2.100
>   libswscale      2.  5.102 /  2.  5.102
>   libswresample   0. 18.100 /  0. 18.100
>   libpostproc    52.  3.100 / 52.  3.100
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
> '.Files/1szqpq/hjmuqD/creation/overlaid-video.mp4':
>   Metadata:
>     major_brand     : isom
>     minor_version   : 512
>     compatible_brands: isomiso2avc1mp41
>     encoder         : Lavf55.33.100
>   Duration: 00:00:44.91, start: 0.000000, bitrate: 443 kb/s
>     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], 392 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn,
> 59.94 tbc (default)
>     Metadata:
>       handler_name    : VideoHandler
>     Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
> fltp, 128 kb/s (default)
>     Metadata:
>       handler_name    : SoundHandler
> Input #1, matroska,webm, from
> '.Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm':
>   Metadata:
>     encoder         : libebml v1.2.0 + libmatroska v1.1.0
>     creation_time   : 2014-04-09 06:16:15
>   Duration: 00:00:11.33, start: 0.000000, bitrate: 996 kb/s
>     Stream #1:0: Video: vp8, yuv420p, 274x486, SAR 1:1 DAR 137:243, 30.05
> fps, 30.05 tbr, 1k tbn, 1k tbc (default)
>     Stream #1:1: Audio: vorbis, 48000 Hz, stereo, fltp (default)
> [libx264 @ 0x104000600] using SAR=1/1
> [libx264 @ 0x104000600] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> AVX
> [libx264 @ 0x104000600] profile High, level 3.1
> [libx264 @ 0x104000600] 264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft
> 2003-2014 - 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=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=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 '.Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4':
>   Metadata:
>     major_brand     : isom
>     minor_version   : 512
>     compatible_brands: isomiso2avc1mp41
>     encoder         : Lavf55.33.100
>     Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
> 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 29.97 tbc (default)
>     Stream #0:1(und): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
> 44100 Hz, stereo, s16, 128 kb/s (default)
>     Metadata:
>       handler_name    : SoundHandler
> Stream mapping:
>   Stream #0:0 (h264) -> overlay:main (graph 0)
>   Stream #1:0 (vp8) -> overlay:overlay (graph 0)
>   overlay (graph 0) -> Stream #0:0 (libx264)
>   Stream #0:1 -> #0:1 (aac -> libvo_aacenc)
> Press [q] to stop, [?] for help
> [Parsed_overlay_0 @ 0x103800000] [framesync @ 0x1038000e8] Buffer queue
> overflow, dropping.
>     Last message repeated 624 times
> frame= 1345 fps=181 q=-1.0 Lsize=    2130kB time=00:00:44.81 bitrate=
> 389.4kbits/s dup=624 drop=0
> video:1851kB audio:251kB subtitle:0 data:0 global headers:0kB muxing
> overhead 1.348262%
> [libx264 @ 0x104000600] frame I:8     Avg QP:13.93  size: 30628
> [libx264 @ 0x104000600] frame P:402   Avg QP:17.93  size:  2802
> [libx264 @ 0x104000600] frame B:935   Avg QP:20.63  size:   560
> [libx264 @ 0x104000600] consecutive B-frames:  2.4% 14.4%  1.1% 82.1%
> [libx264 @ 0x104000600] mb I  I16..4:  9.0% 73.4% 17.6%
> [libx264 @ 0x104000600] mb P  I16..4:  1.7%  1.7%  0.3%  P16..4: 10.9%
>  2.3%  0.9%  0.0%  0.0%    skip:82.2%
> [libx264 @ 0x104000600] mb B  I16..4:  0.1%  0.1%  0.0%  B16..8:  7.6%
>  0.4%  0.0%  direct: 0.1%  skip:91.7%  L0:46.5% L1:50.8% BI: 2.8%
> [libx264 @ 0x104000600] 8x8 transform intra:53.1% inter:78.0%
> [libx264 @ 0x104000600] coded y,uvDC,uvAC intra: 31.4% 36.8% 7.9% inter:
> 1.1% 2.3% 0.0%
> [libx264 @ 0x104000600] i16 v,h,dc,p: 31% 27%  7% 35%
> [libx264 @ 0x104000600] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 41% 20% 23%  3%  3%
>  3%  3%  3%  3%
> [libx264 @ 0x104000600] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 25% 14%  3%  4%
>  4%  4%  3%  3%
> [libx264 @ 0x104000600] i8c dc,h,v,p: 74% 12% 11%  2%
> [libx264 @ 0x104000600] Weighted P-Frames: Y:0.0% UV:0.0%
> [libx264 @ 0x104000600] ref P L0: 68.4%  8.7% 17.1%  5.8%
> [libx264 @ 0x104000600] ref B L0: 85.5% 13.7%  0.8%
> [libx264 @ 0x104000600] ref B L1: 97.5%  2.5%
> [libx264 @ 0x104000600] kb/s:337.78
>

From wally at voosen.eu  Thu Apr 10 12:24:16 2014
From: wally at voosen.eu (wally)
Date: Thu, 10 Apr 2014 12:24:16 +0200
Subject: [FFmpeg-user] mpg to mp4 request
In-Reply-To: <loom.20140409T211826-560@post.gmane.org>
References: <53444185.50609@voosen.eu>
	<loom.20140409T211826-560@post.gmane.org>
Message-ID: <53467150.8090505@voosen.eu>

Hello Carl Eugen,

thank you for reply!

I tried the obvious commandline with a fragment of my sourcefile and the
result was acceptable. sizereduction and quality was OK and i can play on
the SAT Receiver.

After this test i tried with the entire sourcefile (3.4 GB). It results 
in a 1.3GB mp4 file
which doesn't work on the receiver. Message: "Video resolution not 
supported"

But the mp4 of the fragment works.
ffmpeg -i fragment.mp4
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 
720x576 [SAR 64:45 DAR 16:9], 1890 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 
tbc (default)

ffmpeg -i full.mp4
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 
720x576 [SAR 64:45 DAR 16:9], 1890 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 
tbc (default)

For me all this is not obvious :)

wally


On 04/09/2014 09:22 PM, Carl Eugen Hoyos wrote:
> wally <wally <at> voosen.eu> writes:
>
>> Which ffmpeg commandline do this job ?
> Did you test the obvious?
> $ ffmpeg -i source.mpg -vcodec libx264 -vb 1.9M -acodec aac
> -strict -2 -ab 90k out.mp4
>
>>     Metadata:
>>       major_brand     : mp42
>>       minor_version   : 0
>>       compatible_brands: isommp42
> FFmpeg only writes isom files, you can overwrite this,
> but I am not sure that makes the output file mp42.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


From nichot20 at yahoo.com  Thu Apr 10 12:48:24 2014
From: nichot20 at yahoo.com (Tim Nicholson)
Date: Thu, 10 Apr 2014 11:48:24 +0100
Subject: [FFmpeg-user] Urgent problem with ffmpeg installation on OS X
 10.9.2, Xcode Version 5.1 (5B130a)
In-Reply-To: <loom.20140410T092713-130@post.gmane.org>
References: <CF68DCCE.21EA4%dieter.herzmann@topix.ch>	<53463EC7.5080101@yahoo.com>
	<loom.20140410T092713-130@post.gmane.org>
Message-ID: <534676F8.8050500@yahoo.com>

On 10/04/14 08:28, Carl Eugen Hoyos wrote:
> Tim Nicholson <nichot20 <at> yahoo.com> writes:
> 
>> I suggest you try:-
>>
>>
> http://hexeract.wordpress.com/2009/04/12/how-to-compile-ffmpegmplayer-for-macosx/
> 
> The script seems to have several problems, what 
> specific "tweaks" does it show that seem useful to you?
> 
Well 3rd party lib issues as much as anything, for example:-

Hacking the configure script for x264 so it will compile with the latest
clang.

> And does openal really work for? I wasn't successful...

I don't build with that so cannot comment.

> 
> Carl Eugen
> [..]


-- 
Tim

From researchiteng at gmail.com  Thu Apr 10 14:11:55 2014
From: researchiteng at gmail.com (ReSearchIT Eng)
Date: Thu, 10 Apr 2014 15:11:55 +0300
Subject: [FFmpeg-user] loglevel debug - is it possible to get even more?
Message-ID: <CAMndFjmXg0q1eVXVL9XoYTsBiEeGK-GVaECGPTmXBeSsSSegtA@mail.gmail.com>

Hi,
I would like to know how to get much more details from the steps
ffmpeg is taking while processing the data. It would be nice to see
which are all the steps it took to make the result.

E.g. I am sending for audio coding a wav of 64Khz to be encoded to
amrwb encoder which accepts only 16Khz.
I am confident ffmpeg does resampling (downsampling in this case), and
I would like to know which of the filters it used for this task,
parameters, etc.

Is there a way to get this information?

Thanks,
Sebastian.

From hdlmnate at gmail.com  Thu Apr 10 13:54:41 2014
From: hdlmnate at gmail.com (hdn8)
Date: Thu, 10 Apr 2014 04:54:41 -0700 (PDT)
Subject: [FFmpeg-user] Optimal workflow for concatenating (reliably)
Message-ID: <1397130881377-4664731.post@n4.nabble.com>

I have a web application that for a user-given video file concatenates a
short intro and outro, trims based on user-selected in and out points and
overlays a watermark.  In order to make this work I have the following flow
(not included code for generating the intro/outro videos):

1. Separate audio and video 
--  I found that during concatenation the audio tracks sometimes became
unsynchronized, also this separation allows for normalizing the levels to
match the intro and outro music levels.  

2. Overlay watermark and/or subtitles, apply trimming. 
-- Note: This is output to .mp4 (and not .mpg directly) so we can keep a
copy of this file for later use, and discard the original saving disk space.    
-- command:  ffmpeg -i input_video.mp4 -r 25 -acodec libfaac -ac 2 -ar 48000
-ab 128k -vcodec libx264 -q:v 0 -vf
scale=iw*sar:ih[src];movie=watermark.png,scale=iw/1.49882903981:ih/1.49882903981[watermark];[src][watermark]overlay=main_w-overlay_w-10:10,fade=in:0:25
-metadata:s:v:0 rotate=0 -ss 2 -t 23 -y main.mp4

3. Create mpg containers for videos to be concatenated
-- I did not have any reliable success with other formats for concatenation
except with mpg containers, important that the settings and sizes are the
same for all three videos.  
-- command:  
ffmpeg -i main.mp4 -r 25 -an -s 854x480 -q:v 0 -y main_prepconcat.mpg

4. Concatenate videos
-- command:  ffmpeg -i concat:intro.mpg|main_prepconcat.mpg|outro.mpg -q:v 0
-y concat_tmp.mpg

5. Concatenate audios (and normalize)
-- This I do in python, not ffmpeg, normalize and concatenate the audio to
give a file: audio_concat.wav

6. Final encode,  encode output video with given concatenated video and
audio tracks  
-- command: ffmpeg -i concat_tmp.mpg -i audio_concat.wav -vcodec libx264
-q:v 0 -g 50 -y join_process.mp4

7. Faststart (put moov atom at front for faster web instantiation)
qtfaststart join_process.mp4

Im now revisiting this code and looking to see if there are any options for
improvement, better efficiency.  I found the concatenation process difficult
originally, trying many different formats and options before settling on
this workflow, in the next iteration we will add some more editing
functionality which will put more demand on the concatenation and trimming
processes, so if anyone can share their best practices thanks in advance!





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From phpdev at ehrhardt.nl  Thu Apr 10 15:17:53 2014
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Thu, 10 Apr 2014 15:17:53 +0200
Subject: [FFmpeg-user] Optimal workflow for concatenating (reliably)
References: <1397130881377-4664731.post@n4.nabble.com>
Message-ID: <mm3dk9tq73nq90em1i7q4av1e48dppidff@4ax.com>

hdn8 in gmane.comp.video.ffmpeg.user (Thu, 10 Apr 2014 04:54:41 -0700 (PDT)):
>I have a web application that for a user-given video file concatenates a
>short intro and outro, trims based on user-selected in and out points and
>overlays a watermark.

I am doing more or less the same. It is on Windows, but the same should be
possible on other OSs. It is in one pass:

ffmpeg.exe -i intro.mp4 -i main.mp4 -i outtro.mp4 \
	-filter_complex \
	"[0:v]setpts=PTS-STARTPTS[v0]; \
	[0:a]asetpts=PTS-STARTPTS,ebur128=metadata=1,volume=metadata=lavfi.r128.I[a0]; \
	[1:v]setpts=PTS-STARTPTS[v1]; \
	[1:a]asetpts=PTS-STARTPTS,ebur128=metadata=1,volume=metadata=lavfi.r128.I[a1]; \
	[2:v]setpts=PTS-STARTPTS[v2]; \
	[2:a]asetpts=PTS-STARTPTS,ebur128=metadata=1,volume=metadata=lavfi.r128.I[a2]; \
	[v0][a0][v1][a1][v2][a2]concat=n=3:v=1:a=1[v][a3];[a3]ebur128=metadata=1[a]" \
	-map [v] -map [a] -movflags +faststart \
	-s 384x288 -sws_flags bicubic -vcodec libx264 -b:v 320k -r 25 \
	-strict -2 -acodec libfaac -ab 96k -ar 48000 -ac 2 \
	-x264opts \

global_header=1:partitions=p8x8+b8x8+i8x8:level_idc=30:cabac=0:subq=3:qp_min=0:qp_max=51:qp_step=4:me=dia:subme=0:mixed_refs=0:me_range=16:chroma_me=1:trellis=0:8x8dct=0:cqm=flat:chroma_qp_offset=0:nr=0:keyint=25:min_keyint=12:scenecut=0:ratetol=1.0:qcomp=0.60:ip_factor=1.4:weightp=0:fast_pskip=1:frameref=1:bframes=0:mbtree=1:rc_lookahead=15:sliced_threads=0:threads=8\
	-y join_process.mp4

The 'volume=metadata=lavfi.r128.I' is the audio normalizing part. You will
have to patch the FFmpeg sources for that:
https://github.com/Jan-E/MyMpeg/blob/master/ffmpeg_patches/volnorm_new.patch

I tried to convince the FFmpeg developers to add this to the FFmpeg core,
but got answers like "single-pass volume normalization is not possible."
and "using the intermediate value of r128.I or its equivalent from
volumedetect for volume normalization is WRONG.."

This was the original idea:
http://comments.gmane.org/gmane.comp.video.ffmpeg.devel/159589

And this the sequential:
http://comments.gmane.org/gmane.comp.video.ffmpeg.devel/166113

Reliable? Yes. Over 10000 videos have been concatenated/normalized by this
process.

Jan


From hdlmnate at gmail.com  Thu Apr 10 17:07:34 2014
From: hdlmnate at gmail.com (hdn8)
Date: Thu, 10 Apr 2014 08:07:34 -0700 (PDT)
Subject: [FFmpeg-user] Optimal workflow for concatenating (reliably)
In-Reply-To: <mm3dk9tq73nq90em1i7q4av1e48dppidff@4ax.com>
References: <1397130881377-4664731.post@n4.nabble.com>
	<mm3dk9tq73nq90em1i7q4av1e48dppidff@4ax.com>
Message-ID: <1397142454677-4664734.post@n4.nabble.com>

Hi Jan, thanks for pointing me in an optimal direction, the -filter_complex
wasnt around when I made this app, from this it looks probably we can do
this in one pass and save loads of time and resources, avoid having to
create mpg containers etc.  Interesting debate over the "1-pass"
normalization, I also think that introducing an 'approximated' normalization
into the core would be useful in certain scenarios.     



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From phpdev at ehrhardt.nl  Thu Apr 10 18:28:45 2014
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Thu, 10 Apr 2014 18:28:45 +0200
Subject: [FFmpeg-user] Optimal workflow for concatenating (reliably)
References: <1397130881377-4664731.post@n4.nabble.com>
	<mm3dk9tq73nq90em1i7q4av1e48dppidff@4ax.com>
	<1397142454677-4664734.post@n4.nabble.com>
Message-ID: <84hdk99v099oo4097sba8apdfqtrquja29@4ax.com>

hdn8 in gmane.comp.video.ffmpeg.user (Thu, 10 Apr 2014 08:07:34 -0700
(PDT)):
>Interesting debate over the "1-pass" normalization, I also think that
>introducing an 'approximated' normalization into the core would be
>useful in certain scenarios.     

I never really understood the resistance against it. Maybe I stepped on
somebody's toes. Feel free to use my patch. I am keeping it up-to-date
with git head.

https://github.com/Jan-E/MyMpeg/blob/master/ffmpeg_patches/volnorm_new.patch

In some cases you have no choice and have to do it 1-pass, for instance
when transcoding a live-stream.

Jan


From george at nsup.org  Thu Apr 10 20:34:11 2014
From: george at nsup.org (Nicolas George)
Date: Thu, 10 Apr 2014 20:34:11 +0200
Subject: [FFmpeg-user] Optimal workflow for concatenating (reliably)
In-Reply-To: <84hdk99v099oo4097sba8apdfqtrquja29@4ax.com>
References: <1397130881377-4664731.post@n4.nabble.com>
	<mm3dk9tq73nq90em1i7q4av1e48dppidff@4ax.com>
	<1397142454677-4664734.post@n4.nabble.com>
	<84hdk99v099oo4097sba8apdfqtrquja29@4ax.com>
Message-ID: <20140410183411.GA747@phare.normalesup.org>

Le primidi 21 germinal, an CCXXII, Jan Ehrhardt a ?crit?:
> hdn8 in gmane.comp.video.ffmpeg.user (Thu, 10 Apr 2014 08:07:34 -0700
> (PDT)):
> >Interesting debate over the "1-pass" normalization, I also think that
> >introducing an 'approximated' normalization into the core would be
> >useful in certain scenarios.     
> 
> I never really understood the resistance against it. Maybe I stepped on
> somebody's toes.

There is no resistance against a good approximated 1-pass normalization
filter. In fact, there is need for one, an want for it too. So much so that
a reasonably correct one would probably be accepted.

Your proposal was not that, though: it was just wrong. I already have
explained why it is wrong, I can explain again if it is necessary.

Using the immediate loudness to normalize will cause audible distortion,
such as background noise becoming suddenly much louder when people stop
talking.

The short-term loudness will do a bit better on that, but still not good
enough: its reaction to changes is too asymmetrical, first very fast then
slower.

The integrated loudness has a completely different issue: its reaction time
is very fast on the beginning and very slow at the end. That means that a
fanfare opening will cause the following content to be much too quiet for a
long time, while a fanfare ending will not be normalized at all.

I do not know why you have not noticed those effects yet on your encodings.
Maybe you are lucky enough to work with content that is already normalized
enough so that it will not matter. Or maybe you just did not pay enough
attention. That does not matter.

The correct way of doing this is to start with the immediate loudness and
apply a de-noising filter on the signal. That requires a certain amount of
look-ahead, but libavfilter is perfectly capable of doing that.

The de-noising is tricky, though: it must remove the random changes, such as
someone stopping talking for half a second for dramatic effect, but the
sharp transitions must remain sharp. Signal processing is not my forte, but
if someone has pointers to a good smoothing algorithm with those properties,
the infrastructure around it is not difficult at all, and I already have
some code.

Regards,

-- 
  Nicolas George
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From cc at outlook.be  Wed Apr  9 22:21:53 2014
From: cc at outlook.be (Helios)
Date: Wed, 9 Apr 2014 13:21:53 -0700 (PDT)
Subject: [FFmpeg-user] How to write video info in top of screenshot?
Message-ID: <1397074913601-4664723.post@n4.nabble.com>

Hello,

i need help writing text at the top of the outpot screenshot

i want it like this 
http://www.thumbnailme.com/wp-content/themes/wp-coda/screenshots/09_Sample.jpg?rand=696292515

i want to create box at the top and write text to it YOU DONT  need to write
the info just teach how to write text like the picture with any text

i use this code to make screenshot



thank you 



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From rongyan236 at gmail.com  Thu Apr 10 05:11:45 2014
From: rongyan236 at gmail.com (Grace Ryan)
Date: Thu, 10 Apr 2014 11:11:45 +0800
Subject: [FFmpeg-user] A question about fft kernel in FFmpeg
Message-ID: <CAOJqnPQf+WTMDz56Q2ThsR3JVAL0tXgNgDArUEZPOzQAPDcPug@mail.gmail.com>

Hi, Everyone

I notice that the function imdct_half_altivec() in the file
libavcodec/ppc/fft_altivec.c uses ff_fft_calc_altivec(). I wonder what is
the data format (FFTComplex* z) of the input data and output data of the
function ff_fft_calc_altivec()? For example, for 32-point FFT, it is Re0,
Re1, Re2, Re3...Re31, Im0, Im1, Im2...Im31 or something else?

Best,
Rong Yan

From llee040 at sbcglobal.net  Thu Apr 10 22:14:42 2014
From: llee040 at sbcglobal.net (L. Lee)
Date: Thu, 10 Apr 2014 15:14:42 -0500
Subject: [FFmpeg-user] Missing packages for building ffmpeg 2.1 with
 Homebrew for Mac in Mavericks 10.9.2
Message-ID: <CF6C6159.20FEA%llee040@sbcglobal.net>

Using Homebrew, today I finally compiled and installed ffmpeg 2.1 for the
first time since upgrading my Mac to Mavericks. The Homebrew formula for
ffmpeg 2.1 that I used came from
https://gist.github.com/victusfate/5969117.   I found that many dependencies
needed to be removed then reinstalled. Here is the entire command and
options that finally allowed success:

??
brew install ffmpeg2 --with-fdk-aac --disable-ffserver --enable-gpl
--enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex
--enable-librtmp --enable-libtheora --enable-libvpx --enable-pthreads
--enable-x11grab --enable-libx264 --enable-nonfree --enable-version3
--with-ffplay --with-frei0r --with-libvpx  --with-openjpeg --with-opus
--with-rtmpdump --with-schroedinger --with-speex --enable-libfaad
--with-tools
??

Included below is a list of 3 options that I had previously used that today
caused compiling to fail, and the errors reported for each. I tried
everything I could think of, but still couldn?t make any of the 3 work.
Uninstalling and reinstalling libvorbis didn?t help, the same was true in
the case of freetype , and I don?t know what Homebrew package might affect
?libvo-aacenc". Can anybody tell me what packages I can try installing or
reinstalling or anything else I can do to satisfy the dependencies
specified? Thanks.

?with-freetype ERROR: freetype2 not found
?with-libvo-aacenc ERROR: libvo_aacenc not found
?with-libvorbis ERROR: libvorbis not found

Laine Lee



From cehoyos at ag.or.at  Thu Apr 10 22:32:19 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 10 Apr 2014 20:32:19 +0000 (UTC)
Subject: [FFmpeg-user] loglevel debug - is it possible to get even more?
References: <CAMndFjmXg0q1eVXVL9XoYTsBiEeGK-GVaECGPTmXBeSsSSegtA@mail.gmail.com>
Message-ID: <loom.20140410T223118-559@post.gmane.org>

ReSearchIT Eng <researchiteng <at> gmail.com> writes:

> I am confident ffmpeg does resampling (downsampling in 
> this case), and I would like to know which of the filters 
> it used for this task, parameters, etc.

(I believe this is what the console shows.)
Command line and complete, uncut console output missing.

Carl Eugen


From dashing.meng at gmail.com  Fri Apr 11 03:18:07 2014
From: dashing.meng at gmail.com (littlebat)
Date: Fri, 11 Apr 2014 09:18:07 +0800
Subject: [FFmpeg-user] How to write video info in top of screenshot?
In-Reply-To: <1397074913601-4664723.post@n4.nabble.com>
References: <1397074913601-4664723.post@n4.nabble.com>
Message-ID: <20140411091807.405eef1fa04e0b89ef9ed0ba@gmail.com>

On Wed, 9 Apr 2014 13:21:53 -0700 (PDT)
Helios <cc at outlook.be> wrote:

> Hello,
> 
> i need help writing text at the top of the outpot screenshot
> 
> i want it like this 
> http://www.thumbnailme.com/wp-content/themes/wp-coda/screenshots/09_Sample.jpg?rand=696292515
> 
> i want to create box at the top and write text to it YOU DONT  need
> to write the info just teach how to write text like the picture with
> any text
> 
> i use this code to make screenshot
> 

If you only operate some pictures, you can try the commands "convert",
"composite" those are in "ImageMagick" package in Linux.

If you need operate on video stream, read the ffmpeg documentation
about "video filter"(http://ffmpeg.org/ffmpeg-filters.html#overlay-1)

littlebat

From lou at lrcd.com  Fri Apr 11 04:39:05 2014
From: lou at lrcd.com (Lou)
Date: Thu, 10 Apr 2014 18:39:05 -0800
Subject: [FFmpeg-user] How to write video info in top of screenshot?
Message-ID: <1397183945.5258.105243269.1C9A05DC@webmail.messagingengine.com>

On Wed, 9 Apr 2014 13:21:53 -0700 (PDT)
Helios <cc at outlook.be> wrote:

> Hello,
> 
> i need help writing text at the top of the outpot screenshot
> 
> i want it like this 
> http://www.thumbnailme.com/wp-content/themes/wp-coda/screenshots/09_Sample.jpg?rand=696292515
> 
> i want to create box at the top and write text to it YOU DONT  need to write
> the info just teach how to write text like the picture with any text
> 
> i use this code to make screenshot

You forgot your code. Maybe something like this:

ffmpeg -i input -vf \
"[0:v]select=gt(scene\,0.4), \
scale=120:-1, \
tile, \
pad=iw:ih+100:0:oh-ih, \
drawtext=fontfile=/path/to/font.ttf:textfile=/path/to/file.txt:x=5:y=5[v]"
\
-map "[v]" -vframes 1 -qscale:v 3 preview.jpg

See FFmpeg filtering documentation for more info:
https://ffmpeg.org/ffmpeg-filters.html

From asdfg1945 at gmail.com  Fri Apr 11 07:59:57 2014
From: asdfg1945 at gmail.com (Arachnia)
Date: Thu, 10 Apr 2014 22:59:57 -0700 (PDT)
Subject: [FFmpeg-user] out of memory
Message-ID: <1397195997556-4664742.post@n4.nabble.com>

When I converted a rtsp stream to ts and m3u8, I found that memory used by
ffmpeg grow up continuously and process was killed by kernel at last.
Maybe a memory overflow?

My command is 

ffmpeg -f s16le -i /dev/zero -f rtsp -rtsp_transport tcp -i
'rtsp://192.168.0.12:554/rtsptest' -vcodec libx264 -tune zerolatency -crf 25
-b:v 64k -s 352x288 -profile:v baseline -pix_fmt yuv420p -acodec aac -b:a 1k
-ar 22050 -strict -2 -flags -global_header -map 0 -map 1 -f segment
-segment_time 10 -segment_list_size 2 -segment_list test.m3u8
-segment_list_type m3u8 -segment_format mpegts 240p%08d.ts

Output is normal.

system log:
Apr 11 11:36:29 Crtmp kernel: Out of memory: Kill process 8704 (ffmpeg)
score 940 or sacrifice child
Apr 11 11:36:29 Crtmp kernel: Killed process 8704, UID 0, (ffmpeg)
total-vm:6047156kB, anon-rss:3747432kB, file-rss:2392kB

ffmpeg version: 2.1.4

Anything help?



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From cehoyos at ag.or.at  Fri Apr 11 08:58:47 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 11 Apr 2014 06:58:47 +0000 (UTC)
Subject: [FFmpeg-user] out of memory
References: <1397195997556-4664742.post@n4.nabble.com>
Message-ID: <loom.20140411T085751-259@post.gmane.org>

Arachnia <asdfg1945 <at> gmail.com> writes:

> ffmpeg -f s16le -i /dev/zero -f rtsp -rtsp_transport tcp 
> -i 'rtsp://192.168.0.12:554/rtsptest' -vcodec libx264 
> -tune zerolatency -crf 25 -b:v 64k -s 352x288 
> -profile:v baseline -pix_fmt yuv420p -acodec aac -b:a 1k
> -ar 22050 -strict -2 -flags -global_header -map 0 -map 1 
> -f segment -segment_time 10 -segment_list_size 2 
> -segment_list test.m3u8 -segment_list_type m3u8 
> -segment_format mpegts 240p%08d.ts
> 
> Output is normal.

No, it's missing.

Please try again with current git head, something 
similar was just fixed.

Carl Eugen


From goga777 at bk.ru  Fri Apr 11 19:21:23 2014
From: goga777 at bk.ru (Goga777)
Date: Fri, 11 Apr 2014 21:21:23 +0400
Subject: [FFmpeg-user] [aac @ 0xaf6d000] Reserved SBR extensions is not
	implemented.
Message-ID: <E1WYf8i-0004rt-Il.goga777-bk-ru@smtp34.i.mail.ru>

Hi

how to solve the problem with that warnings ?

Goga



/usr/local/bin/ffmpeg -i http://193.242.178.200:8000/iv.aacp tt.avi
ffmpeg version git-2014-04-09-d8ab7f3 Copyright (c) 2000-2014 the FFmpeg developers
  built on Apr 11 2014 21:08:45 with gcc 4.8 (Debian 4.8.2-1)
  configuration: 
  libavutil      52. 75.100 / 52. 75.100
  libavcodec     55. 58.103 / 55. 58.103
  libavformat    55. 36.102 / 55. 36.102
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
[aac @ 0xaf6d000] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xaf6d000] decode_pce: Input buffer exhausted before END element found
Input #0, aac, from 'http://193.242.178.200:8000/iv.aacp':
  Duration: N/A, bitrate: 34 kb/s
    Stream #0:0: Audio: aac, 44100 Hz, stereo, fltp, 34 kb/s
File 'tt.avi' already exists. Overwrite ? [y/N] y
Output #0, avi, to 'tt.avi':
  Metadata:
    ISFT            : Lavf55.36.102
    Stream #0:0: Audio: ac3 ([0] [0][0] / 0x2000), 44100 Hz, stereo, fltp, 192 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (aac -> ac3)
Press [q] to stop, [?] for help
[aac @ 0xaf6d000] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0xaf6d000] decode_pce: Input buffer exhausted before END element found
Error while decoding stream #0:0: Operation not permitted
[aac @ 0xaf6d000] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0xaf6d000] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0xaf6d000] Reserved SBR extensions is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0xaf6d000] If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list.
[aac @ 0xaf6d000] Expected to read 80 PS bits actually read 101.
[aac @ 0xaf6d000] Expected to read 86 PS bits actually read 105.
[aac @ 0xaf6d000] Expected to read 102 PS bits actually read 122.
    Last message repeated 1 times
[aac @ 0xaf6d000] Expected to read 86 PS bits actually read 100.
[aac @ 0xaf6d000] Expected to read 102 PS bits actually read 117.
[aac @ 0xaf6d000] illegal iid
^Cze=    1060kB time=00:00:44.61 bitrate= 194.7kbits/s 

From cehoyos at ag.or.at  Fri Apr 11 19:30:26 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 11 Apr 2014 17:30:26 +0000 (UTC)
Subject: [FFmpeg-user] mpg to mp4 request
References: <53444185.50609@voosen.eu>
	<loom.20140409T211826-560@post.gmane.org>
	<53467150.8090505@voosen.eu>
Message-ID: <loom.20140411T192948-32@post.gmane.org>

wally <wally <at> voosen.eu> writes:

> After this test i tried with the entire sourcefile (3.4 GB). 
> It results in a 1.3GB mp4 file which doesn't work on the 
> receiver.

Did you test if this is just an effect of the output file size?

Carl Eugen


From cehoyos at ag.or.at  Fri Apr 11 19:35:30 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 11 Apr 2014 17:35:30 +0000 (UTC)
Subject: [FFmpeg-user] [aac@0xaf6d000] Reserved SBR extensions is not
	implemented.
References: <E1WYf8i-0004rt-Il.goga777-bk-ru@smtp34.i.mail.ru>
Message-ID: <loom.20140411T193425-674@post.gmane.org>

Goga777 <goga777 <at> bk.ru> writes:

> how to solve the problem with that warnings ?

Afaict, this is just triggered by the cut 
initial frames, nothing that can be "fixed" 
(except silencing the warnings).

Carl Eugen


From cehoyos at ag.or.at  Fri Apr 11 19:46:53 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 11 Apr 2014 17:46:53 +0000 (UTC)
Subject: [FFmpeg-user] [aac@0xaf6d000] Reserved SBR extensions is not
	implemented.
References: <E1WYf8i-0004rt-Il.goga777-bk-ru@smtp34.i.mail.ru>
Message-ID: <loom.20140411T194616-555@post.gmane.org>

Goga777 <goga777 <at> bk.ru> writes:

> how to solve the problem with that warnings ?

... I opened ticket #3552.

Thank you for the sample, Carl Eugen


From asdfg1945 at gmail.com  Fri Apr 11 08:04:44 2014
From: asdfg1945 at gmail.com (Arachnia)
Date: Thu, 10 Apr 2014 23:04:44 -0700 (PDT)
Subject: [FFmpeg-user] loglevel debug - is it possible to get even more?
In-Reply-To: <CAMndFjmXg0q1eVXVL9XoYTsBiEeGK-GVaECGPTmXBeSsSSegtA@mail.gmail.com>
References: <CAMndFjmXg0q1eVXVL9XoYTsBiEeGK-GVaECGPTmXBeSsSSegtA@mail.gmail.com>
Message-ID: <1397196284024-4664743.post@n4.nabble.com>

Maybe option "-loglevel debug" can help?



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/loglevel-debug-is-it-possible-to-get-even-more-tp4664732p4664743.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From xanadu at apost.plala.or.jp  Fri Apr 11 20:42:54 2014
From: xanadu at apost.plala.or.jp (=?utf-8?B?5a6u5p2RIOWFrOeUtw==?=)
Date: Sat, 12 Apr 2014 03:42:54 +0900
Subject: [FFmpeg-user] 2 problem with ffmpeg and x265
Message-ID: <87DD789D-1693-4BA1-87E3-DD336FCD1017@apost.plala.or.jp>

Hi, I?m now trying to build ffmpeg with static x265.  But configure script rejects with error.  The error is "ERROR: x265 not found?

I?m using Mac OS X 10.9.2  and LLVM version 5.1 (clang-503.0.38) (based on LLVM 3.4svn)

Here?s a part of config.log

check_pkg_config x265 x265.h x265_encoder_encode
pkg-config --exists --print-errors x265
check_func_headers x265.h x265_encoder_encode -I/Volumes/ffmpeg_compile/include -L/Volumes/ffmpeg_compile/lib -lx265
check_ld cc -I/Volumes/ffmpeg_compile/include -L/Volumes/ffmpeg_compile/lib -lx265
check_cc -I/Volumes/ffmpeg_compile/include -L/Volumes/ffmpeg_compile/lib
BEGIN /var/folders/qk/x9k9r6f955d75dnv5dd2rgm00000gn/T//ffconf.EjOwyBdL.c
    1	#include <x265.h>
    2	long check_x265_encoder_encode(void) { return (long) x265_encoder_encode; }
    3	int main(void) { return 0; }
END /var/folders/qk/x9k9r6f955d75dnv5dd2rgm00000gn/T//ffconf.EjOwyBdL.c
gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -DPIC -std=c99 -mdynamic-no-pic -fomit-frame-pointer -fPIC -pthread -I/Volumes/ffmpeg_compile/include -L/Volumes/ffmpeg_compile/lib -c -o /var/folders/qk/x9k9r6f955d75dnv5dd2rgm00000gn/T//ffconf.UUCyIpQ5.o /var/folders/qk/x9k9r6f955d75dnv5dd2rgm00000gn/T//ffconf.EjOwyBdL.c
clang: warning: argument unused during compilation: '-L/Volumes/ffmpeg_compile/lib'
gcc -Wl,-dynamic,-search_paths_first -I/Volumes/ffmpeg_compile/include -L/Volumes/ffmpeg_compile/lib -o /var/folders/qk/x9k9r6f955d75dnv5dd2rgm00000gn/T//ffconf.d9N4vA5m /var/folders/qk/x9k9r6f955d75dnv5dd2rgm00000gn/T//ffconf.UUCyIpQ5.o -lx265 -lm -lbz2 -lz -pthread
clang: warning: argument unused during compilation: '-pthread'
Undefined symbols for architecture x86_64:
  "std::__1::locale::use_facet(std::__1::locale::id&) const", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::__1::ios_base::getloc() const", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::__1::basic_string<char, std::__1::char_traits<char>, std::__1::allocator<char> >::__init(unsigned long, char)", referenced from:
      std::__1::ostreambuf_iterator<char, std::__1::char_traits<char> > std::__1::__pad_and_output<char, std::__1::char_traits<char> >(std::__1::ostreambuf_iterator<char, std::__1::char_traits<char> >, char const*, char const*, char const*, std::__1::ios_base&, char) in libx265.a(TComRom.cpp.o)
  "std::__1::basic_string<char, std::__1::char_traits<char>, std::__1::allocator<char> >::~basic_string()", referenced from:
      std::__1::ostreambuf_iterator<char, std::__1::char_traits<char> > std::__1::__pad_and_output<char, std::__1::char_traits<char> >(std::__1::ostreambuf_iterator<char, std::__1::char_traits<char> >, char const*, char const*, char const*, std::__1::ios_base&, char) in libx265.a(TComRom.cpp.o)
  "std::__1::basic_ostream<char, std::__1::char_traits<char> >::put(char)", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
  "std::__1::basic_ostream<char, std::__1::char_traits<char> >::flush()", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
  "std::__1::basic_ostream<char, std::__1::char_traits<char> >::sentry::sentry(std::__1::basic_ostream<char, std::__1::char_traits<char> >&)", referenced from:
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::__1::basic_ostream<char, std::__1::char_traits<char> >::sentry::~sentry()", referenced from:
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::__1::basic_ostream<char, std::__1::char_traits<char> >::operator<<(int)", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
  "std::__1::cerr", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
  "std::__1::ctype<char>::id", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::__1::locale::~locale()", referenced from:
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::__1::ios_base::__set_badbit_and_consider_rethrow()", referenced from:
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::__1::ios_base::clear(unsigned int)", referenced from:
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "std::terminate()", referenced from:
      ___clang_call_terminate in libx265.a(TComRom.cpp.o)
      ___clang_call_terminate in libx265.a(bitcost.cpp.o)
      ___clang_call_terminate in libx265.a(encoder.cpp.o)
      ___clang_call_terminate in libx265.a(threadpool.cpp.o)
      ___clang_call_terminate in libx265.a(frameencoder.cpp.o)
      ___clang_call_terminate in libx265.a(TComPic.cpp.o)
      ___clang_call_terminate in libx265.a(slicetype.cpp.o)
      ...
  "vtable for __cxxabiv1::__class_type_info", referenced from:
      typeinfo for x265_encoder in libx265.a(encoder.cpp.o)
      typeinfo for x265::FrameFilter in libx265.a(encoder.cpp.o)
      typeinfo for x265::TComPicYuv in libx265.a(TComPicYuv.cpp.o)
      typeinfo for x265::ThreadPool in libx265.a(threadpool.cpp.o)
      typeinfo for x265::SEI in libx265.a(frameencoder.cpp.o)
      typeinfo for x265::FrameFilter in libx265.a(frameencoder.cpp.o)
      typeinfo for x265::TComBitIf in libx265.a(frameencoder.cpp.o)
      ...
  NOTE: a missing vtable usually means the first non-inline virtual member function has no definition.
  "vtable for __cxxabiv1::__si_class_type_info", referenced from:
      typeinfo for x265::Encoder in libx265.a(encoder.cpp.o)
      typeinfo for x265::ThreadPoolImpl in libx265.a(threadpool.cpp.o)
      typeinfo for x265::PoolThread in libx265.a(threadpool.cpp.o)
      typeinfo for x265::SEIPictureTiming in libx265.a(frameencoder.cpp.o)
      typeinfo for x265::SEIRecoveryPoint in libx265.a(frameencoder.cpp.o)
      typeinfo for x265::SEIGradualDecodingRefreshInfo in libx265.a(frameencoder.cpp.o)
      typeinfo for x265::SEIDisplayOrientation in libx265.a(frameencoder.cpp.o)
      ...
  NOTE: a missing vtable usually means the first non-inline virtual member function has no definition.
  "vtable for __cxxabiv1::__vmi_class_type_info", referenced from:
      typeinfo for x265::FrameEncoder in libx265.a(frameencoder.cpp.o)
      typeinfo for x265::TEncSbac in libx265.a(TEncSbac.cpp.o)
      typeinfo for x265::MotionEstimate in libx265.a(motion.cpp.o)
  NOTE: a missing vtable usually means the first non-inline virtual member function has no definition.
  "operator delete[](void*)", referenced from:
      x265::BitCost::destroy() in libx265.a(bitcost.cpp.o)
      x265::Encoder::create() in libx265.a(encoder.cpp.o)
      x265::Encoder::destroy() in libx265.a(encoder.cpp.o)
      x265::TComPicSym::destroy() in libx265.a(TComPicSym.cpp.o)
      x265::FrameEncoder::destroy() in libx265.a(frameencoder.cpp.o)
      x265::FrameEncoder::init(x265::Encoder*, int) in libx265.a(frameencoder.cpp.o)
      x265::FrameEncoder::compressFrame() in libx265.a(frameencoder.cpp.o)
      ...
  "operator delete(void*)", referenced from:
      _x265_encoder_open_15 in libx265.a(api.cpp.o)
      x265::Encoder::~Encoder() in libx265.a(encoder.cpp.o)
      x265::Encoder::create() in libx265.a(encoder.cpp.o)
      x265::Encoder::destroy() in libx265.a(encoder.cpp.o)
      x265::Encoder::encode(bool, x265_picture const*, x265_picture*, x265::NALUnitEBSP**) in libx265.a(encoder.cpp.o)
      x265::FrameFilter::~FrameFilter() in libx265.a(encoder.cpp.o)
      x265::TComPicSym::create(int, int, int, unsigned int, unsigned int) in libx265.a(TComPicSym.cpp.o)
      ...
  "operator new[](unsigned long)", referenced from:
      x265::BitCost::setQP(unsigned int) in libx265.a(bitcost.cpp.o)
      x265::BitCost::CalculateLogs() in libx265.a(bitcost.cpp.o)
      x265::Encoder::create() in libx265.a(encoder.cpp.o)
      x265::TComPicSym::create(int, int, int, unsigned int, unsigned int) in libx265.a(TComPicSym.cpp.o)
      x265::FrameEncoder::init(x265::Encoder*, int) in libx265.a(frameencoder.cpp.o)
      x265::FrameEncoder::compressFrame() in libx265.a(frameencoder.cpp.o)
      x265::TComLoopFilter::create(unsigned int) in libx265.a(TComLoopFilter.cpp.o)
      ...
  "operator new(unsigned long)", referenced from:
      _x265_encoder_open_15 in libx265.a(api.cpp.o)
      x265::Encoder::create() in libx265.a(encoder.cpp.o)
      x265::Encoder::encode(bool, x265_picture const*, x265_picture*, x265::NALUnitEBSP**) in libx265.a(encoder.cpp.o)
      x265::TComPicSym::create(int, int, int, unsigned int, unsigned int) in libx265.a(TComPicSym.cpp.o)
      x265::TComPicSym::allocSaoParam(x265::TComSampleAdaptiveOffset*) in libx265.a(TComPicSym.cpp.o)
      x265::ThreadPool::allocThreadPool(int) in libx265.a(threadpool.cpp.o)
      x265::FrameEncoder::compressFrame() in libx265.a(frameencoder.cpp.o)
      ...
  "___cxa_begin_catch", referenced from:
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
      ___clang_call_terminate in libx265.a(TComRom.cpp.o)
      ___clang_call_terminate in libx265.a(bitcost.cpp.o)
      ___clang_call_terminate in libx265.a(encoder.cpp.o)
      ___clang_call_terminate in libx265.a(threadpool.cpp.o)
      ___clang_call_terminate in libx265.a(frameencoder.cpp.o)
      ___clang_call_terminate in libx265.a(TComPic.cpp.o)
      ...
  "___cxa_end_catch", referenced from:
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
  "___cxa_pure_virtual", referenced from:
      vtable for x265::Thread in libx265.a(threading.cpp.o)
      vtable for x265::WaveFront in libx265.a(wavefront.cpp.o)
  "___dynamic_cast", referenced from:
      x265::JobProvider::flush() in libx265.a(threadpool.cpp.o)
  "___gxx_personality_v0", referenced from:
      _x265_encoder_open_15 in libx265.a(api.cpp.o)
      Dwarf Exception Unwind Info (__eh_frame) in libx265.a(api.cpp.o)
      x265::ScanGenerator::GetNextIndex(unsigned int, unsigned int) in libx265.a(TComRom.cpp.o)
      std::__1::basic_ostream<char, std::__1::char_traits<char> >& std::__1::operator<<<std::__1::char_traits<char> >(std::__1::basic_ostream<char, std::__1::char_traits<char> >&, char const*) in libx265.a(TComRom.cpp.o)
      std::__1::ostreambuf_iterator<char, std::__1::char_traits<char> > std::__1::__pad_and_output<char, std::__1::char_traits<char> >(std::__1::ostreambuf_iterator<char, std::__1::char_traits<char> >, char const*, char const*, char const*, std::__1::ios_base&, char) in libx265.a(TComRom.cpp.o)
      Dwarf Exception Unwind Info (__eh_frame) in libx265.a(TComRom.cpp.o)
      x265::BitCost::setQP(unsigned int) in libx265.a(bitcost.cpp.o)
      ...
ld: symbol(s) not found for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
ERROR: x265 not found


And also, I can build ffmpeg with shared x265 but this time dyld: Library not loaded error occurred.

$ which ffmpeg
/Volumes/ffmpeg_compile/bin/ffmpeg
$ ffmpeg 
dyld: Library not loaded: libx265.15.dylib
  Referenced from: /Volumes/ffmpeg_compile/bin/ffmpeg
  Reason: image not found
Trace/BPT trap: 5
$ otool -L /Volumes/ffmpeg_compile/bin/ffmpeg
/Volumes/ffmpeg_compile/bin/ffmpeg:
	/System/Library/Frameworks/QTKit.framework/Versions/A/QTKit (compatibility version 1.0.0, current version 1.0.0)
	/System/Library/Frameworks/Foundation.framework/Versions/C/Foundation (compatibility version 300.0.0, current version 1056.13.0)
	/System/Library/Frameworks/QuartzCore.framework/Versions/A/QuartzCore (compatibility version 1.2.0, current version 1.8.0)
	/System/Library/Frameworks/CoreFoundation.framework/Versions/A/CoreFoundation (compatibility version 150.0.0, current version 855.14.0)
	/System/Library/Frameworks/VideoDecodeAcceleration.framework/Versions/A/VideoDecodeAcceleration (compatibility version 1.0.0, current version 1.0.0)
	libx265.15.dylib (compatibility version 15.0.0, current version 0.9.0)
	/usr/lib/libSystem.B.dylib (compatibility version 1.0.0, current version 1197.1.1)
	/System/Library/Frameworks/CoreVideo.framework/Versions/A/CoreVideo (compatibility version 1.2.0, current version 1.8.0)
	/usr/lib/libobjc.A.dylib (compatibility version 1.0.0, current version 228.0.0)


if I put libx265.15.dylib to /usr/local/lib, ffmpeg can execute.

My final aim is to build ffmpeg with static x265 library.  But as a part of try and error, I have encountered 2 problems.  Please solve them...

From ben at cardinalpeak.com  Fri Apr 11 21:25:53 2014
From: ben at cardinalpeak.com (Ben Mesander)
Date: Fri, 11 Apr 2014 13:25:53 -0600
Subject: [FFmpeg-user] concat muxer with two video files with two different
	resolutions
Message-ID: <CANgFRXB=pDf-t-bB5YG8uRa2bCWyy+0qNVim45kemqJxFSm5eg@mail.gmail.com>

Hello ffmpeg user mailing list,

  Today I tried to concatenate two h.264 video files with the concat muxer.
The two files have differing input resolutions (848x480 and 1280x720). I
would like the resulting video to be 640x480. I used today's static 32-bit
build from ffmpeg.org.

  Here's ffprobe output for the two files:

ben at ben-Latitude-D820:~/vid$ ./ffprobe 117_2463.MOV
ffprobe version N-62162-gec8789a Copyright (c) 2007-2014 the FFmpeg
developers
  built on Apr  5 2014 05:14:18 with gcc 4.6 (Debian 4.6.3-1)
  configuration: --prefix=/root/ffmpeg-static/32bit --arch=x86_32
--extra-cflags='-m32 -I/root/ffmpeg-static/32bit/include -static'
--extra-ldflags='-m32 -L/root/ffmpeg-static/32bit/lib -static'
--extra-libs='-lxml2 -lexpat -lfreetype' --enable-static --disable-shared
--disable-ffserver --disable-doc --enable-bzlib --enable-zlib
--enable-postproc --enable-runtime-cpudetect --enable-libx264 --enable-gpl
--enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-gray
--enable-libass --enable-libfreetype --enable-libopenjpeg --enable-libspeex
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-version3
--enable-libvpx
  libavutil      52. 74.100 / 52. 74.100
  libavcodec     55. 57.100 / 55. 57.100
  libavformat    55. 36.101 / 55. 36.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '117_2463.MOV':
  Metadata:
    major_brand     : qt
    minor_version   : 0
    compatible_brands: qt
    creation_time   : 2014-03-20 12:21:10
    comment         : KODAK Zi8 Pocket Video Camera
    comment-eng     : KODAK Zi8 Pocket Video Camera
  Duration: 00:00:15.02, start: 0.000000, bitrate: 4799 kb/s
    Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p,
848x480 [SAR 1:1 DAR 53:30], 4654 kb/s, 29.97 fps, 29.97 tbr, 90k tbn,
59.94 tbc (default)
    Metadata:
      creation_time   : 2014-03-20 12:21:10
      handler_name    : Ambarella AVC
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo,
fltp, 127 kb/s (default)
    Metadata:
      creation_time   : 2014-03-20 12:21:10
      handler_name    : Ambarella AAC
ben at ben-Latitude-D820:~/vid$ ./ffprobe 117_2466.MOV
ffprobe version N-62162-gec8789a Copyright (c) 2007-2014 the FFmpeg
developers
  built on Apr  5 2014 05:14:18 with gcc 4.6 (Debian 4.6.3-1)
  configuration: --prefix=/root/ffmpeg-static/32bit --arch=x86_32
--extra-cflags='-m32 -I/root/ffmpeg-static/32bit/include -static'
--extra-ldflags='-m32 -L/root/ffmpeg-static/32bit/lib -static'
--extra-libs='-lxml2 -lexpat -lfreetype' --enable-static --disable-shared
--disable-ffserver --disable-doc --enable-bzlib --enable-zlib
--enable-postproc --enable-runtime-cpudetect --enable-libx264 --enable-gpl
--enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-gray
--enable-libass --enable-libfreetype --enable-libopenjpeg --enable-libspeex
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-version3
--enable-libvpx
  libavutil      52. 74.100 / 52. 74.100
  libavcodec     55. 57.100 / 55. 57.100
  libavformat    55. 36.101 / 55. 36.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '117_2466.MOV':
  Metadata:
    major_brand     : qt
    minor_version   : 0
    compatible_brands: qt
    creation_time   : 2014-04-11 11:56:44
    comment         : KODAK Zi8 Pocket Video Camera
    comment-eng     : KODAK Zi8 Pocket Video Camera
  Duration: 00:01:26.39, start: 0.000000, bitrate: 4248 kb/s
    Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], 4111 kb/s, 29.97 fps, 29.97 tbr, 90k tbn,
59.94 tbc (default)
    Metadata:
      creation_time   : 2014-04-11 11:56:44
      handler_name    : Ambarella AVC
    Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo,
fltp, 127 kb/s (default)
    Metadata:
      creation_time   : 2014-04-11 11:56:44
      handler_name    : Ambarella AAC


I tried creating a foo.txt like so:

ben at ben-Latitude-D820:~/vid$ cat foo.txt
file '117_2463.MOV'
file '117_2466.MOV'

And here is the ffmpeg command line & some of the thousands of lines of
output:

./ffmpeg -f concat -i foo.txt -strict -2 -s 640x480  -vcodec libx264
-preset fast -crf 20 -acodec aac out.mp4
ffmpeg version N-62162-gec8789a Copyright (c) 2000-2014 the FFmpeg
developers
  built on Apr  5 2014 05:14:18 with gcc 4.6 (Debian 4.6.3-1)
  configuration: --prefix=/root/ffmpeg-static/32bit --arch=x86_32
--extra-cflags='-m32 -I/root/ffmpeg-static/32bit/include -static'
--extra-ldflags='-m32 -L/root/ffmpeg-static/32bit/lib -static'
--extra-libs='-lxml2 -lexpat -lfreetype' --enable-static --disable-shared
--disable-ffserver --disable-doc --enable-bzlib --enable-zlib
--enable-postproc --enable-runtime-cpudetect --enable-libx264 --enable-gpl
--enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-gray
--enable-libass --enable-libfreetype --enable-libopenjpeg --enable-libspeex
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-version3
--enable-libvpx
  libavutil      52. 74.100 / 52. 74.100
  libavcodec     55. 57.100 / 55. 57.100
  libavformat    55. 36.101 / 55. 36.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
[concat @ 0xa3903e0] Estimating duration from bitrate, this may be
inaccurate
Input #0, concat, from 'foo.txt':
  Duration: 00:00:00.07, start: 0.000000, bitrate: 4 kb/s
    Stream #0:0: Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 848x480
[SAR 1:1 DAR 53:30], 4654 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:1: Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp,
127 kb/s
File 'out.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 0xa3a0c60] using SAR=53/40
[libx264 @ 0xa3a0c60] using cpu capabilities: MMX2 SSE2Fast SSSE3 Cache64
[libx264 @ 0xa3a0c60] profile High, level 3.0
[libx264 @ 0xa3a0c60] 264 - core 129 r2230 1cffe9f - H.264/MPEG-4 AVC codec
- Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1
ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=6 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=3 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=1 keyint=250 keyint_min=25 scenecut=40
intra_refresh=0 rc_lookahead=30 rc=crf mbtree=1 crf=20.0 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         : Lavf55.36.101
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
640x480 [SAR 53:40 DAR 53:30], q=-1--1, 30k tbn, 29.97 tbc
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo,
fltp, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> libx264)
  Stream #0:1 -> #0:1 (aac -> aac)
Press [q] to stop, [?] for help
frame=   40 fps=0.0 q=26.0 size=      29kB time=00:00:01.38 bitrate=
170.0kbits/s    frame=   59 fps= 57 q=26.0 size=      67kB time=00:00:02.02
bitrate= 272.7kbits/s    frame=   79 fps= 50 q=26.0 size=     105kB
time=00:00:02.68 bitrate= 319.6kbits/s    frame=   99 fps= 47 q=26.0
size=     140kB time=00:00:03.34 bitrate= 342.4kbits/s    frame=  119 fps=
45 q=26.0 size=     174kB time=00:00:04.03 bitrate= 353.2kbits/s    frame=
137 fps= 44 q=26.0 size=     207kB time=00:00:04.62 bitrate=
366.1kbits/s    frame=  156 fps= 43 q=26.0 size=     241kB time=00:00:05.26
bitrate= 374.9kbits/s    frame=  175 fps= 42 q=26.0 size=     276kB
time=00:00:05.88 bitrate= 383.8kbits/s    frame=  195 fps= 41 q=26.0
size=     310kB time=00:00:06.57 bitrate= 386.7kbits/s    frame=  215 fps=
41 q=26.0 size=     345kB time=00:00:07.23 bitrate= 390.4kbits/s    frame=
235 fps= 41 q=26.0 size=     381kB time=00:00:07.89 bitrate=
395.0kbits/s    frame=  253 fps= 40 q=26.0 size=     413kB time=00:00:08.49
bitrate= 398.8kbits/s    frame=  274 fps= 41 q=26.0 size=     451kB
time=00:00:09.17 bitrate= 403.1kbits/s    frame=  289 fps= 40 q=26.0
size=     516kB time=00:00:09.70 bitrate= 435.8kbits/s    frame=  309 fps=
40 q=26.0 size=     563kB time=00:00:10.36 bitrate= 445.2kbits/s    frame=
329 fps= 40 q=26.0 size=     613kB time=00:00:11.00 bitrate=
456.1kbits/s    frame=  343 fps= 39 q=26.0 size=     649kB time=00:00:11.49
bitrate= 462.5kbits/s    frame=  359 fps= 39 q=26.0 size=     693kB
time=00:00:12.03 bitrate= 471.8kbits/s    frame=  375 fps= 38 q=26.0
size=     744kB time=00:00:12.56 bitrate= 485.1kbits/s    frame=  393 fps=
38 q=26.0 size=     797kB time=00:00:13.16 bitrate= 495.8kbits/s    frame=
412 fps= 38 q=26.0 size=     842kB time=00:00:13.80 bitrate=
499.5kbits/s    frame=  429 fps= 38 q=26.0 size=     884kB time=00:00:14.37
bitrate= 503.8kbits/s    [h264 @ 0xac826c0] left block unavailable for
requested intra4x4 mode -1 at 0 1
[h264 @ 0xac826c0] error while decoding MB 0 1, bytestream 209211
[h264 @ 0xac826c0] concealing 1586 DC, 1586 AC, 1586 MV errors in I frame
frame=  448 fps= 38 q=26.0 size=     921kB time=00:00:14.84 bitrate=
508.1kbits/s    [h264 @ 0xacc7600] concealing 1515 DC, 1515 AC, 1515 MV
errors in B frame
[h264 @ 0xacc7ca0] concealing 1199 DC, 1199 AC, 1199 MV errors in B frame
[h264 @ 0xac826c0] left block unavailable for requested intra mode at 0 9
[h264 @ 0xac826c0] error while decoding MB 0 9, bytestream 62961
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1162 DC, 1162 AC, 1162 MV errors in P frame
[h264 @ 0xacc7600] concealing 1502 DC, 1502 AC, 1502 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1534 DC, 1534 AC, 1534 MV errors in B frame
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1405 DC, 1405 AC, 1405 MV errors in P frame
[h264 @ 0xacc7600] concealing 1429 DC, 1429 AC, 1429 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1414 DC, 1414 AC, 1414 MV errors in B frame
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1525 DC, 1525 AC, 1525 MV errors in P frame
[h264 @ 0xacc7600] concealing 1521 DC, 1521 AC, 1521 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1503 DC, 1503 AC, 1503 MV errors in B frame
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1482 DC, 1482 AC, 1482 MV errors in P frame
[h264 @ 0xacc7600] concealing 1364 DC, 1364 AC, 1364 MV errors in B frame
frame=  461 fps= 37 q=26.0 size=     956kB time=00:00:15.44 bitrate=
507.3kbits/s    [h264 @ 0xacc7ca0] concealing 1202 DC, 1202 AC, 1202 MV
errors in B frame
[h264 @ 0xac826c0] left block unavailable for requested intra4x4 mode -1 at
0 1
[h264 @ 0xac826c0] error while decoding MB 0 1, bytestream 222145
[h264 @ 0xac826c0] concealing 1586 DC, 1586 AC, 1586 MV errors in I frame
[h264 @ 0xacc7600] concealing 1323 DC, 1323 AC, 1323 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1369 DC, 1369 AC, 1369 MV errors in B frame
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1174 DC, 1174 AC, 1174 MV errors in P frame
[h264 @ 0xacc7600] concealing 1366 DC, 1366 AC, 1366 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1302 DC, 1302 AC, 1302 MV errors in B frame
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1417 DC, 1417 AC, 1417 MV errors in P frame
[h264 @ 0xacc7600] concealing 1481 DC, 1481 AC, 1481 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1136 DC, 1136 AC, 1136 MV errors in B frame
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1516 DC, 1516 AC, 1516 MV errors in P frame
[h264 @ 0xacc7600] concealing 960 DC, 960 AC, 960 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1319 DC, 1319 AC, 1319 MV errors in B frame
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1488 DC, 1488 AC, 1488 MV errors in P frame
[h264 @ 0xacc7600] concealing 1241 DC, 1241 AC, 1241 MV errors in B frame
frame=  476 fps= 37 q=26.0 size=     985kB time=00:00:15.93 bitrate=
506.7kbits/

many thousands of lines substantively similar to the above and then:

[h264 @ 0xac826c0] left block unavailable for requested intra mode at 0 5
[h264 @ 0xac826c0] error while decoding MB 0 5, bytestream 5477
[h264 @ 0xac826c0] Cannot use next picture in error concealment
[h264 @ 0xac826c0] concealing 1374 DC, 1374 AC, 1374 MV errors in P frame
[h264 @ 0xacc7600] concealing 1334 DC, 1334 AC, 1334 MV errors in B frame
[h264 @ 0xacc7ca0] concealing 1482 DC, 1482 AC, 1482 MV errors in B frame
frame= 3039 fps= 42 q=26.0 size=    5502kB time=00:01:41.13 bitrate=
445.6kbits/s    frame= 3039 fps= 42 q=-1.0 Lsize=    5674kB
time=00:01:41.33 bitrate= 458.7kbits/s
video:3981kB audio:1583kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 1.977703%
[libx264 @ 0xa3a0c60] frame I:26    Avg QP:17.30  size: 10018
[libx264 @ 0xa3a0c60] frame P:859   Avg QP:19.38  size:  2324
[libx264 @ 0xa3a0c60] frame B:2154  Avg QP:20.15  size:   844
[libx264 @ 0xa3a0c60] consecutive B-frames:  4.4%  2.2%  2.8% 90.6%
[libx264 @ 0xa3a0c60] mb I  I16..4: 73.1% 14.1% 12.8%
[libx264 @ 0xa3a0c60] mb P  I16..4:  6.2%  1.9%  1.5%  P16..4: 14.8%  3.9%
2.0%  0.0%  0.0%    skip:69.6%
[libx264 @ 0xa3a0c60] mb B  I16..4:  1.0%  0.2%  0.2%  B16..8:  6.7%  1.7%
0.2%  direct: 5.6%  skip:84.4%  L0:50.6% L1:43.1% BI: 6.2%
[libx264 @ 0xa3a0c60] 8x8 transform intra:18.2% inter:43.5%
[libx264 @ 0xa3a0c60] coded y,uvDC,uvAC intra: 24.1% 27.7% 16.6% inter:
3.2% 8.4% 0.9%
[libx264 @ 0xa3a0c60] i16 v,h,dc,p: 80% 17%  2%  1%
[libx264 @ 0xa3a0c60] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 37% 22% 26%  2%  2%
2%  2%  2%  5%
[libx264 @ 0xa3a0c60] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 30% 21%  3%  3%
3%  4%  3%  4%
[libx264 @ 0xa3a0c60] i8c dc,h,v,p: 46% 22% 31%  1%
[libx264 @ 0xa3a0c60] Weighted P-Frames: Y:0.5% UV:0.3%
[libx264 @ 0xa3a0c60] ref P L0: 63.8% 36.2%
[libx264 @ 0xa3a0c60] ref B L0: 78.0% 22.0%
[libx264 @ 0xa3a0c60] ref B L1: 90.5%  9.5%
[libx264 @ 0xa3a0c60] kb/s:321.56

In the resulting output mp4 file, the first video file appears as the
input, correctly sized to 640x480. The second video file shows up as mostly
grey with random green macroblocks and flickery random pixels here and
there.

So it appears an (undocumented? I don't see it) feature of the concat muxer
is the input files must be the same resolution, so this approach simply
won't work?

What I'm looking for is the simplest way to accomplish the goal of resizing
N h.264/aac files with differing sizes and concatenating them into a single
h.264/aac file in mp4 format. I think I can see how to accomplish this with
many, many named pipes, for instance I could fire up an ffmpeg on each
source video file and have it resize to the desired size and write the
audio and video streams to named pipes. Then I could 'cat' all the audio
and video pipes together respectively and feed them into another ffmpeg
instance which could reencode the video as h.264 and write it out in an mp4
container. This is a lot of named pipes to be setting up and tearing down
and seems like a fairly complex shell script with potentially a lot of edge
cases to handle in the face of corrupt or missing or zero length input
files, etc. I don't know if filtergraphs could help me here.

Any advice? Is the behavior I'm seeing expected, or a bug?

Thank you!

Regards,
Ben


-- 
*Ben Mesander*

(303)570-1606 | Email <email at cardinalpeak.com> |
vCard<http://www.cardinalpeak.com/vcard/bmesander.vcf>
 | Web <http://www.cardinalpeak.com/> | Company
Blog<http://www.cardinalpeak.com/blog>
 | LinkedIn<https://cardinalpeak.basecamphq.com/projects/6118601/file/79455231/Linked%20In%20url>

From shortloginsareeasilyremembered at gmail.com  Fri Apr 11 22:54:11 2014
From: shortloginsareeasilyremembered at gmail.com (=?KOI8-R?B?4czFy9PBzsTSIOvVy8zJzg==?=)
Date: Sat, 12 Apr 2014 00:54:11 +0400
Subject: [FFmpeg-user] Usage of AVFrame::motion_val
Message-ID: <CAB5FfcbtJ2WudRijKmCj0mhymKqoFMBuf19ySQPuf3=anxTxCA@mail.gmail.com>

Hello ffmpeg users!

How can I get and set specific motion vectors using the
AVFrame::motion_val? I do not understand the given example and the type of
motion_val. Is it an array of array of pointers? But why does it have
length of 2?

int16_t (*motion_val<http://ffmpeg.org/doxygen/1.2/structAVFrame.html#a8a9ff3a2d5f4191511f516bf0254300e>
[2])[2];

example:
int mv_sample_log2= 4 -
motion_subsample_log2<http://ffmpeg.org/doxygen/1.2/structAVFrame.html#a7cf84898ba50661889cbca26a528be0a>
;
int mb_width= (width<http://ffmpeg.org/doxygen/1.2/structAVFrame.html#a1e71ce60cedd5f3b6811714a9f7f9e0a>
+15)>>4;
int mv_stride= (mb_width << mv_sample_log2) + 1;
motion_val<http://ffmpeg.org/doxygen/1.2/structAVFrame.html#a8a9ff3a2d5f4191511f516bf0254300e>[direction][x
+ y<http://ffmpeg.org/doxygen/1.2/avf__showspectrum_8c.html#aa4f0d3eebc3c443f9be81bf48561a217>*mv_stride][0->mv_x,
1->mv_y];

From mr.artichaut at gmail.com  Sat Apr 12 16:55:20 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Sat, 12 Apr 2014 16:55:20 +0200
Subject: [FFmpeg-user] libavfilter : using amix
Message-ID: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>

Hi,

Using ffmpeg as a library, I'm trying to mix 2 audio files. I'm on it for
weeks now, I know I'm close but there is still something that doesn't work.

My graph is setup with 2 abuffers, 1 amix filter and 1 abuffersink. All are
linked. I read the first input file and write it into the first buffer, I
do the same for the second input file into the second buffer and then I
read the sink and write it into the output file.

But all I have at the end is noise; but not random noise : I can recognize
the rhythm of my 2 inputs. I have no error when I run the program.

My buffers and sink are correctly configured, as are my decoding and
encoding functions because when I try a more simple thing (read one file,
pushing it into the graph made only of a buffer and a sink and then pulling
data out of the sink into a file) it works.

My test program is here :
https://gist.github.com/MrArtichaut/0cd9147441b51113db4d
It's a collection of functions extracted from examples because for now I
just want to validate a solution.

If someone know how to use amix (or libavfilter in general), I would be
very grateful if it could point to me what I'm missing.

Thank you very much.

From cehoyos at ag.or.at  Sat Apr 12 17:43:34 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 12 Apr 2014 15:43:34 +0000 (UTC)
Subject: [FFmpeg-user] feed ffmpeg with quicktime reference?
References: <CAO3XUFzK6z2LaP7QoCgW1BHWZabxHU+Ldw6uBZPNM2x2m-pgeQ@mail.gmail.com>
	<loom.20140406T210159-818@post.gmane.org>
	<CAO3XUFx1OVYAFWjORZfCubbBDczRxNN1gmvVdKiPrGGvGJXc0g@mail.gmail.com>
	<loom.20140406T223025-979@post.gmane.org>
Message-ID: <loom.20140412T174149-601@post.gmane.org>

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

> Johannes Ramson <j.ramson <at> gmail.com> writes:
> 
> > Yes, i checked it twice.
> 
> What does ls show for the directory in question?

Ping.
What is the content for the directory that 
FFmpeg's output prints. (And how can there be 
an absolute path on Windows not starting with a 
disk letter?)

> Please don't top-post here, it is considered rude.

Consider using Google if you have problems 
to understand this...

Consider uploading only the mov file containing the 
reference, I am curious how the drive letter is 
stored.

Carl Eugen


From john_q61 at yahoo.com  Sat Apr 12 18:00:49 2014
From: john_q61 at yahoo.com (John Q)
Date: Sat, 12 Apr 2014 09:00:49 -0700 (PDT)
Subject: [FFmpeg-user] av_interleaved_write_frame(): Unknown error
Message-ID: <1397318449.99578.YahooMailNeo@web122105.mail.ne1.yahoo.com>

I am new to ffmpeg. I am trying to stream my webcam using ffmpeg (x264 encoding) on a windows machine to ffserver on a linux server and restream it. I am getting the following error:?av_interleaved_write_frame(): Unknown error
I checked about it on the net and came across mails which said it could be related to version mismatch between ffmpeg and ffserver. I made sure that all libraries that are used by ffmpeg and ffserver have the same version. I am still getting this error. Appreciate any help in solving this issue.

Warm regards
John

My ffmpeg command is as follows -

.\ffmpeg.exe -f dshow -i video="Webcam C170":audio="Microphone (3- Webcam C170)" -an -g 94 -vcodec libx264 -b:v 512k -r 30 -f flv http://AA.BB.CC.DD:8090/feed1.ffm

[dshow @ 0003f240] real-time buffer 101% full! frame dropped!
? ? Last message repeated 1 times
No pixel format specified, yuv422p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 02563520] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 02563520] profile High 4:2:2, level 3.0, 4:2:2 8-bit
[libx264 @ 02563520] 264 - core 140 r2377 1ca7bb9 - 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 chro
ma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 thre
ads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compa
t=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 o
pen_gop=0 weightp=2 keyint=94 keyint_min=9 scenecut=40 intra_refresh=0 rc_lookahead=4
0 rc=abr mbtree=1 bitrate=512 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_rat
io=1.40 aq=1:1.00
Output #0, flv, to 'http://184.173.147.99:8090/feed1.ffm':
? Metadata:
? ? encoder ? ? ? ? : Lavf55.22.100
? ? Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv422p, 640x480, q=-
1--1, 512 kb/s, 1k tbn, 30 tbc
Stream mapping:
? Stream #0:0 -> #0:0 (rawvideo -> libx264)
Press [q] to stop, [?] for help
frame= ? 59 fps=9.0 q=28.0 size= ? ? ? 3kB time=00:00:00.46 bitrate= ?58.6kbits/s
av_interleaved_write_frame(): Unknown error

In my /etc/ffserver.conf I have the following settings -


<Feed feed1.ffm>

File /tmp/feed1.ffm
##File /mnt/vmtmpfs/feed1.ffm
FileMaxSize 50M


ACL allow 127.0.0.1
ACL allow 0.0.0.0 255.255.255.255

</Feed>

<Stream live.flv>
? ? ? ? Format flv
? ? ? ? Feed feed1.ffm

? ? ? ? VideoCodec libx264
? ? ? ? VideoFrameRate 30
? ? ? ? VideoBitRate 512
? ? ? ? VideoSize 320x240
? ? ? ? AVOptionVideo crf 23
? ? ? ? AVOptionVideo preset medium
? ? ? ? # for more info on crf/preset options, type: x264 --help
? ? ? ? AVOptionVideo flags +global_header
? ? ? ? # NoAudio

? ? ? ? AudioCodec aac
? ? ? ? Strict -2
? ? ? ? AudioBitRate 128
? ? ? ? AudioChannels 2
? ? ? ? AudioSampleRate 44100
? ? ? ? AVOptionAudio flags +global_header
? ? ? ? ACL ALLOW 0.0.0.0 255.255.255.255
</Stream>

From reuben.m at gmail.com  Sat Apr 12 20:07:11 2014
From: reuben.m at gmail.com (Reuben Martin)
Date: Sat, 12 Apr 2014 13:07:11 -0500
Subject: [FFmpeg-user] post-encode stream mapping
In-Reply-To: <loom.20140409T101937-790@post.gmane.org>
References: <CAC9da+6gOMN5VqbBKU4TMXx8uX_irTraLcySio4nHPsrEGTEuQ@mail.gmail.com>
	<loom.20140409T101937-790@post.gmane.org>
Message-ID: <2029554.2B3XIHRhBp@subterfuge>

On Wednesday, April 09, 2014 08:20:06 AM Carl Eugen Hoyos wrote:
> Reuben Martin <reuben.m <at> gmail.com> writes:
> > Is it possible to map an encoded stream to multiple
> > muxers for the purpose of encoding once for multiple
> > outputs?
> 
> Did you test the tee muxer?
> 
 
I looked at the tee muxer, but it seems to have limits in how it can be used from the CLI. Perhaps I don't understand how to use it correctly.

Here's what I use currently:

ffmpeg -fflags +genpts -i rtmp://src/strm \
    -c:a libfdk_aac -vbr 3 -ac 2 -ar 48000 \
    -c:v libx264 -g 31 -crf 20 -profile:v high -preset:v medium -trellis 2 -s 1280x720 -f flv -rtmp_live live rtmp://127.0.0.1/publish/hd \
    -c:a libfdk_aac -vbr 3 -ac 2 -ar 48000 \
    -c:v libx264 -g 31 -crf 22 -profile:v high -preset:v medium -trellis 2 -s 640x360 -f flv  -rtmp_live live rtmp://127.0.0.1/publish/sd \
    -c:a libfdk_aac -vbr 3 -ac 2 -ar 48000 \
    -c:v libx264 -g 31 -crf 24 -profile:v high -preset:v fast -trellis 2 -s 320x180 -f flv -rtmp_live live rtmp://127.0.0.1/publish/ld



I'm not sure how I could use tee to only encode audio once. In theory it should work, but how I would instruct the CLI to do that mapping is not clear.

-Reuben

From george at nsup.org  Sun Apr 13 00:26:31 2014
From: george at nsup.org (Nicolas George)
Date: Sun, 13 Apr 2014 00:26:31 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
Message-ID: <20140412222631.GA31402@phare.normalesup.org>

Le tridi 23 germinal, an CCXXII, Mr Artichaut a ?crit?:
> My test program is here :
> https://gist.github.com/MrArtichaut/0cd9147441b51113db4d
> It's a collection of functions extracted from examples because for now I
> just want to validate a solution.

Your program reads the first file whole and feeds it to the graph, then
reads the second file whole and feeds it to the graph, and only afterwards
read from the filter. That will require enormous amounts of memory, and
possibly fail since libavfilter is not designed for data accumulation in the
middle of filter graphs.

When working with a graph with several inputs, you need to read from the
output, and if it fails due to input being required, add to the correct
input (not both!).

Regards,

-- 
  Nicolas George
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From domenic.disorbo at gdc.com  Fri Apr 11 22:09:55 2014
From: domenic.disorbo at gdc.com (domenic.disorbo at gdc.com)
Date: Fri, 11 Apr 2014 16:09:55 -0400
Subject: [FFmpeg-user] ffmpeg Install problems
Message-ID: <OF1CBA5AE8.5703016F-ON85257CB7.006DBBFC-85257CB7.006EC5A6@gdc.com>

I am following a set of installation instructions and when I do the 
following  I get an error 


wget http://ffmpeg.org/releases/ffmpeg-1.1.3.tar.gz
tar -zxvf ffmpeg-1.1.3.tar.gz
cd ffmpeg-1.1.3
./configure --enable-gpl --enable-libass --enable-libfaac 
--enable-libfdk-aac \
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb 
\
--enable-libspeex --enable-librtmp --enable-libtheora --enable-libvorbis \
--enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree 
--enable-version3 \
--enable-libxvid --enable-libgsm


I get the following error 

ERROR: librtmp 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 tried several times 


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From ekarcher.klein at gmail.com  Sun Apr 13 01:52:30 2014
From: ekarcher.klein at gmail.com (Eric Klein)
Date: Sat, 12 Apr 2014 19:52:30 -0400
Subject: [FFmpeg-user] [FFSERVER] Configuration reload
Message-ID: <CAMN2r+hefwrMyLvtzjC3ZZNRxWEzNF1jiBq1y8h3ufHLurrZcg@mail.gmail.com>

Hi,

I was wondering if there is any way to reload the ffserver configuration
without interrupting the stream ?

Regards

-- 
__________________________________________

KLEIN Eric
Epitech student (france) / UQAR student(canada)
+33 621 621 831 (FR)
+1 418-712-1337 (CA)

From gui456 at yopmail.com  Sat Apr 12 17:01:17 2014
From: gui456 at yopmail.com (Chabloz)
Date: Sat, 12 Apr 2014 08:01:17 -0700 (PDT)
Subject: [FFmpeg-user] avformat_open_input with an https url
Message-ID: <1397314877496-4664753.post@n4.nabble.com>

Hello,

I'm trying to get frames in c++ with av_read_frame from an https video
stream but as soon as I give an https url to the function
avformat_open_input the fuction return a negatif value.

for exemple: av_fromat_input(&pFormatCtx, "https://r1---sn...", NULL, NULL);

will always return a negatif value.
Anyhelp here would be great.
Thanks



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/avformat-open-input-with-an-https-url-tp4664753.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From wkevils at gmail.com  Sat Apr 12 09:07:03 2014
From: wkevils at gmail.com (Kevin Wilson)
Date: Sat, 12 Apr 2014 10:07:03 +0300
Subject: [FFmpeg-user] converting to non HD avi with ffmpeg
In-Reply-To: <CAGXs5wXbA2cVRcZh-_Nw4m+nXkBMUOtbTyvv9DOKxLthRwgYBw@mail.gmail.com>
References: <CAGXs5wWJsFyH_LAC-5F32YtZFNVJ-nwKDKAxYGgaoYruPg5G6A@mail.gmail.com>
	<20120911103539.6717085f@lrcd.com>
	<CAGXs5wUAanJdMDSnS38cOZpNPU8=gD=zeOEDhivz8H02ErRMrw@mail.gmail.com>
	<50516350.4080100@gmail.com>
	<CAGXs5wXbA2cVRcZh-_Nw4m+nXkBMUOtbTyvv9DOKxLthRwgYBw@mail.gmail.com>
Message-ID: <CAGXs5wVXbsX+pxFxyJQqxENoi784v2HCnZtB7XHHADTE8E6Bkw@mail.gmail.com>

Hi,

The command I am used in the past (which Andy suggested, see below) does
not work with ffmpeg on Fedora 20, with ffmpeg-2.1.1-1. I am trying to
convert an mp4 file to a non HD mpeg file. It *must* be non-HD since I want
to play it on a non-HD DVD player
I am getting various errors on various flags.

The command I am using is (ubunutuArm.mp4 is an HD video file):
ffmpeg -i ubunutuArm.mp4 -target ntsc-dvd -bf 2 -flags2 ivlc -dc 10 -b
8000000 -maxrate 9000000 -bufsize 224kB -acodec mp2 -ac 2 -ab 128000 -vf
scale=720:480 ubunutuArmNonHD.mpg

And it says an error about ivlc (and vlc is installed). Googling for that,
I saw some suggestion to try -flags2 intra_vlc -non_linear_quant; I am not
sure it is right, but I tried it, and then it says and error about -dc 10;
I tried omitting -dc 10 and then it said an error about "Unable to parse
option value "-b"". and so on

I would appreciate if someone can tell what should be the command to
convert
an mp4 to a non HD file with this new ffmpeg application.

Regards,
Kevin



On Thu, Sep 13, 2012 at 9:13 PM, Kevin Wilson <wkevils at gmail.com> wrote:

> Andy,
> This worked !
>
> Thanxs.
>
> On Thu, Sep 13, 2012 at 7:38 AM, Andy Civil <andycivil at gmail.com> wrote:
> > On 2012-09-12 10:18 PM, Kevin Wilson wrote:
> >>
> >> Hi, Lou.
> >>
> >> Thanks!
> >>
> >> I have tried:
> >>   ffmpeg -i ubunutuArm.webm  -filter:v scale=640:-1 -q:v 3
> >> ubunutuArmNonHD.avi
> >>
> >
> > I have made a DVD file in the past, can you try this? However, I notice
> that
> > some syntax has changed in the last months, forgive me if some of these
> > parameters need their syntax updating...
> >
> > ffmpeg -i ubunutuArm.webm -target ntsc-dvd -bf 2 -flags2 ivlc -dc 10 -b
> > 8000000 -maxrate 9000000 -bufsize 224kB -acodec mp2 -ac 2 -ab 128000 -vf
> > scale=720:480 ubunutuArmNonHD.mpg
> >
> > --
> > Andy
> >
> > _______________________________________________
> > ffmpeg-user mailing list
> > ffmpeg-user at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From john_q61 at yahoo.com  Sun Apr 13 09:19:17 2014
From: john_q61 at yahoo.com (John Q)
Date: Sun, 13 Apr 2014 00:19:17 -0700 (PDT)
Subject: [FFmpeg-user] av_interleaved_write_frame(): Unknown error
In-Reply-To: <1397318449.99578.YahooMailNeo@web122105.mail.ne1.yahoo.com>
References: <1397318449.99578.YahooMailNeo@web122105.mail.ne1.yahoo.com>
Message-ID: <1397373557.54356.YahooMailNeo@web122104.mail.ne1.yahoo.com>



I am new to ffmpeg. I am trying to stream my webcam using ffmpeg (x264 encoding) on a windows machine to ffserver on a linux server and restream it. I am getting the following error:?av_interleaved_write_frame(): Unknown error
I checked about it on the net and came across mails which said it could be related to version mismatch between ffmpeg and ffserver. I made sure that all libraries that are used by ffmpeg and ffserver have the same version. I am still getting this error. Appreciate any help in solving this issue.

Warm regards
John

My ffmpeg command is as follows -

.\ffmpeg.exe -f dshow -i video="Webcam C170":audio="Microphone (3- Webcam C170)" -an -g 94 -vcodec libx264 -b:v 512k -r 30 -f flv http://AA.BB.CC.DD:8090/feed1.ffm

[dshow @ 0003f240] real-time buffer 101% full! frame dropped!
? ? Last message repeated 1 times
No pixel format specified, yuv422p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 02563520] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 02563520] profile High 4:2:2, level 3.0, 4:2:2 8-bit
[libx264 @ 02563520] 264 - core 140 r2377 1ca7bb9 - 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 chro
ma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 thre
ads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compa
t=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 o
pen_gop=0 weightp=2 keyint=94 keyint_min=9 scenecut=40 intra_refresh=0 rc_lookahead=4
0 rc=abr mbtree=1 bitrate=512 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_rat
io=1.40 aq=1:1.00
Output #0, flv, to 'http://184.173.147.99:8090/feed1.ffm':
? Metadata:
? ? encoder ? ? ? ? : Lavf55.22.100
? ? Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv422p, 640x480, q=-
1--1, 512 kb/s, 1k tbn, 30 tbc
Stream mapping:
? Stream #0:0 -> #0:0 (rawvideo -> libx264)
Press [q] to stop, [?] for help
frame= ? 59 fps=9.0 q=28.0 size= ? ? ? 3kB time=00:00:00.46 bitrate= ?58.6kbits/s
av_interleaved_write_frame(): Unknown error

In my /etc/ffserver.conf I have the following settings -


<Feed feed1.ffm>

File /tmp/feed1.ffm
##File /mnt/vmtmpfs/feed1.ffm
FileMaxSize 50M


ACL allow 127.0.0.1
ACL allow 0.0.0.0 255.255.255.255

</Feed>

<Stream live.flv>
? ? ? ? Format flv
? ? ? ? Feed feed1.ffm

? ? ? ? VideoCodec libx264
? ? ? ? VideoFrameRate 30
? ? ? ? VideoBitRate 512
? ? ? ? VideoSize 320x240
? ? ? ? AVOptionVideo crf 23
? ? ? ? AVOptionVideo preset medium
? ? ? ? # for more info on crf/preset options, type: x264 --help
? ? ? ? AVOptionVideo flags +global_header
? ? ? ? # NoAudio

? ? ? ? AudioCodec aac
? ? ? ? Strict -2
? ? ? ? AudioBitRate 128
? ? ? ? AudioChannels 2
? ? ? ? AudioSampleRate 44100
? ? ? ? AVOptionAudio flags +global_header
? ? ? ? ACL ALLOW 0.0.0.0 255.255.255.255
</Stream>

From mr.artichaut at gmail.com  Sun Apr 13 10:30:22 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Sun, 13 Apr 2014 10:30:22 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <20140412222631.GA31402@phare.normalesup.org>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
Message-ID: <CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>

Thank you, I will do that.

>When working with a graph with several inputs, you need to read from the
>output, and if it fails due to input being required, add to the correct
>input (not both!).

How do I know which input is required when reading from the sink ? Because
it is after the amix filter, my sink only has one input.

I was doing that because my inputs will always be small audio files (less
than 20s) ans there will never be more than 3 of them.

From george at nsup.org  Sun Apr 13 10:42:06 2014
From: george at nsup.org (Nicolas George)
Date: Sun, 13 Apr 2014 10:42:06 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
Message-ID: <20140413084205.GA18477@phare.normalesup.org>

Le quartidi 24 germinal, an CCXXII, Mr Artichaut a ?crit?:
> How do I know which input is required when reading from the sink ? Because
> it is after the amix filter, my sink only has one input.

The av_buffersrc_get_nb_failed_requests() function is exactly for that.

Regards,

-- 
  Nicolas George
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From battistel at gmail.com  Sun Apr 13 14:58:27 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Sun, 13 Apr 2014 14:58:27 +0200
Subject: [FFmpeg-user] Possible issue with asf muxer
In-Reply-To: <CAHss8oES8SmM3f9U57_ArOu4HHTVqX8UER8Aq87Xz1UPU1+dhw@mail.gmail.com>
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
	<loom.20140402T114402-194@post.gmane.org>
	<CAHss8oES8SmM3f9U57_ArOu4HHTVqX8UER8Aq87Xz1UPU1+dhw@mail.gmail.com>
Message-ID: <CAHss8oGvzSRF=GiifwNRnRO9MGbX2RdExtJa2Q_FxVR-uyvxQw@mail.gmail.com>

I'd like to know if you've been able to reproduce the issue.

Additional information: I've seen that intermission2.asf has a very bad
interleaving, something like:

(V=video packet, A=audio packet)
VVVVVVVVVVVVVVVVVVVVVVVV...AAAAAAAAAAAAAAAAAAAAAAA...and so on for more
than half of file. Then in the end it turn into normality VAVAVAVA....

Remuxing the file into asf again reorders packets correctly.
(VAVAVAVAV...till the end).

ffmpeg -i intermission2.asf -c copy intermission3.asf


let me know,
thanks,
MB



2014-04-03 18:34 GMT+02:00 Massimo Battistel <battistel at gmail.com>:

> I tested this command line (without -ac 2 -ar 48000 -f asf)
>> with your sample and the output file plays fine with WMP
>> (and vlc) afaict.
>>
>
> I've tested the same command line (the one without -ac 2 -ar 48000 -f asf)
> with latest ffmpeg Zeranoe builds and vlc and I still see the issue (both
> ffplay and vlc 2.1.3, wmp is fine).
> Video freezes for the first 8 seconds.
> My platform is Windows 7 64bit.
>
>
> How does converting the output file with ffmpeg work?
>
>
> I don't understand. Can you elaborate?
>
>
> Thanks,
> MB
>



-- 
Massimo Battistel.

From cehoyos at ag.or.at  Sun Apr 13 16:41:59 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 13 Apr 2014 14:41:59 +0000 (UTC)
Subject: [FFmpeg-user] Possible issue with asf muxer
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
	<loom.20140402T114402-194@post.gmane.org>
	<CAHss8oES8SmM3f9U57_ArOu4HHTVqX8UER8Aq87Xz1UPU1+dhw@mail.gmail.com>
	<CAHss8oGvzSRF=GiifwNRnRO9MGbX2RdExtJa2Q_FxVR-uyvxQw@mail.gmail.com>
Message-ID: <loom.20140413T164056-915@post.gmane.org>

Massimo Battistel <battistel <at> gmail.com> writes:

> I'd like to know if you've been able to reproduce the issue.

I opened ticket #3555
Thank you for being insistent, I originally misunderstood 
the issue!

For future reports: Please remember that top-posting is 
strongly discouraged here and please fix your quoting.

Carl Eugen


From cehoyos at ag.or.at  Sun Apr 13 16:44:00 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 13 Apr 2014 14:44:00 +0000 (UTC)
Subject: [FFmpeg-user]
	=?utf-8?q?avformat=5Fopen=5Finput_with_an_https_url?=
References: <1397314877496-4664753.post@n4.nabble.com>
Message-ID: <loom.20140413T164305-327@post.gmane.org>

Chabloz <gui456 <at> yopmail.com> writes:

> av_fromat_input(&pFormatCtx, "https://r1---sn...", NULL, NULL);
> 
> will always return a negatif value.

Did you test with ffmpeg (the application)?
What does the console output report?
What version of FFmpeg are you testing 
and how does your configure line look like?

Carl Eugen


From cehoyos at ag.or.at  Sun Apr 13 16:47:57 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 13 Apr 2014 14:47:57 +0000 (UTC)
Subject: [FFmpeg-user] ffmpeg Install problems
References: <OF1CBA5AE8.5703016F-ON85257CB7.006DBBFC-85257CB7.006EC5A6@gdc.com>
Message-ID: <loom.20140413T164415-972@post.gmane.org>

 <domenic.disorbo <at> gdc.com> writes:

> wget http://ffmpeg.org/releases/ffmpeg-1.1.3.tar.gz

This is old and unsupported.
If you are not a distributor, current FFmpeg git head 
is always recommended.

> ERROR: librtmp not found

> check_pkg_config librtmp librtmp/rtmp.h RTMP_Socket
> ERROR: librtmp not found

Did you install librtmp?
(Why do you need it, there is a native implementation?)

Carl Eugen


From wally at voosen.eu  Sun Apr 13 19:47:53 2014
From: wally at voosen.eu (wally)
Date: Sun, 13 Apr 2014 19:47:53 +0200
Subject: [FFmpeg-user] mpg to mp4 request
In-Reply-To: <loom.20140411T192948-32@post.gmane.org>
References: <53444185.50609@voosen.eu>	<loom.20140409T211826-560@post.gmane.org>	<53467150.8090505@voosen.eu>
	<loom.20140411T192948-32@post.gmane.org>
Message-ID: <534ACDC9.4040906@voosen.eu>

Hello Carl Eugen,

sorry for delay, i had not internet access last days.

I have no idea how to test if it is only a size effect.
But i achieved acceptable results with this command:
ffmpeg -i source.mpg -vcodec *mpeg4* -vb 1.9M -acodec aac -strict -2 -ab 
90k target.mp4

Of course i do not know why mpeg4 works and libx264 does not.

I can live with this but little smaller results would be nice.

wally




On 04/11/2014 07:30 PM, Carl Eugen Hoyos wrote:
> wally <wally <at> voosen.eu> writes:
>
>> After this test i tried with the entire sourcefile (3.4 GB).
>> It results in a 1.3GB mp4 file which doesn't work on the
>> receiver.
> Did you test if this is just an effect of the output file size?
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


From george at nsup.org  Sun Apr 13 20:15:45 2014
From: george at nsup.org (Nicolas George)
Date: Sun, 13 Apr 2014 20:15:45 +0200
Subject: [FFmpeg-user] concat muxer with two video files with two
	different resolutions
In-Reply-To: <CANgFRXB=pDf-t-bB5YG8uRa2bCWyy+0qNVim45kemqJxFSm5eg@mail.gmail.com>
References: <CANgFRXB=pDf-t-bB5YG8uRa2bCWyy+0qNVim45kemqJxFSm5eg@mail.gmail.com>
Message-ID: <20140413181545.GA28986@phare.normalesup.org>

Le duodi 22 germinal, an CCXXII, Ben Mesander a ?crit?:
>   Today I tried to concatenate two h.264 video files with the concat muxer.
> The two files have differing input resolutions (848x480 and 1280x720). I
> would like the resulting video to be 640x480. I used today's static 32-bit
> build from ffmpeg.org.

> In the resulting output mp4 file, the first video file appears as the
> input, correctly sized to 640x480. The second video file shows up as mostly
> grey with random green macroblocks and flickery random pixels here and
> there.
> 
> So it appears an (undocumented? I don't see it) feature of the concat muxer
> is the input files must be the same resolution, so this approach simply
> won't work?

I can reproduce the problem. I suspect it has something to do with the
global extradata, I must look into it and understand what is going on.

You probably do not want to create a file that has resolution changes in the
middle: you will be transcoding. Therefore, I suspect the concat filter
would be more adapted to your needs.

Regards,

-- 
  Nicolas George
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From battistel at gmail.com  Mon Apr 14 19:53:01 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Mon, 14 Apr 2014 19:53:01 +0200
Subject: [FFmpeg-user]  Stream blocks when system time changes
Message-ID: <CAHss8oFSZb0vjw3qR8O=U36zhpPWqaVSNGoDywhHS2o64mxUOg@mail.gmail.com>

hello,
with the following command line you generate a stream to localhost:

ffmpeg -re -i "anyvideo" -vcodec libx264 -pix_fmt yuv420p -b:v 1M  -acodec
libvo_aacenc -b:a 64k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1400

Open the previous stream with VLC, then put system clock 1 minute back.
Stream will block almost immediately.

I would like to know if it is possible to tell ffmpeg to use a different
clock reference or if there are workarounds for this issue.

This was just an example, the real case is daylight saving time (and I
can't disable it).

Thanks,
MB

From olivier.pis.langlois at transport.alstom.com  Mon Apr 14 20:21:22 2014
From: olivier.pis.langlois at transport.alstom.com (LANGLOIS Olivier PIS -EXT)
Date: Mon, 14 Apr 2014 18:21:22 +0000
Subject: [FFmpeg-user] Stream blocks when system time changes
In-Reply-To: <CAHss8oFSZb0vjw3qR8O=U36zhpPWqaVSNGoDywhHS2o64mxUOg@mail.gmail.com>
References: <CAHss8oFSZb0vjw3qR8O=U36zhpPWqaVSNGoDywhHS2o64mxUOg@mail.gmail.com>
Message-ID: <89F8E4011FD7234ABFE48744E300CBC206C285F4@041-DB3MPN1-091.041d.mgd.msft.net>

> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Massimo Battistel
> Sent: Monday, April 14, 2014 1:53 PM
> To: FFmpeg user questions
> Subject: [FFmpeg-user] Stream blocks when system time changes
>
> hello,
> with the following command line you generate a stream to localhost:
>
> ffmpeg -re -i "anyvideo" -vcodec libx264 -pix_fmt yuv420p -b:v 1M  -acodec
> libvo_aacenc -b:a 64k -ac 2 -ar 48000 -f mpegts udp://127.0.0.1:1400
>
> Open the previous stream with VLC, then put system clock 1 minute back.
> Stream will block almost immediately.
>
> I would like to know if it is possible to tell ffmpeg to use a different clock
> reference or if there are workarounds for this issue.
>
> This was just an example, the real case is daylight saving time (and I can't
> disable it).
>
> Thanks,
> MB

Hi Massimo,

I am currently doing some work to propose to ffmpeg dev to robustify the project in that aspect. I can point you out where to fetch v1 of my patch if you want to try it out in your setup but before I have 2 questions.

1. On which platform are you running ffmpeg (my patch only address the issue on Linux)?
2. Would you be able to send the stream on a different machine (in order be sure that the freeze is in ffmpeg and not in the player...)?

Please ignore the confidentiality notice below.
It is automatically added without my consent.



________________________________
CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium.

From george at nsup.org  Mon Apr 14 20:50:31 2014
From: george at nsup.org (Nicolas George)
Date: Mon, 14 Apr 2014 20:50:31 +0200
Subject: [FFmpeg-user] Stream blocks when system time changes
In-Reply-To: <CAHss8oFSZb0vjw3qR8O=U36zhpPWqaVSNGoDywhHS2o64mxUOg@mail.gmail.com>
References: <CAHss8oFSZb0vjw3qR8O=U36zhpPWqaVSNGoDywhHS2o64mxUOg@mail.gmail.com>
Message-ID: <20140414185031.GA7579@phare.normalesup.org>

Le quintidi 25 germinal, an CCXXII, Massimo Battistel a ?crit?:
> This was just an example, the real case is daylight saving time (and I
> can't disable it).

Daylight saving time should not affect ffmpeg at all (except for a few
features to include the current time in subtitles / overlay text). If it
does, it means that something is very wrong in your system setup.

Can you describe precisely what your system is and what happens when DST
begins?

Regards,

-- 
  Nicolas George
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From digital.h at gmail.com  Mon Apr 14 21:19:56 2014
From: digital.h at gmail.com (Ivan Hoffmann)
Date: Mon, 14 Apr 2014 16:19:56 -0300
Subject: [FFmpeg-user] ffmetadata issue
Message-ID: <CADvHueYmz010Jo6QFLbmY6Fu-Pzj8NshSE-MAej-wh3=ywA0tw@mail.gmail.com>

Hi all,

I'm trying to get metadata output to a file. I'm using the following
command, as stated in ffmpeg's manual:

ffmpeg -i INPUT_FILE -f ffmetadata output.txt

But in the resulting output.txt file all I get is ;FFMETADATA1 and nothing
more, even though I get the metadata printed in console.

What am I doing wrong?

Any help would be appreciated.

*     .:: Iv?n Hoffmann ::.*
~{ Artesano del Tresd? }~
     digital.h at gmail.com
       @digitalhoffmann

From battistel at gmail.com  Mon Apr 14 21:26:32 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Mon, 14 Apr 2014 21:26:32 +0200
Subject: [FFmpeg-user] Stream blocks when system time changes
In-Reply-To: <20140414185031.GA7579@phare.normalesup.org>
References: <CAHss8oFSZb0vjw3qR8O=U36zhpPWqaVSNGoDywhHS2o64mxUOg@mail.gmail.com>
	<20140414185031.GA7579@phare.normalesup.org>
Message-ID: <CAHss8oGKH8z2pVrKBOac2-LhwCRGyr9B=Qd98i6uGZojLeADUQ@mail.gmail.com>

Thanks to all for quick answers.

For Nicolas:

forget what I told about DST. I did not change the date, just the hour. I
made I test now and DST does not give problems to ffmpeg.


For Olivier:

1. Unfortunely my platform is Windows 7 64 bit.
2. I made the test. The freeze is not in the player, is in ffmpeg.



Anyway it would be useful to have another clock reference to avoid that
small clock adjustment could freeze a stream.


Thanks,
MB

From slanaconda at gmail.com  Mon Apr 14 23:49:24 2014
From: slanaconda at gmail.com (Phunkadilic)
Date: Mon, 14 Apr 2014 14:49:24 -0700 (PDT)
Subject: [FFmpeg-user] Force concat to ignore corrupt files?
Message-ID: <1397512164694-4664775.post@n4.nabble.com>

I'm currently using this command to concat individual video files every
night:

"ffmpeg.exe -f concat -i C:\path\video-files.txt -c copy
C:\Desitination\path\concat-file.mp4"

I've noticed that if there's a single corrupt .mp4 file, the command will
stop and concat only the files that were good before hitting the corrupt
file.  Is there a way/command to have ffmpeg to either skip any corrupt
files or to force through and ignore errors?  

This script runs automatically nightly and I'd rather not have to babysit
these files.  If there's one or two corrupt files, I'd rather them just be
skipped and not worry about having to go through and remove the corrupt
files and re-do the process manually.

Thanks in advance!



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Force-concat-to-ignore-corrupt-files-tp4664775.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From kirill89 at gmail.com  Tue Apr 15 09:47:54 2014
From: kirill89 at gmail.com (Byte)
Date: Tue, 15 Apr 2014 14:47:54 +0700
Subject: [FFmpeg-user] FFmpeg stops at once after start streaming
Message-ID: <CAGN63oPr1fFNh-k2rR1RY6+7xQB5Asd+8Efw_PkCYSAtOFngjw@mail.gmail.com>

I use FFmpeg like this:

    ffmpeg -i "http://xxx/mpegts" -map 0:2 -map 0:0 -vcodec copy -acodec
copy -f flv rtmp://localhost/live/xxx

At once after start i see this:

    ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c)
2000-2013 the Libav developers
    built on Feb  6 2014 20:56:59 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.
    [h264 @ 0xdec460] non-existing PPS referenced
    [h264 @ 0xdec460] non-existing PPS 0 referenced
    [h264 @ 0xdec460] decode_slice_header error
    [h264 @ 0xdec460] no frame!
    ... (previous 4 lines repeats 5-6 times)
    [mpegts @ 0xde27a0] max_analyze_duration reached
    [mpegts @ 0xde27a0] Estimating duration from bitrate, this may be
inaccurate
    Input #0, mpegts, from 'http://xxx/mpegts':
      Duration: N/A, start: 63077.826856, bitrate: 255 kb/s
      Program 1
        Stream #0.0[0x44](eng): Audio: aac, 48000 Hz, stereo, s16, 122 kb/s
        Stream #0.1[0x45](rus): Audio: aac, 48000 Hz, stereo, s16, 132 kb/s
        Stream #0.2[0x46]: Video: h264 (Constrained Baseline), yuv420p,
720x576 [PAR 64:45 DAR 16:9], 26.51 fps, 25 tbr, 90k tbn, 50 tbc
    Output #0, flv, to 'rtmp://localhost/live/xxx':
      Metadata:
        encoder         : Lavf53.21.1
        Stream #0.0: Video: [7][0][0][0] / 0x0007, yuv420p, 720x576 [PAR
64:45 DAR 16:9], q=2-31, 1k tbn, 25 tbc
        Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, 122 kb/s
    Stream mapping:
      Stream #0.2 -> #0.0
      Stream #0.0 -> #0.1
    Press ctrl-c to stop encoding
    developer at myhost:

And it is all. Process stops. In RTMP server i see new connection what at
once disconnected.
What may be wrong?

Thanks.

From cehoyos at ag.or.at  Tue Apr 15 10:36:33 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 15 Apr 2014 08:36:33 +0000 (UTC)
Subject: [FFmpeg-user] FFmpeg stops at once after start streaming
References: <CAGN63oPr1fFNh-k2rR1RY6+7xQB5Asd+8Efw_PkCYSAtOFngjw@mail.gmail.com>
Message-ID: <loom.20140415T103436-378@post.gmane.org>

Byte <kirill89 <at> gmail.com> writes:

>     ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1

This is not FFmpeg but an intentionally broken version 
with several hundred known bugs not present in FFmpeg, 
some of them security relevant. Please understand that 
we cannot support this here.
See http://ffmpeg.org/download.html for supported versions.
See http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html 
for more information.

Carl Eugen



From neve_capricorn at gmx.de  Tue Apr 15 12:32:37 2014
From: neve_capricorn at gmx.de (neve_capricorn at gmx.de)
Date: Tue, 15 Apr 2014 12:32:37 +0200
Subject: [FFmpeg-user] AVFoundation audio device
Message-ID: <534D0AC5.4060804@gmx.de>

the latest git version that i compiled for osx works great with the 
avfoundation devices.

ffmpeg -f avfoundation -video_device_index 0 -i "default" out.mpg

captures fine videoframes into an mpg stream. the underlaying 
avfoundation core audio is not possible as i seen, so i tried:

ffmpeg -f avfoundation -audio_device_index 0 -i "default" out.mpg

a short look in the avfoundation source seems that it?s only for video 
right now.

is there a way how i can capture a audio stream with ffmpeg under osx 
10.9 together with avfoundation to one muxed mpg or similar file container?

thx for help
tom


From battistel at gmail.com  Tue Apr 15 14:55:47 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Tue, 15 Apr 2014 14:55:47 +0200
Subject: [FFmpeg-user]  Understanding multiple aspect ratios
Message-ID: <CAHss8oH7mf-Tp_SQNPqRkz9CEftY=y86JDyu3EVGR8RHX2T=Rg@mail.gmail.com>

Hello,
I have some files that show multiple aspect ratios if probed with ffprobe.
I would like to know how to interpret these results.

The following example gives me 4:3 and 295:216. While I'm expecting 4:3 is
the correct one, it looks that scale filter is using the second value
(295:216).

ffprobe itv.mov

ffprobe version N-62058-gfd2bcfc Copyright (c) 2007-2014 the FFmpeg
developers
  built on Apr  2 2014 22:09:31 with gcc 4.8.2 (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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 71.100 / 52. 71.100
  libavcodec     55. 56.107 / 55. 56.107
  libavformat    55. 36.101 / 55. 36.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'itv.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 537199360
    compatible_brands: qt
    creation_time   : 2008-10-03 16:14:46
  Duration: 00:01:10.00, start: 0.000000, bitrate: 30337 kb/s
    Stream #0:0(eng): Audio: pcm_s16le (lpcm / 0x6D63706C), 48000 Hz,
stereo, s16, 1536 kb/s (default)
    Metadata:
      creation_time   : 2008-10-03 16:14:46
      handler_name    : Apple Alias Data Handler
    Stream #0:1(eng): Subtitle: mov_text (text / 0x74786574), 720x20, 0 kb/s
    Metadata:
      creation_time   : 2008-10-03 16:22:31
      handler_name    : Apple Alias Data Handler
    Stream #0:2(eng): Video: dvvideo (dvcp / 0x70637664), yuv420p, 720x576
[SAR 16:15 DAR 4:3], 28800 kb/s, SAR 59:54 DAR 295:216, 25 fps, 25 tbr, 25
tbn, 25 tbc (default)
    Metadata:
      creation_time   : 2008-10-03 16:58:34
      handler_name    : Apple Alias Data Handler


thanks,
MB

From ckp21082 at gmail.com  Tue Apr 15 15:01:38 2014
From: ckp21082 at gmail.com (chirag parikh)
Date: Tue, 15 Apr 2014 18:31:38 +0530
Subject: [FFmpeg-user] Understanding multiple aspect ratios
In-Reply-To: <CAHss8oH7mf-Tp_SQNPqRkz9CEftY=y86JDyu3EVGR8RHX2T=Rg@mail.gmail.com>
References: <CAHss8oH7mf-Tp_SQNPqRkz9CEftY=y86JDyu3EVGR8RHX2T=Rg@mail.gmail.com>
Message-ID: <CAKr8CtNE7u8fZM1G8wJzCiLi3SrDHFUcatyMWug+7YerR-O6Kw@mail.gmail.com>

unsubscribe  me


On Tue, Apr 15, 2014 at 6:25 PM, Massimo Battistel <battistel at gmail.com>wrote:

> Hello,
> I have some files that show multiple aspect ratios if probed with ffprobe.
> I would like to know how to interpret these results.
>
> The following example gives me 4:3 and 295:216. While I'm expecting 4:3 is
> the correct one, it looks that scale filter is using the second value
> (295:216).
>
> ffprobe itv.mov
>
> ffprobe version N-62058-gfd2bcfc Copyright (c) 2007-2014 the FFmpeg
> developers
>   built on Apr  2 2014 22:09:31 with gcc 4.8.2 (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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc
> --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264
> --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
>   libavutil      52. 71.100 / 52. 71.100
>   libavcodec     55. 56.107 / 55. 56.107
>   libavformat    55. 36.101 / 55. 36.101
>   libavdevice    55. 11.100 / 55. 11.100
>   libavfilter     4.  3.100 /  4.  3.100
>   libswscale      2.  6.100 /  2.  6.100
>   libswresample   0. 18.100 /  0. 18.100
>   libpostproc    52.  3.100 / 52.  3.100
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'itv.mov':
>   Metadata:
>     major_brand     : qt
>     minor_version   : 537199360
>     compatible_brands: qt
>     creation_time   : 2008-10-03 16:14:46
>   Duration: 00:01:10.00, start: 0.000000, bitrate: 30337 kb/s
>     Stream #0:0(eng): Audio: pcm_s16le (lpcm / 0x6D63706C), 48000 Hz,
> stereo, s16, 1536 kb/s (default)
>     Metadata:
>       creation_time   : 2008-10-03 16:14:46
>       handler_name    : Apple Alias Data Handler
>     Stream #0:1(eng): Subtitle: mov_text (text / 0x74786574), 720x20, 0
> kb/s
>     Metadata:
>       creation_time   : 2008-10-03 16:22:31
>       handler_name    : Apple Alias Data Handler
>     Stream #0:2(eng): Video: dvvideo (dvcp / 0x70637664), yuv420p, 720x576
> [SAR 16:15 DAR 4:3], 28800 kb/s, SAR 59:54 DAR 295:216, 25 fps, 25 tbr, 25
> tbn, 25 tbc (default)
>     Metadata:
>       creation_time   : 2008-10-03 16:58:34
>       handler_name    : Apple Alias Data Handler
>
>
> thanks,
> MB
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From jshupert at pps-inc.com  Tue Apr 15 15:35:42 2014
From: jshupert at pps-inc.com (Jim Shupert)
Date: Tue, 15 Apr 2014 09:35:42 -0400
Subject: [FFmpeg-user] 6720x1200 proRes to wmv
Message-ID: <534D35AE.5030304@pps-inc.com>

An intresting matter - just thought I would share.

Goal a 6720x1200 proRes into a wmv

once upon a time i *thought I had seen where the minimum size supported 
was 80x80 & the max was 4096x4096 ... but I cannot find anything 
(google) now.

I Can make 3360 x 600 1680 x 300.

I have tried declaring the frm size and pix_fmt ... but anytime I try to 
make the size of 6720x1200 i see difficulties.


I find it interesting that the mov reports to be :

SAR 6720:6720

ffmpeg -i
/media/data/p/Barrier_4_Rough_5.mov  -b:v 20000k -vcodec wmv2  -s 3360x600
-ar 48000 -b:a 224k -ac 2 -y /media/data/p/Barrier_4_Rough_5-21j.wmv
ffmpeg version git-2013-06-03-b4d4ef5 Copyright (c) 2000-2013 the FFmpeg
developers
   built on Jun  3 2013 16:50:07 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
   configuration: --extra-libs=-ldl --enable-gpl --enable-libass
--enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libspeex --enable-libtheora
--enable-libvorbis --enable-libopus --enable-libvpx --enable-x11grab
--enable-libx264 --enable-nonfree --enable-version3
   libavutil      52. 34.100 / 52. 34.100
   libavcodec     55. 15.100 / 55. 15.100
   libavformat    55.  8.102 / 55.  8.102
   libavdevice    55.  2.100 / 55.  2.100
   libavfilter     3. 74.101 /  3. 74.101
   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
'/media/data/p/Barrier_4_Rough_5.mov':
   Metadata:
     major_brand     : qt
     minor_version   : 537199360
     compatible_brands: qt
     creation_time   : 2014-04-11 19:17:15
   Duration: 00:01:56.45, start: 0.000000, bitrate: 670036 kb/s
     Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le,
6720x1200, 668428 kb/s, SAR 6720:6720 DAR 28:5, 23.98 fps, 23.98 tbr,
24k tbn, 24k tbc
     Metadata:
       creation_time   : 2014-04-11 19:17:15
       handler_name    : Apple Alias Data Handler
       timecode        : 00:00:07:12
     Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz,
stereo, s16, 1536 kb/s
     Metadata:
       creation_time   : 2014-04-11 19:17:15
       handler_name    : Apple Alias Data Handler
       timecode        : 00:00:07:12
     Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
     Metadata:
       creation_time   : 2014-04-11 19:17:15
       handler_name    : Apple Alias Data Handler
       timecode        : 00:00:07:12
Output #0, asf, to '/media/data/p/Barrier_4_Rough_5-21j.wmv':
   Metadata:
     major_brand     : qt
     minor_version   : 537199360
     compatible_brands: qt
     WM/EncodingSettings: Lavf55.8.102
     Stream #0:0(eng): Video: wmv2 (WMV2 / 0x32564D57), yuv420p, 3360x600
[SAR 1:1 DAR 28:5], q=2-31, 20000 kb/s, 1k tbn, 23.98 tbc
     Metadata:
       creation_time   : 2014-04-11 19:17:15
       handler_name    : Apple Alias Data Handler
       timecode        : 00:00:07:12
     Stream #0:1(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz,
stereo, fltp, 224 kb/s
     Metadata:
       creation_time   : 2014-04-11 19:17:15
       handler_name    : Apple Alias Data Handler
       timecode        : 00:00:07:12
Stream mapping:
   Stream #0:0 -> #0:0 (prores -> wmv2)
   Stream #0:1 -> #0:1 (pcm_s16le -> wmav2)
Press [q] to stop, [?] for help
warning, clipping 1 dct coefficients to -255..255:00:20.95
bitrate=15899.2kbits/s
warning, clipping 1 dct coefficients to -255..255:00:20.95
bitrate=16016.5kbits/s
frame=  491 fps=9.3 q=2.0 Lsize=   41048kB time=00:00:20.95
bitrate=16050.8kbits/s
video:40016kB audio:574kB subtitle:0 global headers:0kB muxing overhead
1.129518%


one gets some of this

warning, clipping 1 dct coefficients to -255..255:00:20.95
bitrate=29186.8kbits/s
warning, clipping 1 dct coefficients to -255..255:00:20.95
bitrate=29412.8kbits/s
[wmv2 @ 0x38ad720] warning, clipping 1 dct coefficients to -255..255
     Last message repeated 5 times
warning, clipping 1 dct coefficients to -255..255:00:20.95
bitrate=29566.8kbits/s
warning, clipping 1 dct coefficients to -255..255:00:20.95
bitrate=29802.6kbits/s
warning, clipping 1 dct coefficients to -255..255:00:21.97
bitrate=28917.1kbits/s
warning, clipping 1 dct coefficients to -255..255:00:23.97
bitrate=28238.4kbits/s
warning, clipping 1 dct coefficients to -255..255:00:24.96
bitrate=27660.8kbits/s
[wmv2 @ 0x38ad720] warning, clipping 1 dct coefficients to -255..255

which I would be interested in knowing more about that
In VLC the color is all shited , like the colorspace is wrong.

I can play in ffplay if I
ffplay -vf "scale=1680:300" -i /media/data/p/Barrier_4_Rough_5-30_j.wmv
and the color is normal

Mostly this is just bit of fun I thought I would share ... see what 
others might say.

and If you are wondering WHY someone wishes for a wmv that is 
6720x1200.....???
I am told they wish to play it in a power point.   wow ---- really
so winders all the way.
I was not successful in finding win info on max supported frm size for wmp.

cheers.

jS
It is the kinda thing ....that makes one smile

From andoru.blah at gmail.com  Wed Apr 16 01:57:41 2014
From: andoru.blah at gmail.com (Andoru)
Date: Tue, 15 Apr 2014 23:57:41 +0000 (UTC)
Subject: [FFmpeg-user] Determined to not find libvpx
Message-ID: <loom.20140416T014435-385@post.gmane.org>

Sorry if this is "noobish" problem, but I followed this compiling guide for
Debian: http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide 

I went through it entirely, making sure I get all the paths right, but
ffmpeg doesn't seem to want to find libvpx.
Here's config.log:
http://s000.tinyupload.com/index.php?file_id=00062031169542193809

I've checked the log and there is indeed 
~/ffmpeg_build/includes/vpx/vpx-decoder.h and all the other include files,
so I have no idea what went wrong.


From nickrobbins at yahoo.com  Wed Apr 16 02:47:01 2014
From: nickrobbins at yahoo.com (Nicholas Robbins)
Date: Tue, 15 Apr 2014 17:47:01 -0700 (PDT)
Subject: [FFmpeg-user] Strange -vf fps and -r behavior
Message-ID: <1397609221.55202.YahooMailBasic@web160804.mail.bf1.yahoo.com>

So I have a mkv that I pulled from a DVD. It thinks it is "30" fps but it is really "24". ffprobe gives 

Stream #0:0(eng): Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9 DAR 4:3], 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc

but if I actually count the frames I get 23.97 fps. Ok so when I transcode it to libx264 with the command 

ffmpeg -i IN.mkv -map 0 -c:a copy -c:s copy -c:v libx264 -crf 25 -preset ultrafast -vf fps=fps=24000/1001 OUT.mkv

Some parts of it now appear to be at 12 fps. Stepping frame by frame it appears that in those sections the video is at 24 fps but each frame is doubled.?

If I extract the parts that do this and run them through the command on their own, no problem. Is this a bug, or am I doing something wrong?

I can make the problem go away if I use -r 24000/1001 instead, but my understanding is that -vf fps is preferred to -r.

Console output of "-vf fps=fps=24000/1001" version follows:

ffmpeg version N-62473-g79d82a6 Copyright (c) 2000-2014 the FFmpeg developers
  built on Apr 15 2014 20:04:32 with gcc 4.7.3 (Gentoo Hardened 4.7.3-r1 p1.4, pie-0.5.5)
  configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-
ar --optflags='-O2 -pipe -march=native -fomit-frame-pointer' --extra-cflags='-O2 -pipe -march=native -fomit-frame-pointer' --extra-cxxflags='-O2 -pipe -march=native -fomit-frame-pointer' --disable-
static --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --enable-nonfree --disable-indev=v4l2 --disable-outdev=v4l2 --disable-indev=alsa --disable-indev=oss 
--disable-indev=jack --disable-outdev=alsa --disable-outdev=oss --disable-outdev=sdl --enable-bzlib --enable-runtime-cpudetect --enable-debug --disable-doc --disable-gnutls --enable-hardcoded-table
s --enable-iconv --enable-network --disable-openssl --disable-ffplay --disable-vaapi --disable-vdpau --disable-xlib --enable-zlib --disable-libvo-aacenc --disable-libvo-amrwbenc --enable-libmp3lame
 --disable-libaacplus --enable-libfaac --disable-libtheora --disable-libtwolame --disable-libwavpack --disable-libwebp --enable-libx264 --disable-libx265 --enable-libxvid --disable-libcdio --disabl
e-libiec61883 --disable-libdc1394 --disable-libcaca --disable-openal --disable-opengl --disable-libv4l2 --disable-libpulse --disable-x11grab --disable-libflite --disable-frei0r --disable-fontconfig
 --disable-ladspa --disable-libass --disable-libfreetype --disable-libsoxr --enable-pthreads --disable-libopencore-amrwb --disable-libopencore-amrnb --disable-libfdk-aac --disable-libopenjpeg --dis
able-libbluray --disable-libcelt --disable-libgme --disable-libgsm --disable-libmodplug --disable-libopus --disable-libquvi --disable-librtmp --disable-libssh --disable-libschroedinger --disable-li
bspeex --disable-libvorbis --disable-libvpx --disable-libzvbi --disable-armv5te --disable-armv6 --disable-armv6t2 --disable-neon --disable-vfp --disable-mips32r2 --disable-mipsdspr1 --disable-mipsd
spr2 --disable-mipsfpu --disable-altivec --disable-amd3dnow --disable-amd3dnowext --disable-avx --disable-avx2 --disable-fma3 --disable-fma4 --disable-sse3 --disable-sse4 --disable-sse42 --enable-pic --cpu=host
  libavutil      52. 76.100 / 52. 76.100
  libavcodec     55. 58.103 / 55. 58.103
  libavformat    55. 37.100 / 55. 37.100
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libavresample   1.  2.  0 /  1.  2.  0
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, matroska,webm, from '/var/srv/media/videos/NO-BACKUP-holding/transcoding/2014-04-15-Tue-20-38/Law.and.Order.Criminal.Intent.2001.S03E02.film.mkv':
  Metadata:
    encoder         : libmakemkv v1.8.8 (1.3.0/1.4.1) x86_64-pc-linux-gnu
    creation_time   : 2014-02-14 03:09:14
  Duration: 00:43:56.37, start: 0.000000, bitrate: 5981 kb/s
    Chapter #0.0: start 0.000000, end 615.748467
    Metadata:
      title           : Chapter 01
    Chapter #0.1: start 615.748467, end 1140.539400
    Metadata:
      title           : Chapter 02
    Chapter #0.2: start 1140.539400, end 1677.475800
    Metadata:
      title           : Chapter 03
    Chapter #0.3: start 1677.475800, end 2636.367067
    Metadata:
      title           : Chapter 04
    Stream #0:0(eng): Video: mpeg2video (Main), yuv420p(tv, smpte170m), 720x480 [SAR 8:9 DAR 4:3], max. 9800 kb/s, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc
    Stream #0:1(eng): Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
    Metadata:
      title           : Stereo
    Stream #0:2(eng): Subtitle: dvd_subtitle, 720x480
    Stream #0:3(spa): Subtitle: dvd_subtitle, 720x480
    Stream #0:4(fre): Subtitle: dvd_subtitle, 720x480
[libx264 @ 0x34b9e753d0] using SAR=8/9
[libx264 @ 0x34b9e753d0] using cpu capabilities: MMX2 SSE2Fast SSSE3 Cache64
[libx264 @ 0x34b9e753d0] profile Constrained Baseline, level 3.0
[libx264 @ 0x34b9e753d0] 264 - core 120 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=23 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=25.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
Output #0, matroska, to '/tmp/transcode//Law.and.Order.Criminal.Intent.2001.S03E02.film.mkv':
  Metadata:
    encoder         : Lavf55.37.100
    Chapter #0.0: start 0.000000, end 615.748467
    Metadata:
      title           : Chapter 01
    Chapter #0.1: start 615.748467, end 1140.539400
    Metadata:
      title           : Chapter 02
    Chapter #0.2: start 1140.539400, end 1677.475800
    Metadata:
      title           : Chapter 03
    Chapter #0.3: start 1677.475800, end 2636.367067
    Metadata:
      title           : Chapter 04
    Stream #0:0(eng): Video: h264 (libx264) (H264 / 0x34363248), yuv420p, 720x480 [SAR 8:9 DAR 4:3], q=-1--1, 1k tbn, 23.98 tbc
    Stream #0:1(eng): Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, 192 kb/s
    Metadata:
      title           : Stereo
    Stream #0:2(eng): Subtitle: dvd_subtitle, 720x480
    Stream #0:3(spa): Subtitle: dvd_subtitle, 720x480
    Stream #0:4(fre): Subtitle: dvd_subtitle, 720x480
Stream mapping:
  Stream #0:0 -> #0:0 (mpeg2video -> libx264)
  Stream #0:1 -> #0:1 (copy)
  Stream #0:2 -> #0:2 (copy)
  Stream #0:3 -> #0:3 (copy)
  Stream #0:4 -> #0:4 (copy)
Press [q] to stop, [?] for help
frame=63210 fps=158 q=-1.0 Lsize=  375421kB time=00:43:56.38 bitrate=1166.5kbits/s    
video:306658kB audio:61790kB subtitle:5896kB other streams:0kB global headers:2kB muxing overhead: 0.287864%
[libx264 @ 0x34b9e753d0] frame I:253   Avg QP:21.47  size: 28960
[libx264 @ 0x34b9e753d0] frame P:62957 Avg QP:24.39  size:  4871
[libx264 @ 0x34b9e753d0] mb I  I16..4: 100.0%  0.0%  0.0%
[libx264 @ 0x34b9e753d0] mb P  I16..4:  5.2%  0.0%  0.0%  P16..4: 40.6%  0.0%  0.0%  0.0%  0.0%    skip:54.2%
[libx264 @ 0x34b9e753d0] coded y,uvDC,uvAC intra: 34.8% 29.2% 9.3% inter: 18.1% 9.5% 0.4%
[libx264 @ 0x34b9e753d0] i16 v,h,dc,p: 42% 27% 19% 12%
[libx264 @ 0x34b9e753d0] i8c dc,h,v,p: 49% 20% 25%  5%
[libx264 @ 0x34b9e753d0] kb/s:952.87
bench: utime=753.019s
bench: maxrss=27660kB


From mrfun.china at gmail.com  Wed Apr 16 04:25:08 2014
From: mrfun.china at gmail.com (YIRAN LI)
Date: Wed, 16 Apr 2014 12:25:08 +1000
Subject: [FFmpeg-user] any side effect by --disable-filters
Message-ID: <CAN16yyMwpjQkQaKiWRLtNMB=4DyoGF1E05e3=1UCjhFpszANfg@mail.gmail.com>

Hi,

May I know if there's any bad effect if I use --disable-filters to
configure ffmpeg. Does it disable internal filters used by ffmpeg so that
some files need filters will not be able to be opened/decoded correctly?

Thanks

From james.darnley at gmail.com  Wed Apr 16 10:33:23 2014
From: james.darnley at gmail.com (James Darnley)
Date: Wed, 16 Apr 2014 10:33:23 +0200
Subject: [FFmpeg-user] Determined to not find libvpx
In-Reply-To: <loom.20140416T014435-385@post.gmane.org>
References: <loom.20140416T014435-385@post.gmane.org>
Message-ID: <534E4053.2070808@gmail.com>

On 2014-04-16 01:57, Andoru wrote:
> Sorry if this is "noobish" problem, but I followed this compiling guide for
> Debian: http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide 
> 
> I went through it entirely, making sure I get all the paths right, but
> ffmpeg doesn't seem to want to find libvpx.
> Here's config.log:
> http://s000.tinyupload.com/index.php?file_id=00062031169542193809
> 
> I've checked the log and there is indeed 
> ~/ffmpeg_build/includes/vpx/vpx-decoder.h and all the other include files,
> so I have no idea what went wrong.

> --extra-cflags=-I/home/andoru/ffmpeg/include

This appears to be missing an 's' if the path in your email is correct.


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From cehoyos at ag.or.at  Wed Apr 16 13:00:48 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 16 Apr 2014 11:00:48 +0000 (UTC)
Subject: [FFmpeg-user] any side effect by --disable-filters
References: <CAN16yyMwpjQkQaKiWRLtNMB=4DyoGF1E05e3=1UCjhFpszANfg@mail.gmail.com>
Message-ID: <loom.20140416T130015-726@post.gmane.org>

YIRAN LI <mrfun.china <at> gmail.com> writes:

> May I know if there's any bad effect if I use 
> --disable-filters to configure ffmpeg.

You didn't tell us your usecase but you will probably 
need scale and aresample.

Carl Eugen


From cehoyos at ag.or.at  Wed Apr 16 13:04:41 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 16 Apr 2014 11:04:41 +0000 (UTC)
Subject: [FFmpeg-user] 6720x1200 proRes to wmv
References: <534D35AE.5030304@pps-inc.com>
Message-ID: <loom.20140416T130229-455@post.gmane.org>

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

> ffmpeg -i Barrier_4_Rough_5.mov -b:v 20000k -vcodec wmv2

Some encoders need -mbd 2 with low quantisers / high 
bitrates, this is not resolution-dependent afaict.
(Should be reproducible with very low resolutions 
and -qscale 2)

This is primarily a limitation of the formats (not of 
FFmpeg) although you could argue that it should choose 
-mbd 2 in such cases (I don't think they can be easily 
detected).

Carl Eugen


From cehoyos at ag.or.at  Wed Apr 16 13:02:21 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 16 Apr 2014 11:02:21 +0000 (UTC)
Subject: [FFmpeg-user] Strange -vf fps and -r behavior
References: <1397609221.55202.YahooMailBasic@web160804.mail.bf1.yahoo.com>
Message-ID: <loom.20140416T130053-866@post.gmane.org>

Nicholas Robbins <nickrobbins <at> yahoo.com> writes:

> I can make the problem go away if I use -r 24000/1001 instead, 
> but my understanding is that -vf fps is preferred to -r.

Please see tickets #1578 and #2674.
(If fps had only advantages, -r would invoke -vf fps)

If you believe the documentation is misleading please 
consider sending a documentation patch!

Carl Eugen


From cehoyos at ag.or.at  Wed Apr 16 13:05:31 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 16 Apr 2014 11:05:31 +0000 (UTC)
Subject: [FFmpeg-user] Understanding multiple aspect ratios
References: <CAHss8oH7mf-Tp_SQNPqRkz9CEftY=y86JDyu3EVGR8RHX2T=Rg@mail.gmail.com>
Message-ID: <loom.20140416T130447-6@post.gmane.org>

Massimo Battistel <battistel <at> gmail.com> writes:

> I have some files that show multiple aspect ratios

One is the codec aspect ratio, one the container 
aspect ratio (or in other words: The encoding 
application is probably buggy).

Carl Eugen


From cehoyos at ag.or.at  Wed Apr 16 13:07:17 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 16 Apr 2014 11:07:17 +0000 (UTC)
Subject: [FFmpeg-user] Force concat to ignore corrupt files?
References: <1397512164694-4664775.post@n4.nabble.com>
Message-ID: <loom.20140416T130548-676@post.gmane.org>

Phunkadilic <slanaconda <at> gmail.com> writes:

> "ffmpeg.exe -f concat -i C:\path\video-files.txt -c copy
> C:\Desitination\path\concat-file.mp4"
> 
> I've noticed that if there's a single corrupt .mp4 file, 
> the command will stop and concat only the files that 
> were good before hitting the corrupt file.

I suspect this cannot be fixed (depending on what you mean) 
but please describe a testcase.
(minimal video-files.txt, two ffmpeg -i outputs and the 
output of the failing concat command).

Carl Eugen


From lifeamit at gmail.com  Wed Apr 16 13:35:31 2014
From: lifeamit at gmail.com (Amit Agarwal)
Date: Wed, 16 Apr 2014 17:05:31 +0530
Subject: [FFmpeg-user] Buffer overflow problem in ffmpeg
In-Reply-To: <CAPMi8WbQiXo4LV2aTRPAJjTt2ufOg9Q-OGm5cTTLAQMsLBqBgw@mail.gmail.com>
References: <CAPMi8WYitukWiyS_oARDjphHuTnLmfHPDRWF2CyeKY_=oxGCZg@mail.gmail.com>
	<CAPMi8WbQiXo4LV2aTRPAJjTt2ufOg9Q-OGm5cTTLAQMsLBqBgw@mail.gmail.com>
Message-ID: <CAPMi8Wb6Z4FwbmvqVu92ZUeSsUcJh4eSCbqMxhbw66oOKJOQeA@mail.gmail.com>

I am still stuck in this bug, didn't get any help online as well.

Could some one please help me on this ?


On Thu, Apr 10, 2014 at 3:19 PM, Amit Agarwal <lifeamit at gmail.com> wrote:

> Hi,
>
> Could some one please give me some hint on how can I fix this ?
>
>
> On Wed, Apr 9, 2014 at 1:11 PM, Amit Agarwal <lifeamit at gmail.com> wrote:
>
>> I am trying to overlay a video onto another video by using
>> filter_complex. But it doesn't do overlay properly and says 'Buffer queue
>> overflow'. It works fine if I remove delay by not using -itsoffset. Please
>> let me know what can I do to overcome this problem. I am using latest
>> version of ffmpeg.
>> It is 2.2 N-62296-g4484152.
>>
>> *Here is the command I am trying. *
>> ffmpeg -i .Files/1szqpq/hjmuqD/creation/overlaid-video.mp4 -itsoffset
>> 22.952 -i
>> .Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm
>> -filter_complex "overlay=x='if(gte(t,34.28),NAN,0)':y=0" -y
>> .Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4
>>
>>
>> *And here is the output log.*
>>
>> ffmpeg version 2.2-tessus Copyright (c) 2000-2014 the FFmpeg developers
>>   built on Mar 24 2014 22:25:11 with llvm-gcc 4.2.1 (LLVM build 2336.1.00)
>>   configuration: --prefix=/Users/tessus/data/ext/ffmpeg/sw --as=yasm
>> --extra-version=tessus --disable-shared --enable-static --disable-ffplay
>> --enable-gpl --enable-pthreads --enable-postproc --enable-libmp3lame
>> --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid
>> --enable-libspeex --enable-bzlib --enable-zlib --enable-libopencore-amrnb
>> --enable-libopencore-amrwb --enable-libxavs --enable-version3
>> --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvpx
>> --enable-libgsm --enable-libopus --enable-libmodplug --enable-fontconfig
>> --enable-libfreetype --enable-libass --enable-libbluray --enable-filters
>> --enable-runtime-cpudetect
>>   libavutil      52. 66.100 / 52. 66.100
>>   libavcodec     55. 52.102 / 55. 52.102
>>   libavformat    55. 33.100 / 55. 33.100
>>   libavdevice    55. 10.100 / 55. 10.100
>>   libavfilter     4.  2.100 /  4.  2.100
>>   libswscale      2.  5.102 /  2.  5.102
>>   libswresample   0. 18.100 /  0. 18.100
>>   libpostproc    52.  3.100 / 52.  3.100
>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
>> '.Files/1szqpq/hjmuqD/creation/overlaid-video.mp4':
>>   Metadata:
>>     major_brand     : isom
>>     minor_version   : 512
>>     compatible_brands: isomiso2avc1mp41
>>     encoder         : Lavf55.33.100
>>   Duration: 00:00:44.91, start: 0.000000, bitrate: 443 kb/s
>>     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
>> 1280x720 [SAR 1:1 DAR 16:9], 392 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn,
>> 59.94 tbc (default)
>>     Metadata:
>>       handler_name    : VideoHandler
>>     Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
>> fltp, 128 kb/s (default)
>>     Metadata:
>>       handler_name    : SoundHandler
>> Input #1, matroska,webm, from
>> '.Files/1szqpq/hjmuqD/creation/07eda82db409f6c212ffa829287d9c0d_clip.webm':
>>   Metadata:
>>     encoder         : libebml v1.2.0 + libmatroska v1.1.0
>>     creation_time   : 2014-04-09 06:16:15
>>   Duration: 00:00:11.33, start: 0.000000, bitrate: 996 kb/s
>>     Stream #1:0: Video: vp8, yuv420p, 274x486, SAR 1:1 DAR 137:243, 30.05
>> fps, 30.05 tbr, 1k tbn, 1k tbc (default)
>>     Stream #1:1: Audio: vorbis, 48000 Hz, stereo, fltp (default)
>> [libx264 @ 0x104000600] using SAR=1/1
>> [libx264 @ 0x104000600] using cpu capabilities: MMX2 SSE2Fast SSSE3
>> SSE4.2 AVX
>> [libx264 @ 0x104000600] profile High, level 3.1
>> [libx264 @ 0x104000600] 264 - core 142 - H.264/MPEG-4 AVC codec -
>> Copyleft 2003-2014 - 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=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=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 '.Files/1szqpq/hjmuqD/creation/overlaid-video-2.mp4':
>>   Metadata:
>>     major_brand     : isom
>>     minor_version   : 512
>>     compatible_brands: isomiso2avc1mp41
>>     encoder         : Lavf55.33.100
>>     Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
>> 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 11988 tbn, 29.97 tbc (default)
>>     Stream #0:1(und): Audio: aac (libvo_aacenc) ([64][0][0][0] / 0x0040),
>> 44100 Hz, stereo, s16, 128 kb/s (default)
>>     Metadata:
>>       handler_name    : SoundHandler
>> Stream mapping:
>>   Stream #0:0 (h264) -> overlay:main (graph 0)
>>   Stream #1:0 (vp8) -> overlay:overlay (graph 0)
>>   overlay (graph 0) -> Stream #0:0 (libx264)
>>   Stream #0:1 -> #0:1 (aac -> libvo_aacenc)
>> Press [q] to stop, [?] for help
>> [Parsed_overlay_0 @ 0x103800000] [framesync @ 0x1038000e8] Buffer queue
>> overflow, dropping.
>>     Last message repeated 624 times
>> frame= 1345 fps=181 q=-1.0 Lsize=    2130kB time=00:00:44.81 bitrate=
>> 389.4kbits/s dup=624 drop=0
>> video:1851kB audio:251kB subtitle:0 data:0 global headers:0kB muxing
>> overhead 1.348262%
>> [libx264 @ 0x104000600] frame I:8     Avg QP:13.93  size: 30628
>> [libx264 @ 0x104000600] frame P:402   Avg QP:17.93  size:  2802
>> [libx264 @ 0x104000600] frame B:935   Avg QP:20.63  size:   560
>> [libx264 @ 0x104000600] consecutive B-frames:  2.4% 14.4%  1.1% 82.1%
>> [libx264 @ 0x104000600] mb I  I16..4:  9.0% 73.4% 17.6%
>> [libx264 @ 0x104000600] mb P  I16..4:  1.7%  1.7%  0.3%  P16..4: 10.9%
>>  2.3%  0.9%  0.0%  0.0%    skip:82.2%
>> [libx264 @ 0x104000600] mb B  I16..4:  0.1%  0.1%  0.0%  B16..8:  7.6%
>>  0.4%  0.0%  direct: 0.1%  skip:91.7%  L0:46.5% L1:50.8% BI: 2.8%
>> [libx264 @ 0x104000600] 8x8 transform intra:53.1% inter:78.0%
>> [libx264 @ 0x104000600] coded y,uvDC,uvAC intra: 31.4% 36.8% 7.9% inter:
>> 1.1% 2.3% 0.0%
>> [libx264 @ 0x104000600] i16 v,h,dc,p: 31% 27%  7% 35%
>> [libx264 @ 0x104000600] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 41% 20% 23%  3%
>>  3%  3%  3%  3%  3%
>> [libx264 @ 0x104000600] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 39% 25% 14%  3%
>>  4%  4%  4%  3%  3%
>> [libx264 @ 0x104000600] i8c dc,h,v,p: 74% 12% 11%  2%
>> [libx264 @ 0x104000600] Weighted P-Frames: Y:0.0% UV:0.0%
>> [libx264 @ 0x104000600] ref P L0: 68.4%  8.7% 17.1%  5.8%
>> [libx264 @ 0x104000600] ref B L0: 85.5% 13.7%  0.8%
>> [libx264 @ 0x104000600] ref B L1: 97.5%  2.5%
>> [libx264 @ 0x104000600] kb/s:337.78
>>
>
>

From jshupert at pps-inc.com  Wed Apr 16 16:32:02 2014
From: jshupert at pps-inc.com (Jim Shupert)
Date: Wed, 16 Apr 2014 10:32:02 -0400
Subject: [FFmpeg-user] 6720x1200 proRes to wmv
In-Reply-To: <loom.20140416T130229-455@post.gmane.org>
References: <534D35AE.5030304@pps-inc.com>
	<loom.20140416T130229-455@post.gmane.org>
Message-ID: <534E9462.5040804@pps-inc.com>


> Some encoders need -mbd 2 with low quantisers / high
> bitrates, this is not resolution-dependent afaict.
> (Should be reproducible with very low resolutions
> and -qscale 2)
>
> This is primarily a limitation of the formats (not of
> FFmpeg) although you could argue that it should choose
> -mbd 2 in such cases (I don't think they can be easily
> detected).
>
> Carl Eugen
>
thanks brother,
yes I was able to make the warnings [ ...warning, clipping 1 dct 
coefficients ... ]
  go away with a   "  -qmin 3  "

I wonder what is the diff of

-qscale 2  & -qmin 2


This matter has  brought me to a greater understanding of such matters
thanks for the -mbd
( -mbd <int> E.V.. macroblock decision algorithm (high quality mode)
... though I will confess I am a bit fuzzy on what that -- means 
...exactly.. )

and on format / player support for Very large frm size...

It is an interesting matter -- total supported frame size in different 
players.
QT (win&mac) seems very forgiving - a high cieling
VLC , mplayer , win media player all tend to hollar  " 6720 x 1200 is to 
big " & fail .

one wonders how this will change with the coming of 4k etc.

thanks again



From george at nsup.org  Wed Apr 16 16:38:24 2014
From: george at nsup.org (Nicolas George)
Date: Wed, 16 Apr 2014 16:38:24 +0200
Subject: [FFmpeg-user] Force concat to ignore corrupt files?
In-Reply-To: <1397512164694-4664775.post@n4.nabble.com>
References: <1397512164694-4664775.post@n4.nabble.com>
Message-ID: <20140416143824.GA19727@phare.normalesup.org>

Le quintidi 25 germinal, an CCXXII, Phunkadilic a ?crit?:
> I'm currently using this command to concat individual video files every
> night:
> 
> "ffmpeg.exe -f concat -i C:\path\video-files.txt -c copy
> C:\Desitination\path\concat-file.mp4"
> 
> I've noticed that if there's a single corrupt .mp4 file, the command will
> stop and concat only the files that were good before hitting the corrupt
> file.  Is there a way/command to have ffmpeg to either skip any corrupt
> files or to force through and ignore errors?  
> 
> This script runs automatically nightly and I'd rather not have to babysit
> these files.  If there's one or two corrupt files, I'd rather them just be
> skipped and not worry about having to go through and remove the corrupt
> files and re-do the process manually.

I may be wrong, but it seems to me like a very specific use case.

I suggest you test if the file is corrupt while constructing your concat
script. Something like that would probably do the trick (with extra shell
escaping of course):

    for f in $files; do
      if ffprobe -loglevel warning $f; then
	printf 'file %s\n' $f >> $concat
      else
	corrupted=$((corrupted+1))
      fi
    done

That way, you can do smart decisions depending on the amount of corrupted
files, for example. Depending on the degree of corruption you want to guard
against, you may want to read the file completely (-show_packets), or maybe
even decode it (-show_frames). Note that since the concat demuxer works at
demuxer level, it could not guard against problems at the decoding level
anyway.

Regards,

-- 
  Nicolas George
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From cehoyos at ag.or.at  Wed Apr 16 16:39:32 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 16 Apr 2014 14:39:32 +0000 (UTC)
Subject: [FFmpeg-user] 6720x1200 proRes to wmv
References: <534D35AE.5030304@pps-inc.com>
	<loom.20140416T130229-455@post.gmane.org>
	<534E9462.5040804@pps-inc.com>
Message-ID: <loom.20140416T163737-341@post.gmane.org>

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

> It is an interesting matter -- total supported frame 
> size in different players.
> QT (win&mac) seems very forgiving - a high cieling
> VLC , mplayer , win media player all tend to hollar 
> " 6720 x 1200 is to big " & fail .

I tested the following:
$ ffmpeg -f lavfi -i testsrc=6720x1200 -qscale 2 out.avi

MPlayer (and vlc) has no problems with the output file.

Carl Eugen


From george at nsup.org  Wed Apr 16 17:14:05 2014
From: george at nsup.org (Nicolas George)
Date: Wed, 16 Apr 2014 17:14:05 +0200
Subject: [FFmpeg-user] Could segments from one mpg have different audio
	offsets?
In-Reply-To: <20140406181123.63705de5@zooty>
References: <20140406164316.183c64e5@zooty>
	<loom.20140406T232729-367@post.gmane.org>
	<20140406181123.63705de5@zooty>
Message-ID: <20140416151405.GB26227@phare.normalesup.org>

Le septidi 17 germinal, an CCXXII, Tom Horsley a ?crit?:
> What does the concat filter do if the files being
> concatenated have different offsets?

I do not understand the question. The concat filter does not work on files
but on groups of streams; these groups are called segments in the
documentation. They probably come from files, but not necessarily the same.
I do not know what you call offset.

The documentation says:

# For this filter to work correctly, all segments must start at timestamp 0.

And also:

# The concat filter will use the duration of the longest stream in each
# segment (except the last one), and if necessary pad shorter audio streams
# with silence.

(For video streams, it will only adjust the timestamps to create a gap
before the next segment to leave time for the other streams to finish; the
fps filter can later duplicate the last frame if necessary.)

Regards,

-- 
  Nicolas George
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From nickrobbins at yahoo.com  Wed Apr 16 17:38:11 2014
From: nickrobbins at yahoo.com (Nicholas Robbins)
Date: Wed, 16 Apr 2014 08:38:11 -0700 (PDT)
Subject: [FFmpeg-user] Strange -vf fps and -r behavior
In-Reply-To: <loom.20140416T130053-866@post.gmane.org>
References: <1397609221.55202.YahooMailBasic@web160804.mail.bf1.yahoo.com>
	<loom.20140416T130053-866@post.gmane.org>
Message-ID: <1397662691.30137.YahooMailNeo@web160805.mail.bf1.yahoo.com>

> On Wednesday, April 16, 2014 7:10 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> > Nicholas Robbins <nickrobbins <at> yahoo.com> writes:
> 
>>  I can make the problem go away if I use -r 24000/1001 instead, 
>>  but my understanding is that -vf fps is preferred to -r.
> 
> Please see tickets #1578 and #2674.
> (If fps had only advantages, -r would invoke -vf fps)

I'm not sure this is quite the same as those tickets, but they did bring my attention to the 'round' option in the fps filter. Adding round=zero seems to have solved it, I think.

> If you believe the documentation is misleading please 
> consider sending a documentation patch!

It wasn't the documentation that gave me that impression. I suppose it was just people recommending -vf fps to solve various problems.

> 
> Carl Eugen

From slanaconda at gmail.com  Wed Apr 16 17:25:15 2014
From: slanaconda at gmail.com (Phunkadilic)
Date: Wed, 16 Apr 2014 08:25:15 -0700 (PDT)
Subject: [FFmpeg-user] Force concat to ignore corrupt files?
In-Reply-To: <20140416143824.GA19727@phare.normalesup.org>
References: <1397512164694-4664775.post@n4.nabble.com>
	<20140416143824.GA19727@phare.normalesup.org>
Message-ID: <A7E50DA7-8729-48AB-8AFE-47C6BD5D6B7D@gmail.com>

Thanks for these great tips.  Basically it's like this.  I'm using the concat command for daily merging of video capture files (.mp4).  Every now and then I'll get one or two video files that will have some time of error or will be incomplete.  The concat command see's these files and simply stops the concat process instead of skipping them and continuing on to the next good file in the list.  This is what I would to fix.  I want the concat function to continue on instead of getting stuck and exiting the first time it hits a bad file.

I'll have to try the ffprobe command to see if that does the trick. 

-Dilan
_________________________________
"NO OBSTACLES, ONLY CHALLENGES!" -Groove

On Apr 16, 2014, at 8:39 AM, Nicolas George [via FFmpeg-users] <ml-node+s933282n4664793h96 at n4.nabble.com> wrote:

> Le quintidi 25 germinal, an CCXXII, Phunkadilic a ?crit :
> 
> > I'm currently using this command to concat individual video files every 
> > night: 
> > 
> > "ffmpeg.exe -f concat -i C:\path\video-files.txt -c copy 
> > C:\Desitination\path\concat-file.mp4" 
> > 
> > I've noticed that if there's a single corrupt .mp4 file, the command will 
> > stop and concat only the files that were good before hitting the corrupt 
> > file.  Is there a way/command to have ffmpeg to either skip any corrupt 
> > files or to force through and ignore errors?   
> > 
> > This script runs automatically nightly and I'd rather not have to babysit 
> > these files.  If there's one or two corrupt files, I'd rather them just be 
> > skipped and not worry about having to go through and remove the corrupt 
> > files and re-do the process manually.
> I may be wrong, but it seems to me like a very specific use case. 
> 
> I suggest you test if the file is corrupt while constructing your concat 
> script. Something like that would probably do the trick (with extra shell 
> escaping of course): 
> 
>     for f in $files; do 
>       if ffprobe -loglevel warning $f; then 
>         printf 'file %s\n' $f >> $concat 
>       else 
>         corrupted=$((corrupted+1)) 
>       fi 
>     done 
> 
> That way, you can do smart decisions depending on the amount of corrupted 
> files, for example. Depending on the degree of corruption you want to guard 
> against, you may want to read the file completely (-show_packets), or maybe 
> even decode it (-show_frames). Note that since the concat demuxer works at 
> demuxer level, it could not guard against problems at the decoding level 
> anyway. 
> 
> Regards, 
> 
> -- 
>   Nicolas George 
> 
> _______________________________________________ 
> ffmpeg-user mailing list 
> [hidden email] 
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> 
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> 
> 
> If you reply to this email, your message will be added to the discussion below:
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From jshupert at pps-inc.com  Wed Apr 16 20:42:41 2014
From: jshupert at pps-inc.com (Jim Shupert)
Date: Wed, 16 Apr 2014 14:42:41 -0400
Subject: [FFmpeg-user] Fwd: Re:  6720x1200 proRes to wmv
In-Reply-To: <534EB397.5040305@pps-inc.com>
References: <534EB397.5040305@pps-inc.com>
Message-ID: <534ECF21.4020905@pps-inc.com>




-------- Original Message --------
Subject: 	Re: [FFmpeg-user] 6720x1200 proRes to wmv
Date: 	Wed, 16 Apr 2014 12:45:11 -0400
From: 	Jim Shupert <jshupert at pps-inc.com>
To: 	FFmpeg user questions <ffmpeg-user at ffmpeg.org>



On 4/16/2014 10:39 AM, Carl Eugen Hoyos wrote:
> Jim Shupert <jshupert <at> pps-inc.com> writes:
>
>> It is an interesting matter -- total supported frame
>> size in different players.
>> QT (win&mac) seems very forgiving - a high cieling
>> VLC , mplayer , win media player all tend to hollar
>> " 6720 x 1200 is to big " & fail .
> I tested the following:
> $ ffmpeg -f lavfi -i testsrc=6720x1200 -qscale 2 out.avi
>
> MPlayer (and vlc) has no problems with the output file.
>
>
Thanks Carl,

I tried that here

ffmpeg -f lavfi -i testsrc=6720x1200 -qscale 2 /media/data/p/jimiTst1.avi
ffmpeg version git-2013-06-03-b4d4ef5 Copyright (c) 2000-2013 the FFmpeg
developers
    built on Jun  3 2013 16:50:07 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
    configuration: --extra-libs=-ldl --enable-gpl --enable-libass
--enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libspeex --enable-libtheora
--enable-libvorbis --enable-libopus --enable-libvpx --enable-x11grab
--enable-libx264 --enable-nonfree --enable-version3
    libavutil      52. 34.100 / 52. 34.100
    libavcodec     55. 15.100 / 55. 15.100
    libavformat    55.  8.102 / 55.  8.102
    libavdevice    55.  2.100 / 55.  2.100
    libavfilter     3. 74.101 /  3. 74.101
    libswscale      2.  3.100 /  2.  3.100
    libswresample   0. 17.102 /  0. 17.102
    libpostproc    52.  3.100 / 52.  3.100
Input #0, lavfi, from 'testsrc=6720x1200':
    Duration: N/A, start: 0.000000, bitrate: N/A
      Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 6720x1200
[SAR 1:1 DAR 28:5], 25 tbr, 25 tbn, 25 tbc
Please use -q:a or -q:v, -qscale is ambiguous
Output #0, avi, to '/media/data/p/jimiTst1.avi':
    Metadata:
      ISFT            : Lavf55.8.102
      Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 6720x1200 [SAR
1:1 DAR 28:5], q=2-31, 200 kb/s, 25 tbn, 25 tbc
Stream mapping:
    Stream #0:0 -> #0:0 (rawvideo -> mpeg4)
Press [q] to stop, [?] for help
frame=  158 fps=7.0 q=2.0 Lsize=    3739kB time=00:00:06.32
bitrate=4846.8kbits/s
video:3730kB audio:0kB subtitle:0 global headers:0kB muxing overhead
0.252163%
Received signal 2: terminating.

---
but for me that yield a file that plays badly in VLC [ 2.0.8 Twoflower ]
a diag pops up of
Your video output acceleration driver does not support the required
resolution: 6720x1200 pixels. The maximum supported resolution is
2046x1200.
Video output acceleration will be disabled. However, rendering videos with
overly large resolution may cause severe performance degration.

and my mplayer -- just failes to show anything

MPlayer svn r34540 (Ubuntu), built with gcc-4.6 (C) 2000-2012 MPlayer Team


i do a 'simpler' test

   ffmpeg -f lavfi -i testsrc  -qscale 2 /media/data/p/jimiTst2.avi

ffmpeg version git-2013-06-03-b4d4ef5 Copyright (c) 2000-2013 the FFmpeg
developers
    built on Jun  3 2013 16:50:07 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
    configuration: --extra-libs=-ldl --enable-gpl --enable-libass
--enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libspeex --enable-libtheora
--enable-libvorbis --enable-libopus --enable-libvpx --enable-x11grab
--enable-libx264 --enable-nonfree --enable-version3
    libavutil      52. 34.100 / 52. 34.100
    libavcodec     55. 15.100 / 55. 15.100
    libavformat    55.  8.102 / 55.  8.102
    libavdevice    55.  2.100 / 55.  2.100
    libavfilter     3. 74.101 /  3. 74.101
    libswscale      2.  3.100 /  2.  3.100
    libswresample   0. 17.102 /  0. 17.102
    libpostproc    52.  3.100 / 52.  3.100
Input #0, lavfi, from 'testsrc':
    Duration: N/A, start: 0.000000, bitrate: N/A
      Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 320x240
[SAR 1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc

there is a warning
Please use -q:a or -q:v, -qscale is ambiguous

Output #0, avi, to '/media/data/p/jimiTst2.avi':
    Metadata:
      ISFT            : Lavf55.8.102
      Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240 [SAR
1:1 DAR 4:3], q=2-31, 200 kb/s, 25 tbn, 25 tbc
Stream mapping:
    Stream #0:0 -> #0:0 (rawvideo -> mpeg4)
Press [q] to stop, [?] for help
frame= 3856 fps=586 q=2.0 Lsize=    6578kB time=00:02:34.24 bitrate=
349.3kbits/s
video:6480kB audio:0kB subtitle:0 global headers:0kB muxing overhead
1.509310%
Received signal 2: terminating.

------

and that  , of course is just fine

so why do you- have no problems?





From cehoyos at ag.or.at  Wed Apr 16 21:21:30 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 16 Apr 2014 19:21:30 +0000 (UTC)
Subject: [FFmpeg-user] Fwd: Re:  6720x1200 proRes to wmv
References: <534EB397.5040305@pps-inc.com> <534ECF21.4020905@pps-inc.com>
Message-ID: <loom.20140416T211810-512@post.gmane.org>

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

> but for me that yield a file that plays badly in VLC 
> [ 2.0.8 Twoflower ] a diag pops up of
> Your video output acceleration driver does not support 
> the required resolution: 6720x1200 pixels. The maximum 
> supported resolution is 2046x1200.
> Video output acceleration will be disabled.

This doesn't look like a problem with vlc to me.
(But your vlc version is unsupported, I tested 2.1.x)

> MPlayer svn r34540 (Ubuntu), built with gcc-4.6 (C) 
> 2000-2012 MPlayer Team

> so why do you- have no problems?

You are not showing the complete MPlayer console 
output but I suspect you are not testing MPlayer 
(at least not a supported version).
Otoh, MPlayer may have the same driver problem as 
vlc, but again, this isn't something that can be 
fixed in MPlayer.
It works with my 35? GPU, so I don't think 
extraordinary hardware is required.
(GT 520)

Carl Eugen


From andoru.blah at gmail.com  Wed Apr 16 21:57:26 2014
From: andoru.blah at gmail.com (Andoru)
Date: Wed, 16 Apr 2014 19:57:26 +0000 (UTC)
Subject: [FFmpeg-user] Determined to not find libvpx
References: <loom.20140416T014435-385@post.gmane.org>
	<534E4053.2070808@gmail.com>
Message-ID: <loom.20140416T215558-80@post.gmane.org>


> > I've checked the log and there is indeed 
> > ~/ffmpeg_build/includes/vpx/vpx-decoder.h and all the other include files,
> > so I have no idea what went wrong.
> 
> > --extra-cflags=-I/home/andoru/ffmpeg/include
> 
> This appears to be missing an 's' if the path in your email is correct.

My bad, I meant ~/ffmpeg_build/include/vpx/vpx-decoder.h
I double-checked, the path is correct.



From james.darnley at gmail.com  Wed Apr 16 23:10:09 2014
From: james.darnley at gmail.com (James Darnley)
Date: Wed, 16 Apr 2014 23:10:09 +0200
Subject: [FFmpeg-user] Determined to not find libvpx
In-Reply-To: <loom.20140416T215558-80@post.gmane.org>
References: <loom.20140416T014435-385@post.gmane.org>	<534E4053.2070808@gmail.com>
	<loom.20140416T215558-80@post.gmane.org>
Message-ID: <534EF1B1.5030505@gmail.com>

On 2014-04-16 21:57, Andoru wrote:
> 
>>> I've checked the log and there is indeed 
>>> ~/ffmpeg_build/includes/vpx/vpx-decoder.h and all the other include files,
>>> so I have no idea what went wrong.
>>
>>> --extra-cflags=-I/home/andoru/ffmpeg/include
>>
>> This appears to be missing an 's' if the path in your email is correct.
> 
> My bad, I meant ~/ffmpeg_build/include/vpx/vpx-decoder.h
> I double-checked, the path is correct.

Oh wait...  I'm an idiot.  Never mind the small 's', what about the
'_build' part?


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From andoru.blah at gmail.com  Thu Apr 17 04:28:09 2014
From: andoru.blah at gmail.com (Andoru)
Date: Thu, 17 Apr 2014 02:28:09 +0000 (UTC)
Subject: [FFmpeg-user] Determined to not find libvpx
References: <loom.20140416T014435-385@post.gmane.org>	<534E4053.2070808@gmail.com>
	<loom.20140416T215558-80@post.gmane.org>
	<534EF1B1.5030505@gmail.com>
Message-ID: <loom.20140417T041525-625@post.gmane.org>


> >>> I've checked the log and there is indeed 
> >>> ~/ffmpeg_build/includes/vpx/vpx-decoder.h and all the other include files,
> >>> so I have no idea what went wrong.
> >>
> >>> --extra-cflags=-I/home/andoru/ffmpeg/include
> >>
> >> This appears to be missing an 's' if the path in your email is correct.
> > 
> > My bad, I meant ~/ffmpeg_build/include/vpx/vpx-decoder.h
> > I double-checked, the path is correct.
> 
> Oh wait...  I'm an idiot.  Never mind the small 's', what about the
> '_build' part?


Nope, I'm the idiot, haha. I had had the configure line pointing out to the
wrong path as you noticed, but now it complains that it can't find opus,
here's what it says in the log:
____________________________________________________________________
check_pkg_config opus opus_multistream.h opus_multistream_decoder_create
pkg-config --exists --print-errors opus
Package opus was not found in the pkg-config search path.
Perhaps you should add the directory containing `opus.pc'
to the PKG_CONFIG_PATH environment variable
No package 'opus' found
ERROR: opus not found
_____________________________________________________________________

Indeed, there's no such file anywhere under ~/ffmpeg_build. I tried to make
Opus again without the --disable-shared parameter and it didn't work.


From ddhung at vasc.com.vn  Thu Apr 17 05:23:49 2014
From: ddhung at vasc.com.vn (Dang Duc Hung)
Date: Thu, 17 Apr 2014 10:23:49 +0700
Subject: [FFmpeg-user] Automatic sync video and audio on Mpeg2ts transport
	via UDP IP
Message-ID: <008101cf59ec$763ee120$62bca360$@vasc.com.vn>

I?m trying to find solution to automatic Automatic sync video and audio on Mpeg2ts transport via UDP multicast IP beacause now some channels video IP have this problem. When this problem occur, I often manually re-run the proccess ffmpeg on Linux server.

 

Thank you so much.

 

 

Best Regards

??ng ??c H?ng

 


From toronto.ffmpeg at gmail.com  Thu Apr 17 17:53:16 2014
From: toronto.ffmpeg at gmail.com (Dennis P)
Date: Thu, 17 Apr 2014 11:53:16 -0400
Subject: [FFmpeg-user] Job offer in Toronto.
Message-ID: <CAHvfWLcKeGz5vNV_rHhc3ywoWBJPX+7gL8SaEXGvftpf8SqbCw@mail.gmail.com>

A mature company is looking for a video engineer with FFmpeg / Linux
knowledge.

Needs to understand video processing requirements and build workflow to
accommodate business needs:
- transcoding multiple file formats
- transmux all common file formats
- understanding of all common video and audio compression formats and their
implementation
- thorough understanding of MPEG2-TS format
- knowledge of FFmpeg filters and filter chains
- knowledge of ffmpeg/mp4box/tsmuxer tools
- some knowledge of live streaming technologies is preferred
- College/University education is an asset

Please forward you resumes to this email address:
toronto.ffmpeg  _at_  gmail



Thank you
Dennis

From xanadu at apost.plala.or.jp  Thu Apr 17 18:26:01 2014
From: xanadu at apost.plala.or.jp (=?iso-2022-jp?B?GyRCNVxCPBsoQiAbJEI4eENLGyhC?=)
Date: Fri, 18 Apr 2014 01:26:01 +0900
Subject: [FFmpeg-user] Enabling libx265
Message-ID: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>

Hello, I'm trying to build ffmpeg enabling libx265 on Mac (10.8.5 and 10.9.2)

I build x265 as following:
$ hg clone https://bitbucket.org/multicoreware/x265
$ cd x265/build/linux/
$ ./make-Makefiles.bash

at the time ccmake execute, I select "/Volumes/ffmpeg_compile" as CMAKE_INSTALL_PREFIX.

$ make && make install

As a result, I get following:
-- Install configuration: "Release"
-- Installing: /Volumes/ffmpeg_compile/lib/libx265.a
-- Installing: /Volumes/ffmpeg_compile/include/x265.h
-- Installing: /Volumes/ffmpeg_compile/include/x265_config.h
-- Installing: /Volumes/ffmpeg_compile/lib/libx265.0.9.dylib
-- Installing: /Volumes/ffmpeg_compile/lib/libx265.16.dylib
-- Installing: /Volumes/ffmpeg_compile/lib/libx265.dylib
-- Installing: /Volumes/ffmpeg_compile/lib/pkgconfig/x265.pc
-- Installing: /Volumes/ffmpeg_compile/bin/x265

Here's contents of x265.pc

prefix=/Volumes/ffmpeg_compile
exec_prefix=${prefix}
libdir=${exec_prefix}/lib
includedir=${prefix}/include

Name: x265
Description: H.265/HEVC video encoder
Version: 0.9
Libs: -L${libdir} -lx265
Libs.private: -lstdc++ -l/usr/lib/clang/4.2/lib/darwin/libclang_rt.osx.a
Cflags: -I${includedir}


Now I remove *.dylib files to build ffmpeg as static binary.

$ git clone git://source.ffmpeg.org/ffmpeg.git
$ cd ffmpeg
$ ./configure --as=yasm --enable-gpl --enable-libx265

then ffmpeg configure script exits with "ERROR: x265 not found" error.

then I put back  *.dylib files to /Volumes/ffmpeg_compile/lib/ directory.

Retry

$ cd ffmpeg
$ ./configure --as=yasm --enable-gpl --enable-libx265

this time ffmpeg configure script passes successfully.  So:

$ make

ffmpeg seems to build successfully, but:

$ ./ffmpeg -encoders
dyld: Library not loaded: libx265.16.dylib
  Referenced from: /Volumes/ffmpeg_compile/compile/ffmpeg/./ffmpeg
  Reason: image not found
Trace/BPT trap: 5

$ otool -L ./ffmpeg
./ffmpeg:
	/System/Library/Frameworks/CoreFoundation.framework/Versions/A/CoreFoundation (compatibility version 150.0.0, current version 744.19.0)
	/System/Library/Frameworks/VideoDecodeAcceleration.framework/Versions/A/VideoDecodeAcceleration (compatibility version 1.0.0, current version 1.0.0)
	/System/Library/Frameworks/QuartzCore.framework/Versions/A/QuartzCore (compatibility version 1.2.0, current version 1.8.0)
	libx265.16.dylib (compatibility version 16.0.0, current version 0.9.0)
	/usr/lib/libSystem.B.dylib (compatibility version 1.0.0, current version 169.3.0)
	/System/Library/Frameworks/CoreVideo.framework/Versions/A/CoreVideo (compatibility version 1.2.0, current version 1.8.0)

This ffmpeg seems deepens on libx265.16dylib not /Volumes/ffmpeg_compile/lib/libx265.16dylib.  So I put x265.*.dylib into /usr/local/lib, then ffmpeg can execute.

$ ./ffmpeg -encoders | grep 265
ffmpeg version N-62495-g197fe39 Copyright (c) 2000-2014 the FFmpeg developers
  built on Apr 18 2014 00:37:18 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --as=yasm --enable-gpl --enable-libx265
  libavutil      52. 76.100 / 52. 76.100
  libavcodec     55. 58.103 / 55. 58.103
  libavformat    55. 37.100 / 55. 37.100
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
 V..... libx265              libx265 H.265 / HEVC (codec hevc)



As a result, I have 2 questions:
1.Why ffmpeg configure script exits with error when I try to build ffmpeg with x265.a only? (without x265.*.dylib)
2.With  x265.*.dylib, it seems ffmpeg can build, but funny dependency occur.  Is there any wrong in building x265?



From devcorwin at gmail.com  Thu Apr 17 17:28:14 2014
From: devcorwin at gmail.com (corwin)
Date: Thu, 17 Apr 2014 18:28:14 +0300
Subject: [FFmpeg-user] Streaming two sources merged to one RTMP stream
Message-ID: <581F3108-CEA1-43C1-A87B-983657B9D831@gmail.com>

I have two video sources (one is webcam and another is video captured from the screen) and I want to stream it into one RTMP stream using ffmpeg. When I?m trying to run command:

ffmpeg -y -f vfwcap -i 0 -f dshow -i video="screen-capture-recorder" -r 25 -s 320x240 -vcodec libx264  -f flv "rtmp://example.com?  

Of course ?example.com? should be another source. So when I?m running this command and then looking this stream using ffplay on the end side I can see only one stream.

Can you help me and tell what am I doing wrong?

From Cecil at decebal.nl  Thu Apr 17 19:42:10 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Thu, 17 Apr 2014 19:42:10 +0200
Subject: [FFmpeg-user] Distorted audio in screencast
Message-ID: <877g6ndf9p.fsf@Equus.Decebal.nl>

I have a problem with a zooming sound:
    https://www.facebook.com/photo.php?v=619250241480694

I did not have this in the past as far as I know. But this can also be
because in the past I only used the microphone of the web-cam (and on
another computer). Which is a Logitech, Inc. Webcam Pro 9000.

The command I use is:
ffmpeg -async 1                         \
       -f alsa                          \
       -i "plughw:1"                    \
       -f x11grab -s "800x600"          \
       -r 25                            \
       -i :0.0                          \
       -acodec libmp3lame               \
       -ac 1                            \
       -ar 22050                        \
       -ab 64k                          \
       -qscale:video 4                  \
       testing.avi

Also sometimes the audio is a little out of sync and there are sometimes
sharp sounds. What should I change?

The version of ffmpeg I use is:
    ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers

Until recently I used a camera to make my video's, but I changed to a
screencast for two reasons:
- My camera does not have an external microphone connection
- I need pictures in my video and do not know how to replace video with
  a picture

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From rogerdpack2 at gmail.com  Thu Apr 17 21:29:15 2014
From: rogerdpack2 at gmail.com (Roger Pack)
Date: Thu, 17 Apr 2014 13:29:15 -0600
Subject: [FFmpeg-user] Streaming two sources merged to one RTMP stream
In-Reply-To: <581F3108-CEA1-43C1-A87B-983657B9D831@gmail.com>
References: <581F3108-CEA1-43C1-A87B-983657B9D831@gmail.com>
Message-ID: <CAL1QdWf68g-bK6aekHmio7AS+RyicsQJ39Vzox2V6YAhztB+zQ@mail.gmail.com>

On 4/17/14, corwin <devcorwin at gmail.com> wrote:
> I have two video sources (one is webcam and another is video captured from
> the screen) and I want to stream it into one RTMP stream using ffmpeg. When
> I?m trying to run command:
>
> ffmpeg -y -f vfwcap -i 0 -f dshow -i video="screen-capture-recorder" -r 25
> -s 320x240 -vcodec libx264  -f flv "rtmp://example.com?
>
> Of course ?example.com? should be another source. So when I?m running this
> command and then looking this stream using ffplay on the end side I can see
> only one stream.

How do you want the 2 streams to appear at the receivign end? side by
side? as different mpeg streams?

From olivier.pis.langlois at transport.alstom.com  Thu Apr 17 22:04:40 2014
From: olivier.pis.langlois at transport.alstom.com (LANGLOIS Olivier PIS -EXT)
Date: Thu, 17 Apr 2014 20:04:40 +0000
Subject: [FFmpeg-user] Distorted audio in screencast
In-Reply-To: <877g6ndf9p.fsf@Equus.Decebal.nl>
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
Message-ID: <89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>


> -----Original Message-----
> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
> bounces at ffmpeg.org] On Behalf Of Cecil Westerhof
> Sent: Thursday, April 17, 2014 1:42 PM
> To: ffmpeg-user at ffmpeg.org
> Subject: [FFmpeg-user] Distorted audio in screencast
>
> I have a problem with a zooming sound:
>     https://www.facebook.com/photo.php?v=619250241480694
>
> I did not have this in the past as far as I know. But this can also be because in
> the past I only used the microphone of the web-cam (and on another
> computer). Which is a Logitech, Inc. Webcam Pro 9000.
>
> The command I use is:
> ffmpeg -async 1                         \
>        -f alsa                          \
>        -i "plughw:1"                    \
>        -f x11grab -s "800x600"          \
>        -r 25                            \
>        -i :0.0                          \
>        -acodec libmp3lame               \
>        -ac 1                            \
>        -ar 22050                        \
>        -ab 64k                          \
>        -qscale:video 4                  \
>        testing.avi
>
> Also sometimes the audio is a little out of sync and there are sometimes
> sharp sounds. What should I change?
>
ffmpeg inputs are opened sequentially. I have found that opening slow input devices first could help with AV sync. OTOH, if I understand correctly your setup, you display a webcam output on screen and grab that with x11grab. This for sure add some latency in the video signal. There must be a way to manually add a ts offset on the cmdline to compensate for that.

I think that your sharp sounds are caused by saturation. If you had ALSA buffer overruns, ffmpeg would report it and it sounds more like clipping. There are 2 things that I can think that you can play with fix this problem:

1. Check your mic preamp level with the alsamixer app.
2. Use "hw:1" instead of "plughw:1" to let ffmpeg do the signal adaptation.

What plughw is doing is to ask ALSA to adapt device hw params to what the app is asking (sampling rate, bits per sample, etc). You could check if ffmpeg is handling this better.


________________________________
CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium.

From cehoyos at ag.or.at  Fri Apr 18 05:16:23 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 18 Apr 2014 03:16:23 +0000 (UTC)
Subject: [FFmpeg-user] Enabling libx265
References: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>
Message-ID: <loom.20140418T051500-274@post.gmane.org>

?? ?? <xanadu <at> apost.plala.or.jp> writes:

> -- Installing: /Volumes/ffmpeg_compile/lib/libx265.a
> -- Installing: /Volumes/ffmpeg_compile/include/x265.h

> $ ./configure --as=yasm --enable-gpl --enable-libx265

This is missing --extra-cflags=-I/Volumes/ffmpeg_compile/include 
--extra-ldflags=-L/Volumes/ffmpeg_compile/lib

Carl Eugen


From Cecil at decebal.nl  Fri Apr 18 08:28:09 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Fri, 18 Apr 2014 08:28:09 +0200
Subject: [FFmpeg-user] Distorted audio in screencast
In-Reply-To: <89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	(LANGLOIS Olivier's message of "Thu, 17 Apr 2014 20:04:40 +0000")
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
Message-ID: <87d2gfb18m.fsf@Equus.Decebal.nl>

Op Thursday 17 Apr 2014 22:04 CEST schreef LANGLOIS Olivier:

>
>> -----Original Message-----
>> From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-
>> bounces at ffmpeg.org] On Behalf Of Cecil Westerhof
>> Sent: Thursday, April 17, 2014 1:42 PM
>> To: ffmpeg-user at ffmpeg.org
>> Subject: [FFmpeg-user] Distorted audio in screencast
>>
>> I have a problem with a zooming sound:
>> https://www.facebook.com/photo.php?v=619250241480694
>>
>> I did not have this in the past as far as I know. But this can also
>> be because in the past I only used the microphone of the web-cam
>> (and on another computer). Which is a Logitech, Inc. Webcam Pro
>> 9000.
>>
>> The command I use is:
>> ffmpeg -async 1                         \
>> -f alsa                          \
>> -i "plughw:1"                    \
>> -f x11grab -s "800x600"          \
>> -r 25                            \
>> -i :0.0                          \
>> -acodec libmp3lame               \
>> -ac 1                            \
>> -ar 22050                        \
>> -ab 64k                          \
>> -qscale:video 4                  \
>> testing.avi
>>
>> Also sometimes the audio is a little out of sync and there are
>> sometimes sharp sounds. What should I change?
>>
> ffmpeg inputs are opened sequentially. I have found that opening
> slow input devices first could help with AV sync. OTOH, if I
> understand correctly your setup, you display a webcam output on
> screen and grab that with x11grab.

Correct, I use this because I want to integrate some graphics in  my
video. Up till now I found this the best way (for me).

But it is probably the problem, because when I use:
    ffmpeg -async 1                \
           -f alsa                 \
           -i plughw:1             \
           -acodec libmp3lame      \
           -ac 1                   \
           -ar 22050               \
           -ab 64k                 \
           $(date +%F_%T).mp3

It sounds OK. But when I do the previous command, my computer is not
very busy. So I do not really understand the problem. Is there a way
to easily mix the video from my camera with the audio from my webcam
and then also replace some parts of the video with video? That would
really solve my problem.


> This for sure add some latency in
> the video signal. There must be a way to manually add a ts offset on
> the cmdline to compensate for that.
>
> I think that your sharp sounds are caused by saturation. If you had
> ALSA buffer overruns, ffmpeg would report it and it sounds more like
> clipping. There are 2 things that I can think that you can play with
> fix this problem:
>
> 1. Check your mic preamp level with the alsamixer app.

This gives:
    This sound device does not have any playback controls. 


> 2. Use "hw:1" instead of "plughw:1" to let ffmpeg do the signal adaptation.

That gives:
    [alsa @ 0x149caa0] cannot set channel count to 2 (Invalid argument)
    hw:1: Input/output error


> What plughw is doing is to ask ALSA to adapt device hw params to
> what the app is asking (sampling rate, bits per sample, etc). You
> could check if ffmpeg is handling this better.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From cehoyos at ag.or.at  Fri Apr 18 10:49:34 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 18 Apr 2014 08:49:34 +0000 (UTC)
Subject: [FFmpeg-user] Distorted audio in screencast
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140418T104819-308@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:

> I use this because I want to integrate some graphics 
> in my video.

Did you test the overlay filter?

> > 2. Use "hw:1" instead of "plughw:1" to let ffmpeg 
> > do the signal adaptation.
> 
> That gives:
>     [alsa  <at>  0x149caa0] cannot set channel count 
>     to 2 (Invalid argument) hw:1: Input/output error

Move "-ac 1" in front of "-i"

Carl Eugen


From Cecil at decebal.nl  Fri Apr 18 13:14:09 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Fri, 18 Apr 2014 13:14:09 +0200
Subject: [FFmpeg-user] Distorted audio in screencast
In-Reply-To: <loom.20140418T104819-308@post.gmane.org> (Carl Eugen Hoyos's
	message of "Fri, 18 Apr 2014 08:49:34 +0000 (UTC)")
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
	<loom.20140418T104819-308@post.gmane.org>
Message-ID: <87ioq6anzy.fsf@Equus.Decebal.nl>

Op Friday 18 Apr 2014 10:49 CEST schreef Carl Eugen Hoyos:

> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>
>> I use this because I want to integrate some graphics 
>> in my video.
>
> Did you test the overlay filter?
>
>>> 2. Use "hw:1" instead of "plughw:1" to let ffmpeg 
>>> do the signal adaptation.
>>
>> That gives:
>> [alsa  <at>  0x149caa0] cannot set channel count 
>> to 2 (Invalid argument) hw:1: Input/output error
>
> Move "-ac 1" in front of "-i"

Same error. But I found something interesting. When using the internal
web-cam for video and the external web-cam for audio the sound is much
better. So it looks like fetching video interferes with fetching
audio.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From cehoyos at ag.or.at  Fri Apr 18 13:22:57 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 18 Apr 2014 11:22:57 +0000 (UTC)
Subject: [FFmpeg-user] Distorted audio in screencast
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
	<loom.20140418T104819-308@post.gmane.org>
	<87ioq6anzy.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140418T132226-240@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:

> > Move "-ac 1" in front of "-i"
> 
> Same error.

Please provide the failing command line together with 
the complete, uncut console output.

Carl Eugen


From mailinglist at podiumbv.nl  Fri Apr 18 16:51:20 2014
From: mailinglist at podiumbv.nl (Podium B.V.)
Date: Fri, 18 Apr 2014 16:51:20 +0200
Subject: [FFmpeg-user] VP9 Enc. seems OK but Chromium,
	FireFox and VLC won't play
Message-ID: <53513BE8.4020309@podiumbv.nl>

Hi all,

I'm experimenting again with VP9. Our first goal is to use
it local as a proxy for our (very) big video files. So I tried
to encode my files the following way:

###########

VAR_BUILD="ffmpeg-git-20140417-64bit-static";
VAR_INPUT="/home/<user>/Videos/MuxedCLIP0001.mov";
VAR_OUTPUT="$VAR_INPUT.webm";
VAR_FONTFILE="/usr/share/fonts/truetype/ubuntu-font-family/UbuntuMono-B.ttf";

time /home/<user>/Videos/ffmpeg/$VAR_BUILD/ffmpeg \
-i "$VAR_INPUT" \
-codec:v libvpx-vp9 -strict experimental -quality good -cpu-used 4 -qmin 
50 -qmax 50 -pix_fmt yuv420p \
-map 0:0 -filter:v:0 "yadif=0:-1:0, scale=536:288, setsar=1/1, 
crop=524:288:6:0, hue=0:1.1, drawtext=fontfile=$VAR_FONTFILE: 
timecode='00\:00\:00\:00': r=25: \ x=(w-tw)/2: y=h-(2*lh): 
fontcolor=white: fontsize=22: box=1: boxcolor=0x00000000 at 1" \
-codec:a libvorbis -b:a 96k \
-map 0:1 -filter:a:0 "pan=stereo|c0=c0|c1=c1" \
"$VAR_OUTPUT"

--> Scroll down for complete output <--

###########

from the forums, internet and the WebM website I understand:

Product Support (http://www.webmproject.org/vp9/)

Google Chrome
VP9 decode support was first enabled by default in Google Chrome 29 Dev 
channel (r206883) on 2013-06-26.

Mozilla Firefox
VP9 decode support was first added to Firefox Aurora (?pre-beta?) 
nightly builds on 2013-12-06.

VLC
Experimental VP9 decode support was added to VLC in version 2.1.2.

FFMpeg / Libav
Search the FFMpeg codebase for recent libvpx-related commits
Search the Libav codebase for recent libvpx-related commits

BUT with the files FFMpeg did encode for me non of these "players" play 
the video.
They DO show a video-frame and when I scroll trough the timeline the 
video-frame
of that moment in time become visible! But no sound and it's not playing...

Anyone any ideas ??
Thanks!

###########

$ bash makeVP9proxy.sh
ffmpeg version N-42139-g443261c- http://johnvansickle.com/ffmpeg/ 
Copyright (c) 2000-2014 the FFmpeg developers
built on Apr 17 2014 02:24:41 with gcc 4.8 (Debian 4.8.2-19)
configuration: --enable-gpl --enable-version3 --disable-shared 
--disable-debug --enable-runtime-cpudetect --enable-libmp3lame 
--enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex 
--enable-libvorbis --enable-libvpx --enable-libfreetype 
--enable-fontconfig --enable-libxvid --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc 
--enable-libvo-amrwbenc --enable-gray --enable-libopenjpeg 
--enable-libopus --disable-ffserver
libavutil 52. 76.100 / 52. 76.100
libavcodec 55. 58.103 / 55. 58.103
libavformat 55. 37.100 / 55. 37.100
libavdevice 55. 13.100 / 55. 13.100
libavfilter 4. 4.100 / 4. 4.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 
'/home/<user>/Videos/MuxedCLIP0001.mov':
Metadata:
major_brand : qt
minor_version : 512
compatible_brands: qt
encoder : Lavf55.19.104
Duration: 00:00:23.76, start: 0.000000, bitrate: 38082 kb/s
Stream #0:0(eng): Video: mpeg2video (Main) (xdv7 / 0x37766478), 
yuv420p(tv, bt709), 1440x1080 [SAR 4:3 DAR 16:9], 35008 kb/s, 25 fps, 25 
tbr, 12800 tbn, 50 tbc (default)
Metadata:
handler_name : DataHandler
timecode : 14:54:55:06
Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, 
s16, 768 kb/s (default)
Metadata:
handler_name : DataHandler
Stream #0:2(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, 
s16, 768 kb/s
Metadata:
handler_name : DataHandler
Stream #0:3(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, 
s16, 768 kb/s
Metadata:
handler_name : DataHandler
Stream #0:4(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, mono, 
s16, 768 kb/s
Metadata:
handler_name : DataHandler
Stream #0:5(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
Metadata:
handler_name : DataHandler
timecode : 14:54:55:06
[Parsed_pan_0 @ 0x421bb60] Pure channel mapping detected: 0 M
[libvpx-vp9 @ 0x4195920] v1.3.0-2384-g68dcdde
Output #0, webm, to '/home/<user>/Videos/MuxedCLIP0001.mov.webm':
Metadata:
major_brand : qt
minor_version : 512
compatible_brands: qt
encoder : Lavf55.37.100
Stream #0:0(eng): Video: vp9 (libvpx-vp9), yuv420p, 524x288 [SAR 1:1 DAR 
131:72], q=50-50, 200 kb/s, 1k tbn, 25 tbc (default)
Metadata:
handler_name : DataHandler
timecode : 14:54:55:06
Stream #0:1(eng): Audio: vorbis (libvorbis), 48000 Hz, stereo, fltp, 96 
kb/s (default)
Metadata:
handler_name : DataHandler
Stream mapping:
Stream #0:0 -> #0:0 (mpeg2video -> libvpx-vp9)
Stream #0:1 -> #0:1 (pcm_s16le -> libvorbis)
Press [q] to stop, [?] for help
frame= 594 fps= 38 q=0.0 Lsize= 263kB time=00:00:23.80 bitrate= 90.7kbits/s
video:40kB audio:207kB subtitle:0kB other streams:0kB global headers:4kB 
muxing overhead: 6.699785%

real 0m15.656s
user 0m24.745s
sys 0m0.416s



From Cecil at decebal.nl  Fri Apr 18 17:54:56 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Fri, 18 Apr 2014 17:54:56 +0200
Subject: [FFmpeg-user] Distorted audio in screencast
In-Reply-To: <loom.20140418T132226-240@post.gmane.org> (Carl Eugen Hoyos's
	message of "Fri, 18 Apr 2014 11:22:57 +0000 (UTC)")
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
	<loom.20140418T104819-308@post.gmane.org>
	<87ioq6anzy.fsf@Equus.Decebal.nl>
	<loom.20140418T132226-240@post.gmane.org>
Message-ID: <8761m6aazz.fsf@Equus.Decebal.nl>

Op Friday 18 Apr 2014 13:22 CEST schreef Carl Eugen Hoyos:

> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>
>>> Move "-ac 1" in front of "-i"
>>
>> Same error.
>
> Please provide the failing command line together with 
> the complete, uncut console output.

ffmpeg -async 1                         \
       -f alsa                          \
       -i hw:1                          \
       -f x11grab -s 1280x1024          \
       -r 25                            \
       -ac 1                            \
       -i :0.0                          \
       -acodec libmp3lame               \
       -ar 22050                        \
       -ab 64k                          \
       -qscale:video 4                  \
       testing.avi
ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Dec 17 2013 19:47:47 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 --enable-gnutls
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101
  libavformat    55. 19.104 / 55. 19.104
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
[alsa @ 0x12e5aa0] cannot set channel count to 2 (Invalid argument)
hw:1: Input/output error

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From cehoyos at ag.or.at  Fri Apr 18 18:44:38 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 18 Apr 2014 16:44:38 +0000 (UTC)
Subject: [FFmpeg-user] Distorted audio in screencast
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
	<loom.20140418T104819-308@post.gmane.org>
	<87ioq6anzy.fsf@Equus.Decebal.nl>
	<loom.20140418T132226-240@post.gmane.org>
	<8761m6aazz.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140418T184343-504@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:



> ffmpeg -async 1                         \

>        -f alsa                          \

>        -i hw:1                          \



>        -ac 1                            \



Move "-ac 1" in front of "-i"



Carl Eugen




From cehoyos at ag.or.at  Fri Apr 18 18:46:47 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 18 Apr 2014 16:46:47 +0000 (UTC)
Subject: [FFmpeg-user] VP9 Enc. seems OK but Chromium,
	FireFox and VLC won't play
References: <53513BE8.4020309@podiumbv.nl>
Message-ID: <loom.20140418T184522-156@post.gmane.org>

Podium B.V. <mailinglist <at> podiumbv.nl> writes:



> BUT with the files FFMpeg did encode for me non of 

> these "players" play the video.



Did you test with ffplay?



> They DO show a video-frame and when I scroll trough 

> the timeline the video-frame of that moment in time 

> become visible! But no sound and it's not playing...



Are you reporting a performance issue?



Carl Eugen




From eclipse7 at gmx.net  Fri Apr 18 19:45:38 2014
From: eclipse7 at gmx.net (Alexander Strasser)
Date: Fri, 18 Apr 2014 19:45:38 +0200
Subject: [FFmpeg-user] OpenSSL Heartbeat bug
Message-ID: <20140418174538.GA4779@akuma.local>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Our server hosting the Trac issue tracker was vulnerable to the attack
against OpenSSL known as "heartbleed". The OpenSSL software library was
updated on 7th of April, shortly after the vulnerability was publicly
disclosed. We have changed the private keys (and certificates) for all
FFmpeg servers. The new SHA1 fingerprints are:

ffmpeg.org:      d0 4c 1f d0 08 f6 e0 24 f0 2c 31 de 4d 01 45 04 32 2e 36 29
trac.ffmpeg.org: 2a 1c d7 a5 7e 39 6a bc c3 55 22 88 ba 2a cd e0 1f c1 9f 6e

We encourage you to read up on "OpenSSL heartbleed"[1]. It is possible
that login data for the issue tracker was exposed to people exploiting
this security hole. You might want to change your password in the tracker
and everywhere else you used that same password.


[1] For example here: https://www.schneier.com/blog/archives/2014/04/heartbleed.html
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1

iEYEARECAAYFAlNRZMIACgkQp9ile6h25Y9wVACfRYx19HPswWI9HvKXr/fh7bkF
rKcAn1Yfyz0PPktsZXs4KGw9az7egMTV
=Pi0n
-----END PGP SIGNATURE-----

From spencer.graves at prodsyse.com  Fri Apr 18 19:59:40 2014
From: spencer.graves at prodsyse.com (Spencer Graves)
Date: Fri, 18 Apr 2014 10:59:40 -0700
Subject: [FFmpeg-user] OpenSSL Heartbeat bug
In-Reply-To: <20140418174538.GA4779@akuma.local>
References: <20140418174538.GA4779@akuma.local>
Message-ID: <5351680C.8080508@prodsyse.com>

On 4/18/2014 10:45 AM, Alexander Strasser wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Our server hosting the Trac issue tracker was vulnerable to the attack
> against OpenSSL known as "heartbleed". The OpenSSL software library was
> updated on 7th of April, shortly after the vulnerability was publicly
> disclosed. We have changed the private keys (and certificates) for all
> FFmpeg servers. The new SHA1 fingerprints are:
>
> ffmpeg.org:      d0 4c 1f d0 08 f6 e0 24 f0 2c 31 de 4d 01 45 04 32 2e 36 29
> trac.ffmpeg.org: 2a 1c d7 a5 7e 39 6a bc c3 55 22 88 ba 2a cd e0 1f c1 9f 6e
>
> We encourage you to read up on "OpenSSL heartbleed"[1]. It is possible
> that login data for the issue tracker was exposed to people exploiting
> this security hole. You might want to change your password in the tracker
> and everywhere else you used that same password.

or a similar password.  There are criminal organizations coding bots to 
guess password based on bits of information collected from social 
networking sites like Facebook, genealogical databases, etc.


       Spencer

> [1] For example here: https://www.schneier.com/blog/archives/2014/04/heartbleed.html
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1
>
> iEYEARECAAYFAlNRZMIACgkQp9ile6h25Y9wVACfRYx19HPswWI9HvKXr/fh7bkF
> rKcAn1Yfyz0PPktsZXs4KGw9az7egMTV
> =Pi0n
> -----END PGP SIGNATURE-----
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

From Cecil at decebal.nl  Fri Apr 18 22:32:11 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Fri, 18 Apr 2014 22:32:11 +0200
Subject: [FFmpeg-user] Distorted audio in screencast
In-Reply-To: <loom.20140418T184343-504@post.gmane.org> (Carl Eugen Hoyos's
	message of "Fri, 18 Apr 2014 16:44:38 +0000 (UTC)")
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
	<loom.20140418T104819-308@post.gmane.org>
	<87ioq6anzy.fsf@Equus.Decebal.nl>
	<loom.20140418T132226-240@post.gmane.org>
	<8761m6aazz.fsf@Equus.Decebal.nl>
	<loom.20140418T184343-504@post.gmane.org>
Message-ID: <871twu9y5w.fsf@Equus.Decebal.nl>

Op Friday 18 Apr 2014 18:44 CEST schreef Carl Eugen Hoyos:

> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>
>
>
>> ffmpeg -async 1                         \
>
>> -f alsa                          \
>
>> -i hw:1                          \
>
>
>
>> -ac 1                            \
>
>
>
> Move "-ac 1" in front of "-i"

Eh, it was:
ffmpeg -async 1                         \
       -f alsa                          \
       -i hw:1                          \
       -f x11grab -s 1280x1024          \
       -r 25                            \
       -ac 1                            \
       -i :0.0                          \
       -acodec libmp3lame               \
       -ar 22050                        \
       -ab 64k                          \
       -qscale:video 4                  \
       testing.avi

There are two -i's and I put it before the second. :-(

When I make it:
ffmpeg -async 1                         \
       -f alsa                          \
       -ac 1                            \
       -i hw:1                          \
       -f x11grab -s 1280x1024          \
       -r 25                            \
       -i :0.0                          \
       -acodec libmp3lame               \
       -ar 22050                        \
       -ab 64k                          \
       -qscale:video 4                  \
       testing.avi

it works, but I still have the distorted sound. So it still does not
work satisfactory. So I just use one web-cam for the video and the
other for audio. Not optimal, but it works.

By the way, with the above command the audio only starts after 3
seconds.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From xanadu at apost.plala.or.jp  Fri Apr 18 23:29:46 2014
From: xanadu at apost.plala.or.jp (=?utf-8?B?5a6u5p2RIOWFrOeUtw==?=)
Date: Sat, 19 Apr 2014 06:29:46 +0900
Subject: [FFmpeg-user] Enabling libx265
In-Reply-To: <loom.20140418T051500-274@post.gmane.org>
References: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>
	<loom.20140418T051500-274@post.gmane.org>
Message-ID: <34A48F6F-E481-47AF-B49B-1ED18ECF49B0@apost.plala.or.jp>

Hello Carl-san, 

2014/04/18 12:16?Carl Eugen Hoyos <cehoyos at ag.or.at> ?????

> ?? ?? <xanadu <at> apost.plala.or.jp> writes:
> 
>> -- Installing: /Volumes/ffmpeg_compile/lib/libx265.a
>> -- Installing: /Volumes/ffmpeg_compile/include/x265.h
> 
>> $ ./configure --as=yasm --enable-gpl --enable-libx265
> 
> This is missing --extra-cflags=-I/Volumes/ffmpeg_compile/include 
> --extra-ldflags=-L/Volumes/ffmpeg_compile/lib

I  have added ?extra-cflags and ?extra-ldflags to my configure option.  But the result was same. 
 i.e.
1.can?t configure with static x265 library 
2.with shard library, funny dependency still remain.

I have clean installed Mac OS many times but the result was same.
Now I?m thinking there are something wrong with building x265.  Since  x265 binary also can?t execute in a installed path.  
(/Volumes/ffmpeg_compile/bin). It only can execute on build path (/Volumes/ffmpeg_compile/compile/x265/build/linux)


I?m tired?

Miya

From Cecil at decebal.nl  Sat Apr 19 12:27:21 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 12:27:21 +0200
Subject: [FFmpeg-user] Removing audio noise
Message-ID: <87tx9p8vhy.fsf@Equus.Decebal.nl>

At the moment I am using:
ffmpeg -async 1
       -f alsa
       -ac 1
       -i hw:1,0,0
       -f x11grab -s 1920x1080
       -r 25
       -i :0.0
       -acodec libmp3lame
       -ar 22050
       -ab 64k
       -qscale:video 4
       2014-04-19_12:24.avi

This works reasonably good. The only problem is that there is some
white noise in the audio. Is there a way to filter this away?

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From Cecil at decebal.nl  Sat Apr 19 12:31:42 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 12:31:42 +0200
Subject: [FFmpeg-user] Is there a better way to do this
Message-ID: <87ppkd8vap.fsf@Equus.Decebal.nl>

I am using:
ffmpeg -async 1
       -f alsa
       -ac 1
       -i hw:1,0,0
       -f x11grab -s 1920x1080
       -r 25
       -i :0.0
       -acodec libmp3lame
       -ar 22050
       -ab 64k
       -qscale:video 4
       2014-04-19_12:24.avi

Works reasonably well. But in the man page I do not see the parameter
'-ab' described. Should I use something else for this?


I also get:
[swscaler @ 0x13d3100] deprecated pixel format used, make sure you did set range correctly

What am I doing wrong here?


And if I could do this otherwise better I would like to hear about it.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From h.reindl at thelounge.net  Sat Apr 19 12:40:13 2014
From: h.reindl at thelounge.net (Reindl Harald)
Date: Sat, 19 Apr 2014 12:40:13 +0200
Subject: [FFmpeg-user] Is there a better way to do this
In-Reply-To: <87ppkd8vap.fsf@Equus.Decebal.nl>
References: <87ppkd8vap.fsf@Equus.Decebal.nl>
Message-ID: <5352528D.4050403@thelounge.net>



Am 19.04.2014 12:31, schrieb Cecil Westerhof:
> I am using:
> ffmpeg -async 1
>        -f alsa
>        -ac 1
>        -i hw:1,0,0
>        -f x11grab -s 1920x1080
>        -r 25
>        -i :0.0
>        -acodec libmp3lame
>        -ar 22050
>        -ab 64k
>        -qscale:video 4
>        2014-04-19_12:24.avi
> 
> Works reasonably well. But in the man page I do not see the parameter
> '-ab' described. Should I use something else for this?

ab = AudioBitrate
http://en.wikipedia.org/wiki/MP3

for spoken audio 64 is fine, for music it is crap and
should be at least 128

> I also get:
> [swscaler @ 0x13d3100] deprecated pixel format used, make sure you did set range correctly
> 
> What am I doing wrong here?

just a warning

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From mr.artichaut at gmail.com  Sat Apr 19 12:44:08 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Sat, 19 Apr 2014 12:44:08 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <20140413084205.GA18477@phare.normalesup.org>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
Message-ID: <CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>

Thank.

I've done as you said but still have the same issue. By setting a DEBUG
level for logs and dumping the graph I have this :
https://gist.github.com/MrArtichaut/b4d958b36ff2fdbb64ca

What's is this resampler auto-inserted in the graph ? Maybe my issue came
from that and I have to insert one myself with my own params ?

Thank you.

From Cecil at decebal.nl  Sat Apr 19 12:52:58 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 12:52:58 +0200
Subject: [FFmpeg-user] Is there a better way to do this
In-Reply-To: <5352528D.4050403@thelounge.net> (Reindl Harald's message of
	"Sat, 19 Apr 2014 12:40:13 +0200")
References: <87ppkd8vap.fsf@Equus.Decebal.nl> <5352528D.4050403@thelounge.net>
Message-ID: <87lhv18ub9.fsf@Equus.Decebal.nl>

Op Saturday 19 Apr 2014 12:40 CEST schreef Reindl Harald:

> Am 19.04.2014 12:31, schrieb Cecil Westerhof:
>> I am using:
>> ffmpeg -async 1
>> -f alsa
>> -ac 1
>> -i hw:1,0,0
>> -f x11grab -s 1920x1080
>> -r 25
>> -i :0.0
>> -acodec libmp3lame
>> -ar 22050
>> -ab 64k
>> -qscale:video 4
>> 2014-04-19_12:24.avi
>>
>> Works reasonably well. But in the man page I do not see the
>> parameter '-ab' described. Should I use something else for this?
>
> ab = AudioBitrate
> http://en.wikipedia.org/wiki/MP3

Why is it then not in the documentation?


> for spoken audio 64 is fine, for music it is crap and
> should be at least 128

At the moment it is for a screen-cast, so only spoken audio.


>> I also get: [swscaler @ 0x13d3100] deprecated pixel format used,
>> make sure you did set range correctly
>>
>> What am I doing wrong here?
>
> just a warning

OK, nothing to worry about then.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From h.reindl at thelounge.net  Sat Apr 19 13:01:50 2014
From: h.reindl at thelounge.net (Reindl Harald)
Date: Sat, 19 Apr 2014 13:01:50 +0200
Subject: [FFmpeg-user] Is there a better way to do this
In-Reply-To: <87lhv18ub9.fsf@Equus.Decebal.nl>
References: <87ppkd8vap.fsf@Equus.Decebal.nl>	<5352528D.4050403@thelounge.net>
	<87lhv18ub9.fsf@Equus.Decebal.nl>
Message-ID: <5352579E.8020800@thelounge.net>



Am 19.04.2014 12:52, schrieb Cecil Westerhof:
> Op Saturday 19 Apr 2014 12:40 CEST schreef Reindl Harald:
> 
>> Am 19.04.2014 12:31, schrieb Cecil Westerhof:
>>> I am using:
>>> ffmpeg -async 1
>>> -f alsa
>>> -ac 1
>>> -i hw:1,0,0
>>> -f x11grab -s 1920x1080
>>> -r 25
>>> -i :0.0
>>> -acodec libmp3lame
>>> -ar 22050
>>> -ab 64k
>>> -qscale:video 4
>>> 2014-04-19_12:24.avi
>>>
>>> Works reasonably well. But in the man page I do not see the
>>> parameter '-ab' described. Should I use something else for this?
>>
>> ab = AudioBitrate
>> http://en.wikipedia.org/wiki/MP3
> 
> Why is it then not in the documentation?

other question: why do you set params don't know what they are?

because it is disapproved, exists because of compatibility
as well as "vb" for video-bitrate while recent versions
prefer -b:v and -b:a, ffmpeg --help is a good start in doubt

a simle google search answers that in the search results
https://www.google.com/search?q=ffmpeg+%22-ab%22

Video options:
-vframes number     set the number of video frames to record
-r rate             set frame rate (Hz value, fraction or abbreviation)
-s size             set frame size (WxH or abbreviation)
-aspect aspect      set aspect ratio (4:3, 16:9 or 1.3333, 1.7777)
-bits_per_raw_sample number  set the number of bits per raw sample
-vn                 disable video
-vcodec codec       force video codec ('copy' to copy stream)
-timecode hh:mm:ss[:;.]ff  set initial TimeCode value.
-pass n             select the pass number (1 to 3)
-vf filter_graph    set video filters
-b bitrate          video bitrate (please use -b:v)
-dn                 disable data

Audio options:
-aframes number     set the number of audio frames to record
-aq quality         set audio quality (codec-specific)
-ar rate            set audio sampling rate (in Hz)
-ac channels        set number of audio channels
-an                 disable audio
-acodec codec       force audio codec ('copy' to copy stream)
-vol volume         change audio volume (256=normal)
-af filter_graph    set audio filters

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From codecomplete at free.fr  Sat Apr 19 13:32:11 2014
From: codecomplete at free.fr (Gilles)
Date: Sat, 19 Apr 2014 13:32:11 +0200
Subject: [FFmpeg-user] Turning MP3 into MP4 and join with second MP4 file?
Message-ID: <hvl4l9965gjm9gt8ucrn500qa8ehur942k@4ax.com>

Hello

I have recordings of a radio show, whose first part is only available
as an MP3 audio file while the second parts is available as an MP4
audiovideo file.

I'm using ffmpeg version N-61821-gc05065a on Windows.

I used ffmpeg to turn the MP3 file and a PNG file into an MP4 file,
converted the second MP4 file to match the audiovideo settings of the
first MP4, but it doesn't work and neitherMP4Joiner nor My MP4Box GUI
are able to join the two MP4 files.

****************************************************************

Here's the infos from MediaCodec and the two ffmpeg commands I used:

========== Turning MP3 into MP4 video:
INPUT
Complete name                            : C:\first.half.mp3
Format                                   : MPEG Audio
File size                                : 9.50 MiB
Duration                                 : 10mn 22s
Overall bit rate mode                    : Constant
Overall bit rate                         : 128 Kbps
Writing library                          : LAME3.99r

Audio
Format                                   : MPEG Audio
Format version                           : Version 1
Format profile                           : Layer 3
Mode                                     : Joint stereo
Duration                                 : 10mn 22s
Bit rate mode                            : Constant
Bit rate                                 : 128 Kbps
Channel(s)                               : 2 channels
Sampling rate                            : 48.0 KHz
Compression mode                         : Lossy
Stream size                              : 9.50 MiB (100%)
Writing library                          : LAME3.99r
Encoding settings                        : -m j -V 4 -q 3 -lowpass 17
-b 128

ffmpeg.exe -loop 1 -i speaker.png -i first.half.mp3 -shortest -vf
scale=640:480 -c:v libx264 -tune stillimage -c:a copy
radio.show.first.half.mp4

OUTPUT
Complete name                            :
C:\radio.show.first.half.mp4
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : isom
File size                                : 13.3 MiB
Duration                                 : 10mn 24s
Overall bit rate                         : 178 Kbps
Writing application                      : Lavf55.35.101

Video
ID                                       : 1
Format                                   : AVC
Format/Info                              : Advanced Video Codec
Format profile                           : High 4:4:4 Predictive at L3.0
Format settings, CABAC                   : Yes
Format settings, ReFrames                : 4 frames
Codec ID                                 : avc1
Codec ID/Info                            : Advanced Video Coding
Duration                                 : 10mn 24s
Bit rate                                 : 45.2 Kbps
Width                                    : 640 pixels
Height                                   : 480 pixels
Display aspect ratio                     : 4:3
Frame rate mode                          : Constant
Frame rate                               : 25.000 fps
Color space                              : YUV
Chroma subsampling                       : 4:4:4
Bit depth                                : 8 bits
Scan type                                : Progressive
Bits/(Pixel*Frame)                       : 0.006
Stream size                              : 3.36 MiB (25%)
Writing library                          : x264 core 142 r2409 d6b4e63
Encoding settings                        : cabac=1 / ref=3 /
deblock=1:-3:-3 / analyse=0x3:0x113 / me=hex / subme=7 / psy=1 /
psy_rd=2.00:0.70 / 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=3 / 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=1:1.20

Audio
ID                                       : 2
Format                                   : MPEG Audio
Format version                           : Version 1
Format profile                           : Layer 3
Mode                                     : Joint stereo
Mode extension                           : MS Stereo
Codec ID                                 : 6B
Duration                                 : 10mn 22s
Bit rate mode                            : Constant
Bit rate                                 : 128 Kbps
Channel(s)                               : 2 channels
Sampling rate                            : 48.0 KHz
Compression mode                         : Lossy
Stream size                              : 9.50 MiB (72%)


========== Converting MP4 to identical settings found in first file:
INPUT
Complete name                            : C:\second.half.mp4
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : isom
File size                                : 118 MiB
Duration                                 : 19mn 41s
Overall bit rate mode                    : Variable
Overall bit rate                         : 835 Kbps
Tagged date                              : UTC 2014-04-17 08:37:51
Writing application                      : Lavf55.19.104

Video
ID                                       : 1
Format                                   : AVC
Format/Info                              : Advanced Video Codec
Format profile                           : High at L3.0
Format settings, CABAC                   : Yes
Format settings, ReFrames                : 4 frames
Codec ID                                 : avc1
Codec ID/Info                            : Advanced Video Coding
Duration                                 : 19mn 41s
Bit rate                                 : 700 Kbps
Width                                    : 848 pixels
Height                                   : 480 pixels
Display aspect ratio                     : 16:9
Frame rate mode                          : Constant
Frame rate                               : 25.000 fps
Color space                              : YUV
Chroma subsampling                       : 4:2:0
Bit depth                                : 8 bits
Scan type                                : Progressive
Bits/(Pixel*Frame)                       : 0.069
Stream size                              : 98.5 MiB (84%)
Writing library                          : x264 core 140 r2377M
1ca7bb9

Audio
ID                                       : 2
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : LC
Codec ID                                 : 40
Duration                                 : 19mn 41s
Duration_LastFrame                       : -11ms
Bit rate mode                            : Variable
Bit rate                                 : 128 Kbps
Maximum bit rate                         : 800 Kbps
Channel(s)                               : 2 channels
Channel positions                        : Front: L R
Sampling rate                            : 44.1 KHz
Compression mode                         : Lossy
Stream size                              : 18.0 MiB (15%)

ffmpeg.exe -i second.half.mp4  -shortest -s:v 640x480 -c:v libx264
-c:a  libmp3lame -b:a 128k -r:a 48000 radio.show.second.half.mp4

OUTPUT:
Complete name                            :
C:\radio.show.second.half.mp4
Format                                   : MPEG-4
Format profile                           : Base Media
Codec ID                                 : isom
File size                                : 69.6 MiB
Duration                                 : 19mn 41s
Overall bit rate                         : 494 Kbps
Writing application                      : Lavf55.35.101

Video
ID                                       : 1
Format                                   : AVC
Format/Info                              : Advanced Video Codec
Format profile                           : High at L3.0
Format settings, CABAC                   : Yes
Format settings, ReFrames                : 4 frames
Codec ID                                 : avc1
Codec ID/Info                            : Advanced Video Coding
Duration                                 : 19mn 41s
Bit rate                                 : 359 Kbps
Width                                    : 640 pixels
Height                                   : 480 pixels
Display aspect ratio                     : 16:9
Frame rate mode                          : Constant
Frame rate                               : 25.000 fps
Color space                              : YUV
Chroma subsampling                       : 4:2:0
Bit depth                                : 8 bits
Scan type                                : Progressive
Bits/(Pixel*Frame)                       : 0.047
Stream size                              : 50.5 MiB (73%)
Writing library                          : x264 core 142 r2409 d6b4e63
Encoding settings                        : 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=3 / 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

Audio
ID                                       : 2
Format                                   : MPEG Audio
Format version                           : Version 1
Format profile                           : Layer 3
Mode                                     : Joint stereo
Mode extension                           : MS Stereo
Codec ID                                 : 6B
Duration                                 : 19mn 41s
Duration_LastFrame                       : -1ms
Bit rate mode                            : Constant
Bit rate                                 : 128 Kbps
Channel(s)                               : 2 channels
Sampling rate                            : 44.1 KHz
Compression mode                         : Lossy
Stream size                              : 18.0 MiB (26%)
Writing library                          : LAME3.99.5

****************************************************************

The first file:
aspect ratio 4:3
sample frequency 48000Hz

The second file:
aspect ratio 1:1,77
sample frequency 44100Hz

Any help appreciated,
Thank you.


From george at nsup.org  Sat Apr 19 13:35:13 2014
From: george at nsup.org (Nicolas George)
Date: Sat, 19 Apr 2014 13:35:13 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
Message-ID: <20140419113513.GA14903@phare.normalesup.org>

Le decadi 30 germinal, an CCXXII, Mr Artichaut a ?crit?:
> I've done as you said but still have the same issue. By setting a DEBUG
> level for logs and dumping the graph I have this :
> https://gist.github.com/MrArtichaut/b4d958b36ff2fdbb64ca

Without the code, there is no way of knowing what you are doing wrong.

> What's is this resampler auto-inserted in the graph ?

amix can only work with float samples (hat is one of the reasons I
discourage people of using it; use amerge and pan, they support many
formats); since your input has integer samples, resample is inserted to do
the conversion.

> Maybe my issue came from that and I have to insert one myself with my own
> params ?

No, that is completely unrelated.

Regards,

-- 
  Nicolas George
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From cehoyos at ag.or.at  Sat Apr 19 13:42:05 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 19 Apr 2014 11:42:05 +0000 (UTC)
Subject: [FFmpeg-user] Turning MP3 into MP4 and join with second MP4
	file?
References: <hvl4l9965gjm9gt8ucrn500qa8ehur942k@4ax.com>
Message-ID: <loom.20140419T134114-466@post.gmane.org>

Gilles <codecomplete <at> free.fr> writes:

> Here's the infos from MediaCodec and the two 
> ffmpeg commands I used:

I don't remember a report where MediaInfo was of any 
help but the complete, uncut console output for 
your failing FFmpeg command line is needed if you 
want support here.

Carl Eugen


From cehoyos at ag.or.at  Sat Apr 19 13:43:09 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 19 Apr 2014 11:43:09 +0000 (UTC)
Subject: [FFmpeg-user] Is there a better way to do this
References: <87ppkd8vap.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140419T134220-207@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:

> I am using:

[...]

Complete, uncut console output missing.

I am always using -ab, it has the advantage 
that you can easily test older versions without 
changing the command.

Carl Eugen


From codecomplete at free.fr  Sat Apr 19 14:23:55 2014
From: codecomplete at free.fr (Gilles)
Date: Sat, 19 Apr 2014 14:23:55 +0200
Subject: [FFmpeg-user] Turning MP3 into MP4 and join with second MP4
	file?
References: <hvl4l9965gjm9gt8ucrn500qa8ehur942k@4ax.com>
	<loom.20140419T134114-466@post.gmane.org>
Message-ID: <sjo4l9t9mn2rtpvvgpbutk7shvb596b6pi@4ax.com>

On Sat, 19 Apr 2014 11:42:05 +0000 (UTC), Carl Eugen Hoyos
<cehoyos at ag.or.at> wrote:

>I don't remember a report where MediaInfo was of any 
>help but the complete, uncut console output for 
>your failing FFmpeg command line is needed if you 
>want support here.

Since the report files are pretty big, I zipped and uploaded them
here:
http://pastelink.me/dl/76ae5d

Thank you.


From Cecil at decebal.nl  Sat Apr 19 14:36:19 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 14:36:19 +0200
Subject: [FFmpeg-user] Is there a better way to do this
In-Reply-To: <5352579E.8020800@thelounge.net> (Reindl Harald's message of
	"Sat, 19 Apr 2014 13:01:50 +0200")
References: <87ppkd8vap.fsf@Equus.Decebal.nl> <5352528D.4050403@thelounge.net>
	<87lhv18ub9.fsf@Equus.Decebal.nl> <5352579E.8020800@thelounge.net>
Message-ID: <87ha5p8pj0.fsf@Equus.Decebal.nl>

Op Saturday 19 Apr 2014 13:01 CEST schreef Reindl Harald:

>
>
> Am 19.04.2014 12:52, schrieb Cecil Westerhof:
>> Op Saturday 19 Apr 2014 12:40 CEST schreef Reindl Harald:
>>
>>> Am 19.04.2014 12:31, schrieb Cecil Westerhof:
>>>> I am using:
>>>> ffmpeg -async 1
>>>> -f alsa
>>>> -ac 1
>>>> -i hw:1,0,0
>>>> -f x11grab -s 1920x1080
>>>> -r 25
>>>> -i :0.0
>>>> -acodec libmp3lame
>>>> -ar 22050
>>>> -ab 64k
>>>> -qscale:video 4
>>>> 2014-04-19_12:24.avi
>>>>
>>>> Works reasonably well. But in the man page I do not see the
>>>> parameter '-ab' described. Should I use something else for this?
>>>
>>> ab = AudioBitrate
>>> http://en.wikipedia.org/wiki/MP3
>>
>> Why is it then not in the documentation?
>
> other question: why do you set params don't know what they are?

Why do you think I do not know what it means? I just wondered why it
was not in the documentation.


> because it is disapproved, exists because of compatibility
> as well as "vb" for video-bitrate while recent versions
> prefer -b:v and -b:a, ffmpeg --help is a good start in doubt
>
> a simle google search answers that in the search results
> https://www.google.com/search?q=ffmpeg+%22-ab%22

If you look at the results, you see that -ab is used a lot there. So
if it is disapproved, your query is not a very helpful one.

Also if it is disapproved, not talking about it is not going to help.
Saying why it is disapproved and what to use instead is going to help.

By the way, I changed:
    -ab 64k
to:
    -b:a 128k

I doubled the rate also, because I saw that video took a lot more as
audio, so increasing audio bitrate does not have a big influence and
when I capture more as only voice a change is not meeded.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From Cecil at decebal.nl  Sat Apr 19 14:48:04 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 14:48:04 +0200
Subject: [FFmpeg-user] Listing devices
Message-ID: <87d2gd8ozf.fsf@Equus.Decebal.nl>

In a script I want to set the default device to the web-cam microphone
if it is connected. With:
    man ffmpeg-devices
I see:
    ffmpeg -list_devices true -f dshow -i dummy

But when executing this I get:
    ffmpeg -list_devices true -f dshow -i dummy
    ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
      built on Dec 17 2013 19:47:47 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 --enable-gnutls
      libavutil      52. 48.101 / 52. 48.101
      libavcodec     55. 39.101 / 55. 39.101
      libavformat    55. 19.104 / 55. 19.104
      libavdevice    55.  5.100 / 55.  5.100
      libavfilter     3. 90.100 /  3. 90.100
      libavresample   1.  1.  0 /  1.  1.  0
      libswscale      2.  5.101 /  2.  5.101
      libswresample   0. 17.104 /  0. 17.104
      libpostproc    52.  3.100 / 52.  3.100
    Unrecognized option 'list_devices'.
    Error splitting the argument list: Option not found

So there is an error in the documentation. Also: how do I get a list
of possible input devices?

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From cehoyos at ag.or.at  Sat Apr 19 14:59:22 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 19 Apr 2014 12:59:22 +0000 (UTC)
Subject: [FFmpeg-user] Listing devices
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140419T145832-533@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:

> ffmpeg -list_devices true -f dshow -i dummy
> ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers

This is old.

What does ffmpeg -formats|grep dshow report?

Carl Eugen


From cehoyos at ag.or.at  Sat Apr 19 15:01:42 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 19 Apr 2014 13:01:42 +0000 (UTC)
Subject: [FFmpeg-user] Listing devices
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140419T150050-900@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:

>     ffmpeg -list_devices true -f dshow -i dummy
>     ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
>       built on Dec 17 2013 19:47:47 with gcc 4.7 (SUSE Linux)

dshow is a Windows device.

Sorry for the first mail, Carl Eugen


From Cecil at decebal.nl  Sat Apr 19 15:12:39 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 15:12:39 +0200
Subject: [FFmpeg-user] Listing devices
In-Reply-To: <loom.20140419T145832-533@post.gmane.org> (Carl Eugen Hoyos's
	message of "Sat, 19 Apr 2014 12:59:22 +0000 (UTC)")
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
	<loom.20140419T145832-533@post.gmane.org>
Message-ID: <878ur18nug.fsf@Equus.Decebal.nl>

Op Saturday 19 Apr 2014 14:59 CEST schreef Carl Eugen Hoyos:

> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>
>> ffmpeg -list_devices true -f dshow -i dummy
>> ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
>
> This is old.
>
> What does ffmpeg -formats|grep dshow report?

ffmpeg -formats|grep dshow
ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
  built on Dec 17 2013 19:47:47 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 --enable-gnutls
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101
  libavformat    55. 19.104 / 55. 19.104
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From cehoyos at ag.or.at  Sat Apr 19 15:15:30 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 19 Apr 2014 13:15:30 +0000 (UTC)
Subject: [FFmpeg-user] Turning MP3 into MP4 and join with second MP4
	file?
References: <hvl4l9965gjm9gt8ucrn500qa8ehur942k@4ax.com>
Message-ID: <loom.20140419T151301-233@post.gmane.org>

Gilles <codecomplete <at> free.fr> writes:

> ffmpeg.exe -loop 1 -i speaker.png -i first.half.mp3 
> -shortest -vf scale=640:480 -c:v libx264 
> -tune stillimage -c:a copy radio.show.first.half.mp4

I suspect the output of this command will be sufficient 
to help you.

Please understand that for all questions that you ask 
on this mailing list, it is necessary that you post 
both your failing command line and the complete, uncut 
console output here on the mailing list.

report output is generally unneeded, requesting the 
download of compressed output is not welcome (as far 
as I am concerned).

Carl Eugen


From mr.artichaut at gmail.com  Sat Apr 19 15:16:31 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Sat, 19 Apr 2014 15:16:31 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <20140419113513.GA14903@phare.normalesup.org>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
Message-ID: <CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>

Amerge gives me something similar :/ So, it's

My last version is here :
https://gist.github.com/MrArtichaut/0cd9147441b51113db4d (it's a test
program so yeah, a lot of ugly things I known :) )

When opening my inputs, I also have this in my logs, but I've read that
it's not a real issue :
[wav @ 0x10280ea00] parser not found for codec pcm_s16le, packets or times
may be invalid.

Thank.


2014-04-19 13:35 GMT+02:00 Nicolas George <george at nsup.org>:

> Le decadi 30 germinal, an CCXXII, Mr Artichaut a ?crit :
> > I've done as you said but still have the same issue. By setting a DEBUG
> > level for logs and dumping the graph I have this :
> > https://gist.github.com/MrArtichaut/b4d958b36ff2fdbb64ca
>
> Without the code, there is no way of knowing what you are doing wrong.
>
> > What's is this resampler auto-inserted in the graph ?
>
> amix can only work with float samples (hat is one of the reasons I
> discourage people of using it; use amerge and pan, they support many
> formats); since your input has integer samples, resample is inserted to do
> the conversion.
>
> > Maybe my issue came from that and I have to insert one myself with my own
> > params ?
>
> No, that is completely unrelated.
>
> Regards,
>
> --
>   Nicolas George
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

From george at nsup.org  Sat Apr 19 15:54:14 2014
From: george at nsup.org (Nicolas George)
Date: Sat, 19 Apr 2014 15:54:14 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
Message-ID: <20140419135414.GA25657@phare.normalesup.org>

Le decadi 30 germinal, an CCXXII, Mr Artichaut a ?crit?:
> Amerge gives me something similar :/ So, it's

That is normal, since the conversion is not the origin of your problem. But
my guess is that with amerge and pan, you do not find the auto-inserted
resamplers.

> My last version is here :
> https://gist.github.com/MrArtichaut/0cd9147441b51113db4d (it's a test
> program so yeah, a lot of ugly things I known :) )

Your use of input_to_read is commented out. If it still does not work, I
suggest you log, at each call to av_buffersrc_write_frame(), how many
samples you provide on what input, along with the sample rate:

    "add %d samples on input %d (%d Hz, time=%f, ttime=%f)",
      	frame->nb_samples, i, codec[i]->sample_rate,
	(double)frame->nb_samples / codec[i]->sample_rate,
	(double)(total_samples[i] += frame->nb_samples) / codec[i]->sample_rate

and do the same when getting frames from the sink. The basic rule is that
your program should be adding roughly the same duration of data on each
input.

> When opening my inputs, I also have this in my logs, but I've read that
> it's not a real issue :
> [wav @ 0x10280ea00] parser not found for codec pcm_s16le, packets or times
> may be invalid.

Probably not a problem, PCM does not require a parser.

> 2014-04-19 13:35 GMT+02:00 Nicolas George <george at nsup.org>:

Please remember not to top-post.

Regards,

-- 
  Nicolas George
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From Cecil at decebal.nl  Sat Apr 19 16:19:59 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 16:19:59 +0200
Subject: [FFmpeg-user] Listing devices
In-Reply-To: <loom.20140419T150050-900@post.gmane.org> (Carl Eugen Hoyos's
	message of "Sat, 19 Apr 2014 13:01:42 +0000 (UTC)")
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
	<loom.20140419T150050-900@post.gmane.org>
Message-ID: <874n1p8kq8.fsf@Equus.Decebal.nl>

Op Saturday 19 Apr 2014 15:01 CEST schreef Carl Eugen Hoyos:

> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>
>> ffmpeg -list_devices true -f dshow -i dummy
>> ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers
>> built on Dec 17 2013 19:47:47 with gcc 4.7 (SUSE Linux)
>
> dshow is a Windows device.

But even with that removed, I get the same message.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From cehoyos at ag.or.at  Sat Apr 19 16:23:56 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 19 Apr 2014 14:23:56 +0000 (UTC)
Subject: [FFmpeg-user] Listing devices
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
	<loom.20140419T150050-900@post.gmane.org>
	<874n1p8kq8.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140419T162314-486@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:

> >> ffmpeg -list_devices true -f dshow -i dummy

> > dshow is a Windows device.
> 
> But even with that removed, I get the same message.

Because the option only makes sense for dshow (and openal).

Carl Eugen


From mr.artichaut at gmail.com  Sat Apr 19 16:27:45 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Sat, 19 Apr 2014 16:27:45 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <20140419135414.GA25657@phare.normalesup.org>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
	<20140419135414.GA25657@phare.normalesup.org>
Message-ID: <CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>

The numbers seems good :
add 1024 samples on input 0 (44100 Hz, time=0.023220, ttime=5.410249)
add 1024 samples on input 1 (44100 Hz, time=0.023220, ttime=5.410249)
remove 1024 samples from sink (44100 Hz, time=0.023220, ttime=5.410249)

The issue is that, when my input0, which is the shortest, is finished, I
don't pull any data from the sink anymore. Every av_buffersink_get_frame
failed telling me that it needs data from input0. If I continue to write
data from input 1 it doesn't use it and instead keeps asking for data from
input 0.

My amix filter is configured with "duration=longest" but it's the same
thing with shortest.

From george at nsup.org  Sat Apr 19 16:54:48 2014
From: george at nsup.org (Nicolas George)
Date: Sat, 19 Apr 2014 16:54:48 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
	<20140419135414.GA25657@phare.normalesup.org>
	<CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>
Message-ID: <20140419145448.GA15349@phare.normalesup.org>

Le decadi 30 germinal, an CCXXII, Mr Artichaut a ?crit?:
> The issue is that, when my input0, which is the shortest, is finished, I
> don't pull any data from the sink anymore. Every av_buffersink_get_frame
> failed telling me that it needs data from input0. If I continue to write
> data from input 1 it doesn't use it and instead keeps asking for data from
> input 0.

If the buffer problem happens at the end, that probably means you forgot to
send the EOF null frame.

Regards,

-- 
  Nicolas George
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From mr.artichaut at gmail.com  Sat Apr 19 17:22:03 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Sat, 19 Apr 2014 17:22:03 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <20140419145448.GA15349@phare.normalesup.org>
References: <CAMu2HxchffOpD-UavZWDYnazRRWMaBd6-PCUbD0_tP4qcOHp=A@mail.gmail.com>
	<20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
	<20140419135414.GA25657@phare.normalesup.org>
	<CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>
	<20140419145448.GA15349@phare.normalesup.org>
Message-ID: <CAMu2HxdLpq32wzSxifqhPkiD8ExHgu3x8Cv3+bwiJRwDmq2R8A@mail.gmail.com>

ffmpeg is full of suprise :) I've never seen this "EOF null frame" in the
examples.

Ok, so now it doesn't ask for data from input0 forever, that's good. But my
issue is still there. :/  It must be when I initialize the codecs and
formats ?

I've updated my code :
https://gist.github.com/MrArtichaut/0cd9147441b51113db4d

Thank you very much for your time. If one day I success in mixing those
files, I think I will post it somewhere for the generations to come :)

From Cecil at decebal.nl  Sat Apr 19 18:11:25 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 18:11:25 +0200
Subject: [FFmpeg-user] Listing devices
In-Reply-To: <loom.20140419T162314-486@post.gmane.org> (Carl Eugen Hoyos's
	message of "Sat, 19 Apr 2014 14:23:56 +0000 (UTC)")
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
	<loom.20140419T150050-900@post.gmane.org>
	<874n1p8kq8.fsf@Equus.Decebal.nl>
	<loom.20140419T162314-486@post.gmane.org>
Message-ID: <87zjjh7102.fsf@Equus.Decebal.nl>

Op Saturday 19 Apr 2014 16:23 CEST schreef Carl Eugen Hoyos:

> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>
>>>> ffmpeg -list_devices true -f dshow -i dummy
>
>>> dshow is a Windows device.
>>
>> But even with that removed, I get the same message.
>
> Because the option only makes sense for dshow (and openal).

OK, what is then the way to get the devices in Linux?

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From codecomplete at free.fr  Sat Apr 19 18:27:31 2014
From: codecomplete at free.fr (Gilles)
Date: Sat, 19 Apr 2014 18:27:31 +0200
Subject: [FFmpeg-user] Turning MP3 into MP4 and join with second MP4
	file?
References: <hvl4l9965gjm9gt8ucrn500qa8ehur942k@4ax.com>
	<loom.20140419T151301-233@post.gmane.org>
Message-ID: <8u85l9p30jfsdmatsq272f55gm16n286ek@4ax.com>

On Sat, 19 Apr 2014 13:15:30 +0000 (UTC), Carl Eugen Hoyos
<cehoyos at ag.or.at> wrote:
>report output is generally unneeded, requesting the 
>download of compressed output is not welcome (as far 
>as I am concerned).

Didn't you ask for the output?

CEH > I don't remember a report where MediaInfo was of any 
help but the complete, uncut console output for 
your failing FFmpeg command line is needed if you 
want support here.


From george at nsup.org  Sat Apr 19 18:35:35 2014
From: george at nsup.org (Nicolas George)
Date: Sat, 19 Apr 2014 18:35:35 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <CAMu2HxdLpq32wzSxifqhPkiD8ExHgu3x8Cv3+bwiJRwDmq2R8A@mail.gmail.com>
References: <20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
	<20140419135414.GA25657@phare.normalesup.org>
	<CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>
	<20140419145448.GA15349@phare.normalesup.org>
	<CAMu2HxdLpq32wzSxifqhPkiD8ExHgu3x8Cv3+bwiJRwDmq2R8A@mail.gmail.com>
Message-ID: <20140419163535.GA21238@phare.normalesup.org>

Le decadi 30 germinal, an CCXXII, Mr Artichaut a ?crit?:
> ffmpeg is full of suprise :) I've never seen this "EOF null frame" in the
> examples.

I checked and this is true. The examples are bogus. Thanks for helping to
notice it.

> Ok, so now it doesn't ask for data from input0 forever, that's good. But my
> issue is still there. :/  It must be when I initialize the codecs and
> formats ?

I do not know. Can you show the console log produced by your program (you
can trim when there are a lot of similar lines, leaving only a few at the
beginning and the end)?

Regards,

-- 
  Nicolas George
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From mr.artichaut at gmail.com  Sat Apr 19 18:46:28 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Sat, 19 Apr 2014 18:46:28 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <20140419163535.GA21238@phare.normalesup.org>
References: <20140412222631.GA31402@phare.normalesup.org>
	<CAMu2HxfX1tR3cuW6Jt_FQfaPc0uPEYYVM_sFhY5NGLTwk47S0g@mail.gmail.com>
	<20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
	<20140419135414.GA25657@phare.normalesup.org>
	<CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>
	<20140419145448.GA15349@phare.normalesup.org>
	<CAMu2HxdLpq32wzSxifqhPkiD8ExHgu3x8Cv3+bwiJRwDmq2R8A@mail.gmail.com>
	<20140419163535.GA21238@phare.normalesup.org>
Message-ID: <CAMu2Hxc3cDkw2T-Y-zOf=u6otJ4KEGzqNEhhNBdSPX6d+e=rTw@mail.gmail.com>

Here is all I have :
https://gist.github.com/MrArtichaut/7e9c8c3cffec36030137

From Cecil at decebal.nl  Sat Apr 19 22:20:20 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sat, 19 Apr 2014 22:20:20 +0200
Subject: [FFmpeg-user] Listing devices
In-Reply-To: <87zjjh7102.fsf@Equus.Decebal.nl> (Cecil Westerhof's message of
	"Sat, 19 Apr 2014 18:11:25 +0200")
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
	<loom.20140419T150050-900@post.gmane.org>
	<874n1p8kq8.fsf@Equus.Decebal.nl>
	<loom.20140419T162314-486@post.gmane.org>
	<87zjjh7102.fsf@Equus.Decebal.nl>
Message-ID: <87vbu56ph7.fsf@Equus.Decebal.nl>

Op Saturday 19 Apr 2014 18:11 CEST schreef Cecil Westerhof:

> Op Saturday 19 Apr 2014 16:23 CEST schreef Carl Eugen Hoyos:
>
>> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>>
>>>>> ffmpeg -list_devices true -f dshow -i dummy
>>
>>>> dshow is a Windows device.
>>>
>>> But even with that removed, I get the same message.
>>
>> Because the option only makes sense for dshow (and openal).
>
> OK, what is then the way to get the devices in Linux?

At the moment I use the following:
    command='
    ffmpeg -y
           -async 1
           -f alsa
           -ac 1
           -i '"${MIC_WEBCAM}"'
           -acodec libmp3lame
           -ar 22050
           -b:a 128k
           -t 1 dummy.mp3'
    if ${command} 2>/dev/null ; then
        micToUse="${MIC_WEBCAM}"
    else
        micToUse="${MIC_INTERN}"
    fi

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From behnam.heydarshahi at gmail.com  Sat Apr 19 08:31:32 2014
From: behnam.heydarshahi at gmail.com (Behnam Heydarshahi)
Date: Sat, 19 Apr 2014 11:01:32 +0430
Subject: [FFmpeg-user] Encoding a Multi-Rate Container for Helix Server
Message-ID: <CAGKGyfmrnevc98n=uyrKxt7Hj752LcHqMHAKNDzs8uU=_cF6Fg@mail.gmail.com>

I want to Encode a Multi-Rate mp4 container for helix server. Though they
are multi-rate in MediaInfo application, Helix server doesn't recognize and
treat my encoded movies as Multi-Rate.

Has anybody had any experience with encoding Multi-Rate for Helix Server?
If so, can you please suggest me an FFmpeg command?

From linuxrojas at gmail.com  Sat Apr 19 22:31:17 2014
From: linuxrojas at gmail.com (Frank R)
Date: Sat, 19 Apr 2014 14:31:17 -0600
Subject: [FFmpeg-user] configure err with cygwin
Message-ID: <CAEaup1kxxVrMaFXnH1C92Jiy0aX3MTrLx0WKkYPO4hH-N2tF+g@mail.gmail.com>

I am gettng error with configure script and libaacplus on cygwin

sh-4.1$ ./configure --enable-gpl --enable-libx264 --enable-libmp3lame
--enable-nonfree --enable-libaacplus
ERROR: libaacplus >= 2.0.0 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.


config.log -- issues

BEGIN /tmp/ffconf.kRIFjT2g.m
    1   #include <AVFoundation/AVFoundation.h>
    2   int main(void) { return 0; }
END /tmp/ffconf.kRIFjT2g.m
gcc -Werror=missing-prototypes -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64
-D_LARGEFILE_SOURCE -U__STRICT_ANSI__ -std=c99 -fomit-frame-pointer
-pthread -c -o /tmp/ffconf.GKihfmky.o /tmp/ffconf.kRIFjT2g.m
gcc: error: spawn: No such file or directory
check_lib aacplus.h aacplusEncOpen -laacplus
check_header aacplus.h
check_cpp
BEGIN /tmp/ffconf.BEilMTlr.c
    1   #include <aacplus.h>
    2   int x;
END /tmp/ffconf.BEilMTlr.c
gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE
-U__STRICT_ANSI__ -std=c99 -fomit-frame-pointer -pthread -E -o
/tmp/ffconf.GKihfmky.o /tmp/ffconf.BEilMTlr.c
check_func aacplusEncOpen -laacplus
check_ld cc -laacplus
check_cc
BEGIN /tmp/ffconf.BEilMTlr.c
    1   extern int aacplusEncOpen();
    2   int main(void){ aacplusEncOpen(); }
END /tmp/ffconf.BEilMTlr.c
gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE
-U__STRICT_ANSI__ -std=c99 -fomit-frame-pointer -pthread -c -o
/tmp/ffconf.GKihfmky.o /tmp/ffconf.BEilMTlr.c
gcc -Wl,--as-needed -o /tmp/ffconf.aUjwgq7T.exe /tmp/ffconf.GKihfmky.o
-laacplus -lm -lbz2 -lz -pthread -lpsapi -ladvapi32 -lshell32
/usr/lib/gcc/i686-pc-cygwin/4.8.2/../../../../i686-pc-cygwin/bin/ld:
cannot find -laacplus
collect2: error: ld returned 1 exit status
ERROR: libaacplus >= 2.0.0 not found

From anshul.ffmpeg at gmail.com  Sat Apr 19 23:06:48 2014
From: anshul.ffmpeg at gmail.com (Anshul)
Date: Sun, 20 Apr 2014 02:36:48 +0530
Subject: [FFmpeg-user] configure err with cygwin
In-Reply-To: <CAEaup1kxxVrMaFXnH1C92Jiy0aX3MTrLx0WKkYPO4hH-N2tF+g@mail.gmail.com>
References: <CAEaup1kxxVrMaFXnH1C92Jiy0aX3MTrLx0WKkYPO4hH-N2tF+g@mail.gmail.com>
Message-ID: <4786780e-535e-44dd-a7a9-6815049709ff@email.android.com>

Frank R <linuxrojas at gmail.com> wrote:
>I am gettng error with configure script and libaacplus on cygwin
>
>sh-4.1$ ./configure --enable-gpl --enable-libx264 --enable-libmp3lame
>--enable-nonfree --enable-libaacplus
>ERROR: libaacplus >= 2.0.0 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.
>
>
>config.log -- issues
>
>BEGIN /tmp/ffconf.kRIFjT2g.m
>    1   #include <AVFoundation/AVFoundation.h>
>    2   int main(void) { return 0; }
>END /tmp/ffconf.kRIFjT2g.m
>gcc -Werror=missing-prototypes -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64
>-D_LARGEFILE_SOURCE -U__STRICT_ANSI__ -std=c99 -fomit-frame-pointer
>-pthread -c -o /tmp/ffconf.GKihfmky.o /tmp/ffconf.kRIFjT2g.m
>gcc: error: spawn: No such file or directory
>check_lib aacplus.h aacplusEncOpen -laacplus
>check_header aacplus.h
>check_cpp
>BEGIN /tmp/ffconf.BEilMTlr.c
>    1   #include <aacplus.h>
>    2   int x;
>END /tmp/ffconf.BEilMTlr.c
>gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE
>-U__STRICT_ANSI__ -std=c99 -fomit-frame-pointer -pthread -E -o
>/tmp/ffconf.GKihfmky.o /tmp/ffconf.BEilMTlr.c
>check_func aacplusEncOpen -laacplus
>check_ld cc -laacplus
>check_cc
>BEGIN /tmp/ffconf.BEilMTlr.c
>    1   extern int aacplusEncOpen();
>    2   int main(void){ aacplusEncOpen(); }
>END /tmp/ffconf.BEilMTlr.c
>gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE
>-U__STRICT_ANSI__ -std=c99 -fomit-frame-pointer -pthread -c -o
>/tmp/ffconf.GKihfmky.o /tmp/ffconf.BEilMTlr.c
>gcc -Wl,--as-needed -o /tmp/ffconf.aUjwgq7T.exe /tmp/ffconf.GKihfmky.o
>-laacplus -lm -lbz2 -lz -pthread -lpsapi -ladvapi32 -lshell32
>/usr/lib/gcc/i686-pc-cygwin/4.8.2/../../../../i686-pc-cygwin/bin/ld:
>cannot find -laacplus
>collect2: error: ld returned 1 exit status
>ERROR: libaacplus >= 2.0.0 not found
>_______________________________________________
>ffmpeg-user mailing list
>ffmpeg-user at ffmpeg.org
>http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Isnt it clear that you need to install libaacplus. Give path of libaacplus or keep that library where it can be found by cygbin.
-- 
Sent from my Android device with K-9 Mail. Please excuse my brevity.

From dashing.meng at gmail.com  Sun Apr 20 03:52:22 2014
From: dashing.meng at gmail.com (littlebat)
Date: Sun, 20 Apr 2014 09:52:22 +0800
Subject: [FFmpeg-user] Removing audio noise
In-Reply-To: <87tx9p8vhy.fsf@Equus.Decebal.nl>
References: <87tx9p8vhy.fsf@Equus.Decebal.nl>
Message-ID: <20140420095222.200426a7f8d80d284728e170@gmail.com>

On Sat, 19 Apr 2014 12:27:21 +0200
Cecil Westerhof <Cecil at decebal.nl> wrote:

> At the moment I am using:
> ffmpeg -async 1
>        -f alsa
>        -ac 1
>        -i hw:1,0,0
>        -f x11grab -s 1920x1080
>        -r 25
>        -i :0.0
>        -acodec libmp3lame
>        -ar 22050
>        -ab 64k
>        -qscale:video 4
>        2014-04-19_12:24.avi
> 
> This works reasonably good. The only problem is that there is some
> white noise in the audio. Is there a way to filter this away?
> 

Maybe this article will help:
How To Do Noise Reduction Using ffmpeg And sox:
http://www.zoharbabin.com/how-to-do-noise-reduction-using-ffmpeg-and-sox/

From lou at lrcd.com  Sun Apr 20 08:15:04 2014
From: lou at lrcd.com (Lou)
Date: Sat, 19 Apr 2014 22:15:04 -0800
Subject: [FFmpeg-user] Removing audio noise
In-Reply-To: <87tx9p8vhy.fsf@Equus.Decebal.nl>
References: <87tx9p8vhy.fsf@Equus.Decebal.nl>
Message-ID: <1397974504.28045.108316677.3DF4ACD9@webmail.messagingengine.com>

On Sat, Apr 19, 2014, at 02:27 AM, Cecil Westerhof wrote:
> The only problem is that there is some white noise in
> the audio. Is there a way to filter this away?

Do you know the cause of the white noise?
Does the white noise also occur in arecord?

From Cecil at decebal.nl  Sun Apr 20 10:06:06 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sun, 20 Apr 2014 10:06:06 +0200
Subject: [FFmpeg-user] Removing audio noise
In-Reply-To: <20140420095222.200426a7f8d80d284728e170@gmail.com> (littlebat's
	message of "Sun, 20 Apr 2014 09:52:22 +0800")
References: <87tx9p8vhy.fsf@Equus.Decebal.nl>
	<20140420095222.200426a7f8d80d284728e170@gmail.com>
Message-ID: <87r44s77dd.fsf@Equus.Decebal.nl>

Op Sunday 20 Apr 2014 03:52 CEST schreef littlebat:

> On Sat, 19 Apr 2014 12:27:21 +0200
> Cecil Westerhof <Cecil at decebal.nl> wrote:
>
>> At the moment I am using:
>> ffmpeg -async 1
>> -f alsa
>> -ac 1
>> -i hw:1,0,0
>> -f x11grab -s 1920x1080
>> -r 25
>> -i :0.0
>> -acodec libmp3lame
>> -ar 22050
>> -ab 64k
>> -qscale:video 4
>> 2014-04-19_12:24.avi
>>
>> This works reasonably good. The only problem is that there is some
>> white noise in the audio. Is there a way to filter this away?
>>
>
> Maybe this article will help: How To Do Noise Reduction Using ffmpeg
> And sox:
> http://www.zoharbabin.com/how-to-do-noise-reduction-using-ffmpeg-and-sox/

Certainly did. Had to remove the sameq of-course. ;-)

I made the following function (fatal and showMessage are functions I
defined):
    function removeWhiteNoise {
      if [[ ${#} -ne 2 ]] ; then
        fatal "${FUNCNAME} <INPUT> <OUTPUT>"
        return
      fi

      local -r REM_WHITE_NOISE_INPUT_FILE="${1}"; shift
      local -r REM_WHITE_NOISE_OUTPUT_FILE="${1}"; shift

      showMessage "${DEBUG}" "Create video only"
      ffmpeg                                    \
          -y                                    \
          -i "${REM_WHITE_NOISE_INPUT_FILE}"    \
          -qscale:video 0 -an                   \
          tmpvid.mp4 2>/dev/null
      showMessage "${DEBUG}" "Create audio only"
      ffmpeg                                    \
          -y                                    \
          -i "${REM_WHITE_NOISE_INPUT_FILE}"    \
          -qscale:audio 0                       \
          tmpaud.wav 2>/dev/null
      showMessage "${DEBUG}" "Create noise file"
      ffmpeg                                    \
          -y                                    \
          -i "${REM_WHITE_NOISE_INPUT_FILE}"    \
          -vn -ss 00:00:00 -t 00:00:01          \
          noiseaud.wav 2>/dev/null
      showMessage "${DEBUG}" "Create noise profile"
      sox noiseaud.wav -n noiseprof noise.prof
      showMessage "${DEBUG}" "Create cleaned audio"
      sox tmpaud.wav tmpaud-clean.wav noisered noise.prof 0.21
      showMessage "${DEBUG}" "Create cleaned file"
      ffmpeg                    \
          -y                                    \
          -i tmpaud-clean.wav   \
          -i tmpvid.mp4         \
          -qscale:video 0       \
          -qscale:audio 0       \
          "${REM_WHITE_NOISE_OUTPUT_FILE}" 2>/dev/null
      showMessage "${DEBUG}" "Remove temp files"
      rm tmpvid.mp4 tmpaud.wav noiseaud.wav noise.prof tmpaud-clean.wav
      showMessage "${DEBUG}" "Ready"
    }

This works reasonably good. It sounds a little bit like you are
talking into a tin, but I am not using equipment that costs thousands
of dollars, so I should accept it I think.

It take a reasonable amount of time. For a seven minute audio I get
the following output:
    09:47:14: Going to remove white-noise
    09:47:14: Create video only
    09:48:34: Create audio only
    09:48:35: Create noise file
    09:48:35: Create noise profile
    09:48:35: Create cleaned audio
    sox WARN noisered: noisered clipped 23 samples; decrease volume?
    sox WARN dither: dither clipped 21 samples; decrease volume?
    09:48:41: Create cleaned file
    09:49:24: Remove temp files
    09:49:24: Ready
    09:49:24: Ready removing white-noise

So it takes about a third of the length of the video. I have to play a
little bit more, becuase of the two warnings I get. But I am already a
far way on the right track.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From Cecil at decebal.nl  Sun Apr 20 10:18:45 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sun, 20 Apr 2014 10:18:45 +0200
Subject: [FFmpeg-user] Removing audio noise
In-Reply-To: <1397974504.28045.108316677.3DF4ACD9@webmail.messagingengine.com>
	(lou@lrcd.com's message of "Sat, 19 Apr 2014 22:15:04 -0800")
References: <87tx9p8vhy.fsf@Equus.Decebal.nl>
	<1397974504.28045.108316677.3DF4ACD9@webmail.messagingengine.com>
Message-ID: <87mwfg76sa.fsf@Equus.Decebal.nl>

Op Sunday 20 Apr 2014 08:15 CEST schreef lou at lrcd.com:

> On Sat, Apr 19, 2014, at 02:27 AM, Cecil Westerhof wrote:
>> The only problem is that there is some white noise in
>> the audio. Is there a way to filter this away?
>
> Do you know the cause of the white noise?

No.


> Does the white noise also occur in arecord?

Even more.

But problem is solved with the tip of littlebat.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From Cecil at decebal.nl  Sun Apr 20 10:22:29 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sun, 20 Apr 2014 10:22:29 +0200
Subject: [FFmpeg-user] Removing audio noise
In-Reply-To: <87r44s77dd.fsf@Equus.Decebal.nl> (Cecil Westerhof's message of
	"Sun, 20 Apr 2014 10:06:06 +0200")
References: <87tx9p8vhy.fsf@Equus.Decebal.nl>
	<20140420095222.200426a7f8d80d284728e170@gmail.com>
	<87r44s77dd.fsf@Equus.Decebal.nl>
Message-ID: <87ioq476m2.fsf@Equus.Decebal.nl>

Op Sunday 20 Apr 2014 10:06 CEST schreef Cecil Westerhof:

>> Maybe this article will help: How To Do Noise Reduction Using
>> ffmpeg And sox:
>> http://www.zoharbabin.com/how-to-do-noise-reduction-using-ffmpeg-and-sox/
>
> Certainly did. Had to remove the sameq of-course. ;-)

Even better: I had a problem when fetching video for a screen-cast with
the same web-cam as I fetched audio with. That problem is also solved.
It is a little work to figure out, but with command line utilities
there is a lot possible.

Now I have to find something against keyboard clicks. But that is not
as nearly as important as this problem was.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From Cecil at decebal.nl  Sun Apr 20 10:34:04 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sun, 20 Apr 2014 10:34:04 +0200
Subject: [FFmpeg-user] Distorted audio in screencast
In-Reply-To: <loom.20140418T104819-308@post.gmane.org> (Carl Eugen Hoyos's
	message of "Fri, 18 Apr 2014 08:49:34 +0000 (UTC)")
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
	<loom.20140418T104819-308@post.gmane.org>
Message-ID: <87eh0s762r.fsf@Equus.Decebal.nl>

Op Friday 18 Apr 2014 10:49 CEST schreef Carl Eugen Hoyos:

> Cecil Westerhof <Cecil <at> decebal.nl> writes:
>
>> I use this because I want to integrate some graphics 
>> in my video.
>
> Did you test the overlay filter?

I did not. I have looked at it and that could be very use-full.
Especially when combined with the white-noise filter I have now.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From Cecil at decebal.nl  Sun Apr 20 10:37:09 2014
From: Cecil at decebal.nl (Cecil Westerhof)
Date: Sun, 20 Apr 2014 10:37:09 +0200
Subject: [FFmpeg-user] Distorted audio in screencast
In-Reply-To: <871twu9y5w.fsf@Equus.Decebal.nl> (Cecil Westerhof's message of
	"Fri, 18 Apr 2014 22:32:11 +0200")
References: <877g6ndf9p.fsf@Equus.Decebal.nl>
	<89F8E4011FD7234ABFE48744E300CBC206C2BDEC@041-DB3MPN1-091.041d.mgd.msft.net>
	<87d2gfb18m.fsf@Equus.Decebal.nl>
	<loom.20140418T104819-308@post.gmane.org>
	<87ioq6anzy.fsf@Equus.Decebal.nl>
	<loom.20140418T132226-240@post.gmane.org>
	<8761m6aazz.fsf@Equus.Decebal.nl>
	<loom.20140418T184343-504@post.gmane.org>
	<871twu9y5w.fsf@Equus.Decebal.nl>
Message-ID: <87a9bg75xm.fsf@Equus.Decebal.nl>

Op Friday 18 Apr 2014 22:32 CEST schreef Cecil Westerhof:

> it works, but I still have the distorted sound. So it still does not
> work satisfactory. So I just use one web-cam for the video and the
> other for audio. Not optimal, but it works.

Not necessary anymore: with the function I made for white-noise
reduction, the distorted sound is also removed.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof

From 550532914 at qq.com  Sun Apr 20 12:05:52 2014
From: 550532914 at qq.com (=?utf-8?B?R29vc2U=?=)
Date: Sun, 20 Apr 2014 18:05:52 +0800
Subject: [FFmpeg-user] block issue ,help!help! config err with libx264.
Message-ID: <tencent_6E5C0344180D9B762B4B49A5@qq.com>

firstly, thank your reading and nicely help.
I ve suffered by this block issue for several days.


the config.log is in the attachment.


I? intall ffmpeg on picuntu env, which is 

PicUntu, based on Ubuntu Qantal 12.10, The PicUntu project team announced the release of an Ubuntu-based Linux distribution, specifically developed to run on RK3066 chipset devices. According to the PicUntu team, it can run a company web server, corporate email server, central database server, or content management portal.More info here
Mirror: LinuxUpgradeTool&RKBatchTool-marsboard-HDMI-picuntu-0.9-RC2.2-lubuntu-desktop-v1.2.img.7z






pls give a hand!


thanks
-------------- next part --------------
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From dopelabs at dubstep.fm  Sun Apr 20 12:54:41 2014
From: dopelabs at dubstep.fm (DopeLabs)
Date: Sun, 20 Apr 2014 03:54:41 -0700
Subject: [FFmpeg-user] force/replace colors of waveform (showwaves)? (and
	60fps at 720p)?
Message-ID: <28A1689D-E2EE-485C-B190-EFD1DCF45782@dubstep.fm>

working command im using is below, generates a nice output like

http://dsfm.tv/images/showwaves.png

white and grey seem to be the 2 colors generated for the waveform. 

is there any way to change these colors?

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

also... when -r 60 is specified before -filter_complex, the output is 60 fps, but showwaves still only outputs at 34fps (@720p)(even if r=60 is specified in the filter_complex), but if i change to showwaves=s=720x405, showwaves outputs a proper 60fps... 

not sure at what point showwaves decides to not try for 60fps anymore, but id like to avoid generating a smaller sized true 60fps file, and then scaling the rest of the way to reach my target of 720p... any ideas?

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

console stuffs:



ffmpeg -i http://domiain.com:8000 -filter_complex showwaves=s=1280x720:mode=line -c:v libx264 -preset superfast -b:v 4096k -pix_fmt yuv420p  -c:a aac -strict -2 -b:a 256k -f mp4 outies.mp4


user at BCD2:/# ffmpeg -i http://domain.com:8000 -filter_complex showwaves=s=1280x720:mode=line -c:v libx264 -preset superfast -b:v 4096k -pix_fmt yuv420p -c:a aac -strict -2 -b:a 256k -f mp4 outies.mp4
ffmpeg version N-61744-g0dd4eff Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar 23 2014 19:59:06 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  configuration: --prefix=/usr/local --bindir=/usr/local/bin --datadir=/usr/local/share/ffmpeg --docdir=/usr/local/share/doc/ffmpeg --libdir=/usr/local/lib --shlibdir=/usr/local/lib --incdir=/usr/local/include --mandir=/usr/local/share/man --enable-version3 --enable-x11grab --enable-opengl --enable-opencl --enable-openal --enable-libxvid --enable-libx264 --enable-libwavpack --enable-libvpx --enable-libvorbis --enable-libvo-aacenc --enable-libtheora --enable-libssh --enable-librtmp --enable-libopenjpeg --enable-libmp3lame --enable-libfdk-aac --enable-libfaac --enable-libass --enable-vda --enable-avresample --enable-nonfree --enable-gpl
  libavutil      52. 67.100 / 52. 67.100
  libavcodec     55. 52.103 / 55. 52.103
  libavformat    55. 34.101 / 55. 34.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libavresample   1.  2.  0 /  1.  2.  0
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
[mp3 @ 0x20d8a40] Header missing
Input #0, mp3, from 'http://domain.com:8000':
  Duration: N/A, start: 0.000000, bitrate: 256 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 256 kb/s
[libx264 @ 0x20e5a80] using SAR=1/1
[libx264 @ 0x20e5a80] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
[libx264 @ 0x20e5a80] profile High, level 3.2
[libx264 @ 0x20e5a80] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x3 me=dia subme=1 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 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=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc=abr mbtree=0 bitrate=4096 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 pb_ratio=1.30 aq=1:1.00
Output #0, mp4, to 'outies.mp4':
  Metadata:
    encoder         : Lavf55.34.101
    Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 4096 kb/s, 17640 tbn, 34.45 tbc
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 256 kb/s
Stream mapping:
  Stream #0:0 (mp3) -> showwaves (graph 0)
  showwaves (graph 0) -> Stream #0:0 (libx264)
  Stream #0:0 -> #0:1 (mp3 -> aac)
Press [q] to stop, [?] for help
[mp3 @ 0x20d8a40] Header missing
Error while decoding stream #0:0: Invalid data found when processing input
frame=  605 fps= 63 q=-1.0 Lsize=    8297kB time=00:00:17.60 bitrate=3861.3kbits/s    
video:7728kB audio:554kB subtitle:0 data:0 global headers:0kB muxing overhead 0.191773%
[libx264 @ 0x20e5a80] frame I:13    Avg QP:17.73  size: 15171
[libx264 @ 0x20e5a80] frame P:532   Avg QP:24.46  size: 13691
[libx264 @ 0x20e5a80] frame B:60    Avg QP:23.98  size:  7201
[libx264 @ 0x20e5a80] consecutive B-frames: 80.2% 19.8%  0.0%  0.0%
[libx264 @ 0x20e5a80] mb I  I16..4: 89.2%  0.3% 10.4%
[libx264 @ 0x20e5a80] mb P  I16..4:  9.4%  0.1%  5.6%  P16..4:  3.7%  0.0%  0.0%  0.0%  0.0%    skip:81.1%
[libx264 @ 0x20e5a80] mb B  I16..4:  0.4%  0.0%  0.5%  B16..8:  4.7%  0.0%  0.0%  direct: 0.2%  skip:94.1%  L0:46.2% L1:48.6% BI: 5.2%
[libx264 @ 0x20e5a80] final ratefactor: 21.61
[libx264 @ 0x20e5a80] 8x8 transform intra:0.9% inter:9.5%
[libx264 @ 0x20e5a80] coded y,uvDC,uvAC intra: 27.2% 0.0% 0.0% inter: 2.8% 0.0% 0.0%
[libx264 @ 0x20e5a80] i16 v,h,dc,p: 88% 11%  1%  0%
[libx264 @ 0x20e5a80] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 13% 27%  9%  3%  4%  2%  2% 17%
[libx264 @ 0x20e5a80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 68%  8% 13%  2%  1%  3%  1%  3%  2%
[libx264 @ 0x20e5a80] i8c dc,h,v,p: 100%  0%  0%  0%
[libx264 @ 0x20e5a80] Weighted P-Frames: Y:0.4% UV:0.0%
[libx264 @ 0x20e5a80] kb/s:3604.90

From george at nsup.org  Sun Apr 20 13:57:26 2014
From: george at nsup.org (Nicolas George)
Date: Sun, 20 Apr 2014 13:57:26 +0200
Subject: [FFmpeg-user] force/replace colors of waveform (showwaves)?
 (and 60fps at 720p)?
In-Reply-To: <28A1689D-E2EE-485C-B190-EFD1DCF45782@dubstep.fm>
References: <28A1689D-E2EE-485C-B190-EFD1DCF45782@dubstep.fm>
Message-ID: <20140420115726.GA30793@phare.normalesup.org>

Le primidi 1er flor?al, an CCXXII, DopeLabs a ?crit?:
> also... when -r 60 is specified before -filter_complex, the output is 60
> fps, but showwaves still only outputs at 34fps (@720p)(even if r=60 is
> specified in the filter_complex), but if i change to showwaves=s=720x405,
> showwaves outputs a proper 60fps... 
> 
> not sure at what point showwaves decides to not try for 60fps anymore, but
> id like to avoid generating a smaller sized true 60fps file, and then
> scaling the rest of the way to reach my target of 720p... any ideas?

Your input has 44100 samples per second, showwaves plots 1280 samples per
frame, that makes 44100/1280. = 34.453125 frames per second.

With the 720 resolution, you get 61.25, not 60.

Regards,

-- 
  Nicolas George
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From cehoyos at ag.or.at  Sun Apr 20 16:50:49 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 20 Apr 2014 14:50:49 +0000 (UTC)
Subject: [FFmpeg-user] Listing devices
References: <87d2gd8ozf.fsf@Equus.Decebal.nl>
	<loom.20140419T150050-900@post.gmane.org>
	<874n1p8kq8.fsf@Equus.Decebal.nl>
	<loom.20140419T162314-486@post.gmane.org>
	<87zjjh7102.fsf@Equus.Decebal.nl> <87vbu56ph7.fsf@Equus.Decebal.nl>
Message-ID: <loom.20140420T164639-641@post.gmane.org>

Cecil Westerhof <Cecil <at> decebal.nl> writes:

> > OK, what is then the way to get the devices in 
> > Linux?

$ ffmpeg -formats
lists the demuxers, the muxers, the input and the 
output devices.

> At the moment I use the following:
>     command='
>     ffmpeg -y
>            -async 1
>            -f alsa

Or do you mean "how to list alsa devices"?
$ aplay -l
lists the alsa output devices
$ arecord -l
lists the alsa input devices.

There is of course a semantic difference between the 
FFmpeg devices (for example "alsa") and the alsa 
devices (like "card 1" "device 7" which translates 
to "hw:1,7" in FFmpeg's alsa notation).

Carl Eugen


From george at nsup.org  Sun Apr 20 20:22:06 2014
From: george at nsup.org (Nicolas George)
Date: Sun, 20 Apr 2014 20:22:06 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <CAMu2Hxc3cDkw2T-Y-zOf=u6otJ4KEGzqNEhhNBdSPX6d+e=rTw@mail.gmail.com>
References: <20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
	<20140419135414.GA25657@phare.normalesup.org>
	<CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>
	<20140419145448.GA15349@phare.normalesup.org>
	<CAMu2HxdLpq32wzSxifqhPkiD8ExHgu3x8Cv3+bwiJRwDmq2R8A@mail.gmail.com>
	<20140419163535.GA21238@phare.normalesup.org>
	<CAMu2Hxc3cDkw2T-Y-zOf=u6otJ4KEGzqNEhhNBdSPX6d+e=rTw@mail.gmail.com>
Message-ID: <20140420182206.GA1508@phare.normalesup.org>

Le decadi 30 germinal, an CCXXII, Mr Artichaut a ?crit?:
> Here is all I have :
> https://gist.github.com/MrArtichaut/7e9c8c3cffec36030137

Ok, so your buffer overflow problem is gone, but the garbled music is still
there. Got it.

Your encoder works unconditionally with AV_SAMPLE_FMT_S16, 2 channels and
the same sample rate as the first input file. That is not correct.

For each parameter (sample format, channel count and layout, sample rate)
you must either configure the output encoder to use the same value as the
output of libavfilter or force libavfilter to convert to the format you want
to use.

The latter can be done by using av_opt_set* on abuffersink_ctx after initing
it.

Note that most encoders will only support one or a few formats. For a lot of
encoders, there is a limit on the supported channel layouts, and some
annoying encoders have limits on the sample rate.

Your problem comes from the sample format, I told you amix works with float
while you configure your encoder for 16-bits ints.

I get correct results by adding:

av_opt_set_int_list(abuffersink_ctx, "sample_fmts",
    ((int[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }),
    AV_SAMPLE_FMT_NONE, AV_OPT_SEARCH_CHILDREN);

Regards,

-- 
  Nicolas George
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From codecomplete at free.fr  Mon Apr 21 03:13:09 2014
From: codecomplete at free.fr (Gilles)
Date: Mon, 21 Apr 2014 03:13:09 +0200
Subject: [FFmpeg-user] Turning MP3 into MP4 and join with second MP4
	file?
References: <hvl4l9965gjm9gt8ucrn500qa8ehur942k@4ax.com>
Message-ID: <j3s8l9h355u2c5c8ntrfubth70qm06ql2l@4ax.com>

On Sat, 19 Apr 2014 13:32:11 +0200, Gilles <codecomplete at free.fr>
wrote:
>I used ffmpeg to turn the MP3 file and a PNG file into an MP4 file,
>converted the second MP4 file to match the audiovideo settings of the
>first MP4, but it doesn't work and neitherMP4Joiner nor My MP4Box GUI
>are able to join the two MP4 files.

For others benefit:

1. Using ffmpeg, created first MP4 file by combining MP3 + PNG

ffmpeg.exe -loop 1 -i speaker.png -i first.half.mp3 -shortest -c:v
libx264 -s:v 848x480 -tune stillimage -pix_fmt yuv420p -c:a aac
-strict -2 -f mp4 part1.mp4

2. Using ffmpeg, converted second MP4 file to match settings used by
first file

ffmpeg.exe -report -i second.part.mp4 -shortest -c:v libx264 -pix_fmt
yuv420p -strict experimental -c:a aac -ab 128k -ar 48000 -strict -2 -f
mp4 part.2.mp4

3. Joined both files with Corel VideoStudio Pro X5

4. As output MP4 was three times bigger, used Handbrake to reduce file
size further.

Thank you.


From tmh920811 at 163.com  Mon Apr 21 03:21:44 2014
From: tmh920811 at 163.com (Minhao Tang)
Date: Mon, 21 Apr 2014 09:21:44 +0800 (CST)
Subject: [FFmpeg-user] hevc wpp decoding
Message-ID: <18e88823.24c8.14581dfe056.Coremail.tmh920811@163.com>


Hello, 
    I'm interested in the accelerating effect of wpp decoding applied in hevc decoder. And I've found there exist codes named wpp_entry(balabala...) in ffmpeg-2.2/libavcodec/hevc.c . 
    Thus my question is what command line should I use to activate the wpp decoding? I found that it doesn't enter wpp as default.


Best Regards,
Minhao Tang





--

???
?????????
????? 100084
??(+86)15201524211
Minhao Tang
Department of E.E, Tsinghua University
Beijing?China 100084
Cell(+86)15201524211

From xanadu at apost.plala.or.jp  Mon Apr 21 06:54:51 2014
From: xanadu at apost.plala.or.jp (=?utf-8?B?5a6u5p2RIOWFrOeUtw==?=)
Date: Mon, 21 Apr 2014 13:54:51 +0900
Subject: [FFmpeg-user] Enabling libx265
In-Reply-To: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>
References: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>
Message-ID: <6020031A-5DFD-405F-9E39-71F3B75CEF58@apost.plala.or.jp>


2014/04/18 1:26??? ?? <xanadu at apost.plala.or.jp> ?????

> Now I remove *.dylib files to build ffmpeg as static binary.
> 
> $ git clone git://source.ffmpeg.org/ffmpeg.git
> $ cd ffmpeg
> $ ./configure --as=yasm --enable-gpl --enable-libx265
> 
> then ffmpeg configure script exits with "ERROR: x265 not found" error.

The cause of this issue might be a bug with x265.  It can be solved by:
1.Add -lstdc++ into CFLAGS
or
2.Edit x265.pc as follows:
--- a/x265.pc	2014-04-16 14:50:20.000000000 +0900
+++ b/x265.pc	2014-04-21 10:54:32.000000000 +0900
@@ -6,6 +6,6 @@
 Name: x265
 Description: H.265/HEVC video encoder
 Version: 0.9
-Libs: -L${libdir} -lx265
+Libs: -L${libdir} -lx265 -lstdc++
 Libs.private: -lc++ -l/Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/lib/clang/5.1/lib/darwin/libclang_rt.osx.a
 Cflags: -I${includedir}


// Miya

From ulzii.ict at gmail.com  Mon Apr 21 10:30:38 2014
From: ulzii.ict at gmail.com (Ulziisaikhan L)
Date: Mon, 21 Apr 2014 16:30:38 +0800
Subject: [FFmpeg-user] UDP multicast mosaic stream stops
Message-ID: <CABXT5Gt40AWzFKPFbzRD+rLG=E0u18pj3ttCgZKR0tF5Z9kcHg@mail.gmail.com>

dear all,


I'm streaming mosaic from UDP multicast.
mosaic have 6 and 4 windows but both of them stream normally up to 2 day.
actually 6 streams overloading buffer then I reduced 4 stream but still
terminated. any idea ?


*4 window mosaic stream*
/root/bin/ffmpeg -f mpegts -i 'udp://
238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.15:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.40:8001?fifo_size
=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.45:8001?fifo_size=1000000&overrun_nonfatal=1' -filter_complex
"nullsrc=size=1024x600 [base]; [0:v] setpts=PTS-STARTPTS, scale=512x300
[upperleft
]; [1:v] setpts=PTS-STARTPTS, scale=512x300 [upperright]; [2:v]
setpts=PTS-STARTPTS, scale=512x300 [lowerleft]; [3:v] setpts=PTS-STARTPTS,
scale=512x300 [lowerright]; [base][upperleft] overla
y=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=512 [tmp2];
[tmp2][lowerleft] overlay=shortest=1:y=300 [tmp3]; [tmp3][lowerright]
overlay=shortest=1:x=512:y=300" -strict experime
ntal -vcodec libx264 -preset ultrafast -profile:v main -an -f mpegts udp://
238.1.1.42:8001?pkt_size=1316


*6 windows mosaic stream*
/root/bin/ffmpeg -f mpegts -i 'udp://
238.1.1.111:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.27:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.15:8001?fifo_size
=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.124:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.14:8001?fifo_size=1000000&overrun_nonfatal=1' -i 'udp://
238.1.1.40:8001?fifo_si
ze=1000000&overrun_nonfatal=1' -filter_complex "nullsrc=size=1024x600
[base]; [0:v] setpts=PTS-STARTPTS, scale=341x300 [upperleft]; [1:v]
setpts=PTS-STARTPTS, scale=341x300 [uppermiddle]; [2:
v] setpts=PTS-STARTPTS, scale=341x300 [upperright]; [3:v]
setpts=PTS-STARTPTS, scale=341x300 [lowerleft]; [4:v] setpts=PTS-STARTPTS,
scale=341x300 [lowermiddle]; [5:v] setpts=PTS-STARTPTS, sc
ale=341x300 [lowerright]; [base][upperleft] overlay=shortest=1 [tmp1];
[tmp1] [uppermiddle] overlay=shortest=1:x=341 [tmp2]; [tmp2] [upperright]
overlay=shortest=1:x=682 [tmp3]; [tmp3] [lower
left] overlay=shortest=1:y=300 [tmp4]; [tmp4] [lowermiddle]
overlay=shortest=1:x=341:y=300 [tmp5]; [tmp5] [lowerright]
overlay=shortest=1:x=682:y=300" -strict experimental -vcodec libx264 -pr
eset ultrafast -profile:v main -an -f mpegts udp://
238.1.1.42:8001?pkt_size=1316


thank a lot
Ulzii

From mr.artichaut at gmail.com  Mon Apr 21 11:05:09 2014
From: mr.artichaut at gmail.com (Mr Artichaut)
Date: Mon, 21 Apr 2014 11:05:09 +0200
Subject: [FFmpeg-user] libavfilter : using amix
In-Reply-To: <20140420182206.GA1508@phare.normalesup.org>
References: <20140413084205.GA18477@phare.normalesup.org>
	<CAMu2HxeYjjEbt7WTegDZw0OhVWG-vUGfCgxx7EAD+MJWmFoRcw@mail.gmail.com>
	<20140419113513.GA14903@phare.normalesup.org>
	<CAMu2HxeOc_PMTgdXWchKyPLczK2LOwXf1TyDTRfuxd3kpzAdNg@mail.gmail.com>
	<20140419135414.GA25657@phare.normalesup.org>
	<CAMu2HxeNb8Bozftqrx4B0bu8RPkaEowndzGBJJ8e+owh0wFRNQ@mail.gmail.com>
	<20140419145448.GA15349@phare.normalesup.org>
	<CAMu2HxdLpq32wzSxifqhPkiD8ExHgu3x8Cv3+bwiJRwDmq2R8A@mail.gmail.com>
	<20140419163535.GA21238@phare.normalesup.org>
	<CAMu2Hxc3cDkw2T-Y-zOf=u6otJ4KEGzqNEhhNBdSPX6d+e=rTw@mail.gmail.com>
	<20140420182206.GA1508@phare.normalesup.org>
Message-ID: <CAMu2HxdBD7Pb6=kMNH0R9Djdhz5khmXk-e-ucy1Rd2ozv26zjg@mail.gmail.com>

It works. Thank you very much for your help and your patience.

The output format thing totally makes sense now that I think of it. It
didn't work first so I've also setup the channel layout in the sink and
then it was OK.

I've made a new gist with the working code :
https://gist.github.com/MrArtichaut/11136813

I will clean it a little but it's a good example for anyone trying to use
amix or something similar. Your messages were also very instructive.

Thank you again.

From he_yaming at yahoo.com  Mon Apr 21 20:20:57 2014
From: he_yaming at yahoo.com (Yaming He)
Date: Mon, 21 Apr 2014 11:20:57 -0700 (PDT)
Subject: [FFmpeg-user] How to build 64-bit ffmpeg using msvc toolchain?
Message-ID: <1398104457.98654.YahooMailNeo@web140505.mail.bf1.yahoo.com>

Hello, I am building the ffmpeg by using MSVC toolchain. By
default it generates the 32-bit build but I want have a 64-bit build. I used
the below configuration flags but it always generate the 32-bit binary. Did I
miss anything? How to build a 64-bit ffmpeg with msvc toolcahin?
?
./configure --toolchain=msvc --enable-shared
--disable-static --target_os=win64 --arch=x86_64
?
Thanks.
Jerry

From miro.rovis at croatiafidelis.hr  Mon Apr 21 22:18:28 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Mon, 21 Apr 2014 22:18:28 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
Message-ID: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>

Hello FFmpeg people and users!
(and Hello Gentoo people, indirectly!)

While I heavy-heartedly, since a few hours ago, abandoned all my work 
in
progress because it was programmatically, IMO, that is via portage
arrangements, purposefully, knowingly (but indirectly, not on me 
personally, as
programs do), interrupted. Explanation follows later.

I am reinstalling Gentoo, which is explained here (currently the link 
is
perfectly extant, but that doesn't, with some mods in Gentoo (Gentoo 
people,
you are getting a copy of this email too, you too vow transparency, I 
can't be
wrong about that), [but that doesn't, with some mods in Gentoo] mean 
that it
will remain, and I can assure you that I respect what I/others write, 
as
documents, so it won't be my fault in case in the future it isn't 
there, sorry
for having to provide these caveats)...

I am reinstalling Gentoo, all (and the links pointed to) is (hopefully 
also at
later time) here:
Air-Gapped Gentoo Install, Tentative
https://forums.gentoo.org/viewtopic-t-987268.html

...And in the links pointed from there, all painstakingly checked 
before
submitting them, local and external.

I am writing this to the list for reasons of being as user, victim of a 
few
cunning attempts to be almost forced into using libav instead of real 
ffmpeg.

I notice only now that FFmpeg hosts a mailing list for libav (if my
understanding is correct that that's the library containing avconv 
program, a
fork of ffmpeg), and I actually wish that the compromise worked between 
the
FFmpeg team and the Avconv team.

However, I'm not writing this to Avconv team. They're unnecessarily 
spending my
time, as they already were a year ago when I installed Debian for the 
first
time on another box, and when I found that ffmpeg was deprecated and 
that I had
to use avconv... I want to live without them, even though I wish them 
peace and
prosperity, but not in my yard, and so I don't really want to converse 
with
them. They cost me time, as generally user, who they try to impose 
their
programs on, again and again...

I believe you are familiar with these issues, such as in Debian, where 
ffmpeg
is virtually banned. Let me see if there were public discussions on the 
issue
of Debian and the ffmpeg ban from it (other than from Christan 
Marillat's
deb-multimedia.org , but he was somewhat almost sidelined, and harassed 
by
Zacchiroli the then leader in Debian -- I'm not an insieder, the 
information
was from different places on the Internet, publically accessible)...

But let me see if there were public discussions on the issue of Debian 
and the
ffmpeg ban... on...

(Just I don't want to use the Surveillance Engine itself, directly, 
thanks a lot!)

literal input into ddg,gg:
https://ffmpeg.org/pipermail/ffmpeg-user/ Debian avconv

only a few people asking questions to ffmpeg people about avconv and 
turned
away...

literal input into startpage,com:
https://ffmpeg.org/pipermail/ffmpeg-user/ Debian avconv

basically the same, worth pointing out this reply by Eugene Hoyos:

http://lists.ffmpeg.org/pipermail/ffmpeg-user/2012-December/012223.html
"> avconv version v9_beta2-501-g6d5b009

This application is known to contain several hundred bugs
not present in FFmpeg, some of them security relevant,
please understand that we therefore cannot support it."

But not anything openly telling for the public.

So I am left to conclude for myself.

I want to show my problem openly, on the Gentoo Forums, but I was 
wrongly
banned from the Forums in one occasion, and so I am not in the clear 
that what
I want to publically enquire about and suggest and plead, yes, actually 
plead
to the Gentoo Developers, yes, the top Gentoo brass actually...

...So I am not in the clear that what I want to publically enquire and 
plead
about to Gentoo people, would remain in the open, and this time I don't 
want to
lose my work, just because I may think differently, such as I think and 
express
very openly on NSA spying that is getting a strong hold in GNU/Linux 
through
SELinux, while the honest and very reliable Grsecurity/Pax are being 
sidewayed
(and even so through censorship).

So this time I don't want to write and lose my work.

So my question is, to Gentoo Admins and Moderators and Developers, can 
I openly
ask honest questions on Gentoo Forums about the issue that I broached 
here, and
do you mind about me calling for witnesses in among the FFmpeg Admins,
Moderators and Developers, as well as subscribers?

And can you, FFmpeg team, allow me to openly voice my concern that I 
touched
upon here?

I'm writing in a little hurry, I'm long overdue doing other things, 
basically
using my Gentoo, and not still building it (and pls. notice that the 
hindrance
in question has interfered even with my building that I should be 
finishing
doing), so pls.  forgive syntax and other typoes and errors.

Pls. allow this in the open, it is probably best for this message to be 
on the
FFmpeg Users Mailing list and publically in the archives as all emails 
get to
be, and after getting a reply from Gentoo Admins, hopefully positive, 
opening a
topic on Gentoo Forums and giving my entire explanation, with very 
clear
arguments.

I'm after peaceful solutions, not warrring, but absolutely against 
arrogance in
imposing programs on users,

Miroslav Rovis
Zagreb, Croatia
www.CroatiaFidelis.hr

Post Scriptum: On a final note, I am not certain who among the Gentoo 
team to
contact about this, so it has to be a public link on the 
forums.gentoo.org to
start with, as soon as my email shows in the FFmpeg archives. When I 
think
carefully, that is the sole way that I can do it, I have no time for 
asking and
waiting, and asking and waiting, as sometimes happens with some of the 
Gentoo
people. Once the link is public, there is no need repeating. Much 
easier.

I will, however, send this copy straight to the main Admin of Gentoo 
Forums, at
the same time as I will send this letter to the FFmpeg Users list, so 
that
entire text holds completely fair and true.


From miro.rovis at croatiafidelis.hr  Mon Apr 21 22:58:38 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Mon, 21 Apr 2014 22:58:38 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
Message-ID: <68bb22ca2aeea01cb214d305b8256d0c@croatiafidelis.hr>

On 2014-04-21 22:18, miro.rovis at croatiafidelis.hr wrote:
> Hello FFmpeg people and users!
> (and Hello Gentoo people, indirectly!)
> 
> While I heavy-heartedly, since a few hours ago, abandoned all my work 
> in
> progress because it was programmatically, IMO, that is via portage
> arrangements, purposefully, knowingly (but indirectly, not on me 
> personally, as
> programs do), interrupted. Explanation follows later.
> 
> I am reinstalling Gentoo, which is explained here (currently the link 
> is
> perfectly extant, but that doesn't, with some mods in Gentoo (Gentoo 
> people,
> you are getting a copy of this email too, you too vow transparency, I 
> can't be
> wrong about that), [but that doesn't, with some mods in Gentoo] mean 
> that it
> will remain, and I can assure you that I respect what I/others write, 
> as
> documents, so it won't be my fault in case in the future it isn't 
> there, sorry
> for having to provide these caveats)...
> 
> I am reinstalling Gentoo, all (and the links pointed to) is (hopefully 
> also at
> later time) here:
> Air-Gapped Gentoo Install, Tentative
> https://forums.gentoo.org/viewtopic-t-987268.html
> 
> ...And in the links pointed from there, all painstakingly checked 
> before
> submitting them, local and external.
> 
> I am writing this to the list for reasons of being as user, victim of 
> a few
> cunning attempts to be almost forced into using libav instead of real 
> ffmpeg.
> 
> I notice only now that FFmpeg hosts a mailing list for libav (if my
> understanding is correct that that's the library containing avconv 
> program, a
> fork of ffmpeg), and I actually wish that the compromise worked 
> between the
> FFmpeg team and the Avconv team.
> 
> However, I'm not writing this to Avconv team. They're unnecessarily 
> spending my
> time, as they already were a year ago when I installed Debian for the 
> first
> time on another box, and when I found that ffmpeg was deprecated and 
> that I had
> to use avconv... I want to live without them, even though I wish them 
> peace and
> prosperity, but not in my yard, and so I don't really want to converse 
> with
> them. They cost me time, as generally user, who they try to impose 
> their
> programs on, again and again...
> 
> I believe you are familiar with these issues, such as in Debian, where 
> ffmpeg
> is virtually banned. Let me see if there were public discussions on 
> the issue
> of Debian and the ffmpeg ban from it (other than from Christan 
> Marillat's
> deb-multimedia.org , but he was somewhat almost sidelined, and 
> harassed by
> Zacchiroli the then leader in Debian -- I'm not an insieder, the 
> information
> was from different places on the Internet, publically accessible)...
> 
> But let me see if there were public discussions on the issue of Debian 
> and the
> ffmpeg ban... on...
> 
> (Just I don't want to use the Surveillance Engine itself, directly,
> thanks a lot!)
> 
> literal input into ddg,gg:
> https://ffmpeg.org/pipermail/ffmpeg-user/ Debian avconv
> 
> only a few people asking questions to ffmpeg people about avconv and 
> turned
> away...
> 
> literal input into startpage,com:
> https://ffmpeg.org/pipermail/ffmpeg-user/ Debian avconv
> 
> basically the same, worth pointing out this reply by Eugene Hoyos:
> 
> http://lists.ffmpeg.org/pipermail/ffmpeg-user/2012-December/012223.html
> "> avconv version v9_beta2-501-g6d5b009
> 
> This application is known to contain several hundred bugs
> not present in FFmpeg, some of them security relevant,
> please understand that we therefore cannot support it."
> 
> But not anything openly telling for the public.
> 
> So I am left to conclude for myself.
> 
> I want to show my problem openly, on the Gentoo Forums, but I was 
> wrongly
> banned from the Forums in one occasion, and so I am not in the clear 
> that what
> I want to publically enquire about and suggest and plead, yes, 
> actually plead
> to the Gentoo Developers, yes, the top Gentoo brass actually...
> 
> ...So I am not in the clear that what I want to publically enquire and 
> plead
> about to Gentoo people, would remain in the open, and this time I 
> don't want to
> lose my work, just because I may think differently, such as I think 
> and express
> very openly on NSA spying that is getting a strong hold in GNU/Linux 
> through
> SELinux, while the honest and very reliable Grsecurity/Pax are being 
> sidewayed
> (and even so through censorship).
> 
> So this time I don't want to write and lose my work.
> 
> So my question is, to Gentoo Admins and Moderators and Developers, can 
> I openly
> ask honest questions on Gentoo Forums about the issue that I broached 
> here, and
> do you mind about me calling for witnesses in among the FFmpeg Admins,
> Moderators and Developers, as well as subscribers?
> 
> And can you, FFmpeg team, allow me to openly voice my concern that I 
> touched
> upon here?
> 
> I'm writing in a little hurry, I'm long overdue doing other things, 
> basically
> using my Gentoo, and not still building it (and pls. notice that the 
> hindrance
> in question has interfered even with my building that I should be 
> finishing
> doing), so pls.  forgive syntax and other typoes and errors.
> 
> Pls. allow this in the open, it is probably best for this message to 
> be on the
> FFmpeg Users Mailing list and publically in the archives as all emails 
> get to
> be, and after getting a reply from Gentoo Admins, hopefully positive, 
> opening a
> topic on Gentoo Forums and giving my entire explanation, with very 
> clear
> arguments.
> 
> I'm after peaceful solutions, not warrring, but absolutely against 
> arrogance in
> imposing programs on users,
> 
> Miroslav Rovis
> Zagreb, Croatia
> www.CroatiaFidelis.hr
> 
> Post Scriptum: On a final note, I am not certain who among the Gentoo 
> team to
> contact about this, so it has to be a public link on the 
> forums.gentoo.org to
> start with, as soon as my email shows in the FFmpeg archives. When I 
> think
> carefully, that is the sole way that I can do it, I have no time for 
> asking and
> waiting, and asking and waiting, as sometimes happens with some of the 
> Gentoo
> people. Once the link is public, there is no need repeating. Much 
> easier.
> 
> I will, however, send this copy straight to the main Admin of Gentoo 
> Forums, at
> the same time as I will send this letter to the FFmpeg Users list, so 
> that
> entire text holds completely fair and true.
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

I did, first thing, send the entire of the above text to the main Admin 
of the Gentoo Forums.

And, the title the same, on Gentoo:
Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539612.html
(pls. hurriedly take notice that the link is alive right now, else I 
won't have witnesses that it was alive; happened, happened, it did 
happened to me on Gentoo Forums)
...There is a little longer to wait now:

Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539612.html

Miroslav Rovis
Zagreb, Croatia
www.CroatiaFidelis.hr

From cehoyos at ag.or.at  Mon Apr 21 23:37:04 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 21 Apr 2014 21:37:04 +0000 (UTC)
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
Message-ID: <loom.20140421T232710-490@post.gmane.org>

 <miro.rovis <at> croatiafidelis.hr> writes:

> I notice only now that FFmpeg hosts a mailing list for 
> libav (if my understanding is correct that that's the 
> library containing avconv program, a fork of ffmpeg)

FFmpeg does not host a mailing list for avconv, no.

FFmpeg provides several libraries, among them libavcodec, 
libavformat, libavfilter, libpostproc, libswscale, 
libswresample and libavutil.
All these libraries (and the ffmpeg, ffplay, etc. 
programs) are supported on FFmpeg mailing lists, see 
http://ffmpeg.org/contact.html

> http://lists.ffmpeg.org/pipermail/ffmpeg-user/2012-December/012223.html
> "> avconv version v9_beta2-501-g6d5b009
> 
> This application is known to contain several hundred bugs
> not present in FFmpeg, some of them security relevant,
> please understand that we therefore cannot support it."
> 
> But not anything openly telling for the public.

How can I be more public?
Sorry if I misunderstand but users reported many 
hundred bugs in the last three years (since a group of 
FFmpeg developers tried to take over the project), many 
of these bugs were fixed in FFmpeg but not fixed in 
avconv.

Your email is very difficult to understand (but I am not 
a native speaker), maybe these links help you:
http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html
https://packages.gentoo.org/package/media-video/ffmpeg
(If you cannot use 2.2, please use at least 1.2)
https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=729203
http://anonscm.debian.org/gitweb/?p=collab-maint/ffmpeg.git;a=summary

If you have questions remaining, please ask them!

Carl Eugen


From leonard at kcfchurch.org  Tue Apr 22 00:26:44 2014
From: leonard at kcfchurch.org (Leonard Bogard)
Date: Mon, 21 Apr 2014 15:26:44 -0700
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <68bb22ca2aeea01cb214d305b8256d0c@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<68bb22ca2aeea01cb214d305b8256d0c@croatiafidelis.hr>
Message-ID: <CA+JzsNTn1qX+PwqT9vUrN-cxBP+gZMS=twkcqLeWa76=-yz=wg@mail.gmail.com>

On Mon, Apr 21, 2014 at 1:58 PM, <miro.rovis at croatiafidelis.hr> wrote:

> On 2014-04-21 22:18, miro.rovis at croatiafidelis.hr wrote:
>
>> Hello FFmpeg people and users!
>> (and Hello Gentoo people, indirectly!)
>>
>> While I heavy-heartedly, since a few hours ago, abandoned all my work in
>> progress because it was programmatically, IMO, that is via portage
>> arrangements, purposefully, knowingly (but indirectly, not on me
>> personally, as
>> programs do), interrupted. Explanation follows later.
>>
>> I am reinstalling Gentoo, which is explained here (currently the link is
>> perfectly extant, but that doesn't, with some mods in Gentoo (Gentoo
>> people,
>> you are getting a copy of this email too, you too vow transparency, I
>> can't be
>> wrong about that), [but that doesn't, with some mods in Gentoo] mean that
>> it
>> will remain, and I can assure you that I respect what I/others write, as
>> documents, so it won't be my fault in case in the future it isn't there,
>> sorry
>> for having to provide these caveats)...
>>
>> I am reinstalling Gentoo, all (and the links pointed to) is (hopefully
>> also at
>> later time) here:
>> Air-Gapped Gentoo Install, Tentative
>> https://forums.gentoo.org/viewtopic-t-987268.html
>>
>> ...And in the links pointed from there, all painstakingly checked before
>> submitting them, local and external.
>>
>> I am writing this to the list for reasons of being as user, victim of a
>> few
>> cunning attempts to be almost forced into using libav instead of real
>> ffmpeg.
>>
>> I notice only now that FFmpeg hosts a mailing list for libav (if my
>> understanding is correct that that's the library containing avconv
>> program, a
>> fork of ffmpeg), and I actually wish that the compromise worked between
>> the
>> FFmpeg team and the Avconv team.
>>
>> However, I'm not writing this to Avconv team. They're unnecessarily
>> spending my
>> time, as they already were a year ago when I installed Debian for the
>> first
>> time on another box, and when I found that ffmpeg was deprecated and that
>> I had
>> to use avconv... I want to live without them, even though I wish them
>> peace and
>> prosperity, but not in my yard, and so I don't really want to converse
>> with
>> them. They cost me time, as generally user, who they try to impose their
>> programs on, again and again...
>>
>> I believe you are familiar with these issues, such as in Debian, where
>> ffmpeg
>> is virtually banned. Let me see if there were public discussions on the
>> issue
>> of Debian and the ffmpeg ban from it (other than from Christan Marillat's
>> deb-multimedia.org , but he was somewhat almost sidelined, and harassed
>> by
>> Zacchiroli the then leader in Debian -- I'm not an insieder, the
>> information
>> was from different places on the Internet, publically accessible)...
>>
>> But let me see if there were public discussions on the issue of Debian
>> and the
>> ffmpeg ban... on...
>>
>> (Just I don't want to use the Surveillance Engine itself, directly,
>> thanks a lot!)
>>
>> literal input into ddg,gg:
>> https://ffmpeg.org/pipermail/ffmpeg-user/ Debian avconv
>>
>> only a few people asking questions to ffmpeg people about avconv and
>> turned
>> away...
>>
>> literal input into startpage,com:
>> https://ffmpeg.org/pipermail/ffmpeg-user/ Debian avconv
>>
>> basically the same, worth pointing out this reply by Eugene Hoyos:
>>
>> http://lists.ffmpeg.org/pipermail/ffmpeg-user/2012-December/012223.html
>> "> avconv version v9_beta2-501-g6d5b009
>>
>> This application is known to contain several hundred bugs
>> not present in FFmpeg, some of them security relevant,
>> please understand that we therefore cannot support it."
>>
>> But not anything openly telling for the public.
>>
>> So I am left to conclude for myself.
>>
>> I want to show my problem openly, on the Gentoo Forums, but I was wrongly
>> banned from the Forums in one occasion, and so I am not in the clear that
>> what
>> I want to publically enquire about and suggest and plead, yes, actually
>> plead
>> to the Gentoo Developers, yes, the top Gentoo brass actually...
>>
>> ...So I am not in the clear that what I want to publically enquire and
>> plead
>> about to Gentoo people, would remain in the open, and this time I don't
>> want to
>> lose my work, just because I may think differently, such as I think and
>> express
>> very openly on NSA spying that is getting a strong hold in GNU/Linux
>> through
>> SELinux, while the honest and very reliable Grsecurity/Pax are being
>> sidewayed
>> (and even so through censorship).
>>
>> So this time I don't want to write and lose my work.
>>
>> So my question is, to Gentoo Admins and Moderators and Developers, can I
>> openly
>> ask honest questions on Gentoo Forums about the issue that I broached
>> here, and
>> do you mind about me calling for witnesses in among the FFmpeg Admins,
>> Moderators and Developers, as well as subscribers?
>>
>> And can you, FFmpeg team, allow me to openly voice my concern that I
>> touched
>> upon here?
>>
>> I'm writing in a little hurry, I'm long overdue doing other things,
>> basically
>> using my Gentoo, and not still building it (and pls. notice that the
>> hindrance
>> in question has interfered even with my building that I should be
>> finishing
>> doing), so pls.  forgive syntax and other typoes and errors.
>>
>> Pls. allow this in the open, it is probably best for this message to be
>> on the
>> FFmpeg Users Mailing list and publically in the archives as all emails
>> get to
>> be, and after getting a reply from Gentoo Admins, hopefully positive,
>> opening a
>> topic on Gentoo Forums and giving my entire explanation, with very clear
>> arguments.
>>
>> I'm after peaceful solutions, not warrring, but absolutely against
>> arrogance in
>> imposing programs on users,
>>
>> Miroslav Rovis
>> Zagreb, Croatia
>> www.CroatiaFidelis.hr
>>
>> Post Scriptum: On a final note, I am not certain who among the Gentoo
>> team to
>> contact about this, so it has to be a public link on the
>> forums.gentoo.org to
>> start with, as soon as my email shows in the FFmpeg archives. When I think
>> carefully, that is the sole way that I can do it, I have no time for
>> asking and
>> waiting, and asking and waiting, as sometimes happens with some of the
>> Gentoo
>> people. Once the link is public, there is no need repeating. Much easier.
>>
>> I will, however, send this copy straight to the main Admin of Gentoo
>> Forums, at
>> the same time as I will send this letter to the FFmpeg Users list, so that
>> entire text holds completely fair and true.
>>
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
> I did, first thing, send the entire of the above text to the main Admin of
> the Gentoo Forums.
>
> And, the title the same, on Gentoo:
> Libav (Avconv) Imposition on Users who want FFmpeg
> https://forums.gentoo.org/viewtopic-p-7539612.html
> (pls. hurriedly take notice that the link is alive right now, else I won't
> have witnesses that it was alive; happened, happened, it did happened to me
> on Gentoo Forums)
> ...There is a little longer to wait now:
>
> Libav (Avconv) Imposition on Users who want FFmpeg
> https://forums.gentoo.org/viewtopic-p-7539612.html
>
> Miroslav Rovis
> Zagreb, Croatia
> www.CroatiaFidelis.hr
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

Wow, what an amazing coincidence, the gentoo forums are down.
http://www.downforeveryoneorjustme.com/http://forums.gentoo.org

From h.reindl at thelounge.net  Tue Apr 22 00:29:08 2014
From: h.reindl at thelounge.net (Reindl Harald)
Date: Tue, 22 Apr 2014 00:29:08 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <CA+JzsNTn1qX+PwqT9vUrN-cxBP+gZMS=twkcqLeWa76=-yz=wg@mail.gmail.com>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>	<68bb22ca2aeea01cb214d305b8256d0c@croatiafidelis.hr>
	<CA+JzsNTn1qX+PwqT9vUrN-cxBP+gZMS=twkcqLeWa76=-yz=wg@mail.gmail.com>
Message-ID: <53559BB4.2000403@thelounge.net>

Am 22.04.2014 00:26, schrieb Leonard Bogard:
>> Libav (Avconv) Imposition on Users who want FFmpeg
>> https://forums.gentoo.org/viewtopic-p-7539612.html
>>
>> Miroslav Rovis
>> Zagreb, Croatia
>> www.CroatiaFidelis.hr
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
> 
> Wow, what an amazing coincidence, the gentoo forums are down.
> http://www.downforeveryoneorjustme.com/http://forums.gentoo.org

please strip your quotes and remove signature from them

it's hard to follow a thread if you have to scroll down dozens
of pages to find two lines between a lot of text you already
faced and the list-signature

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From u at pkh.me  Tue Apr 22 00:30:11 2014
From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=)
Date: Tue, 22 Apr 2014 00:30:11 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <CA+JzsNTn1qX+PwqT9vUrN-cxBP+gZMS=twkcqLeWa76=-yz=wg@mail.gmail.com>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<68bb22ca2aeea01cb214d305b8256d0c@croatiafidelis.hr>
	<CA+JzsNTn1qX+PwqT9vUrN-cxBP+gZMS=twkcqLeWa76=-yz=wg@mail.gmail.com>
Message-ID: <20140421223011.GA31405@leki>

On Mon, Apr 21, 2014 at 03:26:44PM -0700, Leonard Bogard wrote:
[...]
> 
> Wow, what an amazing coincidence, the gentoo forums are down.
> http://www.downforeveryoneorjustme.com/http://forums.gentoo.org

http://infra-status.gentoo.org/

The forums.gentoo.org sponsor (Gossamer Threads) is going to perform some
maintenance of the forums machines and do a memory upgrade on Thursday.

Update: Postponed to Monday.

Regards,

[...]

-- 
Cl?ment B.
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From miro.rovis at croatiafidelis.hr  Tue Apr 22 04:45:48 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Tue, 22 Apr 2014 04:45:48 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <loom.20140421T232710-490@post.gmane.org>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<loom.20140421T232710-490@post.gmane.org>
Message-ID: <38199d860399b4c99b811a623646dcd6@croatiafidelis.hr>

On 2014-04-21 23:37, Carl Eugen Hoyos wrote:
> <miro.rovis <at> croatiafidelis.hr> writes:
> 
>> I notice only now that FFmpeg hosts a mailing list for
>> libav (if my understanding is correct that that's the
>> library containing avconv program, a fork of ffmpeg)
> 
> FFmpeg does not host a mailing list for avconv, no.
> 
> FFmpeg provides several libraries, among them libavcodec,
> libavformat, libavfilter, libpostproc, libswscale,
> libswresample and libavutil.
> All these libraries (and the ffmpeg, ffplay, etc.
> programs) are supported on FFmpeg mailing lists, see
> http://ffmpeg.org/contact.html
> 
>> http://lists.ffmpeg.org/pipermail/ffmpeg-user/2012-December/012223.html
>> "> avconv version v9_beta2-501-g6d5b009
>> 
>> This application is known to contain several hundred bugs
>> not present in FFmpeg, some of them security relevant,
>> please understand that we therefore cannot support it."
>> 
>> But not anything openly telling for the public.
> 
> How can I be more public?
> Sorry if I misunderstand but users reported many
> hundred bugs in the last three years (since a group of
> FFmpeg developers tried to take over the project), many
> of these bugs were fixed in FFmpeg but not fixed in
> avconv.
> 
> Your email is very difficult to understand (but I am not
> a native speaker), maybe these links help you:
> http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html

I'll read this again.

> https://packages.gentoo.org/package/media-video/ffmpeg
> (If you cannot use 2.2, please use at least 1.2)
> https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=729203
> http://anonscm.debian.org/gitweb/?p=collab-maint/ffmpeg.git;a=summary

I'll study those too.

> If you have questions remaining, please ask them!
> 
> Carl Eugen
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Carl, in the first place, thank you for your kind reply.

But there were two more replies, of which this one:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021026.html
I don't understand (not even if it was addressed to me or to Leonard 
Bogard

But this I can say.
That page was down, and it is "coincidence", IMO.
I then performed an air-gap manouver which is very costly timewise, so 
it wasn't possible for me to reply earlier. Will be clarified first next 
(see below).

I'll send what I had prepared. I see some people will be interested.
And I see now that the page is up, and that I am probably not banned.
Am I, again, like I already was, for no just reasno, Gentoo people 
(who, if not earlier, will probably read this later)?

So, if not only the page is up, but also I am not banned, then I will 
post my short research that I announced, there, on Gentoo Forums, and as 
far as depends on me, announce promptly on this list when I do so.

Sorry for not very clear meanings, but it is also the reason, that the 
most important part, the tecnhical part is only coming (the second) 
next.

So first next is what I prepared when I thought I wouldn't be allowed 
transparency about what I deem misdeeds against good programs like 
FFmpeg.

Miroslav Rovis
Zagreb, Croatia
www.CroatiaFidelis.hr



From miro.rovis at croatiafidelis.hr  Tue Apr 22 04:50:08 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Tue, 22 Apr 2014 04:50:08 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <38199d860399b4c99b811a623646dcd6@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<loom.20140421T232710-490@post.gmane.org>
	<38199d860399b4c99b811a623646dcd6@croatiafidelis.hr>
Message-ID: <c08bea68c2de190fb7987cb060069dd7@croatiafidelis.hr>

On 2014-04-22 04:45, miro.rovis at croatiafidelis.hr wrote:
> On 2014-04-21 23:37, Carl Eugen Hoyos wrote:
>> <miro.rovis <at> croatiafidelis.hr> writes:
>> 
>>> I notice only now that FFmpeg hosts a mailing list for
>>> libav (if my understanding is correct that that's the
>>> library containing avconv program, a fork of ffmpeg)
>> FFmpeg does not host a mailing list for avconv, no.
>> FFmpeg provides several libraries, among them libavcodec,
>> libavformat, libavfilter, libpostproc, libswscale,
>> libswresample and libavutil.
>> All these libraries (and the ffmpeg, ffplay, etc.
>> programs) are supported on FFmpeg mailing lists, see
>> http://ffmpeg.org/contact.html
>> 
>>> http://lists.ffmpeg.org/pipermail/ffmpeg-user/2012-December/012223.html
>>> "> avconv version v9_beta2-501-g6d5b009
>>> This application is known to contain several hundred bugs
>>> not present in FFmpeg, some of them security relevant,
>>> please understand that we therefore cannot support it."
>>> But not anything openly telling for the public.
>> How can I be more public?
>> Sorry if I misunderstand but users reported many
>> hundred bugs in the last three years (since a group of
>> FFmpeg developers tried to take over the project), many
>> of these bugs were fixed in FFmpeg but not fixed in
>> avconv.
>> Your email is very difficult to understand (but I am not
>> a native speaker), maybe these links help you:
>> http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html
> 
> I'll read this again.
> 
>> https://packages.gentoo.org/package/media-video/ffmpeg
>> (If you cannot use 2.2, please use at least 1.2)
>> https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=729203
>> http://anonscm.debian.org/gitweb/?p=collab-maint/ffmpeg.git;a=summary
> 
> I'll study those too.
> 
>> If you have questions remaining, please ask them!
>> Carl Eugen
>> _______________________________________________
>> ffmpeg-user mailing list
>> ffmpeg-user at ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> 
> Carl, in the first place, thank you for your kind reply.
> 
> But there were two more replies, of which this one:
> https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021026.html
> I don't understand (not even if it was addressed to me or to Leonard 
> Bogard
> 
> But this I can say.
> That page was down, and it is "coincidence", IMO.
> I then performed an air-gap manouver which is very costly timewise,
> so it wasn't possible for me to reply earlier. Will be clarified first
> next (see below).
> 
> I'll send what I had prepared. I see some people will be interested.
> And I see now that the page is up, and that I am probably not banned.
> Am I, again, like I already was, for no just reasno, Gentoo people
> (who, if not earlier, will probably read this later)?
> 
> So, if not only the page is up, but also I am not banned, then I will
> post my short research that I announced, there, on Gentoo Forums, and
> as far as depends on me, announce promptly on this list when I do so.
> 
> Sorry for not very clear meanings, but it is also the reason, that
> the most important part, the tecnhical part is only coming (the
> second) next.
> 
> So first next is what I prepared when I thought I wouldn't be allowed
> transparency about what I deem misdeeds against good programs like
> FFmpeg.
> 
> Miroslav Rovis
> Zagreb, Croatia
> www.CroatiaFidelis.hr
> 
> 
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

==========================
Pls. note that I wrote this much earlier, when the freedom of speech or 
not in this particular instance seemed banned for me (and I still don't 
know if I am not now banned from Gentoo forums)

I'm very tired, here is morning, no sleep yet, so no correcting of text 
below.
==========================

Absolutely fascinating!
I must say I didn't expect a blow so promptly!
The very Impeccable Madame may have gotten alarmed!
(the one mentioned explicite (Latin: explicitly), only once, with a 
three caps
letter word, in the first message)

The Impeccable Madame is starting to be served well in the GNU/Linux 
community
in general, some honest and really moral devs lost the leadership, 
others
somewhat buckled in under pressure, good programs being replaced by the
compromised ones, all happening, not yet completed, but happening even 
in...

...Even in the most perfect of all GNU/Linices, because none has the 
perfection
of the Portage, but the Gentoo GNU/Linux!

What I am trying to say, is, this message below, was deemed in some way
offending. Pls. read and dissuade me, or try and dissuade them, if you 
can, of
the futility of someone's, mine or theirs, attitude, in the civilized 
world
where free speech features shiny and high as requirement:

PASTING:
========

A  critique, as well as enquiry of mine is, beginning of it, here:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021022.html

I'm now waiting for Gentoo Mods, Admins and Devs to give me green lite 
to
openly expose the problem, technically, in terms of what can be found 
on emerge
install's standard outputs, and some other portage things.

I can't wait for long, because I have completely interrupted my 
building of my
Gentoo box, as is available here:

Air-Gapped Gentoo Install, Tentative
https://forums.gentoo.org/viewtopic-t-987268.html

to see.

It is early A.M in the U.S. where most of the deciders, and the Gentoo
Foundation, reside.

An hour or two I have to wait.

After that, I'll simply try and post my text here (the technical 
arguments,
things that happened on my Gentoo box) , so there be no cross-posting, 
but an
attempt at transparent collaboration in giving information with both 
the FFmpeg
team and Gentoo team,

links nowadays work just fine, unless there is censorship, and it would 
be
great all the documents so far in this matter, and the documents that I 
intend
to post, be freely available for time to come.

Thank you,

Miroslav Rovis
Zagreb, Croatia
www.CroatiaFidelis.hr

======
PASTED (and typoes corrected)

But, around half hour ago now, the reader will understand that all this
writing, pasting of links, as well as what you will read me mentioning 
later,
takes time, around half hour ago now, I went online again (I work 
mostly
offline), and checked the second message of mine as it was posted in 
the
archives on FFmpeg, and from there, I clicked the link to the new topic 
I
started on Gentoo Forums:
Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539612.html
  that I gave in that second message of mine to FFmpeg Users mailing 
list that
was, by then, already posted on the FFmpeg Users archives:
[FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021023.html

And, lo and behold, the link opens nothing!

So, to show you that story, and to finish the exposing of murky deals 
against
FFmpeg, after showing that I didn't give you wrong address, but that, 
in all
probability, my new topic of the same title as this thread here, was 
removed
from Gentoo Forums, I now have to prepare the screencasts that I took, 
and that
prove, and very much amplify what I told you so far about censorship on 
Gentoo
Forums, along with a possibility for, if someone should doubt, the 
pcapng
captures (if they're good, can only take them on this online system, 
can't
check them).

The screencasts have sums:
3e923265dc69901248fddda6fe5ac1535648db5fe5cb0b07b71a77c7d7c7209f  
Screen_140421_2140.mkv
6ec392b2c14c4a8a766a1369abeb097da0c53fad215de5758300e05350fc49ad  
Screen_140421_2206.mkv

and the pcapng dumps:
8bc5672ddbf509ddeff0769db4bf488a18aa539e5ffb757bcf5b9dc0b07376c5  
dump_140421_2140.pcapng
55cdd8ca0b851dca79b4b8169eeca668e793503056690d4ea4b6f9977d2681be  
dump_140421_2206.pcapng

I think the ones of the timestamp 140421_2206 (2014-04-2 22:06) should 
suffice.

I wouldn't be spending anymore of anybody's time on this non-technical 
issue as
far as FFmpeg goes.

However, I am among the persecuted bunch in Croatia anyhow, so while I 
would
gladly take this all onto www.CroatiaFidelis.hr, don't you think that 
the
Regime in my country wouldn't collaborate gladly with the Madame 
Impeccable
since we really have a neo commie cliqu? in Croatia that even tortures 
people.

Please read that last line well and take it to heart. Not all of you 
are
deprived of it.

After all, someone did get it from Google, and those can only be such
people as some of those cliqu?s with big cash, such as taxpayers' 
moneys,
complete removal of my Youtube account (some 5 ys work, 500+ videos, 
officially
600,000+ views, where the lie that it was for copyright breaches is 
bigger that
Google itself.

But, I will wait a little to see how you, people from FFmpeg will 
react, and
whether I could at least post how the libav in the emerge process 
usurps the
ffmpeg name (in the program virtual/ffmpeg) to remove ffmpeg completely 
out of
visible options for common users, and how it is programmed into Gentoo 
portage
of these days, to take ffmpeg out of sight, at the installation of 
mplayer in
my concrete case, which I have prepared a text for sending.

I have used ffmpeg daily, and a lot, in some at least six years by now, 
and I
don't only want not to remain myself without it, but also that other 
users not
remain without it as well.

At the moment, I am preparing my responses offline.

I know so so well, that in moments like these, intrusion attempts on my 
systems
intensify.  Before I even try and see what happened online, I 
absolutely have
to pack up all the screencasts, bookmark some things in proper way, and 
use
another clone machine, not this one machine, to go online and try to 
post this
message of mine (such air-gapped preparations do take hours and more).

The screencast I will post when there is a few people, or if there will 
be a
few, who are willing to download it promptly and document in some way 
that they
saw it, then I can try and post it on www.CroatiaFidelis.hr

I am now probably banned from Gentoo...

But of course I don't know what they decided, I sure would very much 
like that
it's only some of the Mods took me out, others said, no let him be, and 
that I
may be back on Gentoo Forums.

Pls. is it acceptable that I post the technical stuff on how, again, 
not able
to express myself better:

...how the libav in the emerge process usurps the ffmpeg name (in the 
program
virtual/ffmpeg) to remove ffmpeg completely out of visible options for 
common
users, and how it is programmed into Gentoo portage of these days, to 
take
ffmpeg out of sight, at the installation of mplayer in my concrete 
case, which
I have prepared a text for sending.

Thanks for the FFmpeg program which is basic in my work!

Miroslav Rovis
Zagreb, Croatia
www.CroatiaFidelis.hr


From miro.rovis at croatiafidelis.hr  Tue Apr 22 05:12:41 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Tue, 22 Apr 2014 05:12:41 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <c08bea68c2de190fb7987cb060069dd7@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<loom.20140421T232710-490@post.gmane.org>
	<38199d860399b4c99b811a623646dcd6@croatiafidelis.hr>
	<c08bea68c2de190fb7987cb060069dd7@croatiafidelis.hr>
Message-ID: <f166c0f62720a55adf214610aaefb308@croatiafidelis.hr>

The technical information is there!
Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539696.html#7539696
Thank you, and also thank you Gentoo people.
If you didn't know, if you give in to NSA, by means of the now dominant 
SELinux, as almost all other Linuces have done, the world has lost so 
much, that you can't even phathom!
Please don't!
Miroslav Rovis
Zagreb, Croatia,
www.CroatiaFidelis.hr

From miro.rovis at croatiafidelis.hr  Tue Apr 22 14:49:33 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Tue, 22 Apr 2014 14:49:33 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <f166c0f62720a55adf214610aaefb308@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<loom.20140421T232710-490@post.gmane.org>
	<38199d860399b4c99b811a623646dcd6@croatiafidelis.hr>
	<c08bea68c2de190fb7987cb060069dd7@croatiafidelis.hr>
	<f166c0f62720a55adf214610aaefb308@croatiafidelis.hr>
Message-ID: <c20b5825fb55af6d58fa87f17c113e9e@croatiafidelis.hr>

On 2014-04-22 05:12, miro.rovis at croatiafidelis.hr wrote:
> The technical information is there!
> Libav (Avconv) Imposition on Users who want FFmpeg
> https://forums.gentoo.org/viewtopic-p-7539696.html#7539696
> Thank you, and also thank you Gentoo people.
> If you didn't know, if you give in to NSA, by means of the now
> dominant SELinux, as almost all other Linuces have done, the world has
> lost so much, that you can't even phathom!
> Please don't!

And there is now so much more legible formatted and filled with 
previously missing links version:
Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539792.html#7539792

I hope the problem is clearly put, and undeniable.
I hope this helps the cause of freedom, yes, no exaggeration.

Miroslav Rovis
Zagreb, Croatia,
www.CroatiaFidelis.hr


From miro.rovis at croatiafidelis.hr  Tue Apr 22 15:05:21 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Tue, 22 Apr 2014 15:05:21 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <c20b5825fb55af6d58fa87f17c113e9e@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<loom.20140421T232710-490@post.gmane.org>
	<38199d860399b4c99b811a623646dcd6@croatiafidelis.hr>
	<c08bea68c2de190fb7987cb060069dd7@croatiafidelis.hr>
	<f166c0f62720a55adf214610aaefb308@croatiafidelis.hr>
	<c20b5825fb55af6d58fa87f17c113e9e@croatiafidelis.hr>
Message-ID: <bc9a8e1ebd55560259f0c20cc161717e@croatiafidelis.hr>

..[snip]..
> Libav (Avconv) Imposition on Users who want FFmpeg
> https://forums.gentoo.org/viewtopic-p-7539792.html#7539792
> 
> I hope the problem is clearly put, and undeniable.
> I hope this helps the cause of freedom, yes, no exaggeration.
> 
> Miroslav Rovis
> Zagreb, Croatia,
> www.CroatiaFidelis.hr
> 
More inconclusiveness:
There now I see two topics completely same, at two different addresses:

Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539696.html

and

Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539792.html

Just in case.

I am done exposing my issue.
I will now take a break from this issue.
I will gladly see if there is any sufficient support that FFmpeg should 
mean FFmpeg, and not something else.
Pls. note:
debian-multimedia.org
of Christian Marillat was badly pestered by Zacchiroli and gave in 
under legal threats and renamed the website:
deb-multimedia.org
Why, if we allow that libav uses the ffmpeg name, double standards.

But, as I said, I am off this topic.
I gave a lot and I am tired now. And have to do other things.
Thank you!
Miroslav Rovis

M.R.


From james.darnley at gmail.com  Tue Apr 22 15:34:42 2014
From: james.darnley at gmail.com (James Darnley)
Date: Tue, 22 Apr 2014 15:34:42 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
Message-ID: <53566FF2.7040202@gmail.com>

On 2014-04-21 22:18, miro.rovis at croatiafidelis.hr wrote:
> snip

If you are looking for language tips, here you go:

User fewer nested brackets.  A sentence should still make sense after
you eliminate/delete each bracketed section.  The example I created here
is limited and repetitive but should show you what I mean

> If you need to explain something (that's fine (but if you need to explain that, consider rewriting), do it), that's fine.
> If you need to explain something (that's fine, do it), that's fine.
> If you need to explain something, that's fine.

See?

Use fewer commas.  Your second paragraph is one entire sentence with so
many commas separating various things that make it very hard to keep
track of your point.

I don't want dissuade you from using English if it isn't your native
language.  You seem to speak it fairly well.  I am no English professor
either.  This is just to help you be more clear about your point.

Carl did give you a couple of links which may help clear up the
situation.  I will just add this.

FFmpeg (two upper-case 'F') is the name of the project/group here at
ffmpeg.org.  You can communicate with it/us at this mailing list.  It
provides these programs: ffmpeg, ffplay, ffserver, ffprobe.  It also
provides these libraries (among others): libavcodec, libavformat.  These
libraries are often referred to with some on these shorthands: libav,
libav*, lavc, lavf.

Libav (with the upper-case 'L') is the name of that other project/group
over at libav.org.  It provides the programs: avconv, avprobe.  It too
provides some libraries: libavcodec, libavformat.

Overall it is pretty confusing for those on the "outside".


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From tevans.uk at googlemail.com  Tue Apr 22 18:18:51 2014
From: tevans.uk at googlemail.com (Tom Evans)
Date: Tue, 22 Apr 2014 17:18:51 +0100
Subject: [FFmpeg-user] block issue ,help!help! config err with libx264.
In-Reply-To: <tencent_6E5C0344180D9B762B4B49A5@qq.com>
References: <tencent_6E5C0344180D9B762B4B49A5@qq.com>
Message-ID: <CAFHbX1KTq1FkEpPV4-Vu-r+KMCOsZbDoJXzjUx2Z+CG=8_0+uA@mail.gmail.com>

On Sun, Apr 20, 2014 at 11:05 AM, Goose <550532914 at qq.com> wrote:
> firstly, thank your reading and nicely help.
> I ve suffered by this block issue for several days.
>
>
> the config.log is in the attachment.
>
>
> I? intall ffmpeg on picuntu env, which is
>
> PicUntu, based on Ubuntu Qantal 12.10, The PicUntu project team announced the release of an Ubuntu-based Linux distribution, specifically developed to run on RK3066 chipset devices. According to the PicUntu team, it can run a company web server, corporate email server, central database server, or content management portal.More info here
> Mirror: LinuxUpgradeTool&RKBatchTool-marsboard-HDMI-picuntu-0.9-RC2.2-lubuntu-desktop-v1.2.img.7z
>
>

> $ head -n 1 config.log ; tail -n 4 config.log
# ./configure --enable-gpl --enable-libfaac --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libtheora --enable-libvorbis --enable-libx264
--enable-libxvid --enable-nonfree --enable-postproc --enable-version3
--enable-x11grab --extra-ldflags=-ldl --ld=-ldl
gcc -c -o /tmp/ffconf.PUHLKT1y.o /tmp/ffconf.CoN8mcQl.c
-ldl -ldl -o /tmp/ffconf.KVd7meiA /tmp/ffconf.PUHLKT1y.o
./configure: 597: ./configure: -ldl: not found
C compiler test failed.

So you explicitly set --extra-ldflags to "-ldl", and add it to
extra-ldflags, and the error is that libdl does not exist on your
system.

The whole "ldl" bits on your config line leads me to believe that you
first had other errors which you tried to fix with shotgun debugging,
adding arbitrary things. This obscures the real error.

Initially, try "./configure" - with no other arguments at all. Does
configure work then? Does it then compile?

Cheers

Tom

From miro.rovis at croatiafidelis.hr  Tue Apr 22 22:49:09 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Tue, 22 Apr 2014 22:49:09 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <53566FF2.7040202@gmail.com>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
Message-ID: <3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>

On 2014-04-22 15:34, James Darnley wrote:
> On 2014-04-21 22:18, miro.rovis at croatiafidelis.hr wrote:
>> snip
> 
> If you are looking for language tips, here you go:
> 
> User fewer nested brackets.  A sentence should still make sense after
> you eliminate/delete each bracketed section.  The example I created 
> here
> is limited and repetitive but should show you what I mean
> 
>> If you need to explain something (that's fine (but if you need to 
>> explain that, consider rewriting), do it), that's fine.
>> If you need to explain something (that's fine, do it), that's fine.
>> If you need to explain something, that's fine.
> 
> See?
> 
> Use fewer commas.  Your second paragraph is one entire sentence with 
> so
> many commas separating various things that make it very hard to keep
> track of your point.
> 
> I don't want dissuade you from using English if it isn't your native
> language.  You seem to speak it fairly well.  I am no English 
> professor
> either.  This is just to help you be more clear about your point.
> 
> Carl did give you a couple of links which may help clear up the
> situation.  I will just add this.
> 
> FFmpeg (two upper-case 'F') is the name of the project/group here at
> ffmpeg.org.  You can communicate with it/us at this mailing list.  It
> provides these programs: ffmpeg, ffplay, ffserver, ffprobe.  It also
> provides these libraries (among others): libavcodec, libavformat.  
> These
> libraries are often referred to with some on these shorthands: libav,
> libav*, lavc, lavf.
> 
> Libav (with the upper-case 'L') is the name of that other 
> project/group
> over at libav.org.  It provides the programs: avconv, avprobe.  It too
> provides some libraries: libavcodec, libavformat.
> 
> Overall it is pretty confusing for those on the "outside".
> 

You are right, James.
And thank you for more explanation.
But it traspires that I understand most of what you say.
I just wouldn't be able to enumerate the libraries and programs so 
well, and if anybody read my post in which I invested so much, it should 
be clear for him from what I wrote there that I do understand the basic 
difference btwn the two now incompatible projects that can't really be 
installed on a system together, because they basically do the same 
things, and one don't coexist with the other.

I suppose, James, that you are one of the FFmpeg team (you can see in 
the completed edition that I gave link to some half day later that I 
used "FFmpeg").

But. I don't get it. Does no one really care that this other project is 
using "ffmpeg" in such confusing way and then gets a lot of users that 
otherwise would be using FFmpeg?

And did anybody notice they multiplied the links to the topic, and you 
know it took me a while to first understand it (it's just e.g. 
symlinking, nothing else, but why?) and then to get over the frustration 
about it...

It is now at least four links, that open the same page... the:
"Libav (Avconv) Imposition on Users who want FFmpeg" page.

So no one cares that the Libav team (so not libav, but Libav, capital 
"L", the only difference, *!@#$&&!!), is taking away your users?

Cheers,
Miroslav Rovis,
Zagreb, Croatia
www.CroatiaFidelis.hr


From thanhnghile.py.bk at gmail.com  Tue Apr 22 12:28:53 2014
From: thanhnghile.py.bk at gmail.com (Thanh-Nghi Le)
Date: Tue, 22 Apr 2014 03:28:53 -0700 (PDT)
Subject: [FFmpeg-user] undefined reference errors when calling
	avdevice_register_all()
Message-ID: <1398162533491-4664883.post@n4.nabble.com>

When calling avdevice_register_all(), I got these errors:

/usr/local/lib/libavfilter.a(avf_showspectrum.o): In function
`config_output':
/opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:170:
undefined reference to `av_rdft_end'
/opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:171:
undefined reference to `av_rdft_init'
/usr/local/lib/libavfilter.a(avf_showspectrum.o): In function
`plot_spectrum_column':
/opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:298:
undefined reference to `av_rdft_calc'
/usr/local/lib/libavfilter.a(avf_showspectrum.o): In function `uninit':
/opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:106:
undefined reference to `av_rdft_end'
/usr/local/lib/libavfilter.a(vf_dctdnoiz.o): In function `dct_block':
/opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/vf_dctdnoiz.c:82: undefined
reference to `av_dct_calc'
/opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/vf_dctdnoiz.c:95: undefined
reference to `av_dct_calc'
/usr/local/lib/libavfilter.a(vf_dctdnoiz.o): In function `idct_block':
[......]

Here is my code:

#include <stdio.h>
#include <libavformat/avformat.h>

int main() {
	av_register_all();
	avdevice_register_all();

	return 0;
}

And I compile it as below:

gcc demo_capture.c -o demo_capture -pthread -lavformat -lavcodec -lavdevice
-lavfilter -lavutil -lswscale -lswresample -lpostproc -lz -lrt -lm -lx264
-lmp3lame -lvpx -lasound -lSDL

If I remove the function avdevice_register_all(), it's OK, no errors appears
then.
So what causes the above errors?



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/undefined-reference-errors-when-calling-avdevice-register-all-tp4664883.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From timothygu99 at gmail.com  Tue Apr 22 06:17:22 2014
From: timothygu99 at gmail.com (Timothy Gu)
Date: Mon, 21 Apr 2014 21:17:22 -0700
Subject: [FFmpeg-user] [FFmpeg-devel] ffmpeg installation enquiry
In-Reply-To: <CAPb1s=fgunjHN+O_EZBq44BCq-348ckOaPZ3zbfNZpM_tC1kNg@mail.gmail.com>
References: <CAPb1s=fgunjHN+O_EZBq44BCq-348ckOaPZ3zbfNZpM_tC1kNg@mail.gmail.com>
Message-ID: <CA+Yu1_Ws3aWJG1P4RhfXMkrLMtxzydhmE+3xR8peUimAawdS=g@mail.gmail.com>

Please do not send this kind of email to FFmpeg-devel mailing list.

On Mon, Apr 21, 2014 at 6:28 AM, Ade Miju <interprinze at gmail.com> wrote:
> Hi
> I have couples of Inquiries. Here are they

> [1] Does ffmpeg.org  has paid support service. What i mean by that is does
> ffmpeg.org has a group of specialist that can be paid to install ffmpeg for
> client, if requires

No, we don't provide paid support services.

For installation instructions see http://ffmpeg.org/download.html

If you would like to hire FFmpeg developers see
http://ffmpeg.org/consulting.html

If you have further questions send them to
https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/

Remember to read http://trac.ffmpeg.org/wiki/MailingListEtiquette
before sending emails.

[...]

Timothy

From u at pkh.me  Wed Apr 23 08:08:24 2014
From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=)
Date: Wed, 23 Apr 2014 08:08:24 +0200
Subject: [FFmpeg-user] undefined reference errors when calling
 avdevice_register_all()
In-Reply-To: <1398162533491-4664883.post@n4.nabble.com>
References: <1398162533491-4664883.post@n4.nabble.com>
Message-ID: <20140423060824.GB30379@leki>

On Tue, Apr 22, 2014 at 03:28:53AM -0700, Thanh-Nghi Le wrote:
> When calling avdevice_register_all(), I got these errors:
> 
> /usr/local/lib/libavfilter.a(avf_showspectrum.o): In function
> `config_output':
> /opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:170:
> undefined reference to `av_rdft_end'
> /opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:171:
> undefined reference to `av_rdft_init'
> /usr/local/lib/libavfilter.a(avf_showspectrum.o): In function
> `plot_spectrum_column':
> /opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:298:
> undefined reference to `av_rdft_calc'
> /usr/local/lib/libavfilter.a(avf_showspectrum.o): In function `uninit':
> /opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/avf_showspectrum.c:106:
> undefined reference to `av_rdft_end'
> /usr/local/lib/libavfilter.a(vf_dctdnoiz.o): In function `dct_block':
> /opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/vf_dctdnoiz.c:82: undefined
> reference to `av_dct_calc'
> /opt/install-ffmpeg/ffmpeg-2.1.4/libavfilter/vf_dctdnoiz.c:95: undefined
> reference to `av_dct_calc'
> /usr/local/lib/libavfilter.a(vf_dctdnoiz.o): In function `idct_block':
> [......]
> 
> Here is my code:
> 
> #include <stdio.h>
> #include <libavformat/avformat.h>
> 
> int main() {
> 	av_register_all();
> 	avdevice_register_all();
> 
> 	return 0;
> }
> 
> And I compile it as below:
> 
> gcc demo_capture.c -o demo_capture -pthread -lavformat -lavcodec -lavdevice
> -lavfilter -lavutil -lswscale -lswresample -lpostproc -lz -lrt -lm -lx264
> -lmp3lame -lvpx -lasound -lSDL
> 
> If I remove the function avdevice_register_all(), it's OK, no errors appears
> then.
> So what causes the above errors?
> 

Works for me here. This looks related to libavfilter dependencies to
libavcodec. How was FFmpeg built?

There was a dependency bug at some point but it was fixed and I checked
it's present in 2.1.4 so it should be fine...

-- 
Cl?ment B.
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From george at nsup.org  Wed Apr 23 09:21:29 2014
From: george at nsup.org (Nicolas George)
Date: Wed, 23 Apr 2014 09:21:29 +0200
Subject: [FFmpeg-user] undefined reference errors when calling
	avdevice_register_all()
In-Reply-To: <20140423060824.GB30379@leki>
References: <1398162533491-4664883.post@n4.nabble.com>
	<20140423060824.GB30379@leki>
Message-ID: <20140423072129.GA20090@phare.normalesup.org>

Le quartidi 4 flor?al, an CCXXII, Cl?ment B?sch a ?crit?:
> > gcc demo_capture.c -o demo_capture -pthread -lavformat -lavcodec -lavdevice
> > -lavfilter -lavutil -lswscale -lswresample -lpostproc -lz -lrt -lm -lx264
> > -lmp3lame -lvpx -lasound -lSDL
> Works for me here. This looks related to libavfilter dependencies to
> libavcodec. How was FFmpeg built?

My guess is: it was built static. This an ultra-FAQ, and not related to
FFmpeg: static libraries must be in dependency order. In doubt, use
pkg-config.

Regards,

-- 
  Nicolas George
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From cehoyos at ag.or.at  Wed Apr 23 14:56:41 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 23 Apr 2014 12:56:41 +0000 (UTC)
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
Message-ID: <loom.20140423T140348-894@post.gmane.org>

 <miro.rovis <at> croatiafidelis.hr> writes:

> I do understand the basic 
> difference btwn the two now incompatible projects

Could you elaborate?
To the best of my knowledge, FFmpeg is a superset of 
avconv, so there should be no incompatibilities.

> that can't really be installed on a system together, 
> because they basically do the same  things, and one 
> don't coexist with the other.

Since that is done (or can be done) on Debian now, I 
(very much so) hope that you are not right;-)

> But. I don't get it. Does no one really care that 
> this other project is using "ffmpeg" in such 
> confusing way and then gets a lot of users that 
> otherwise would be using FFmpeg?

I really don't understand:
Everybody here does care about it and many people do 
as much as they can about it - what do you think we 
should do (additionally)?


From xanadu at apost.plala.or.jp  Wed Apr 23 16:35:03 2014
From: xanadu at apost.plala.or.jp (=?utf-8?B?5a6u5p2RIOWFrOeUtw==?=)
Date: Wed, 23 Apr 2014 23:35:03 +0900
Subject: [FFmpeg-user] Enabling libx265
In-Reply-To: <6020031A-5DFD-405F-9E39-71F3B75CEF58@apost.plala.or.jp>
References: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>
	<6020031A-5DFD-405F-9E39-71F3B75CEF58@apost.plala.or.jp>
Message-ID: <2E584549-DB1F-4A9E-BB2D-85D47DD66D45@apost.plala.or.jp>


2014/04/21 13:54??? ?? <xanadu at apost.plala.or.jp> ?????

> 
> 2014/04/18 1:26??? ?? <xanadu at apost.plala.or.jp> ?????
> 
>> then ffmpeg configure script exits with "ERROR: x265 not found" error.
> 
> The cause of this issue might be a bug with x265.  It can be solved by:
> 1.Add -lstdc++ into CFLAGS
> or
> 2.Edit x265.pc as follows:
> --- a/x265.pc	2014-04-16 14:50:20.000000000 +0900
> +++ b/x265.pc	2014-04-21 10:54:32.000000000 +0900
> @@ -6,6 +6,6 @@
> Name: x265
> Description: H.265/HEVC video encoder
> Version: 0.9
> -Libs: -L${libdir} -lx265
> +Libs: -L${libdir} -lx265 -lstdc++
> Libs.private: -lc++ -l/Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/lib/clang/5.1/lib/darwin/libclang_rt.osx.a
> Cflags: -I${includedir}

After talking with x265 development team, adding "-lstdc++" flags should not be the corrective action.  
Instead, configure ffmpeg with --pkg-config-flags=--static option, this issue is solved.
By adding --static, x265.pc gives -lc++ or -lstdc++ with --cflags to ffmpeg.  

In addition, x265.pc?s Libs.private line contains absolute path of libclang_rt.osx.a (Mac) to be linked.  This was minor bug in x265.  I confirmed ?-l? is removed today.


// Miya

From cehoyos at ag.or.at  Wed Apr 23 17:28:22 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 23 Apr 2014 15:28:22 +0000 (UTC)
Subject: [FFmpeg-user] Possible issue with asf muxer
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
Message-ID: <loom.20140423T172750-634@post.gmane.org>

Massimo Battistel <battistel <at> gmail.com> writes:

> The output file has lots of issue with ffplay: 
> a/v sync, bad audio, ...

This should be fixed in current FFmpeg git head, 
thank you for the report!

Carl Eugen


From renaux.jacky at orange.fr  Wed Apr 23 18:02:17 2014
From: renaux.jacky at orange.fr (jacky)
Date: Wed, 23 Apr 2014 18:02:17 +0200
Subject: [FFmpeg-user] sc_threshold value
Message-ID: <5357E409.4010404@orange.fr>

Hi every body

May be it is a stupid question but I have'nt found answer
sc_treshold value is related to scene detection
leftover it is enabled and sc_threshold 1 disabled
is there others options (0,2,3....)
how much the option 1 degrades quality on amphi courses where there
are slow changes (except during zoom )
do you know some links to get some insights ?
best regards

jacky


From phpdev at ehrhardt.nl  Wed Apr 23 18:23:49 2014
From: phpdev at ehrhardt.nl (Jan Ehrhardt)
Date: Wed, 23 Apr 2014 18:23:49 +0200
Subject: [FFmpeg-user] Enabling libx265
References: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>
	<6020031A-5DFD-405F-9E39-71F3B75CEF58@apost.plala.or.jp>
	<2E584549-DB1F-4A9E-BB2D-85D47DD66D45@apost.plala.or.jp>
Message-ID: <smpfl95t530tabi2i8oi15coe7c6psnqf6@4ax.com>

?? ?? in gmane.comp.video.ffmpeg.user (Wed, 23 Apr 2014 23:35:03 +0900):
>
>> Name: x265
>> Description: H.265/HEVC video encoder
>> Version: 0.9
>> -Libs: -L${libdir} -lx265
>> +Libs: -L${libdir} -lx265 -lstdc++
>> Libs.private: -lc++ -l/Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/lib/clang/5.1/lib/darwin/libclang_rt.osx.a
>> Cflags: -I${includedir}
>
>After talking with x265 development team, adding "-lstdc++" flags should not be the corrective action.  
>Instead, configure ffmpeg with --pkg-config-flags=--static option, this issue is solved.
>By adding --static, x265.pc gives -lc++ or -lstdc++ with --cflags to ffmpeg.  

I did not know the --pkg-config-flags syntax, but this boils down to the
same solution:
http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=5&t=4&start=20#p5338

Put this at the beginning of your ./configure commandline:

pkg_config='pkg-config --static'

This will add the Libs.private to the linker command as well. See man
pkg-config.

Jan


From putta.pv at gmail.com  Wed Apr 23 19:00:23 2014
From: putta.pv at gmail.com (PUTTA SUKESH)
Date: Wed, 23 Apr 2014 22:30:23 +0530
Subject: [FFmpeg-user] How to stream H.264 through RTP with ffmpeg?
Message-ID: <CAPHyg86tVfPLMcnVbOzbqg2fN150NPsF-6P84oFMHkHmhJsEGQ@mail.gmail.com>

I am able to stream live streaming using web cam.I use the ffserver.conf
which is present in ffmpeg source.
commands i used
ffserver -f ffserver.conf &
ffmpeg -f v4l2 -s 320x240 -r 25 -i /dev/video0
http://localhost:8090/feed1.ffm<https://chnmail.hcl.com/owa/redir.aspx?C=vBZG0wmuH0OA4sQ79YuRKcbc9_6rMtEIRqxWE6GqBiAkb66uacX6ynjn55koQWS1rT_6ZOtTk28.&URL=http%3a%2f%2flocalhost%3a8090%2ffeed1.ffm>
ffplay http://localhost:8090/test1.mpg<https://chnmail.hcl.com/owa/redir.aspx?C=vBZG0wmuH0OA4sQ79YuRKcbc9_6rMtEIRqxWE6GqBiAkb66uacX6ynjn55koQWS1rT_6ZOtTk28.&URL=http%3a%2f%2flocalhost%3a8090%2ftest1.mpg>(I
AM ABLE TO PLAY)

now I need to stream h264 file (test.h264) through ffmpeg and ffserver
throuth rtp. what are the configurations I need to do in ffserver.conf.

From miro.rovis at croatiafidelis.hr  Wed Apr 23 21:09:40 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Wed, 23 Apr 2014 21:09:40 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <loom.20140423T140348-894@post.gmane.org>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
Message-ID: <ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>

On 2014-04-23 14:56, Carl Eugen Hoyos wrote:
> <miro.rovis <at> croatiafidelis.hr> writes:
> 
>> I do understand the basic
>> difference btwn the two now incompatible projects
> 
> Could you elaborate?
> To the best of my knowledge, FFmpeg is a superset of
> avconv, so there should be no incompatibilities.
> 
>> that can't really be installed on a system together,
>> because they basically do the same  things, and one
>> don't coexist with the other.
> 
> Since that is done (or can be done) on Debian now, I
> (very much so) hope that you are not right;-)

I'm happy Debian has changed in their attitude! I'm not an
insider, just a passionate user of good and free (read GNU
and compatible) computing, and that probably results from
the huge page that you gave me:
https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=729203
(among others), which I dedicated some time for, but not
enough to reache there. :-)

>> But. I don't get it. Does no one really care that
>> this other project is using "ffmpeg" in such
>> confusing way and then gets a lot of users that
>> otherwise would be using FFmpeg?
> 
> I really don't understand:
> Everybody here does care about it and many people do
> as much as they can about it - what do you think we
> should do (additionally)?

Well, I hope that Gentoo devs are reading this, and
now they won't conclude that FFmpeg people don't care.
But it would be great if some of the FFmpeg leaders or
if you have a board or council or somesuch, sent a request
that you want the name FFmpeg for FFmpeg so users are not
fooled like I was.

Such formal requests are never very easily disregarded, IIUC.

That was what I hoped for, and why I went public and invested
time and effort into it.

I love FFmpeg because it gave me soo much, I couldn't have
done my huge collection without it, and I was trying to give
back by pointing to the issue and hopefully initiating a
righting of this wrong!

Thank you, Carl! Thank you all you FFmpeg people. Pls keep up!
Keep free, keep GNU!

Miroslav Rovis
Zagreb, Croatia
www.CroatiaFidelis.hr

P.S. On another note. Do you know why I was confused, and why
other people certainly are, about libav, and why I thought, in
some of my previous posts in this thread that you made some
compromise with Libav?
Have a look at this page, but not pronto, no. First imagine
that you are you user visiting it first time, or that you heard
and even that you used both ffmpeg and avconv (now marketed as
Libav), but that you don't deal with it daily. OK. Now open
that page on ffmpeg's domain:

http://ffmpeg.org/mailman/listinfo

Do you see:

Libav-user

list in bottom. I know the mimicry is theirs, and intentional, just
like all of their machinations, and I'm not saying much can be done,
but what could be done is: point out there on that page that that
is not the Avcon (now marketed Libav) which is, on Gentoo, not
compatible with FFmpeg, or something better and nicer and clearer.


From finlay at moeraki.com  Wed Apr 23 21:51:03 2014
From: finlay at moeraki.com (John Finlay)
Date: Wed, 23 Apr 2014 12:51:03 -0700
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <loom.20140423T140348-894@post.gmane.org>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>	<53566FF2.7040202@gmail.com>	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
Message-ID: <535819A7.1070304@moeraki.com>

On 4/23/2014 5:56 AM, Carl Eugen Hoyos wrote:
>   <miro.rovis <at> croatiafidelis.hr> writes:
>
>> I do understand the basic
>> difference btwn the two now incompatible projects
> Could you elaborate?
> To the best of my knowledge, FFmpeg is a superset of
> avconv, so there should be no incompatibilities.
>
>
In my experience the incompatibilities between avconv and ffmpeg are 
mostly command line differences e.g map 0.1 vs map 0:1 and differences 
in placement of options e.g before or after the -i options. In terms of 
features ffmpeg is a superset of avconv.

John

From miro.rovis at croatiafidelis.hr  Wed Apr 23 22:30:37 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Wed, 23 Apr 2014 22:30:37 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <535819A7.1070304@moeraki.com>
References: "\"<a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>"
	<53566FF2.7040202@gmail.com>"
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<535819A7.1070304@moeraki.com>
Message-ID: <737c7ee0513485c2214e76f147e9926e@croatiafidelis.hr>

On 2014-04-23 21:51, John Finlay wrote:
> On 4/23/2014 5:56 AM, Carl Eugen Hoyos wrote:
>>   <miro.rovis <at> croatiafidelis.hr> writes:
>> 
>>> I do understand the basic
>>> difference btwn the two now incompatible projects
>> Could you elaborate?
>> To the best of my knowledge, FFmpeg is a superset of
>> avconv, so there should be no incompatibilities.
>> 
>> 
> In my experience the incompatibilities between avconv and ffmpeg are
> mostly command line differences e.g map 0.1 vs map 0:1 and differences
> in placement of options e.g before or after the -i options. In terms
> of features ffmpeg is a superset of avconv.
> 
> John

I am not really a dev to match the discussion in this aspect, as I said 
many times here and on Gentoo Forums, I started this thread to keep 
FFmpeg from being hijacked users from.

Carl Eugen Hoyos, I don't know about you John Finlay, but Carl and 
other developers, please, I believe, but again, I'm not expert, that 
there is an opening in there on that Gentoo Forums thread, and maybe you 
could sort things out, and take back the ffmpeg name to mean only FFmpeg 
and not Libav (Avconv).

Pls look up here what a TomWij Developer kindly wrote, and which is far 
above my undestanding without so much more effert than I can easily find 
time for:

Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-p-7539612.html#7540338
(as well as my reply
https://forums.gentoo.org/viewtopic-p-7539612.html#7540688 )
There, I did what I said there to TomWij in my reply.

Hope the new version FFmpeg is fixed in Gentoo and marvellous as ever!

Miroslav Rovis
www.CroatiaFidelis.hr


From cehoyos at ag.or.at  Wed Apr 23 23:14:27 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 23 Apr 2014 21:14:27 +0000 (UTC)
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>	<53566FF2.7040202@gmail.com>	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<535819A7.1070304@moeraki.com>
Message-ID: <loom.20140423T231038-495@post.gmane.org>

John Finlay <finlay <at> moeraki.com> writes:

> In my experience the incompatibilities between avconv 
> and ffmpeg are mostly command line differences e.g 
> map 0.1 vs map 0:1

This sounds unlikely:
The map 0.1 syntax was dropped from FFmpeg three years 
ago iirc.

> and differences in placement of options e.g before 
> or after the -i options.

This is at least misleading:
There are input options (which have to be placed 
before -i) and output options (which have to be placed 
after the filename following -i). There are input and 
output options with the same name. And there are 
global options where the position does not matter.
That said, I don't think there is any difference 
between FFmpeg and any forks with respect to the 
placement of options.

> In terms of features ffmpeg is a superset of avconv.

This is correct to the best of my knowledge.

Carl Eugen


From cehoyos at ag.or.at  Wed Apr 23 23:23:48 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 23 Apr 2014 21:23:48 +0000 (UTC)
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
Message-ID: <loom.20140423T231538-959@post.gmane.org>

 <miro.rovis <at> croatiafidelis.hr> writes:

> Well, I hope that Gentoo devs are reading this

(Why should they?)

> But it would be great if some of the FFmpeg leaders or
> if you have a board or council or somesuch, sent a request
> that you want the name FFmpeg for FFmpeg so users are not
> fooled like I was.

You may misunderstand:
I stopped counting the number of requests and similar that 
were sent by the FFmpeg project leader and developers a 
very long time ago...

> Such formal requests are never very easily disregarded, IIUC.

I hate to tell you but you are - unfortunately - wrong!

Carl Eugen


From thanhnghile.py.bk at gmail.com  Wed Apr 23 11:24:37 2014
From: thanhnghile.py.bk at gmail.com (Thanh-Nghi Le)
Date: Wed, 23 Apr 2014 02:24:37 -0700 (PDT)
Subject: [FFmpeg-user] undefined reference errors when calling
	avdevice_register_all()
In-Reply-To: <1398162533491-4664883.post@n4.nabble.com>
References: <1398162533491-4664883.post@n4.nabble.com>
Message-ID: <1398245077611-4664892.post@n4.nabble.com>

Yes, I built FFmpeg as static library. The configuration:

$ ./configure --enable-gpl --enable-libmp3lame  --enable-libvpx
--enable-libx264 --enable-nonfree --enable-postproc --enable-version3
$ make
$ sudo checkinstall --pkgname=ffmpeg --pkgversion="5-$(./version.sh)"
--backup=no --deldoc=yes --default

I have added more headers into my code:  demo_capture.c
<http://ffmpeg-users.933282.n4.nabble.com/file/n4664892/demo_capture.c>  
I want to open /dev/video0 but it still throws some more diferrent errors: 
log.txt <http://ffmpeg-users.933282.n4.nabble.com/file/n4664892/log.txt>  



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/undefined-reference-errors-when-calling-avdevice-register-all-tp4664883p4664892.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From miro.rovis at croatiafidelis.hr  Wed Apr 23 23:44:17 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Wed, 23 Apr 2014 23:44:17 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <loom.20140423T231538-959@post.gmane.org>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
	<loom.20140423T231538-959@post.gmane.org>
Message-ID: <47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>

On 2014-04-23 23:23, Carl Eugen Hoyos wrote:
> <miro.rovis <at> croatiafidelis.hr> writes:
> 
>> Well, I hope that Gentoo devs are reading this
> 
> (Why should they?)
> 
>> But it would be great if some of the FFmpeg leaders or
>> if you have a board or council or somesuch, sent a request
>> that you want the name FFmpeg for FFmpeg so users are not
>> fooled like I was.
> 
> You may misunderstand:
> I stopped counting the number of requests and similar that
> were sent by the FFmpeg project leader and developers a
> very long time ago...
> 
>> Such formal requests are never very easily disregarded, IIUC.
> 
> I hate to tell you but you are - unfortunately - wrong!
> 
> Carl Eugen
> 

I see. Sad.
Another sad thing: something somehow made this your reply above on the 
thread, and from the archived mail of mine to which this above is your 
reply aooear wrongly.

What I mean is,
look at the thread:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/thread.html
and open the currently last two (not others, other are OK):

[FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg   
miro.rovis at croatiafidelis.hr
[FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg   Carl 
Eugen Hoyos
[FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg   John 
Finlay
[FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg   
miro.rovis at croatiafidelis.hr
[FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg   Carl 
Eugen Hoyos

I can't represent this better, sorry.
Again, open just the last two, the mine and yours.
Mine appears correctly as:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021048.html

buy what I see under yours is:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021049.html
from the thread

and, if I open (which in this demonstration I suggest to open) mine, 
and click on the next message from that mine, I see again:

what opened up for me from the thread page:

https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021049.html

But that is not this reply of yours, no.

I'm pasting what opens to me:

PASTE START
[FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
Carl Eugen Hoyos cehoyos at ag.or.at
Wed Apr 23 23:14:27 CEST 2014

     Previous message: [FFmpeg-user] Libav (Avconv) Imposition on Users 
who want FFmpeg
     Next message: [FFmpeg-user] [FFmpeg-devel] ffmpeg installation 
enquiry
     Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]

John Finlay <finlay <at> moeraki.com> writes:

> In my experience the incompatibilities between avconv
> and ffmpeg are mostly command line differences e.g
> map 0.1 vs map 0:1

This sounds unlikely:
The map 0.1 syntax was dropped from FFmpeg three years
ago iirc.

> and differences in placement of options e.g before
> or after the -i options.

This is at least misleading:
There are input options (which have to be placed
before -i) and output options (which have to be placed
after the filename following -i). There are input and
output options with the same name. And there are
global options where the position does not matter.
...[snip]...
PASTE END
I'v also screencast this, so there be no one will be
able to say I'm lying:
2abcd85f801c73e59b7caaaf1d16a29e513d270efe924643fbffdf6b962384c3  
Screen_140423_222051_naibd6.mkv
and taking pcapdump:
c89f8a343d04eea607035f5e5f4d3041ee0de5420ffbd53c349839768a42f4ce  
dump_140423_222052_naibd6.pcapng

Miroslav Rovis
Zagreb, Croatia
www.CroatiaFidelis.hr

BTW, I'm using:
  ffmpeg -f pulse -f x11grab -s xga -r 10 -i :0.0 Screen_`date 
+%y%m%d_%H%M%S`_`hostname`.mkv
on Debian (but it's Christian Marillat's FFmpeg)

for screencasting




From tgleason at colorhythm.com  Wed Apr 23 23:47:38 2014
From: tgleason at colorhythm.com (Tom Gleason)
Date: Wed, 23 Apr 2014 17:47:38 -0400
Subject: [FFmpeg-user] ffmpeg stalls
Message-ID: <CAN9Yyd2jiW2+-tXXbFRnoiW8_mB-NhcDKFiEcQfyD77-fY3W=g@mail.gmail.com>

For what reason would FFMPEG stall during a transcode?

I have a small EC2 instance, and can easily transcode 10 seconds of a file,
but when I try to transcode full (say 10 min) files, it always stalls, top
showing 0% cpu. What would be the likely issue?

--
Tom Gleason

From lou at lrcd.com  Thu Apr 24 00:18:04 2014
From: lou at lrcd.com (Lou)
Date: Wed, 23 Apr 2014 14:18:04 -0800
Subject: [FFmpeg-user] ffmpeg stalls
In-Reply-To: <CAN9Yyd2jiW2+-tXXbFRnoiW8_mB-NhcDKFiEcQfyD77-fY3W=g@mail.gmail.com>
References: <CAN9Yyd2jiW2+-tXXbFRnoiW8_mB-NhcDKFiEcQfyD77-fY3W=g@mail.gmail.com>
Message-ID: <20140423141804.7f334399@lrcd.com>

On Wed, 23 Apr 2014 17:47:38 -0400
Tom Gleason <tgleason at colorhythm.com> wrote:

> For what reason would FFMPEG stall during a transcode?
> 
> I have a small EC2 instance, and can easily transcode 10 seconds of a file,
> but when I try to transcode full (say 10 min) files, it always stalls, top
> showing 0% cpu. What would be the likely issue?
> 
> --
> Tom Gleason

Please include your ffmpeg command and the complete console output if
possible.

Does this occur with a particular input or with all inputs? Please
include a sample input if the issue is limited to specific inputs.

Can this issue be duplicated on another system?

From finlay at moeraki.com  Thu Apr 24 00:20:10 2014
From: finlay at moeraki.com (John Finlay)
Date: Wed, 23 Apr 2014 15:20:10 -0700
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <loom.20140423T231038-495@post.gmane.org>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>	<53566FF2.7040202@gmail.com>	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>	<loom.20140423T140348-894@post.gmane.org>	<535819A7.1070304@moeraki.com>
	<loom.20140423T231038-495@post.gmane.org>
Message-ID: <53583C9A.5010903@moeraki.com>

On 4/23/2014 2:14 PM, Carl Eugen Hoyos wrote:
> John Finlay <finlay <at> moeraki.com> writes:
>
>> In my experience the incompatibilities between avconv
>> and ffmpeg are mostly command line differences e.g
>> map 0.1 vs map 0:1
> This sounds unlikely:
> The map 0.1 syntax was dropped from FFmpeg three years
> ago iirc.

The avconv version of ffmpeg in Ubuntu accepts -map 0.1 and -map 0:1 but 
the version of ffmpeg I use and avconv only accept -map 0:1
>> and differences in placement of options e.g before
>> or after the -i options.
> This is at least misleading:
> There are input options (which have to be placed
> before -i) and output options (which have to be placed
> after the filename following -i). There are input and
> output options with the same name. And there are
> global options where the position does not matter.
> That said, I don't think there is any difference
> between FFmpeg and any forks with respect to the
> placement of options.
>

Avconv works with -vf options before the -i options but the version of 
ffmpeg I use seems to ignore -vf options unless they are after the -i 
options.

The version of ffmpeg I use is 1.2.2 and the version of avconv is 0.8.10 
on ubuntu 12.04. I suppose these differences could have been eliminated 
in later versions. My point is that there is confusion between avconv's 
ffmpeg and the real ffmpeg on ubuntu systems that lead to 
incompatibilities in command line use.

John

From xanadu at apost.plala.or.jp  Thu Apr 24 05:00:36 2014
From: xanadu at apost.plala.or.jp (=?utf-8?B?5a6u5p2RIOWFrOeUtw==?=)
Date: Thu, 24 Apr 2014 12:00:36 +0900
Subject: [FFmpeg-user] Enabling libx265
In-Reply-To: <smpfl95t530tabi2i8oi15coe7c6psnqf6@4ax.com>
References: <678C8AA7-A705-4532-9B72-6D71B6AE1069@apost.plala.or.jp>
	<6020031A-5DFD-405F-9E39-71F3B75CEF58@apost.plala.or.jp>
	<2E584549-DB1F-4A9E-BB2D-85D47DD66D45@apost.plala.or.jp>
	<smpfl95t530tabi2i8oi15coe7c6psnqf6@4ax.com>
Message-ID: <EDDE30D3-1FE8-4E23-848A-46D7BBE85226@apost.plala.or.jp>


2014/04/24 1:23?Jan Ehrhardt <phpdev at ehrhardt.nl> ?????

> ?? ?? in gmane.comp.video.ffmpeg.user (Wed, 23 Apr 2014 23:35:03 +0900):
>> 
>> Instead, configure ffmpeg with --pkg-config-flags=--static option, this issue is solved.
>> By adding --static, x265.pc gives -lc++ or -lstdc++ with --cflags to ffmpeg.  
> 
> I did not know the --pkg-config-flags syntax, but this boils down to the
> same solution:
> http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=5&t=4&start=20#p5338
> 
> Put this at the beginning of your ./configure commandline:
> 
> pkg_config='pkg-config --static'
> 
> This will add the Libs.private to the linker command as well. See man
> pkg-config.

Thanks.  I?ve confirmed that this option also solves the issue with Libs.private.


// Miya

From mrukant at gmail.com  Thu Apr 24 05:29:37 2014
From: mrukant at gmail.com (Mrukant Popat)
Date: Wed, 23 Apr 2014 20:29:37 -0700
Subject: [FFmpeg-user] Not able to get constant 25fps for H.264 HD Transport
	Stream after transcoding
Message-ID: <CADDQthvDtxcBAT_h95QjQOefJ=9mnfbBcyjqsp=hfk4LNOBqkw@mail.gmail.com>

Below command line generates a Constant bitrate TS of 5000Kbps , but the
video elementary stream inside does not show 25 fps when using MediaInfo.
Infact when I play it with mplayer, it plays with 50fps. However, VLC plays
with 25fps.

Here is the command line

ffmpeg.exe -loglevel debug -i input.ts -pix_fmt yuv420p -r 25.0 -c:s copy
-c:v libx264 -preset slow -tune film  -x264opts
tff:fps=25.0:bitrate=4500:ref=3:threads=6:no-scenecut
:keyint=50:vbv-bufsize=180:vbv-maxrate=4500:videoformat=pal -c:a libfaac
 -ab 192000 -ar 48000 -f mpegts -mpegts_pmt_start_pid 0x64 -map 0:0 -map
0:1 -map 0:2 -
map 0:3 -streamid 0:1000 -streamid 1:1001 -streamid 2:1002 -streamid 3:1005
-muxrate 5M output.ts > c:\commandoutput.txt 2>&1



Here is the link to detailed output to this command --->
https://drive.google.com/file/d/0B4vc-v98vRCkZ3BIelhGNVBWSE9IbW1kQk5yV1k2bHZBalJR/edit?usp=sharing



I am looking to get video with 25 fps and also the value of 25 fps set in
VUI of SEI inside H.264 stream.


Thank you,
Mrukant Popat

From miro.rovis at croatiafidelis.hr  Thu Apr 24 05:53:18 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Thu, 24 Apr 2014 05:53:18 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
	<loom.20140423T231538-959@post.gmane.org>
	<47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>
Message-ID: <08628653094bed272050866b11fd63dc@croatiafidelis.hr>

...[snip]
> I'v also screencast this, so there be no one will be
> able to say I'm lying:
> 2abcd85f801c73e59b7caaaf1d16a29e513d270efe924643fbffdf6b962384c3
> Screen_140423_222051_naibd6.mkv
> and taking pcapdump:
> c89f8a343d04eea607035f5e5f4d3041ee0de5420ffbd53c349839768a42f4ce
> dump_140423_222052_naibd6.pcapng
...[snip]
> BTW, I'm using:
>  ffmpeg -f pulse -f x11grab -s xga -r 10 -i :0.0 Screen_`date
> +%y%m%d_%H%M%S`_`hostname`.mkv
> on Debian (but it's Christian Marillat's FFmpeg)
> 
> for screencasting
> 
> Miroslav Rovis
> Zagreb, Croatia
> www.CroatiaFidelis.hr

I can be off now (unless I need to reply more). But I have to
tell that I wanted to demonstrate what my (Bash) program Flowstamp
can do, using FFmpeg and ImageMagick and other:

http://www.croatiafidelis.hr/gnu/Flowstamp/

It is a flowstamped screencast of what I wrote in my previous post.
(I was wrong, in most aspects.)

From u at pkh.me  Thu Apr 24 07:47:16 2014
From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=)
Date: Thu, 24 Apr 2014 07:47:16 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <53583C9A.5010903@moeraki.com>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<535819A7.1070304@moeraki.com>
	<loom.20140423T231038-495@post.gmane.org>
	<53583C9A.5010903@moeraki.com>
Message-ID: <20140424054716.GC30379@leki>

On Wed, Apr 23, 2014 at 03:20:10PM -0700, John Finlay wrote:
> On 4/23/2014 2:14 PM, Carl Eugen Hoyos wrote:
> >John Finlay <finlay <at> moeraki.com> writes:
> >
> >>In my experience the incompatibilities between avconv
> >>and ffmpeg are mostly command line differences e.g
> >>map 0.1 vs map 0:1
> >This sounds unlikely:
> >The map 0.1 syntax was dropped from FFmpeg three years
> >ago iirc.
> 
> The avconv version of ffmpeg in Ubuntu accepts -map 0.1 and -map 0:1 but the
> version of ffmpeg I use and avconv only accept -map 0:1
> >>and differences in placement of options e.g before
> >>or after the -i options.
> >This is at least misleading:
> >There are input options (which have to be placed
> >before -i) and output options (which have to be placed
> >after the filename following -i). There are input and
> >output options with the same name. And there are
> >global options where the position does not matter.
> >That said, I don't think there is any difference
> >between FFmpeg and any forks with respect to the
> >placement of options.
> >
> 
> Avconv works with -vf options before the -i options but the version of

No. Not in the recent versions. They originate from that change.

> ffmpeg I use seems to ignore -vf options unless they are after the -i
> options.
> 
> The version of ffmpeg I use is 1.2.2 and the version of avconv is 0.8.10 on

Those two mismatch. ffmpeg 1.2.2 is more recent that libav 0.8.10

-- 
Cl?ment B.
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From pastas4 at gmail.com  Thu Apr 24 08:07:27 2014
From: pastas4 at gmail.com (Dainius (GreatEmerald))
Date: Thu, 24 Apr 2014 09:07:27 +0300
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <08628653094bed272050866b11fd63dc@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
	<loom.20140423T231538-959@post.gmane.org>
	<47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>
	<08628653094bed272050866b11fd63dc@croatiafidelis.hr>
Message-ID: <CABhjJhMfGMmHE61SaqOAk-RzQiwL-=WAPa8zpfGQdZ+N7mXwDg@mail.gmail.com>

I haven't read your post on the forum (as it's still down), but if
it's only about the virtual, then it's confusing, yes, but not really
something they can do much about. Now firstly you are not supposed to
install virtuals by themselves, it's just a virtual package that
others could depend on (depending on virtual/ffmpeg means that the
program doesn't care about whether it's FFmpeg or Libav that is behind
/usr/bin/ffmpeg). The reason why it's called "ffmpeg" is because
FFmpeg was first, and thus everyone depends on a binary of that name.
I'm not sure to what else it could be renamed.

Whether the virtual prefers one over the other I'm not really sure.
The ebuild for virtual/ffmpeg-0.10.3 lists media-video/ffmpeg first,
while virtual/ffmpeg-9 lists media-video/libav first (and the
numbering seems to prefer Libav's versioning). The current stabilised
version is virtual/ffmpeg-0.10.3. Overall dealing with virtuals in
Gentoo is kind of a pain since there is no easy way to specify what
you want it to use under the hood, with the best way to do that is to
pre-emptively install one of the packages the virtual depends on.

If the virtual does now prefer Libav, then it's rather unfortunate,
but I'd rather hear both sides of the argument for that change.
Personally I don't see why they would do that, as FFmpeg being the
superset naturally provides more and there are programs that depend
specifically on FFmpeg (I haven't heard any that specifically depend
on Libav).

Well, that said, there is a bit of a problem in that regard with
GStreamer. GStreamer 1.X only officially supports Libav for some
reason (I think it's because they want to support both and thus opt
for the smaller feature set), and thus when installing it with FFmpeg
Portage outputs a warning. There are also some reported bugs against
GStreamer when using FFmpeg, but then it will probably be fixed in
GStreamer 1.4:
https://bugzilla.gnome.org/show_bug.cgi?id=721077#c24

From cehoyos at ag.or.at  Thu Apr 24 08:49:13 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 24 Apr 2014 06:49:13 +0000 (UTC)
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
	<loom.20140423T231538-959@post.gmane.org>
	<47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>
	<08628653094bed272050866b11fd63dc@croatiafidelis.hr>
	<CABhjJhMfGMmHE61SaqOAk-RzQiwL-=WAPa8zpfGQdZ+N7mXwDg@mail.gmail.com>
Message-ID: <loom.20140424T084700-946@post.gmane.org>

Dainius (GreatEmerald <pastas4 <at> gmail.com> writes:

> If the virtual does now prefer Libav, then it's rather unfortunate,
> but I'd rather hear both sides of the argument for that change.

http://blogs.gentoo.org/lu_zero/2013/01/18/the-case-of-defaults-libav-vs-ffmpeg/
http://aballier.wordpress.com/2013/01/18/ffmpeg-vs-libav-a-distribution-maintainer-point-of-view-almost-two-years-after-the-split/
(It is necessary to read the second one to "understand" 
the first one which is full of the usual lies.)

> Well, that said, there is a bit of a problem in that regard with
> GStreamer.

If you know of any bugs in FFmpeg that affect GStreamer 
please report them to us!
(So far, all reports ended in "you are right, it works 
fine".)

Carl Eugen


From pastas4 at gmail.com  Thu Apr 24 09:49:27 2014
From: pastas4 at gmail.com (Dainius (GreatEmerald))
Date: Thu, 24 Apr 2014 10:49:27 +0300
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <loom.20140424T084700-946@post.gmane.org>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
	<loom.20140423T231538-959@post.gmane.org>
	<47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>
	<08628653094bed272050866b11fd63dc@croatiafidelis.hr>
	<CABhjJhMfGMmHE61SaqOAk-RzQiwL-=WAPa8zpfGQdZ+N7mXwDg@mail.gmail.com>
	<loom.20140424T084700-946@post.gmane.org>
Message-ID: <CABhjJhNEXc1qRr5BN6wjhLQ6JN_ZjuLeUbP7NdM9jtCG=69FVg@mail.gmail.com>

On Thu, Apr 24, 2014 at 9:49 AM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> http://blogs.gentoo.org/lu_zero/2013/01/18/the-case-of-defaults-libav-vs-ffmpeg/
> http://aballier.wordpress.com/2013/01/18/ffmpeg-vs-libav-a-distribution-maintainer-point-of-view-almost-two-years-after-the-split/
> (It is necessary to read the second one to "understand"
> the first one which is full of the usual lies.)

Hmm, I thought this was old news, given that the date is the beginning
of 2013, but then the virtual/ffmpeg changelog says it just takes that
long until "ffmpeg-9" gets stabilised. So this is exactly about that
change, yes. And yeap, it looks like it was changed for no real reason
at all, the official rationale being:

> This change is being done due to various reasons like matching default with Fedora and Debian, or due to fact that some projects which are high-profile (eg sh*tload of people use them) will be probably libav only. One example being gst-libav which is in return required by libreoffice-4 which is due release in about month. To go for least pain for the user we decided to move from default ffmpeg to default libav library.

In other words, "maybe GStreamer will only work with libav" and "some
other distros switched" (and may have since then switched back,
actually; in Debian there are now quite a few people fighting to get
FFmpeg back as the default). Maybe there is enough following in Gentoo
now to also switch back, on the occasion of FFmpeg 2.2 being released?

> If you know of any bugs in FFmpeg that affect GStreamer
> please report them to us!
> (So far, all reports ended in "you are right, it works
> fine".)

No, I don't; the link I posted is actually about a GStreamer bug when
using FFmpeg, not the other way around, as far as I can tell.

From cehoyos at ag.or.at  Thu Apr 24 09:54:46 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Thu, 24 Apr 2014 07:54:46 +0000 (UTC)
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
	<loom.20140423T231538-959@post.gmane.org>
	<47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>
	<08628653094bed272050866b11fd63dc@croatiafidelis.hr>
	<CABhjJhMfGMmHE61SaqOAk-RzQiwL-=WAPa8zpfGQdZ+N7mXwDg@mail.gmail.com>
	<loom.20140424T084700-946@post.gmane.org>
	<CABhjJhNEXc1qRr5BN6wjhLQ6JN_ZjuLeUbP7NdM9jtCG=69FVg@mail.gmail.com>
Message-ID: <loom.20140424T095056-445@post.gmane.org>

Dainius (GreatEmerald <pastas4 <at> gmail.com> writes:

> In other words, "maybe GStreamer will only work with 
> libav" and "some other distros switched"

That is exactly the argumentation that was used.

> (and may have since then switched back,

Users of the "other distros" are held hostage by 
one former FFmpeg developer who was part of the 
group who tried to steal the project and is among 
the people who regularly spread lies about FFmpeg.

> actually; in Debian there are now quite a few 
> people fighting to get FFmpeg back as the default).

There is actually a package that would allow to 
install FFmpeg.
But it appears that it cannot be uploaded because 
of the many security issues in avconv.
(sic! the security team is blocking the upload 
because of the many security issues with the fork)

Carl Eugen


From h.reindl at thelounge.net  Thu Apr 24 11:31:59 2014
From: h.reindl at thelounge.net (Reindl Harald)
Date: Thu, 24 Apr 2014 11:31:59 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <CABhjJhNEXc1qRr5BN6wjhLQ6JN_ZjuLeUbP7NdM9jtCG=69FVg@mail.gmail.com>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>	<53566FF2.7040202@gmail.com>	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>	<loom.20140423T140348-894@post.gmane.org>	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>	<loom.20140423T231538-959@post.gmane.org>	<47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>	<08628653094bed272050866b11fd63dc@croatiafidelis.hr>	<CABhjJhMfGMmHE61SaqOAk-RzQiwL-=WAPa8zpfGQdZ+N7mXwDg@mail.gmail.com>	<loom.20140424T084700-946@post.gmane.org>
	<CABhjJhNEXc1qRr5BN6wjhLQ6JN_ZjuLeUbP7NdM9jtCG=69FVg@mail.gmail.com>
Message-ID: <5358DA0F.1000402@thelounge.net>



Am 24.04.2014 09:49, schrieb Dainius (GreatEmerald):
>> This change is being done due to various reasons like matching 
>> default with Fedora and Debian

this is another lie in that context

* Fedora does *not* ship ffmpeg at all for patent reasons
* rpmfusion does
* rpmfusion does *not* ship libav, still the origin ffmpeg

Jason Garrett-Glaser is spreading lies over years

he even told me on the x264 list that the command "ffmpeg" is
deprecated and ffmpeg is dead and replaced by libav at all

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From miro.rovis at croatiafidelis.hr  Thu Apr 24 13:28:12 2014
From: miro.rovis at croatiafidelis.hr (miro.rovis at croatiafidelis.hr)
Date: Thu, 24 Apr 2014 13:28:12 +0200
Subject: [FFmpeg-user] Libav (Avconv) Imposition on Users who want FFmpeg
In-Reply-To: <08628653094bed272050866b11fd63dc@croatiafidelis.hr>
References: <a9cead821cbb3df245f95928cb51c68c@croatiafidelis.hr>
	<53566FF2.7040202@gmail.com>
	<3e839655845675ecb71e0b27e00fbf11@croatiafidelis.hr>
	<loom.20140423T140348-894@post.gmane.org>
	<ae4432314b910f90daf5aac35a875449@croatiafidelis.hr>
	<loom.20140423T231538-959@post.gmane.org>
	<47efdd47be6bb8948a257e2f5e9cbf25@croatiafidelis.hr>
	<08628653094bed272050866b11fd63dc@croatiafidelis.hr>
Message-ID: <3209b46dca1209db000cd18f4999bdbb@croatiafidelis.hr>

On 2014-04-24 05:53, miro.rovis at croatiafidelis.hr wrote:
> ...[snip]
>> I'v also screencast this, so there be no one will be
>> able to say I'm lying:
>> 2abcd85f801c73e59b7caaaf1d16a29e513d270efe924643fbffdf6b962384c3
>> Screen_140423_222051_naibd6.mkv
>> and taking pcapdump:
>> c89f8a343d04eea607035f5e5f4d3041ee0de5420ffbd53c349839768a42f4ce
>> dump_140423_222052_naibd6.pcapng
> ...[snip]
>> BTW, I'm using:
>>  ffmpeg -f pulse -f x11grab -s xga -r 10 -i :0.0 Screen_`date
>> +%y%m%d_%H%M%S`_`hostname`.mkv
>> on Debian (but it's Christian Marillat's FFmpeg)
>> for screencasting
>> Miroslav Rovis
>> Zagreb, Croatia
>> www.CroatiaFidelis.hr
> 
> I can be off now (unless I need to reply more). But I have to
> tell that I wanted to demonstrate what my (Bash) program Flowstamp
> can do, using FFmpeg and ImageMagick and other:
> 
> http://www.croatiafidelis.hr/gnu/Flowstamp/
> 
> It is a flowstamped screencast of what I wrote in my previous post.
> (I was wrong, in most aspects.)

I would have liked if I could now have gone away because I can't be up 
to the discussion, to incomplete being my understanding, I am not really 
a dev (only a fraction of a dev).

But I must show that the links to Gentoo Forums that I gave were all 
the time consistent, and some have now started to not show.

I have gone now through all the messages as they are archived on:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/thread.html#21052
and freely available to the world.
At some point in the past, all of these opened exactly the same page, 
or, what else, in other words they were all probably (sym)links to the 
same page, that bears the same exact title as this thread that you are 
reading, either by means of your mail agent, or from the web:
https://forums.gentoo.org/viewtopic-p-7539612.html
https://forums.gentoo.org/viewtopic-p-7539612.html#7540338
https://forums.gentoo.org/viewtopic-p-7539612.html#7540688
https://forums.gentoo.org/viewtopic-p-7539696.html
https://forums.gentoo.org/viewtopic-p-7539696.html#7539696
https://forums.gentoo.org/viewtopic-p-7539792.html
https://forums.gentoo.org/viewtopic-p-7539792.html#7539792
https://forums.gentoo.org/viewtopic-t-987268.html
(that title of course being:
Libav (Avconv) Imposition on Users who want FFmpeg

I first noticed that "inconclusiveness" in this post of this thread:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021032.html
But the post on Gentoo I was first given a link as:
https://forums.gentoo.org/viewtopic-p-7539612.html
(when it went down as explained in imediately consecutive posts, and 
why bother
wording any unnecessary suspicion on that)

But I have seen even more addresses that usually were taking over the 
old
address (which took over from the then old), as I was posting, and I 
don't want
to (not just because it's laborious), but if I wanted I could prove so, 
because
I was often screencasting, as I explained here:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021058.html

But this is just not right. Why would someone do so?
I have to now recall that I already noticed that also here:
https://ffmpeg.org/pipermail/ffmpeg-user/2014-April/021035.html
where find:
> And did anybody notice they multiplied the links to the topic, and you
> know it took me a while to first understand it (it's just e.g.
> symlinking, nothing else, but why?) and then to get over the 
> frustration
> about it...

But the problem is some of the links, most notable the very first one 
that was given
to me as I posted on Gentoo Forums, is now not working!

So whoever reads the archives will have problem figuring out what 
happened.

And that is not a *good thing*.

What I get it I try to access the very first address that the topic I 
opened was assigned is:

  phpBB : Critical Error

Error updating sessions table

and nothing shows.

But the link, if one opens the section of the Forums:
Portage & Programming
https://forums.gentoo.org/viewforum-f-8.html
  is currently there under:
Libav (Avconv) Imposition on Users who want FFmpeg
https://forums.gentoo.org/viewtopic-t-989196.html

and whether now the other links open that exact same page, I am sick 
now of making any more trying about it.

Instead, what I posted, and for conciliatory reasons I posted not 
anything that puts anybody in bad light, here:
Flowstamp Demo
http://www.croatiafidelis.hr/gnu/Flowstamp/
is when there was nice, conciliatory talk on this issue in that topic 
on Gentoo Forums.

I am happy that developers, really capable people, have now been 
discussing this issue in the open!

I wish everybody all the best, to all, but to some the prosperity I 
wish for them is neither in my computer not by imposition on users, but 
through fair competition, if they can,

Miroslav Rovis
Zagreb, Croatia,
www.CroatiaFidelis.hr


From tgleason at colorhythm.com  Thu Apr 24 16:02:30 2014
From: tgleason at colorhythm.com (Tom Gleason)
Date: Thu, 24 Apr 2014 10:02:30 -0400
Subject: [FFmpeg-user] ffmpeg stalls
In-Reply-To: <20140423141804.7f334399@lrcd.com>
References: <CAN9Yyd2jiW2+-tXXbFRnoiW8_mB-NhcDKFiEcQfyD77-fY3W=g@mail.gmail.com>
	<20140423141804.7f334399@lrcd.com>
Message-ID: <CAN9Yyd2wOksQA5d6TqDTmHxvgqFdrCf4uZSScXTeOKsaZ65WXw@mail.gmail.com>

Sorry, turns out it was an issue with our use of proc_open on php.
http://www.php.net/manual/en/function.proc-open.php

We had to use 'a' instead of 'w' on the STDERR pipe to accomodate the
length of ffmpeg output, otherwise it got blocked.

$process = @proc_open($command, array(1 => array('pipe', 'w'), 2 =>
array('pipe', 'a')), $pipe, NULL, NULL, array('bypass_shell' => true));

--
Tom Gleason
Senior Software Developer
Colorhythm
http://www.colorhythm.com<http://www.google.com/url?q=http%3A%2F%2Fwww.colorhythm.com&sa=D&sntz=1&usg=AFQjCNHcZwtg8mRjlK3RaR8L4QQndc4KnA>

Main Office: +1 415-399-9921
Fax: +1 415-399-9928
Mobile: +1 347-537-8465

tgleason at colorhythm.com


On Wed, Apr 23, 2014 at 6:18 PM, Lou <lou at lrcd.com> wrote:

> On Wed, 23 Apr 2014 17:47:38 -0400
> Tom Gleason <tgleason at colorhythm.com> wrote:
>
> > For what reason would FFMPEG stall during a transcode?
> >
> > I have a small EC2 instance, and can easily transcode 10 seconds of a
> file,
> > but when I try to transcode full (say 10 min) files, it always stalls,
> top
> > showing 0% cpu. What would be the likely issue?
> >
> > --
> > Tom Gleason
>
> Please include your ffmpeg command and the complete console output if
> possible.
>
> Does this occur with a particular input or with all inputs? Please
> include a sample input if the issue is limited to specific inputs.
>
> Can this issue be duplicated on another system?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From thorswill at hotmail.com  Thu Apr 24 19:52:21 2014
From: thorswill at hotmail.com (Tobie Horswill)
Date: Thu, 24 Apr 2014 13:52:21 -0400
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
Message-ID: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>

I'm having difficulties getting the new AVfoundation input device working under Mac OS X 10.7

I've rebuilt from the latest source which includes the AVFoundation additions with --enable-indev=avfoundation but I still get an : Unknown input format 'avfoundation' whenever I try to use it.

I gather this should work under OS X 10.7+ correct? I unfortunately can't run Mavericks on this computer.

Any input is welcome.

Thanks,

--
TH 		 	   		  

From thanhnghile.py.bk at gmail.com  Thu Apr 24 12:06:11 2014
From: thanhnghile.py.bk at gmail.com (Thanh-Nghi Le)
Date: Thu, 24 Apr 2014 03:06:11 -0700 (PDT)
Subject: [FFmpeg-user] undefined reference errors when calling
	avdevice_register_all()
In-Reply-To: <20140423072129.GA20090@phare.normalesup.org>
References: <1398162533491-4664883.post@n4.nabble.com>
	<20140423060824.GB30379@leki>
	<20140423072129.GA20090@phare.normalesup.org>
Message-ID: <1398333971420-4664917.post@n4.nabble.com>

Yes, I built FFmpeg as static library. The configuration: 

$ ./configure --enable-gpl --enable-libmp3lame  --enable-libvpx
--enable-libx264 --enable-nonfree --enable-postproc --enable-version3 
$ make 
$ sudo checkinstall --pkgname=ffmpeg --pkgversion="5-$(./version.sh)"
--backup=no --deldoc=yes --default 

I have added more headers into my code:  demo_capture.c
<http://ffmpeg-users.933282.n4.nabble.com/file/n4664892/demo_capture.c>  
I want to open /dev/video0 but it still throws some more diferrent errors: 
log.txt <http://ffmpeg-users.933282.n4.nabble.com/file/n4664892/log.txt>  

I guess I included not enough headers or they are not in correct order.



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/undefined-reference-errors-when-calling-avdevice-register-all-tp4664883p4664917.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From bszabolcs at gmail.com  Thu Apr 24 10:44:06 2014
From: bszabolcs at gmail.com (Szabolcs Balogh)
Date: Thu, 24 Apr 2014 11:44:06 +0300
Subject: [FFmpeg-user] Generating HLS stream with ffmpeg programmatically
Message-ID: <CAGRht3i_hWHH2NveCsNoqeJ-+JO8Y3vx1oHj2FBtmpBZ+0wpRg@mail.gmail.com>

Hi,

I'm trying to convert an mp4 file to HLS stream (*.ts files + *.m3u8 file)
on Android using ffmpeg through Android NDK.

If specified the output file as "index.m3u8" and the output format as
"hls". I also specified "hls_time" and "hls_list_size" using
av_opt_set_int. It works properly and generates the *.ts files and the m3u8
file. My problem is, that the m3u8 file contains only the segment name.



*#EXTINF:10.0,segment0001.ts*
But I need something like:



*#EXTINF:10.0,http://www.myserver.com/segment0001.ts
<http://www.myserver.com/segment0001.ts>*

I tried to specify "segment_list_entry_prefix", but it seems that it has no
effect for hls. I checked the source code, and there is no such parameter
for hls.

How can I specify a full URL for the segment name in the m3u8 file?

Thanks in advance,
Szabolcs Balogh

From the.el.tea at gmail.com  Fri Apr 25 04:16:28 2014
From: the.el.tea at gmail.com (Michael LeSauvage)
Date: Thu, 24 Apr 2014 20:16:28 -0600
Subject: [FFmpeg-user] Can ffmpeg overlay time indexed text?
Message-ID: <CAOBwzE+KEzY2ucz4pRciPp-GGkRqWXM1TQJ3AR8boRATVFJC_w@mail.gmail.com>

New user question.  I did searches on this but only found discussions back
from 2011.

I have a video and a file of related data that is time indexed relative to
the video.  It's Excel data now, but I can make it any arbitrary format -
probably something like <vid time>, <data1>, <data2>

The data is not linear and cannot be normalized.  So there might be a data
points at 20.210  |  20.215  |  20.230  | 22.205 for example.

I would like the data for each data point overlaid on video, remaining on
the screen until replaced by the next data point.

Is this a capability I could find in ffmpeg?



Mike

From anatol2002 at gmail.com  Fri Apr 25 07:06:21 2014
From: anatol2002 at gmail.com (Anatol)
Date: Fri, 25 Apr 2014 08:06:21 +0300
Subject: [FFmpeg-user] Generating HLS stream with ffmpeg programmatically
In-Reply-To: <CAGRht3i_hWHH2NveCsNoqeJ-+JO8Y3vx1oHj2FBtmpBZ+0wpRg@mail.gmail.com>
References: <CAGRht3i_hWHH2NveCsNoqeJ-+JO8Y3vx1oHj2FBtmpBZ+0wpRg@mail.gmail.com>
Message-ID: <CAPCGgdQHM-LNbianzePUySCE75BrcMEKTv_ePCPi0TD_J_8pWQ@mail.gmail.com>

M3U8 is just a text file that you can easily parse and change to whatever
you like.


On Thu, Apr 24, 2014 at 11:44 AM, Szabolcs Balogh <bszabolcs at gmail.com>wrote:

> Hi,
>
> I'm trying to convert an mp4 file to HLS stream (*.ts files + *.m3u8 file)
> on Android using ffmpeg through Android NDK.
>
> If specified the output file as "index.m3u8" and the output format as
> "hls". I also specified "hls_time" and "hls_list_size" using
> av_opt_set_int. It works properly and generates the *.ts files and the m3u8
> file. My problem is, that the m3u8 file contains only the segment name.
>
>
>
> *#EXTINF:10.0,segment0001.ts*
> But I need something like:
>
>
>
> *#EXTINF:10.0,http://www.myserver.com/segment0001.ts
> <http://www.myserver.com/segment0001.ts>*
>
> I tried to specify "segment_list_entry_prefix", but it seems that it has no
> effect for hls. I checked the source code, and there is no such parameter
> for hls.
>
> How can I specify a full URL for the segment name in the m3u8 file?
>
> Thanks in advance,
> Szabolcs Balogh
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>

From george at nsup.org  Fri Apr 25 10:06:04 2014
From: george at nsup.org (Nicolas George)
Date: Fri, 25 Apr 2014 10:06:04 +0200
Subject: [FFmpeg-user] Can ffmpeg overlay time indexed text?
In-Reply-To: <CAOBwzE+KEzY2ucz4pRciPp-GGkRqWXM1TQJ3AR8boRATVFJC_w@mail.gmail.com>
References: <CAOBwzE+KEzY2ucz4pRciPp-GGkRqWXM1TQJ3AR8boRATVFJC_w@mail.gmail.com>
Message-ID: <20140425080604.GA14488@phare.normalesup.org>

Le quintidi 5 flor?al, an CCXXII, Michael LeSauvage a ?crit?:
> I have a video and a file of related data that is time indexed relative to
> the video.  It's Excel data now, but I can make it any arbitrary format -
> probably something like <vid time>, <data1>, <data2>
> 
> The data is not linear and cannot be normalized.  So there might be a data
> points at 20.210  |  20.215  |  20.230  | 22.205 for example.
> 
> I would like the data for each data point overlaid on video, remaining on
> the screen until replaced by the next data point.
> 
> Is this a capability I could find in ffmpeg?

Just think to your "time-indexed text" as subtitles, and you will find
plenty of solutions.

If you can process your input to compute the end time for each entry, then I
suggest to use the ASS format, that is the one used internally so you would
be avoiding conversions.

Regards,

-- 
  Nicolas George
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From thilo.borgmann at mail.de  Fri Apr 25 14:13:59 2014
From: thilo.borgmann at mail.de (Thilo Borgmann)
Date: Fri, 25 Apr 2014 14:13:59 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
Message-ID: <535A5187.5030107@mail.de>

Am 24.04.14 19:52, schrieb Tobie Horswill:
> I'm having difficulties getting the new AVfoundation input device working
> under Mac OS X 10.7
> 
> I've rebuilt from the latest source which includes the AVFoundation additions
> with --enable-indev=avfoundation but I still get an : Unknown input format
> 'avfoundation' whenever I try to use it.
> 
> I gather this should work under OS X 10.7+ correct? 

Yes.

> I unfortunately can't run
> Mavericks on this computer.
> 
> Any input is welcome.

Check your config.log, it will most likely be a compiler/framework header issue.

If you can't try another (or updated) compiler, you might be able to switch to
the qtkit input device, still functional up to 10.9.

If in doubt you should make your config.log available for us to see.

-Thilo

From neve_capricorn at gmx.de  Fri Apr 25 14:57:57 2014
From: neve_capricorn at gmx.de (neve_capricorn at gmx.de)
Date: Fri, 25 Apr 2014 14:57:57 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <535A5187.5030107@mail.de>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
	<535A5187.5030107@mail.de>
Message-ID: <535A5BD5.90708@gmx.de>

hi thilo,

can i use av-foundation in ffmpeg to grab audio devices with the 
underlaying coreAudio?

currently i was able to capute from my sight camera videoframes, but can 
i also address an audio device for capturing audio frames under osx 10.9 
with ffmpeg included qtkit or avfoundation?

If yes, how can i grab it?

Thx for help.
Tom

Am 25.04.14 14:13, schrieb Thilo Borgmann:
> Am 24.04.14 19:52, schrieb Tobie Horswill:
>> I'm having difficulties getting the new AVfoundation input device working
>> under Mac OS X 10.7
>>
>> I've rebuilt from the latest source which includes the AVFoundation additions
>> with --enable-indev=avfoundation but I still get an : Unknown input format
>> 'avfoundation' whenever I try to use it.
>>
>> I gather this should work under OS X 10.7+ correct?
> Yes.
>
>> I unfortunately can't run
>> Mavericks on this computer.
>>
>> Any input is welcome.
> Check your config.log, it will most likely be a compiler/framework header issue.
>
> If you can't try another (or updated) compiler, you might be able to switch to
> the qtkit input device, still functional up to 10.9.
>
> If in doubt you should make your config.log available for us to see.
>
> -Thilo
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From t.rapp at noa-audio.com  Fri Apr 25 16:05:09 2014
From: t.rapp at noa-audio.com (Tobias Rapp)
Date: Fri, 25 Apr 2014 16:05:09 +0200
Subject: [FFmpeg-user] Getting MPEG2 video stream bitrate
Message-ID: <20140425160509.6bde5be1e07111b47dbeb50d@noa-audio.com>

Hello,

I am trying to get the bitrate of an MXF/MPEG2 file using ffprobe. The
summary printed to stderr shows the bitrate info "max. 50000 kb/s" but the
value printed to stdout says "bit_rate=N/A".

How can I systematically retrieve the bitrate of an MPEG2 stream?

My command line was:

$ ffprobe -show_streams -select_streams v -probesize 10000000 omneoan_720x576.mxf

stdout/stderr output files are attached.

Best regards,
Tobias
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From cehoyos at ag.or.at  Fri Apr 25 16:20:45 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 25 Apr 2014 14:20:45 +0000 (UTC)
Subject: [FFmpeg-user] Getting MPEG2 video stream bitrate
References: <20140425160509.6bde5be1e07111b47dbeb50d@noa-audio.com>
Message-ID: <loom.20140425T161712-777@post.gmane.org>

Tobias Rapp <t.rapp <at> noa-audio.com> writes:

> I am trying to get the bitrate of an MXF/MPEG2 file using ffprobe. The
> summary printed to stderr shows the bitrate info "max. 50000 kb/s" but the
> value printed to stdout says "bit_rate=N/A".

The stream does not save a (constant or average) bitrate, only the 
maximum bitrate is stored.

> ffprobe version 2.2.1 Copyright (c) 2007-2014 the FFmpeg developers

Unrelated:
Please always test current FFmpeg git head before asking here.

> [mxf  <at>  0x27bd2e0] Field dominance 0 support is not implemented. 
> Update your FFmpeg version to the newest one from Git. If the problem 
> still occurs, it means that your file has a feature which has not been 
> implemented.
> [mxf  <at>  0x27bd2e0] If you want to help, upload a sample of this file to
> ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel 
> mailing list.

And it appears that we need such a sample.
(I don't know much about mxf though.)

Carl Eugen


From krueger at lesspain.de  Fri Apr 25 16:57:34 2014
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 25 Apr 2014 16:57:34 +0200
Subject: [FFmpeg-user] Interlaced frames not recognized as such
Message-ID: <CAEnqZEULyykapJTh6qN2ckxe5B8jbd4ig4BROMO_2KOqLaHZUg@mail.gmail.com>

Summary: Frames from an interlaced file created using ffmpeg's
tinterlace and fieldorder filters are not recognized as interlaced by
ffmpeg. Now I wonder if I used the wrong CL for creating the file or
if interlaced detection fails in this case.

I created an interlaced file by this command line (full output included):

ffmpeg -i color_frames_png_1080_50p.mov -vf
"tinterlace=interleave_top,fieldorder=tff" -vcodec mpeg2video -q 1
-pix_fmt yuv422p -r 25 -y color_frames_mpeg2_1080_50i_tff_3f.mov
ffmpeg version N-62704-g92cc6d5 Copyright (c) 2000-2014 the FFmpeg developers
  built on Apr 25 2014 15:47:16 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --enable-gpl
  libavutil      52. 78.100 / 52. 78.100
  libavcodec     55. 59.100 / 55. 59.100
  libavformat    55. 37.101 / 55. 37.101
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'color_frames_png_1080_50p.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 512
    compatible_brands: qt
    encoder         : Lavf55.24.100
  Duration: 00:00:00.12, start: 0.000000, bitrate: 3202 kb/s
    Stream #0:0(eng): Video: png (png  / 0x20676E70), rgb24, 1920x1080
[SAR 1:1 DAR 16:9], 3151 kb/s, 50 fps, 50 tbr, 12800 tbn, 12800 tbc
(default)
    Metadata:
      handler_name    : DataHandler
Output #0, mov, to 'color_frames_mpeg2_1080_50i_tff_3f.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 512
    compatible_brands: qt
    encoder         : Lavf55.37.101
    Stream #0:0(eng): Video: mpeg2video (m2v1 / 0x3176326D), yuv422p,
1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 12800 tbn, 25 tbc
(default)
    Metadata:
      handler_name    : DataHandler
Stream mapping:
  Stream #0:0 -> #0:0 (png -> mpeg2video)
Press [q] to stop, [?] for help
frame=    3 fps=0.0 q=1.0 Lsize=    1996kB time=00:00:00.08
bitrate=204347.0kbits/s
video:1995kB audio:0kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 0.038675%

And then ran ffprobe on it like this:

ffprobe -show_frames color_frames_mpeg2_1080_50i_tff_3f.mov
ffprobe version N-62704-g92cc6d5 Copyright (c) 2007-2014 the FFmpeg developers
  built on Apr 25 2014 15:47:16 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --enable-gpl
  libavutil      52. 78.100 / 52. 78.100
  libavcodec     55. 59.100 / 55. 59.100
  libavformat    55. 37.101 / 55. 37.101
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'color_frames_mpeg2_1080_50i_tff_3f.mov':
  Metadata:
    major_brand     : qt
    minor_version   : 512
    compatible_brands: qt
    encoder         : Lavf55.37.101
  Duration: 00:00:00.12, start: 0.000000, bitrate: 136231 kb/s
    Stream #0:0(eng): Video: mpeg2video (4:2:2) (xd5e / 0x65356478),
yuv422p(tv), 1920x1080 [SAR 1:1 DAR 16:9], 136178 kb/s, 25 fps, 25
tbr, 12800 tbn, 50 tbc (default)
    Metadata:
      handler_name    : DataHandler
[FRAME]
media_type=video
key_frame=1
pkt_pts=0
pkt_pts_time=0.000000
pkt_dts=0
pkt_dts_time=0.000000
best_effort_timestamp=0
best_effort_timestamp_time=0.000000
pkt_duration=512
pkt_duration_time=0.040000
pkt_pos=36
pkt_size=428246
width=1920
height=1080
pix_fmt=yuv422p
sample_aspect_ratio=1:1
pict_type=I
coded_picture_number=0
display_picture_number=0
interlaced_frame=0
top_field_first=0
repeat_pict=0
[/FRAME]
[FRAME]
media_type=video
key_frame=0
pkt_pts=512
pkt_pts_time=0.040000
pkt_dts=512
pkt_dts_time=0.040000
best_effort_timestamp=512
best_effort_timestamp_time=0.040000
pkt_duration=512
pkt_duration_time=0.040000
pkt_pos=428282
pkt_size=890034
width=1920
height=1080
pix_fmt=yuv422p
sample_aspect_ratio=1:1
pict_type=P
coded_picture_number=1
display_picture_number=0
interlaced_frame=0
top_field_first=0
repeat_pict=0
[/FRAME]
[FRAME]
media_type=video
key_frame=0
pkt_pts=1024
pkt_pts_time=0.080000
pkt_dts=N/A
pkt_dts_time=N/A
best_effort_timestamp=1024
best_effort_timestamp_time=0.080000
pkt_duration=512
pkt_duration_time=0.040000
pkt_pos=1318316
pkt_size=724400
width=1920
height=1080
pix_fmt=yuv422p
sample_aspect_ratio=1:1
pict_type=P
coded_picture_number=2
display_picture_number=0
interlaced_frame=0
top_field_first=0
repeat_pict=0
[/FRAME]

A you can see, none of the frames are recognized as such by ffprobe.
The actual problem I had was that using the api
AVFrame.interlaced_frame was set to 0 which seems incorrect. So I do
not think it is an ffprobe problem. I merely used it to build a
reproducible test case with fftools. The file is reported as
interlaced by ffmbc, mediainfo if anyone cares to know.

Thanks for any hints,

Robert

From cehoyos at ag.or.at  Fri Apr 25 17:27:10 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Fri, 25 Apr 2014 15:27:10 +0000 (UTC)
Subject: [FFmpeg-user] Interlaced frames not recognized as such
References: <CAEnqZEULyykapJTh6qN2ckxe5B8jbd4ig4BROMO_2KOqLaHZUg@mail.gmail.com>
Message-ID: <loom.20140425T172005-500@post.gmane.org>

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

> Summary: Frames from an interlaced file created using ffmpeg's
> tinterlace and fieldorder filters are not recognized as interlaced by
> ffmpeg.

Did you test the idet filter?
Only this filter (of FFmpeg's features) can tell you if the *content* 
of a video stream is interlaced or progressive (the filter can not 
yet identify telecined content, it is probably not very difficult to 
add this).

> Now I wonder if I used the wrong CL for creating the file or
> if interlaced detection fails in this case.

> I created an interlaced file by this command line (full output included):

Just to clarify:
You created a progressive MPEG-2 stream showing interlaced frames.

> ffmpeg -i color_frames_png_1080_50p.mov -vf
> "tinterlace=interleave_top,fieldorder=tff" -vcodec mpeg2video -q 1
> -pix_fmt yuv422p -r 25 -y color_frames_mpeg2_1080_50i_tff_3f.mov

There is an option to create an interlaced MPEG-2 stream but you did 
not use it (-ilme -flags +ildct), so a progressive stream was created...

Unrelated:
Old MEncoder documentation suggests never to use -qscale 1 but 
-qscale 2.

> interlaced_frame=0

which is reported by ffprobe.

Note that ffprobe will report "interlaced_frame" for a (sd) dvb recording 
no matter how progressive the content is.
(Our tv providers always flag "interlaced")

> The file is reported as interlaced by ffmbc, mediainfo if anyone 
> cares to know.

Sounds like a bug to me.

Carl Eugen


From krueger at lesspain.de  Fri Apr 25 17:42:26 2014
From: krueger at lesspain.de (=?UTF-8?Q?Robert_Kr=C3=BCger?=)
Date: Fri, 25 Apr 2014 17:42:26 +0200
Subject: [FFmpeg-user] Interlaced frames not recognized as such
In-Reply-To: <loom.20140425T172005-500@post.gmane.org>
References: <CAEnqZEULyykapJTh6qN2ckxe5B8jbd4ig4BROMO_2KOqLaHZUg@mail.gmail.com>
	<loom.20140425T172005-500@post.gmane.org>
Message-ID: <CAEnqZEUVip1hpP=0R-5b3DLXPniPwU21JN889KkeDszWASnp_g@mail.gmail.com>

On Fri, Apr 25, 2014 at 5:27 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> Robert Kr?ger <krueger <at> lesspain.de> writes:
>
>> Summary: Frames from an interlaced file created using ffmpeg's
>> tinterlace and fieldorder filters are not recognized as interlaced by
>> ffmpeg.
>
> Did you test the idet filter?
> Only this filter (of FFmpeg's features) can tell you if the *content*
> of a video stream is interlaced or progressive (the filter can not
> yet identify telecined content, it is probably not very difficult to
> add this).

I am making the interlaced file artificially on purpose for testing,
so I know what content will be in there in this case and there I am
interlacing progressive frames (again on purpose).

>
>> Now I wonder if I used the wrong CL for creating the file or
>> if interlaced detection fails in this case.
>
>> I created an interlaced file by this command line (full output included):
>
> Just to clarify:
> You created a progressive MPEG-2 stream showing interlaced frames.
>
>> ffmpeg -i color_frames_png_1080_50p.mov -vf
>> "tinterlace=interleave_top,fieldorder=tff" -vcodec mpeg2video -q 1
>> -pix_fmt yuv422p -r 25 -y color_frames_mpeg2_1080_50i_tff_3f.mov
>
> There is an option to create an interlaced MPEG-2 stream but you did
> not use it (-ilme -flags +ildct), so a progressive stream was created...

sh....! Of course, -flags +ildct was missing. Just hadn't done that on
the CL for a long time.

Thanks a lot for the quick help. Works a expected now.

>
> Unrelated:
> Old MEncoder documentation suggests never to use -qscale 1 but
> -qscale 2.

Oh, that's news to me. I have been using 1 whenever I wanted highest
quality possible. I didn't find the values documented however (I just
looked at http://ffmpeg.org/ffmpeg.html).


>
>> interlaced_frame=0
>
> which is reported by ffprobe.
>
> Note that ffprobe will report "interlaced_frame" for a (sd) dvb recording
> no matter how progressive the content is.
> (Our tv providers always flag "interlaced")
>
>> The file is reported as interlaced by ffmbc, mediainfo if anyone
>> cares to know.
>
> Sounds like a bug to me.

I think they use the container metadata and the resulting mov file
indeed has the fiel atom set to interlaced content.

Regards and thanks,

Robert

From asdfg1945 at gmail.com  Fri Apr 25 08:06:08 2014
From: asdfg1945 at gmail.com (Arachnia)
Date: Thu, 24 Apr 2014 23:06:08 -0700 (PDT)
Subject: [FFmpeg-user] duration errors for help
Message-ID: <1398405968411-4664924.post@n4.nabble.com>

I make a test for converting a rtsp stream from a ipcamera to ts and m3u8
recently.

when I only encode a video stream, it works perfectly. My command is

ffmpeg  -f rtsp -rtsp_transport tcp -i 'rtsp://192.168.0.91:554/rtsp'
-vcodec libx264 -crf 24 -b:v 128k -s 352x288 -profile:v baseline -pix_fmt
yuv420p -acodec aac -ac 2 -b:a 1k -ar 22050 -strict -2 -flags -global_header
-map 0 -f segment -segment_time 10 -segment_list_size 2 -segment_list
test.m3u8 -segment_list_type m3u8 -segment_format mpegts live%08d.ts

Ipcam closed audio output, so the command only had a video encoding. I run
the command for 24 hours, it works fine.
Duration every ts file is 10s. 

ffprobe live00008529.ts
ffprobe version 2.2.1 Copyright (c) 2007-2014 the FFmpeg developers
  built on Apr 14 2014 16:45:50 with gcc 4.4.7 (GCC) 20120313 (Red Hat
4.4.7-4)
  configuration: --enable-libx264 --enable-gpl --enable-pthreads
--enable-avfilter --enable-filter=movie --enable-libfaac --enable-nonfree
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mpegts, from 'live00008529.ts':
  Duration: 00:00:09.96, start: 85294.360000, bitrate: 166 kb/s
  Program 1 
    Metadata:
      service_name    : RTSP Session
      service_provider: FFmpeg
    Stream #0:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] /
0x001B), yuv420p, 352x288, 25 fps, 25 tbr, 90k tbn, 50 tbc

When I opened the audio output, it was a nightmare.
I ran the command about one and half an hour, the duration began to changed.

ffprobe 'live00000546.ts
ffprobe version 2.2.1 Copyright (c) 2007-2014 the FFmpeg developers
  built on Apr 22 2014 09:45:19 with gcc 4.4.7 (GCC) 20120313 (Red Hat
4.4.7-4)
  configuration: --enable-libx264 --enable-gpl --enable-pthreads
--enable-avfilter --enable-filter=movie --enable-libfaac --enable-nonfree
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mpegts, from 'live00000546.ts':
  Duration: 00:00:13.27, start: 5456.735789, bitrate: 204 kb/s
  Program 1 
    Metadata:
      service_name    : RTSP Session
      service_provider: FFmpeg
    Stream #0:0[0x100]: Video: h264 (Baseline) ([27][0][0][0] / 0x001B),
yuv420p, 352x288, 25 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x101]: Audio: aac ([15][0][0][0] / 0x000F), 22050 Hz,
stereo, fltp, 2 kb/s

If I wait for more time, the result is like follow
ffprobe live00001021.ts
 Input #0, mpegts, from 'live00001021.ts':
  Duration: 00:00:39.26, start: 10190.309300, bitrate: 21 kb/s

ffprobe live00002021.ts
Input #0, mpegts, from 'live00002027.ts':
  Duration: 26:29:43.68, start: 20219.606444, bitrate: 0 kb/s

And jwplayer can't play at all.
I try same options like '-vbsf h264_mp4toannexb', or '-absf aac_adtstoasc'
but not work.

I don't know how it happed and it has confused me three days.
Anything help.




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From iamnoble2014 at gmail.com  Fri Apr 25 15:37:33 2014
From: iamnoble2014 at gmail.com (sean blow)
Date: Fri, 25 Apr 2014 09:37:33 -0400
Subject: [FFmpeg-user] Concatenate MP4s plus fade in effect command
Message-ID: <CA+Jd8D86gqDAyzdy2M6E9w5mikir5nx0jFaULODoBYNfy0-a8Q@mail.gmail.com>

Hello everyone,

I have three mp4s that I would like to join without re-encoding and I'm
doing the following with:

ffmpeg -i input1.mp4 -c copy -bsf h264_mp4toannexb output1.ts
ffmpeg -i input2.mp4 -c copy -bsf h264_mp4toannexb output2.ts
ffmpeg -i input3.mp4 -c copy -bsf h264_mp4toannexb output2.ts

ffmpeg -i "concat:output1.ts|output2.ts|output3.ts" -c copy -absf
aac_adtstoasc concatAll.mp4

I need help with adding a fade in effect at the beginning of output2.ts and
output3.ts for a duration of 2 seconds each. basically I need to be able to
be able to add a transition effect from video to video.

Thanks in advance for your help.

From katherinegyee at gmail.com  Fri Apr 25 19:29:21 2014
From: katherinegyee at gmail.com (Katherine Yee)
Date: Fri, 25 Apr 2014 13:29:21 -0400
Subject: [FFmpeg-user] Stripping 'actual time of day' from AVI files
Message-ID: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>

To Whom This May Concern,

I was wondering if you knew a way to (I'm not sure if I am wording this
correctly) extract the 'time of day' off of .AVI files. It is apparently
embedded in the audio. I currently use SMPTE to monitor the time, so I know
it's there, but I need to extract ?the time (and navigation), preferably in
a .txt or .doc file, that is embedded in the file. All the different
scripts that I use through ffmpeg only give me a start: 0.0000000 and
duration: 00:01:23:83. Can this be even done? Any kind of help would be
very much appreciated!

Thanks,
Katherine





.:"If you put your mind to it, you can accomplish anything":. Marty McFly

From mrukant at gmail.com  Fri Apr 25 20:03:12 2014
From: mrukant at gmail.com (Mrukant Popat)
Date: Fri, 25 Apr 2014 11:03:12 -0700
Subject: [FFmpeg-user] Not able to get constant 25fps for H.264 HD Transport
	Stream after transcoding
Message-ID: <CADDQthtTU59hRvdsmLSwq95KGgEv0nBThwW7Q9TonMs=7iQN0A@mail.gmail.com>

Dear All,

Below command line generates a Constant bitrate TS of 5000Kbps , but the
video elementary stream inside does not show 25 fps when using MediaInfo.
Infact when I play it with mplayer, it plays with 50fps. However, VLC plays
with 25fps.

Here is the command line

ffmpeg.exe -loglevel debug -i input.ts -pix_fmt yuv420p -r 25.0 -c:s copy
-c:v libx264 -preset slow -tune film  -x264opts
tff:fps=25.0:bitrate=4500:ref=3:threads=6:no-scenecut
:keyint=50:vbv-bufsize=180:vbv-maxrate=4500:videoformat=pal -c:a libfaac
 -ab 192000 -ar 48000 -f mpegts -mpegts_pmt_start_pid 0x64 -map 0:0 -map
0:1 -map 0:2 -
map 0:3 -streamid 0:1000 -streamid 1:1001 -streamid 2:1002 -streamid 3:1005
-muxrate 5M output.ts > c:\commandoutput.txt 2>&1



Here is the link to detailed output to this command --->
https://drive.google.com/file/d/0B4vc-v98vRCkZ3BIelhGNVBWSE9IbW1kQk5yV1k2bHZBalJR/edit?usp=sharing



I am looking to get video with 25 fps and also the value of 25 fps set in
VUI of SEI inside H.264 stream.


Thank you,
Mrukant Popat

From lou at lrcd.com  Fri Apr 25 20:28:53 2014
From: lou at lrcd.com (Lou)
Date: Fri, 25 Apr 2014 10:28:53 -0800
Subject: [FFmpeg-user] Stripping 'actual time of day' from AVI files
In-Reply-To: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
References: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
Message-ID: <20140425102853.5f4e88b9@lrcd.com>

On Fri, 25 Apr 2014 13:29:21 -0400
Katherine Yee <katherinegyee at gmail.com> wrote:

> To Whom This May Concern,
> 
> I was wondering if you knew a way to (I'm not sure if I am wording this
> correctly) extract the 'time of day' off of .AVI files. It is apparently
> embedded in the audio. I currently use SMPTE to monitor the time, so I know
> it's there, but I need to extract ?the time (and navigation), preferably in
> a .txt or .doc file, that is embedded in the file. All the different
> scripts that I use through ffmpeg only give me a start: 0.0000000 and
> duration: 00:01:23:83. Can this be even done? Any kind of help would be
> very much appreciated!
> 
> Thanks,
> Katherine

Please provide a short sample if possible.

I'm not exactly sure what you're referring to, but does the audio sound
like noise? I ask because I have some samples that, if I recall
correctly, contain linear timecode within a channel of the audio stream:

http://avmule.com/samples/deltasub.mp3

Turn your audio down before playing.

From lou at lrcd.com  Fri Apr 25 20:51:12 2014
From: lou at lrcd.com (Lou)
Date: Fri, 25 Apr 2014 10:51:12 -0800
Subject: [FFmpeg-user] Concatenate MP4s plus fade in effect command
In-Reply-To: <CA+Jd8D86gqDAyzdy2M6E9w5mikir5nx0jFaULODoBYNfy0-a8Q@mail.gmail.com>
References: <CA+Jd8D86gqDAyzdy2M6E9w5mikir5nx0jFaULODoBYNfy0-a8Q@mail.gmail.com>
Message-ID: <20140425105112.174691f5@lrcd.com>

On Fri, 25 Apr 2014 09:37:33 -0400
sean blow <iamnoble2014 at gmail.com> wrote:

> I need help with adding a fade in effect at the beginning of output2.ts and
> output3.ts for a duration of 2 seconds each. basically I need to be able to
> be able to add a transition effect from video to video.

Fading will require re-encoding. See the fade, setpts, concat filters:

http://ffmpeg.org/ffmpeg-filters.html#fade
http://ffmpeg.org/ffmpeg-filters.html#setpts_002c-asetpts
http://ffmpeg.org/ffmpeg-filters.html#concat

From katherinegyee at gmail.com  Sat Apr 26 00:48:00 2014
From: katherinegyee at gmail.com (Katherine Yee)
Date: Fri, 25 Apr 2014 18:48:00 -0400
Subject: [FFmpeg-user] Stripping 'actual time of day' from AVI files
In-Reply-To: <20140425102853.5f4e88b9@lrcd.com>
References: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
	<20140425102853.5f4e88b9@lrcd.com>
Message-ID: <CAP4dC35OGMR-mvtmLWqtf3MNVXHE7GcCv_V89zDuryUhtzCgzw@mail.gmail.com>

(previous limit size was over 250K)

Lou,

Yes! Thats the lovely sound that comes out of all the videos that I have.
So supposedly theres time (and navigation) embedded in the audio? I've also
attached a sample of the file that I am working with. I was using ffmpeg in
terminal to try to extract the start time of the video.

    timecode        : 00:00:00;00
    Duration: 00:01:16.51, start: 0.000000, bitrate: 124294 kb/s
    Stream #0:0: Video: rawvideo (Y41B / 0x42313459), yuv411p, 720x480, SAR
8:9 DAR 4:3, 29.97 tbr, 29.97 tbn, 29.97 tbc
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 32000 Hz, stereo, s16p,
96 kb/s

But it just shows timecode: 00:00:00;00 instead of an actual time that I'm
looking for.

Thanks,
Katherine








.:"If you put your mind to it, you can accomplish anything":. Marty McFly


On Fri, Apr 25, 2014 at 2:28 PM, Lou <lou at lrcd.com> wrote:

> On Fri, 25 Apr 2014 13:29:21 -0400
> Katherine Yee <katherinegyee at gmail.com> wrote:
>
> > To Whom This May Concern,
> >
> > I was wondering if you knew a way to (I'm not sure if I am wording this
> > correctly) extract the 'time of day' off of .AVI files. It is apparently
> > embedded in the audio. I currently use SMPTE to monitor the time, so I
> know
> > it's there, but I need to extract ?the time (and navigation), preferably
> in
> > a .txt or .doc file, that is embedded in the file. All the different
> > scripts that I use through ffmpeg only give me a start: 0.0000000 and
> > duration: 00:01:23:83. Can this be even done? Any kind of help would be
> > very much appreciated!
> >
> > Thanks,
> > Katherine
>
> Please provide a short sample if possible.
>
> I'm not exactly sure what you're referring to, but does the audio sound
> like noise? I ask because I have some samples that, if I recall
> correctly, contain linear timecode within a channel of the audio stream:
>
> http://avmule.com/samples/deltasub.mp3
>
> Turn your audio down before playing.
>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: Station395_1.mp3
Type: audio/mpeg
Size: 232049 bytes
Desc: not available
URL: <http://ffmpeg.org/pipermail/ffmpeg-user/attachments/20140425/dfdf8c28/attachment.mp3>

From nlewis at crawford.com  Sat Apr 26 02:02:21 2014
From: nlewis at crawford.com (Nathan Lewis)
Date: Fri, 25 Apr 2014 20:02:21 -0400
Subject: [FFmpeg-user] Stripping 'actual time of day' from AVI files
In-Reply-To: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
References: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
Message-ID: <CACegdb+1o6wKdSEDC2qYE_4XpHgw0nsZi1r-M5WchtHzA9Q7iw@mail.gmail.com>

On Friday, April 25, 2014, Katherine Yee <katherinegyee at gmail.com> wrote:

> To Whom This May Concern,
>
> I was wondering if you knew a way to (I'm not sure if I am wording this
> correctly) extract the 'time of day' off of .AVI files. It is apparently
> embedded in the audio. I currently use SMPTE to monitor the time, so I know
> it's there, but I need to extract the time (and navigation), preferably in
> a .txt or .doc file, that is embedded in the file. All the different
> scripts that I use through ffmpeg only give me a start: 0.0000000 and
> duration: 00:01:23:83. Can this be even done? Any kind of help would be
> very much appreciated!
>
>
I'm not sure if ffmpeg will be able to decode the SMPTE audible timecode
track.  I'll defer to others with more experience with regards to ffmpeg's
capabilities, but you might want to look into libltc and specifically
ltc-tools.

libltc:

http://x42.github.io/libltc/index.html

>From libltc webpage:
"Is there [free] software is using libltc?

yes, ltc-tools <https://github.com/x42/ltc-tools>. It comprises
JACK<http://jackaudio.org/> applications
to generate and decode LTC from live sources and includes tools to read or
write LTC from/to audio-files. xjadeo <http://xjadeo.sf.net/> video-monitor
has been updated to use libltc, and ardour <http://ardour.org/>3 DAW fully
supports chasing and latency free [re]encoding of LTC since svn rev 13390."

Link to ltc-tools:

https://github.com/x42/ltc-tools


Hope this might be useful.

-- 

--

*Nathan Lewis*
Media Engineer


*Crawford Media Services, Inc.*direct: 678.536.4903 | main: 404.876.0333 |
fax: 678.536.4912 |
e:  nlewis at crawford.com | w: www.crawford.com | @crawford_media
6 West Druid Hills Drive, NE Atlanta, GA 30329


*Your Content. Our Solutions.*

From ybrenman at xopnetworks.com  Sat Apr 26 06:56:29 2014
From: ybrenman at xopnetworks.com (Yan Brenman)
Date: Fri, 25 Apr 2014 21:56:29 -0700
Subject: [FFmpeg-user] Need help with the optimal ffmpeg settings to publish
	video
Message-ID: <535B3C7D.2060700@xopnetworks.com>

Reaching out to all ffmpeg gurus,

need any help I can get picking up the most optimal ffmpeg setting for 
publishing captured desktop/screen video.
The type of application we are trying to implement is sort of 
desktop/screen sharing "webinar" used for the company training sessions.
Presenter publishes streaming video of his/here desktop to the 
nginx-rtmp web server which then gets reflected to all clients connected
to the nginx web server hosted website.
For the streaming video publishing side we are planning to use ffmpeg 
(we tried other clients but ffmpeg seems to be providing much
better quality) and for the client side it's going to be either ffplay 
or browser embedded flash player (if we get it working with ffmepg). The
most obvious target platform for this is going to be Microsoft Windows.
With H.264 compression, FLV format and "fast" preset we are getting 
pretty decent quality of the video, but the delay is around 10 seconds
(which is obviously unacceptable). Even with reducing the quality we 
can't get delay lower than 5 seconds (and at that time quality
becomes simply unacceptable).
Clearly there is something we are not setting correctly, because I keep 
hearing about the delay bellow 1 seconds (unfortunately none of
the examples provided with those posting ever works in reality).
I would greatly appreciate any type of help/advise anybody can provide.

Just for reference - here are the settings I am using right now:

*ffmpeg -rtbufsize 300M -f dshow -i video="screen-capture-recorder" -c:v 
libx264 -preset fast -an -f flv rtmp://[server IP 
address]/live/screen_share <http://usn7.xopnetworks.com/live/screen_share>*

Please let me know if there is anything else I can provide.

Thank you very much
Yan


From cehoyos at ag.or.at  Sat Apr 26 08:32:26 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 26 Apr 2014 06:32:26 +0000 (UTC)
Subject: [FFmpeg-user] Stripping 'actual time of day' from AVI files
References: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
	<20140425102853.5f4e88b9@lrcd.com>
	<CAP4dC35OGMR-mvtmLWqtf3MNVXHE7GcCv_V89zDuryUhtzCgzw@mail.gmail.com>
Message-ID: <loom.20140426T083121-440@post.gmane.org>

Katherine Yee <katherinegyee <at> gmail.com> writes:

>     timecode        : 00:00:00;00
>     Duration: 00:01:16.51, start: 0.000000, bitrate: 124294 kb/s
>     Stream #0:0: Video: rawvideo (Y41B / 0x42313459), yuv411p, 720x480, SAR
> 8:9 DAR 4:3, 29.97 tbr, 29.97 tbn, 29.97 tbc
>     Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 32000 Hz, stereo, s16p,
> 96 kb/s

Please provide this avi file.
Either upload to http://www.datafilehost.com/ 
or read https://ffmpeg.org/bugreports.html - 
there is NO filesize limit.

Please remember not to top-post here, Carl Eugen


From phil_rhodes at rocketmail.com  Sat Apr 26 09:25:13 2014
From: phil_rhodes at rocketmail.com (Phil Rhodes)
Date: Sat, 26 Apr 2014 00:25:13 -0700 (PDT)
Subject: [FFmpeg-user] Stripping 'actual time of day' from AVI files
In-Reply-To: <CACegdb+1o6wKdSEDC2qYE_4XpHgw0nsZi1r-M5WchtHzA9Q7iw@mail.gmail.com>
References: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
	<CACegdb+1o6wKdSEDC2qYE_4XpHgw0nsZi1r-M5WchtHzA9Q7iw@mail.gmail.com>
Message-ID: <1398497113.34211.YahooMailNeo@web161506.mail.bf1.yahoo.com>

>?yes, ltc-tools <https://github.com/x42/ltc-tools>. It comprises
>?JACK<http://jackaudio.org/> applications
>?to generate and decode LTC from live sources and includes tools to read or
>?write LTC from/to audio-files.?

I fear that the set up of this sort of environment might be quite complicated. Do we know which operating system Katherine is running? If it's not Linux, this could be a real bear to make work.

I did write a very quick tool to do this under Windows. It wasn't particularly complicated.

P

From priftien at hotmail.com  Sat Apr 26 11:02:48 2014
From: priftien at hotmail.com (enri69)
Date: Sat, 26 Apr 2014 02:02:48 -0700 (PDT)
Subject: [FFmpeg-user] DVB Subtitles On/Off
Message-ID: <1398502968777-4664943.post@n4.nabble.com>

Hi everybody!

Is it possible to stream a local file and a subtitle for that file (in srt
or whatever format) so that in the final stream the subtitles can be
switched on and off.

My problem is that I want to stream to 'UDP multicast (224.1.1.1:1234)'
Video + Audio in a Mkv container + Subtitle in .srt or .ass. But I need
subtitles to be switched On and Off. Until now I have successfully manage to
stream all the above transcoding to mpeg2video, using the dvbsub codec for
subs.

But using Vlc Player or a DVB-C STB I can receive the video stream with all
audio streams and the DVB Subtitle stream, BUT it seems that the sub stream
is empty.

I have tried with ass subs but they are hardcoded with the -vf filter, and I
want the subs to be Dvb Subtitles.

This is what I am using:

ffmpeg -re -i 'input.mpg' -i 'input.sub' -map 0:0 -map 0:1 -map 0:2 -map 0:3
-map 1:0 -vcodec mpeg2video -s 1280x720 -r 25 -b:v 9M -acodec mp2 -ac 2 -b:a
192k -scodec dvbsub -f mpegts udp://224.1.1.1:1234?pkt_size=1316

I am using latest Zeranoe 64bit build on Windows 7 64bit, but I'm willing to
use it from Linux if it is a windows build problem.

Anyone has the answer to this?



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View this message in context: http://ffmpeg-users.933282.n4.nabble.com/DVB-Subtitles-On-Off-tp4664943.html
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From nlewis at crawford.com  Sat Apr 26 17:17:24 2014
From: nlewis at crawford.com (Nathan Lewis)
Date: Sat, 26 Apr 2014 11:17:24 -0400
Subject: [FFmpeg-user] Stripping 'actual time of day' from AVI files
In-Reply-To: <1398497113.34211.YahooMailNeo@web161506.mail.bf1.yahoo.com>
References: <CAP4dC37R_=B=RMJZJMeuwiE13UGJpaLd005Qsr9Z4j=mpQH4qw@mail.gmail.com>
	<CACegdb+1o6wKdSEDC2qYE_4XpHgw0nsZi1r-M5WchtHzA9Q7iw@mail.gmail.com>
	<1398497113.34211.YahooMailNeo@web161506.mail.bf1.yahoo.com>
Message-ID: <CACegdbKZixoPR4_5Uc57buAor+UfPaFu79n337X5XHOMSZU20Q@mail.gmail.com>

On Sat, Apr 26, 2014 at 3:25 AM, Phil Rhodes <phil_rhodes at rocketmail.com>wrote:
>
> I fear that the set up of this sort of environment might be quite
> complicated.
>

Understandably.  Decoding audible SMPTE timecode isn't a trivial task.


> Do we know which operating system Katherine is running? If it's not Linux,
> this could be a real bear to make work.
>

At this point we do not.

I did write a very quick tool to do this under Windows. It wasn't
> particularly complicated.


Care to share your Windows solution?

From cehoyos at ag.or.at  Sat Apr 26 18:38:22 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 26 Apr 2014 16:38:22 +0000 (UTC)
Subject: [FFmpeg-user] DVB Subtitles On/Off
References: <1398502968777-4664943.post@n4.nabble.com>
Message-ID: <loom.20140426T183724-626@post.gmane.org>

enri69 <priftien <at> hotmail.com> writes:

> But using Vlc Player or a DVB-C STB I can receive 
> the video stream with all audio streams and the 
> DVB Subtitle stream, BUT it seems that the sub 
> stream is empty.

Is this reproducible with file output?
If yes, please provide your failing command line 
together with the complete, uncut console output.

Carl Eugen


From priftien at hotmail.com  Sat Apr 26 19:41:28 2014
From: priftien at hotmail.com (enri69)
Date: Sat, 26 Apr 2014 10:41:28 -0700 (PDT)
Subject: [FFmpeg-user] DVB Subtitles On/Off
In-Reply-To: <1398502968777-4664943.post@n4.nabble.com>
References: <1398502968777-4664943.post@n4.nabble.com>
Message-ID: <1398534088901-4664946.post@n4.nabble.com>

Hi!

Now that I am trying with a SUB subtitle which I got from converting a srt
with txt2vobsub I am getting an error....

>ffmpeg -re -i c:\simpson.mkv -i c:\simpson.sub -map 0:0 -map 0:1 -map 0:2
-map
0:3 -map 1:0 -vcodec mpeg2video -s 1280x720 -r 25 -b:v 9M -acodec mp2 -ac 2
-b:a
 192k -scodec dvbsub -f mpegts udp://224.1.1.1:1234?pkt_size=1316
ffmpeg version N-62669-g443936d Copyright (c) 2000-2014 the FFmpeg
developers
  built on Apr 23 2014 22:09:11 with gcc 4.8.2 (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-libvidstab
--enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
--enable-libwavp
ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable
-decklink --enable-zlib
  libavutil      52. 78.100 / 52. 78.100
  libavcodec     55. 59.100 / 55. 59.100
  libavformat    55. 37.101 / 55. 37.101
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, matroska,webm, from 'c:\simpson.mkv':
  Metadata:
    title           : The Simpsons [Season 21] (2009)
    encoder         : libebml v1.2.3 + libmatroska v1.3.0
    creation_time   : 2012-02-06 11:16:19
  Duration: 00:21:31.46, start: 0.000000, bitrate: 4725 kb/s
    Stream #0:0(eng): Video: h264 (High), yuv420p(tv, bt709), 1280x720 [SAR
1:1
DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 2k tbc (default)
    Metadata:
      title           : Release for HU
    Stream #0:1(ita): Audio: mp2, 48000 Hz, stereo, s16p, 224 kb/s (default)
    Metadata:
      title           : Italisht
    Stream #0:2(rus): Audio: ac3, 48000 Hz, stereo, fltp, 256 kb/s
    Metadata:
      title           : AC3 2.0 256 Kbps
    Stream #0:3(eng): Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s
    Metadata:
      title           : AC3 5.1 384 Kbps
Input #1, mpeg, from 'c:\simpson.sub':
  Duration: 00:19:20.98, start: 72.540000, bitrate: 6 kb/s
    Stream #1:0[0x20]: Subtitle: dvd_subtitle
Output #0, mpegts, to 'udp://224.1.1.1:1234?pkt_size=1316':
  Metadata:
    title           : The Simpsons [Season 21] (2009) - Release for HDClub
    encoder         : Lavf55.37.101
    Stream #0:0(eng): Video: mpeg2video, yuv420p, 1280x720 [SAR 1:1 DAR
16:9], q
=2-31, 9000 kb/s, 90k tbn, 25 tbc (default)
    Metadata:
      title           : Release for HU
    Stream #0:1(ita): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s (default)
    Metadata:
      title           : Italisht
    Stream #0:2(rus): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s
    Metadata:
      title           : AC3 2.0 256 Kbps
    Stream #0:3(eng): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s
    Metadata:
      title           : AC3 5.1 384 Kbps
    Stream #0:4: Subtitle: dvb_subtitle (dvbsub), 1280x720
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg2video)
  Stream #0:1 -> #0:1 (mp2 -> mp2)
  Stream #0:2 -> #0:2 (ac3 -> mp2)
  Stream #0:3 -> #0:3 (ac3 -> mp2)
  Stream #1:0 -> #0:4 (dvdsub -> dvbsub)
Press [q] to stop, [?] for help
[mpegts @ 00000000051fa6c0] Encoder did not produce proper pts, making some
up.
[mpegts @ 00000000051fa6c0] Application provided invalid, non monotonically
incr
easing dts to muxer in stream 4: 620640 >= 617130
av_interleaved_write_frame(): Invalid argument
frame=    0 fps=0.0 q=0.0 Lsize=       6kB time=00:00:10.22 bitrate=  
4.4kbits/
s
video:0kB audio:1kB subtitle:6kB other streams:0kB global headers:0kB muxing
ove
rhead: unknown



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From cehoyos at ag.or.at  Sat Apr 26 22:01:53 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 26 Apr 2014 20:01:53 +0000 (UTC)
Subject: [FFmpeg-user] DVB Subtitles On/Off
References: <1398502968777-4664943.post@n4.nabble.com>
	<1398534088901-4664946.post@n4.nabble.com>
Message-ID: <loom.20140426T220126-409@post.gmane.org>

enri69 <priftien <at> hotmail.com> writes:

> Now that I am trying with a SUB subtitle which 
> I got from converting a srt with txt2vobsub I 
> am getting an error....

Is the problem reproducible with file output?

Carl Eugen


From priftien at hotmail.com  Sat Apr 26 22:23:18 2014
From: priftien at hotmail.com (enri69)
Date: Sat, 26 Apr 2014 13:23:18 -0700 (PDT)
Subject: [FFmpeg-user] DVB Subtitles On/Off
In-Reply-To: <loom.20140426T220126-409@post.gmane.org>
References: <1398502968777-4664943.post@n4.nabble.com>
	<1398534088901-4664946.post@n4.nabble.com>
	<loom.20140426T220126-409@post.gmane.org>
Message-ID: <1398543798352-4664948.post@n4.nabble.com>

Yes. This is with file output

>ffmpeg -re -i c:\simpson.mkv -i c:\simpson.sub -map 0:0 -map 0:1 -map 1:0
-vcod
ec mpeg2video -s 720x480 -r 25 -b:v 1M -acodec mp2 -ac 2 -b:a 192k -scodec
dvbsu
b -f mpegts c:\simp.ts
ffmpeg version N-62669-g443936d Copyright (c) 2000-2014 the FFmpeg
developers
  built on Apr 23 2014 22:09:11 with gcc 4.8.2 (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-libvidstab
--enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
--enable-libwavp
ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable
-decklink --enable-zlib
  libavutil      52. 78.100 / 52. 78.100
  libavcodec     55. 59.100 / 55. 59.100
  libavformat    55. 37.101 / 55. 37.101
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, matroska,webm, from 'c:\simpson.mkv':
  Metadata:
    title           : The Simpsons [Season 21] (2009)
    encoder         : libebml v1.3.0 + libmatroska v1.4.1
    creation_time   : 2014-04-26 18:22:05
  Duration: 00:04:00.29, start: 0.000000, bitrate: 5555 kb/s
    Stream #0:0(eng): Video: h264 (High), yuv420p(tv, bt709), 1280x720 [SAR
1:1
DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 2k tbc (default)
    Metadata:
      title           : Release for HU
    Stream #0:1(ita): Audio: mp2, 48000 Hz, stereo, s16p, 224 kb/s (default)
    Metadata:
      title           : Italisht
Input #1, mpeg, from 'c:\simpson.sub':
  Duration: 00:19:20.98, start: 72.540000, bitrate: 6 kb/s
    Stream #1:0[0x20]: Subtitle: dvd_subtitle
File 'c:\simp.ts' already exists. Overwrite ? [y/N] y
Output #0, mpegts, to 'c:\simp.ts':
  Metadata:
    title           : The Simpsons [Season 21] (2009)
    encoder         : Lavf55.37.101
    Stream #0:0(eng): Video: mpeg2video, yuv420p, 720x480 [SAR 32:27 DAR
16:9],
q=2-31, 1000 kb/s, 90k tbn, 25 tbc (default)
    Metadata:
      title           : Release for HU
    Stream #0:1(ita): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s (default)
    Metadata:
      title           : Italisht
    Stream #0:2: Subtitle: dvb_subtitle (dvbsub), 720x480
Stream mapping:
  Stream #0:0 -> #0:0 (h264 -> mpeg2video)
  Stream #0:1 -> #0:1 (mp2 -> mp2)
  Stream #1:0 -> #0:2 (dvdsub -> dvbsub)
Press [q] to stop, [?] for help
[mpegts @ 0000000002aa55e0] Encoder did not produce proper pts, making some
up.
[mpegts @ 0000000002aa55e0] Application provided invalid, non monotonically
incr
easing dts to muxer in stream 2: 620640 >= 617130
av_interleaved_write_frame(): Invalid argument
frame=    0 fps=0.0 q=0.0 Lsize=       6kB time=00:00:10.22 bitrate=  
4.4kbits/
s
video:0kB audio:1kB subtitle:6kB other streams:0kB global headers:0kB muxing
ove
rhead: unknown



If I put a SRT file instead of a SUB it encodes the file but with empty DVB
Subtream in VLC.

Here is the file encoded
https://www.wetransfer.com/downloads/33eee23e9b5b5627866c41d4f33ce01e20140426201855/43145531071191e908bb00294b1282a820140426201855/0c4c2c



--
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From cehoyos at ag.or.at  Sat Apr 26 23:41:14 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sat, 26 Apr 2014 21:41:14 +0000 (UTC)
Subject: [FFmpeg-user] DVB Subtitles On/Off
References: <1398502968777-4664943.post@n4.nabble.com>
	<1398534088901-4664946.post@n4.nabble.com>
	<loom.20140426T220126-409@post.gmane.org>
	<1398543798352-4664948.post@n4.nabble.com>
Message-ID: <loom.20140426T234052-263@post.gmane.org>

enri69 <priftien <at> hotmail.com> writes:

> Yes. This is with file output
> 
> >ffmpeg -re -i c:\simpson.mkv -i c:\simpson.sub

Please provide the input samples.

Carl Eugen


From barsnick at gmx.net  Sun Apr 27 00:35:19 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Sun, 27 Apr 2014 00:35:19 +0200
Subject: [FFmpeg-user] DVB Subtitles On/Off
In-Reply-To: <1398543798352-4664948.post@n4.nabble.com>
References: <1398502968777-4664943.post@n4.nabble.com>
	<1398534088901-4664946.post@n4.nabble.com>
	<loom.20140426T220126-409@post.gmane.org>
	<1398543798352-4664948.post@n4.nabble.com>
Message-ID: <20140426223519.GA19533@sunshine.barsnick.net>

Hi,

On Sat, Apr 26, 2014 at 13:23:18 -0700, enri69 wrote:

[I have shorted this a slight bit]

> Input #0, matroska,webm, from 'c:\simpson.mkv':
>     Stream #0:0(eng): Video: h264 (High), yuv420p(tv, bt709), 1280x720 [SAR
>     Stream #0:1(ita): Audio: mp2, 48000 Hz, stereo, s16p, 224 kb/s (default)
> Input #1, mpeg, from 'c:\simpson.sub':
>     Stream #1:0[0x20]: Subtitle: dvd_subtitle
> Output #0, mpegts, to 'c:\simp.ts':
>     Stream #0:0(eng): Video: mpeg2video, yuv420p, 720x480 [SAR 32:27 DAR 16:9], q=2-31, 1000 kb/s, 90k tbn, 25 tbc (default)
>     Stream #0:1(ita): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s (default)
>     Stream #0:2: Subtitle: dvb_subtitle (dvbsub), 720x480
> Stream mapping:
>   Stream #0:0 -> #0:0 (h264 -> mpeg2video)
>   Stream #0:1 -> #0:1 (mp2 -> mp2)
>   Stream #1:0 -> #0:2 (dvdsub -> dvbsub)

> If I put a SRT file instead of a SUB it encodes the file but with empty DVB
> Subtream in VLC.

Can either of those formats even successfully be muxed into an MPEG-TS?
https://wiki.videolan.org/MPEG/#Accepted_subtitle_codecs

SRT shouldn't work at all, I wonder why ffmpeg doesn't refuse to do so.
I was under the impression (from personal experience, not standards)
that dvdsub shouldn't work, but dvbsub should. Caveat: ffmpeg could
have problems of its own with the latter, see trac #2024.

Moritz

From gliese849b at gmail.com  Sun Apr 27 19:54:32 2014
From: gliese849b at gmail.com (RDP)
Date: Sun, 27 Apr 2014 18:54:32 +0100
Subject: [FFmpeg-user] Error after conversion back to WebM ( under IE 11 )
Message-ID: <CAEUFz7f_2BQt1h1sOMZzTCbq=CqaMX1sdtmwRsvjKKrLngBHMQ@mail.gmail.com>

Greetngs!

Example of what I am experiencing is linked to here:

    http://musicbitz.org/WebMTest/

A sample .webm file is converted to .mp4, then back to .webm, using ffmpeg.
Basic syntax is used with no added flags or other options.

Viewing with Internet Explorer 11, under WInodws 8.1, the first two files play
back tickety-boo, but the conversion back to .webm fails to be
recognised successfully.

Converting most media files to WebM with ffmpeg, seems to produce the
same result.

All ideas and suggestion gratefully received!

Many thanks.

Regards,


Chris.
.

From cehoyos at ag.or.at  Sun Apr 27 20:05:47 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 27 Apr 2014 18:05:47 +0000 (UTC)
Subject: [FFmpeg-user] Error after conversion back to WebM ( under IE 11
	)
References: <CAEUFz7f_2BQt1h1sOMZzTCbq=CqaMX1sdtmwRsvjKKrLngBHMQ@mail.gmail.com>
Message-ID: <loom.20140427T200304-752@post.gmane.org>

RDP <gliese849b <at> gmail.com> writes:

> Viewing with Internet Explorer 11, under WInodws 8.1, 
> the first two files play back tickety-boo, but the 
> conversion back to .webm fails to be recognised 
> successfully.

See ticket #3583

For future questions:
Please always add the failing command line (in your case 
the command line that produces the webm file) together 
with the complete, uncut console output.

I am still surprised that everybody seems to believe IE 
plays webm files...

Carl Eugen


From gliese849b at gmail.com  Sun Apr 27 20:34:06 2014
From: gliese849b at gmail.com (RDP)
Date: Sun, 27 Apr 2014 19:34:06 +0100
Subject: [FFmpeg-user] Error after conversion back to WebM ( under IE 11
	)
In-Reply-To: <loom.20140427T200304-752@post.gmane.org>
References: <CAEUFz7f_2BQt1h1sOMZzTCbq=CqaMX1sdtmwRsvjKKrLngBHMQ@mail.gmail.com>
	<loom.20140427T200304-752@post.gmane.org>
Message-ID: <CAEUFz7ey-JhEkSpWn1njb2=m8Ao_Xi=qoizUSagoGWquC5Z6yQ@mail.gmail.com>

On Sun, Apr 27, 2014 at 7:05 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> RDP <gliese849b <at> gmail.com> writes:
>
>> Viewing with Internet Explorer 11, under WInodws 8.1,
>> the first two files play back tickety-boo, but the
>> conversion back to .webm fails to be recognised
>> successfully.
>
> See ticket #3583
>
> For future questions:
> Please always add the failing command line (in your case
> the command line that produces the webm file) together
> with the complete, uncut console output.
>
> I am still surprised that everybody seems to believe IE
> plays webm files...
>
> Carl Eugen

Car,

The ( very simple and basic ) commands I used are stated on the web
page I linked to.
None of those simple basic commands *fail* with an error of any kind.
I they had, then
you would had got the 'offending' output quoted, here.  Howeveer, the
resulting .webm as
produced using ffmpeg has obvoously 'changed' in some way from the
original, else it
would play, and we wouldn't be having this exchange,

I assume nothing.  I expect nothing.  I'm not 'everybody'.  I don't
use IE  by default.
However, I am investigatbg a behaviour, and would like to locate the
source of the
problem, be that the encoding library or the program implementing it.  Hence my
question.

IE does, for better or worse, play webm files when the correct bits
are in place to help
it along. The link I gave shows that.  Sames goes for some other
browsers, when they
are sutably tweaked.

Should I infer from your message that you consider ffmpeg not to be
'at fault', just because
the command does not throw an error during conversion?

Regards,


Chris.

From cehoyos at ag.or.at  Sun Apr 27 21:15:00 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 27 Apr 2014 19:15:00 +0000 (UTC)
Subject: [FFmpeg-user] Error after conversion back to WebM ( under IE 11
	)
References: <CAEUFz7f_2BQt1h1sOMZzTCbq=CqaMX1sdtmwRsvjKKrLngBHMQ@mail.gmail.com>
	<loom.20140427T200304-752@post.gmane.org>
	<CAEUFz7ey-JhEkSpWn1njb2=m8Ao_Xi=qoizUSagoGWquC5Z6yQ@mail.gmail.com>
Message-ID: <loom.20140427T210211-800@post.gmane.org>

RDP <gliese849b <at> gmail.com> writes:

> On Sun, Apr 27, 2014 at 7:05 PM, Carl Eugen Hoyos wrote:
> > RDP <gliese849b <at> gmail.com> writes:
> >
> >> Viewing with Internet Explorer 11, under WInodws 8.1,
> >> the first two files play back tickety-boo, but the
> >> conversion back to .webm fails to be recognised
> >> successfully.
> >
> > See ticket #3583

    ^^^^^^^^^^^^^^^^

> > For future questions:
> > Please always add the failing command line (in your case
> > the command line that produces the webm file) together
> > with the complete, uncut console output.
> >
> > I am still surprised that everybody seems to believe IE
> > plays webm files...

> The ( very simple and basic ) commands I used are stated 
> on the web page I linked to.

Please always post command lines etc. (everything except 
sample files) on the mailing list, do not use external 
resources, they may disappear.

> None of those simple basic commands *fail* with an 
> error of any kind.

I understand that.
The failing command line in your case is the command 
line that produces a webm file that does not play.
This is what I tried to explain with "in your case 
the command line that produces the webm file".

> I they had, then you would had got the 'offending' 
> output quoted, here. 

The command line is needed no matter if an error is 
shown or not.

> Howeveer the resulting .webm as produced using 
> ffmpeg has obvoously 'changed' in some way from the
> original, else it would play, and we wouldn't be 
> having this exchange,

This is correct (at least that something has changed).

> I assume nothing. I expect nothing. I'm not 
> 'everybody'. I don't use IE  by default.

You are one (of several persons) who - afaict - 
assume that IE plays webm files (it does not).

>From your question, it would have been difficult 
to understand that you are using Google's webm 
plugin (which for some reason requires the mkv 
version to be set to 2 while FFmpeg sets it to 
4 since it uses features only present in version 
4).

[...]

> Should I infer from your message that you consider 
> ffmpeg not to be 'at fault', just because the 
> command does not throw an error during conversion?

No, my message (above) meant that we are aware of 
the problem.

Carl Eugen


From gliese849b at gmail.com  Sun Apr 27 22:32:28 2014
From: gliese849b at gmail.com (RDP)
Date: Sun, 27 Apr 2014 21:32:28 +0100
Subject: [FFmpeg-user] Error after conversion back to WebM ( under IE 11
	)
Message-ID: <CAEUFz7fFiinhUHshLskzejZVLMAFs_m11kvkvGMG36ukZwYVVQ@mail.gmail.com>

On Sun, Apr 27, 2014 at 8:15 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> RDP <gliese849b <at> gmail.com> writes:
>
>> On Sun, Apr 27, 2014 at 7:05 PM, Carl Eugen Hoyos wrote:
>> > RDP <gliese849b <at> gmail.com> writes:
>> >
>> >> Viewing with Internet Explorer 11, under WInodws 8.1,
>> >> the first two files play back tickety-boo, but the
>> >> conversion back to .webm fails to be recognised
>> >> successfully.

> Please always post command lines etc. (everything except
> sample files) on the mailing list, do not use external
> resources, they may disappear.

Understood ( though in this case I'd happily leave them behind
for posterity )

> The failing command line in your case is the command
> line that produces a webm file that does not play.
> This is what I tried to explain with "in your case
> the command line that produces the webm file".

Understood ( we just saw this from a slightly different aspect )

>> If they had, then you would had got the 'offending'
>> output quoted, here.
>
> The command line is needed no matter if an error is
> shown or not.

Understood.

>> I assume nothing. I expect nothing. I'm not
>> 'everybody'. I don't use IE  by default.
>
> You are one (of several persons) who - afaict -
> assume that IE plays webm files (it does not).

There was no intentional ass-out-of-u-and-me from
either side  ;o)  However, with all due respect, maybe
a little pedanty creeping in here?  ;o)   IE does 'play'
webm *with* the necessary assistance.  Otherwise,
I agree with you, no, by default, IE does not play
webm ( yet ).

> From your question, it would have been difficult
> to understand that you are using Google's webm
> plugin (which for some reason requires the mkv
> version to be set to 2 while FFmpeg sets it to
> 4 since it uses features only present in version
> 4).

As it happens a search provided a similar scenario,

https://groups.google.com/a/webmproject.org/forum/#!topic/webm-discuss/oQOU2-txchw
> [...]

and knowing allows patching/fixing.

>> Should I infer from your message that you consider
>> ffmpeg not to be 'at fault', just because the
>> command does not throw an error during conversion?
>
> No, my message (above) meant that we are aware of
> the problem.

In which case, no more needs to be said on the matter.

Cheers for your messages and assistance.  Rest assured,
should I post again, I will border on compendiousness!
;o)


Regards,


Chris,

From cehoyos at ag.or.at  Sun Apr 27 22:52:44 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Sun, 27 Apr 2014 20:52:44 +0000 (UTC)
Subject: [FFmpeg-user] Error after conversion back to WebM ( under IE 11
	)
References: <CAEUFz7fFiinhUHshLskzejZVLMAFs_m11kvkvGMG36ukZwYVVQ@mail.gmail.com>
Message-ID: <loom.20140427T225156-298@post.gmane.org>

RDP <gliese849b <at> gmail.com> writes:

> As it happens a search provided a similar scenario,

Since you cut my message above, you may still have 
missed this:
https://trac.ffmpeg.org/ticket/3583

Carl Eugen


From seandarcy2 at gmail.com  Mon Apr 28 01:03:35 2014
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Sun, 27 Apr 2014 19:03:35 -0400
Subject: [FFmpeg-user] wav concat woes
Message-ID: <ljk2c8$jju$1@ger.gmane.org>

On Fedora 20, trying to concatenate a set of wav files.

cat 01_list.txt
file '01/01.01_test.wav'
file '01/01.02_test.wav'
file '01/01.03_test.wav'
file '01/01.04_test.wav'

ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a 
aac_he -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out

ffmpeg version 2.2.git Copyright (c) 2000-2014 the FFmpeg developers
   built on Apr  2 2014 11:12:31 with gcc 4.8.2 (GCC) 20131212 (Red Hat 
4.8.2-7)
   configuration: --prefix=/usr --bindir=/usr/bin 
--datadir=/usr/share/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man 
--shlibdir=/usr/lib64 --extra-cflags='-Ofast -march=native -mtune=native 
-fopenmp -fomit-frame-pointer -pipe' --enable-avresample --enable-static 
--enable-shared --enable-gray --enable-gpl --enable-nonfree 
--enable-version3 --enable-postproc --enable-avfilter 
--enable-avresample --enable-pthreads --enable-x11grab --enable-gray 
--enable-vaapi --enable-hardcoded-tables --enable-libaacplus 
--enable-avisynth --enable-frei0r --enable-libfdk-aac --enable-libgsm 
--enable-libmp3lame --enable-libopenjpeg --enable-libopus 
--enable-librtmp --enable-libschroedinger --enable-libspeex 
--enable-libtheora --enable-libvo-aacenc --enable-libvorbis 
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libxvid 
--enable-zlib --disable-debug --cpu=amdfam10 --arch=x86_64 --enable-pic 
--enable-libopencv --enable-openssl
   libavutil      52. 71.100 / 52. 71.100
   libavcodec     55. 56.107 / 55. 56.107
   libavformat    55. 36.100 / 55. 36.100
   libavdevice    55. 11.100 / 55. 11.100
   libavfilter     4.  3.100 /  4.  3.100
   libavresample   1.  2.  0 /  1.  2.  0
   libswscale      2.  5.102 /  2.  5.102
   libswresample   0. 18.100 /  0. 18.100
   libpostproc    52.  3.100 / 52.  3.100
[concat @ 0x82de60] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, concat, from '01_list.txt':
   Duration: 00:00:00.00, start: 0.000000, bitrate: 1410 kb/s
     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 
stereo, s16, 1411 kb/s
Output #0, ipod, to 'test2.m4a':
   Metadata:
     encoder         : Lavf55.36.100
     Stream #0:0: Audio: aac (libfdk_aac) (HE-AAC) (mp4a / 0x6134706D), 
44100 Hz, mono, s16, 48 kb/s
Stream mapping:
   Stream #0:0 -> #0:0 (pcm_s16le -> libfdk_aac)
Press [q] to stop, [?] for help
size=     281kB time=00:00:47.81 bitrate=  48.2kbits/s    ^Msize= 
559kB time=00:01:35.27 bitrate=  48.1kbits/s    ^Msize=     838kB 
time=00:02:22.82 bitrate=  48.1kbits/s    ^Msize=    1115kB 
time=00:03:10.05 bitrate=  48.0kbits/s    ^Msize=    1390kB 
time=00:03:57.14 bitrate=  48.0kbits/s    ^Msize=    1667kB 
time=00:04:44.28 bitrate=  48.0kbits/s    ^MDTS -406750698222075776, 
next:301253333 st:0 invalid dropping
PTS -406750698222075776, next:301253333 invalid dropping st:0
DTS -406750698222074752, next:301276552 st:0 invalid dropping
PTS -406750698222074752, next:301276552 invalid dropping st:0
DTS -406750698222073728, next:301299771 st:0 invalid dropping
.............


The other wav files have the same parameters:

ffprobe 01/01.02_test.wav
ffprobe version 2.2.git Copyright (c) 2007-2014 the FFmpeg developers
......................
Input #0, wav, from '01/01.02_test.wav':
   Duration: 00:05:00.13, bitrate: 1411 kb/s
     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 
channels, s16, 1411 kb/s

And I can run the command on each individually.

Any help appreciated.

sean


From jklimek at gmail.com  Mon Apr 28 01:20:21 2014
From: jklimek at gmail.com (John Klimek)
Date: Sun, 27 Apr 2014 19:20:21 -0400
Subject: [FFmpeg-user] Best settings for upscaling 480p to 1080p?
Message-ID: <CAOEv1ka3eNAeW8P=NWtOiWM++XA+kaadSwDDvrvaNEE47VYB+w@mail.gmail.com>

What are some good settings or filters to use when upscaling 480p content
to 1080p?

I usually just let my media player upscale the content to my monitor's
native resolution (1080p), but I'm wondering if I could use FFmpeg to
achieve really good upscaling (and possibly apply a bit of sharpening, etc)

From h.reindl at thelounge.net  Mon Apr 28 01:26:01 2014
From: h.reindl at thelounge.net (Reindl Harald)
Date: Mon, 28 Apr 2014 01:26:01 +0200
Subject: [FFmpeg-user] Best settings for upscaling 480p to 1080p?
In-Reply-To: <CAOEv1ka3eNAeW8P=NWtOiWM++XA+kaadSwDDvrvaNEE47VYB+w@mail.gmail.com>
References: <CAOEv1ka3eNAeW8P=NWtOiWM++XA+kaadSwDDvrvaNEE47VYB+w@mail.gmail.com>
Message-ID: <535D9209.7040906@thelounge.net>



Am 28.04.2014 01:20, schrieb John Klimek:
> What are some good settings or filters to use when upscaling 480p content
> to 1080p?
> 
> I usually just let my media player upscale the content to my monitor's
> native resolution (1080p), but I'm wondering if I could use FFmpeg to
> achieve really good upscaling (and possibly apply a bit of sharpening, etc)

you can't magically scale anything up
how do you imagine the missing informations of the orginial coming back?

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From jklimek at gmail.com  Mon Apr 28 01:35:38 2014
From: jklimek at gmail.com (John Klimek)
Date: Sun, 27 Apr 2014 19:35:38 -0400
Subject: [FFmpeg-user] Best settings for upscaling 480p to 1080p?
In-Reply-To: <535D9209.7040906@thelounge.net>
References: <CAOEv1ka3eNAeW8P=NWtOiWM++XA+kaadSwDDvrvaNEE47VYB+w@mail.gmail.com>
	<535D9209.7040906@thelounge.net>
Message-ID: <CAOEv1kZxbyyfKa+w4iPj-ajGwtGL3q6GAjbPUsBzL6ZRfkOdgQ@mail.gmail.com>

I understand your argument, but something has to "guess" what that missing
information is, right?  If you play a 480p video on a 1080p monitor, then
either the video card or media player is upscaling ("guessing") to the
native resolution.

I was wondering if there was a better method at upscaling than letting the
video card or media player perform it.

I've read that lots of people like the DVDO upscaling products so there
must be a difference in methods and some might be better than others?



On Sun, Apr 27, 2014 at 7:26 PM, Reindl Harald <h.reindl at thelounge.net>wrote:

>
>
> Am 28.04.2014 01:20, schrieb John Klimek:
> > What are some good settings or filters to use when upscaling 480p content
> > to 1080p?
> >
> > I usually just let my media player upscale the content to my monitor's
> > native resolution (1080p), but I'm wondering if I could use FFmpeg to
> > achieve really good upscaling (and possibly apply a bit of sharpening,
> etc)
>
> you can't magically scale anything up
> how do you imagine the missing informations of the orginial coming back?
>
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
>

From seandarcy2 at gmail.com  Mon Apr 28 03:08:14 2014
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Sun, 27 Apr 2014 21:08:14 -0400
Subject: [FFmpeg-user] mono or stereo? ffmpeg converts or phone?
Message-ID: <ljk9lv$97h$1@ger.gmane.org>

I have a bunch of lectures I need to transcode to play from the android 
phone, The lectures are in wav stereo format. The phone has only one 
speaker, so mono.

I want to transcode it for the most compression for a given bitrate; 
I've only got 16gb. So for a bitrate of let's say 48000, am I better off 
converting to mono in ffmpeg ( -ac 1 ) or leaving it stereo and letting 
the phone convert it?

sean


From gliese849b at gmail.com  Mon Apr 28 05:29:13 2014
From: gliese849b at gmail.com (RDP)
Date: Mon, 28 Apr 2014 04:29:13 +0100
Subject: [FFmpeg-user] Error after conversion back to WebM ( under IE 11
	)
In-Reply-To: <loom.20140427T225156-298@post.gmane.org>
References: <CAEUFz7fFiinhUHshLskzejZVLMAFs_m11kvkvGMG36ukZwYVVQ@mail.gmail.com>
	<loom.20140427T225156-298@post.gmane.org>
Message-ID: <CAEUFz7cbEiZdO0Y0xYvivJMuELhLDuyk1MpUVctTpwn5n-Q7xw@mail.gmail.com>

On Sun, Apr 27, 2014 at 9:52 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:
> RDP <gliese849b <at> gmail.com> writes:
>
>> As it happens a search provided a similar scenario,
>
> Since you cut my message above, you may still have
> missed this:
> https://trac.ffmpeg.org/ticket/3583

> Carl Eugen

Noted Carl.  Thanks very much.

I didn't miss it, but admit to not having read it yet (..  I'd also
not tried 1.2 prior to
my - somewhat premature - posting.

Thanks again.

Regards,


Chris.

From gliese849b at gmail.com  Mon Apr 28 06:08:04 2014
From: gliese849b at gmail.com (RDP)
Date: Mon, 28 Apr 2014 05:08:04 +0100
Subject: [FFmpeg-user] wav concat woes
In-Reply-To: <ljk2c8$jju$1@ger.gmane.org>
References: <ljk2c8$jju$1@ger.gmane.org>
Message-ID: <CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>

On Mon, Apr 28, 2014 at 12:03 AM, Sean Darcy <seandarcy2 at gmail.com> wrote:
> On Fedora 20, trying to concatenate a set of wav files.
>
> cat 01_list.txt
> file '01/01.01_test.wav'
> file '01/01.02_test.wav'
> file '01/01.03_test.wav'
> file '01/01.04_test.wav'
>
> ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a aac_he
> -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out
> And I can run the command on each individually.
>..
>..
> Any help appreciated.

As far as I am aware, conjoining .wav files is not that simple.  While
cat joins the files together, it doesn't take into
account the RIFF header at the beginning of each one.  Thus you get a
combined file that thinks it is only as long
as the first sample in the list.  You either need a program that can
join them correctly for you, or you take the headers
into account when joining them, and adjust the parameters of the
resulting file accordingly.

These random links may assist,

  http://www.codeproject.com/Articles/15187/Concatenating-Wave-Files-Using-C

  http://www.codeproject.com/Articles/15187/Concatenating-Wave-Files-Using-C

Regards,


Chris,

From thiles at confex.com  Mon Apr 28 06:49:17 2014
From: thiles at confex.com (Tim Hiles)
Date: Sun, 27 Apr 2014 21:49:17 -0700
Subject: [FFmpeg-user] wav concat woes
In-Reply-To: <CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
References: <ljk2c8$jju$1@ger.gmane.org>
	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
Message-ID: <CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>

> > On Fedora 20, trying to concatenate a set of wav files.
> >
> > cat 01_list.txt
> > file '01/01.01_test.wav'
> > file '01/01.02_test.wav'
> > file '01/01.03_test.wav'
> > file '01/01.04_test.wav'
> >
> > ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a aac_he
> > -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out
> > And I can run the command on each individually.
> >..
> >..
> > Any help appreciated.
>
>
I will preface my comment by saying I'm not sure if this breaks ffmpeg
email list rules by suggesting this, so my apologies in advance if I'm not
supposed to.

I used ffmpeg for almost every video encoding and some audio encoding as
well.  Occasionally I will need to use opensource command line software
called sox.  I'm not endorsing, I'm not on their team, nor a friend of
anyone on the team, etc.  Just mentioning another tool to use.

According to the sox documentation you can use example below to concatenate.

sox ?m music.mp3 voice.wav mixed.flac

 I just tested it out using this:

 c:\sox\sox.exe "WS520148.WAV" "WS520149.WAV" total.wav


As far as I am aware, conjoining .wav files is not that simple.  While
> cat joins the files together, it doesn't take into
> account the RIFF header at the beginning of each one.  Thus you get a
> combined file that thinks it is only as long
> as the first sample in the list.  You either need a program that can
> join them correctly for you, or you take the headers
> into account when joining them, and adjust the parameters of the
> resulting file accordingly.


I won't dispute what Chris said at all because it maybe be true with ffmpeg
but I did ffprobe on the resulting file I got and got this:

c:\ffmpeg\ffmpeg\bin\ffprobe.exe -i total.wav
ffprobe version N-51683-g9dc88ac Copyright (c) 2007-2013 the FFmpeg
developers
  built on Apr  8 2013 21:19:21 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-iconv --enable-libass
--enable-libbluray --enable-libcaca --enable-libfreetyp
e --enable-libgsm --enable-libilbc --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-li
bopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger
--enable-libsoxr --enable-libspeex --enable-libtheo
ra --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libx264
 --enable-libxavs --enable-libxvid --enable-zlib
  libavutil      52. 25.100 / 52. 25.100
  libavcodec     55.  2.100 / 55.  2.100
  libavformat    55.  1.100 / 55.  1.100
  libavdevice    55.  0.100 / 55.  0.100
  libavfilter     3. 49.101 /  3. 49.101
  libswscale      2.  2.100 /  2.  2.100
  libswresample   0. 17.102 /  0. 17.102
  libpostproc    52.  2.100 / 52.  2.100
[wav @ 020df920] max_analyze_duration 5000000 reached at 5015510
microseconds
Input #0, wav, from 'total.wav':
  Duration: 00:01:07.01, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2
channels, s16, 1411 kb/s

In addition, windows media player and windows explorer gives the correct
time duration of the combined wav files.

Tim

From gliese849b at gmail.com  Mon Apr 28 08:24:14 2014
From: gliese849b at gmail.com (RDP)
Date: Mon, 28 Apr 2014 07:24:14 +0100
Subject: [FFmpeg-user] wav concat woes
In-Reply-To: <CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
References: <ljk2c8$jju$1@ger.gmane.org>
	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
	<CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
Message-ID: <CAEUFz7cHD8QEHhXbrk1nttsRyh1YAio43f0UWn2Tn9aOQqkskw@mail.gmail.com>

On Mon, Apr 28, 2014 at 5:49 AM, Tim Hiles <thiles at confex.com> wrote:
>> > On Fedora 20, trying to concatenate a set of wav files.
>> >
>> > cat 01_list.txt
>> > file '01/01.01_test.wav'
>> > file '01/01.02_test.wav'
>> > file '01/01.03_test.wav'
>> > file '01/01.04_test.wav'
>> >
>> > ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a aac_he
>> > -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out
>> > And I can run the command on each individually.
>> >..
>> >..
>> > Any help appreciated.
>>
>>
> I will preface my comment by saying I'm not sure if this breaks ffmpeg
> email list rules by suggesting this, so my apologies in advance if I'm not
> supposed to.
>
> I used ffmpeg for almost every video encoding and some audio encoding as
> well.  Occasionally I will need to use opensource command line software
> called sox.  I'm not endorsing, I'm not on their team, nor a friend of
> anyone on the team, etc.  Just mentioning another tool to use.
>
> According to the sox documentation you can use example below to concatenate.
>
> sox ?m music.mp3 voice.wav mixed.flac

Yes, I was awatre of sox,

    http://sox.sourceforge.net/

I'm sure it breaks no rules to mention it. Sox has a dfferent remit.

> As far as I am aware, conjoining .wav files is not that simple.  While
>> cat joins the files together, it doesn't take into
>> account the RIFF header at the beginning of each one.  Thus you get a
>> combined file that thinks it is only as long
>> as the first sample in the list.  You either need a program that can
>> join them correctly for you, or you take the headers
>> into account when joining them, and adjust the parameters of the
>> resulting file accordingly.
>
>
> I won't dispute what Chris said at all because it maybe be true with ffmpeg
> but I did ffprobe on the resulting file I got and got this:

I was specifically targetting the quoted use of the cat command here,
not ffmpeg.
cat does exactly what it says on the tin, i.e. joins files together.
cat does not diffentiate
between types of file, other than text and binary, and therefore has
no knowledge
of format of a wave file.

Same issue would surface for any other file with a similar header/data layout.
For example,

   cat flash01.swf flash02.swf > flash_combined.swf

simply won't produce a usuable file.

Regards,


Chris.

From cehoyos at ag.or.at  Mon Apr 28 09:04:37 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 07:04:37 +0000 (UTC)
Subject: [FFmpeg-user] mono or stereo? ffmpeg converts or phone?
References: <ljk9lv$97h$1@ger.gmane.org>
Message-ID: <loom.20140428T090223-420@post.gmane.org>

Sean Darcy <seandarcy2 <at> gmail.com> writes:

> So for a bitrate of let's say 48000, am I better off 
> converting to mono in ffmpeg ( -ac 1 ) or leaving it 
> stereo and letting the phone convert it?

You may be better off (and if playback is only mono, you 
don't loose anything) but please understand that you cannot 
double the quality / half the bitrate.
The chosen encoder is much more important, if your phone 
does not support opus, make sure to choose libfdk-aac.

Carl Eugen


From t.rapp at noa-audio.com  Mon Apr 28 09:10:09 2014
From: t.rapp at noa-audio.com (Tobias Rapp)
Date: Mon, 28 Apr 2014 09:10:09 +0200
Subject: [FFmpeg-user] Getting MPEG2 video stream bitrate
In-Reply-To: <loom.20140425T161712-777@post.gmane.org>
References: <20140425160509.6bde5be1e07111b47dbeb50d@noa-audio.com>
	<loom.20140425T161712-777@post.gmane.org>
Message-ID: <20140428091009.54313bb434c616a30f3bc0ea@noa-audio.com>

Carl Eugen Hoyos wrote:
> The stream does not save a (constant or average) bitrate, only the 
> maximum bitrate is stored.

Yes but the stream has a de-facto bitrate defined by the size of each
frame. And if I choose a "probesize" that is too small ffprobe reports:

[mxf @ 0x2f692e0] Stream #0: not enough frames to estimate rate; consider
increasing probesize

So I guess it is estimating the bitrate from frame size but just doen't
report the value. I want to find out if that is caused by an incomplete
arguments list on my side.

> > [mxf  <at>  0x27bd2e0] Field dominance 0 support is not implemented. 
> > Update your FFmpeg version to the newest one from Git. If the problem 
> > still occurs, it means that your file has a feature which has not been 
> > implemented.
> > [mxf  <at>  0x27bd2e0] If you want to help, upload a sample of this file to
> > ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel 
> > mailing list.
> 
> And it appears that we need such a sample.
> (I don't know much about mxf though.)

I had a look at the MXF input file with the mxfdump tool from mxflib and
that specific file does not contain any "field dominance" descriptor (see
attached log). So ffprobe/ffmpeg reports "0" because this is the
initialized value in ffmpeg.

Best regards,
Tobias
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From cehoyos at ag.or.at  Mon Apr 28 09:10:32 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 07:10:32 +0000 (UTC)
Subject: [FFmpeg-user] Best settings for upscaling 480p to 1080p?
References: <CAOEv1ka3eNAeW8P=NWtOiWM++XA+kaadSwDDvrvaNEE47VYB+w@mail.gmail.com>
Message-ID: <loom.20140428T090637-270@post.gmane.org>

John Klimek <jklimek <at> gmail.com> writes:

> What are some good settings or filters to use when 
> upscaling 480p content to 1080p?

These scaler settings improve quality but please understand 
that this takes a lot of processing power both at encoding 
and at decoding time, so it may not be worth the effort:
-sws_flags lanczos+accurate_rnd+full_chroma_int+full_chroma_inp

If you want to apply sharpening, the gaussian filter 
should allow this, I never tried it.

Carl Eugen


From cehoyos at ag.or.at  Mon Apr 28 09:28:41 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 07:28:41 +0000 (UTC)
Subject: [FFmpeg-user] wav concat woes
References: <ljk2c8$jju$1@ger.gmane.org>
	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
Message-ID: <loom.20140428T092701-280@post.gmane.org>

RDP <gliese849b <at> gmail.com> writes:

> On Mon, Apr 28, 2014 at 12:03 AM, Sean Darcy wrote:
> > trying to concatenate a set of wav files.

> As far as I am aware, conjoining .wav files is not 
> that simple.

This is why FFmpeg contains a concat demuxer and a 
concat filter, both are perfectly capable of 
concatenating wav files.

I opened ticket #3598 for the excessive verboseness.

Carl Eugen


From cehoyos at ag.or.at  Mon Apr 28 10:09:45 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 08:09:45 +0000 (UTC)
Subject: [FFmpeg-user] Getting MPEG2 video stream bitrate
References: <20140425160509.6bde5be1e07111b47dbeb50d@noa-audio.com>
	<loom.20140425T161712-777@post.gmane.org>
	<20140428091009.54313bb434c616a30f3bc0ea@noa-audio.com>
Message-ID: <loom.20140428T100343-652@post.gmane.org>

Tobias Rapp <t.rapp <at> noa-audio.com> writes:

> Carl Eugen Hoyos wrote:
> > The stream does not save a (constant or average) 
> > bitrate, only the maximum bitrate is stored.
> 
> Yes but the stream has a de-facto bitrate defined by 
> the size of each frame.

But this is not how the bitrate field in AVCodecContext 
is defined (and I would not rule out that a patch 
changing this will be rejected).

> And if I choose a "probesize" that is too small 
> ffprobe reports:
> 
> [mxf  <at>  0x2f692e0] Stream #0: not enough frames 
> to estimate rate; consider increasing probesize
> 
> So I guess it is estimating the bitrate from frame 
> size but just doen't report the value.

Sorry if you consider the warning misleading (a patch 
fixing this may be welcome) but this is not meant, 
at least not in the mxf-mpegvideo case.

> I want to find out if that is caused by an incomplete
> arguments list on my side.

No, as said mpeg2video files don't contain the 
information you are looking for.
One possibility to get it is to remux the whole 
video stream and calculate the bitrate.

ffmpeg (the application) could probably be changed so 
that it shows the actual input bitrate(s) on eof (after 
conversion). Such a patch may be welcome.

Carl Eugen


From t.rapp at noa-audio.com  Mon Apr 28 11:16:45 2014
From: t.rapp at noa-audio.com (Tobias Rapp)
Date: Mon, 28 Apr 2014 11:16:45 +0200
Subject: [FFmpeg-user] Getting MPEG2 video stream bitrate
In-Reply-To: <loom.20140428T100343-652@post.gmane.org>
References: <20140425160509.6bde5be1e07111b47dbeb50d@noa-audio.com>
	<loom.20140425T161712-777@post.gmane.org>
	<20140428091009.54313bb434c616a30f3bc0ea@noa-audio.com>
	<loom.20140428T100343-652@post.gmane.org>
Message-ID: <20140428111645.2b5a020e1564921e05dc3358@noa-audio.com>

Carl Eugen Hoyos wrote:
> No, as said mpeg2video files don't contain the 
> information you are looking for.
> One possibility to get it is to remux the whole 
> video stream and calculate the bitrate.

I have just found out that the "-show_frames" option of ffprobe displays
"pkt_size" and "pkt_duration_time". So it is possible to calculate an
average bitrate using something like "avg(pkt_size * 8 /
pkt_duration_time)".

Best regards,
Tobias


From barsnick at gmx.net  Mon Apr 28 11:41:38 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Mon, 28 Apr 2014 11:41:38 +0200
Subject: [FFmpeg-user] wav concat woes
In-Reply-To: <CAEUFz7cHD8QEHhXbrk1nttsRyh1YAio43f0UWn2Tn9aOQqkskw@mail.gmail.com>
References: <ljk2c8$jju$1@ger.gmane.org>
	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
	<CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
	<CAEUFz7cHD8QEHhXbrk1nttsRyh1YAio43f0UWn2Tn9aOQqkskw@mail.gmail.com>
Message-ID: <20140428094138.GB17030@sunshine.barsnick.net>

On Mon, Apr 28, 2014 at 07:24:14 +0100, RDP wrote:
> >> > cat 01_list.txt
> >> > file '01/01.01_test.wav'
> >> > file '01/01.02_test.wav'
> >> > file '01/01.03_test.wav'
> >> > file '01/01.04_test.wav'
> >> >
> >> > ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a aac_he
> >> > -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out
> >> > And I can run the command on each individually.

> I was specifically targetting the quoted use of the cat command here,
> not ffmpeg.
> cat does exactly what it says on the tin, i.e. joins files together.
> cat does not diffentiate between types of file, other than text and
> binary, and therefore has no knowledge of format of a wave file.

Chris, where do you see the use of 'cat' to combine files in the
original request? Sean used 'cat' only to show the content of the file
handed to ffmpeg's concat muxer. It's all about ffmpeg here.

Moritz

From cehoyos at ag.or.at  Mon Apr 28 11:44:32 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 09:44:32 +0000 (UTC)
Subject: [FFmpeg-user] wav concat woes
References: <ljk2c8$jju$1@ger.gmane.org>
	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
	<CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
	<CAEUFz7cHD8QEHhXbrk1nttsRyh1YAio43f0UWn2Tn9aOQqkskw@mail.gmail.com>
	<20140428094138.GB17030@sunshine.barsnick.net>
Message-ID: <loom.20140428T114404-751@post.gmane.org>

Moritz Barsnick <barsnick <at> gmx.net> writes:

> Sean used 'cat' only to show the content of the file
> handed to ffmpeg's concat muxer.

demuxer ;-)
^^

Carl Eugen


From kk_konrad at gmx.at  Mon Apr 28 15:48:08 2014
From: kk_konrad at gmx.at (Konrad Karl)
Date: Mon, 28 Apr 2014 15:48:08 +0200
Subject: [FFmpeg-user] mp4 concat demuxer problem
Message-ID: <0LZhoi-1XJd372j4G-00lYCs@mail.gmx.com>

Hi,

I tried to concatenate a few movies (h264/aac mp4, all produced by kdenlive/melt using
the same rendering settings) using the concat demuxer like this on
Fedora 20 x86_64:


ffmpeg -f concat -i concat-test -c copy out.mp4

The output file fails to play after the first input file (mplayer, ffplay, vlc)

Trimmed down samples have already been uploaded to upload.ffmpeg.org/incoming

Contents of the uploaded info file  and uncut console output of one concat command  below.

Konrad


======================= concat-test.info ==========================

Problem description: 

mp4: concat demuxer produces corrupt output after the first input file:

reproduce like this using samples from two different files (1 second duration each):

ffmpeg -f concat -i concat-test -c copy out.mp4

Result: out.mp4 fails to play the contents of the second and all further input files.
(tested with mplayer, ffplay, vlc)
The output container does not matter.(ts with -bsf h264_mp4toannexb, mkv)

then errors similar to this:
[h264 @ 0x7fa9040cb5a0] left block unavailable for requested intra mode at 0 0
[h264 @ 0x7fa9040cb5a0] error while decoding MB 0 0, bytestream 299                                                                                                                                     
[h264 @ 0x7fa9040cb5a0] concealing 8160 DC, 8160 AC, 8160 MV errors in I frame
ad nauseam

There must be some incompatibility between the files because if the same
file is concatenated multiple times if plays fine.

All files have been produced by kdenlive using the same rendering settings.
The samples have been cut from the start of the original files.
(ffmpeg -i infile.mp4 -t 1 -c copy outfile.mp4)

Fedora 20 x86_64, ffmpeg from rpmfusion  repository and also from todays git.

ffmpeg
ffmpeg version 2.1.4 Copyright (c) 2000-2014 the FFmpeg developers
  built on Feb 25 2014 08:24:47 with gcc 4.8.2 (GCC) 20131212 (Red Hat 4.8.2-7)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcdio --enable-libcelt --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101
  libavformat    55. 19.104 / 55. 19.104
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...
------------------
ffmpeg version N-62699-gc7b0890 Copyright (c) 2000-2014 the FFmpeg developers
  built on Apr 28 2014 11:55:48 with gcc 4.8.2 (GCC) 20131212 (Red Hat 4.8.2-7)
  configuration: --prefix=/home/ki/melt/20140428 --enable-gpl --enable-version3 --enable-shared --enable-debug --enable-pthreads --enable-runtime-cpudetect --disable-doc --enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-libx264 --enable-libvpx
  libavutil      52. 78.100 / 52. 78.100
  libavcodec     55. 59.100 / 55. 59.100
  libavformat    55. 37.101 / 55. 37.101
  libavdevice    55. 13.100 / 55. 13.100
  libavfilter     4.  4.100 /  4.  4.100
  libswscale      2.  6.100 /  2.  6.100
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...

==================  attempt to concat two mp4 files =========================================

ffmpeg -f concat  -i concat-test -c copy -y out.mp4
ffmpeg version 2.1.4 Copyright (c) 2000-2014 the FFmpeg developers
  built on Feb 25 2014 08:24:47 with gcc 4.8.2 (GCC) 20131212 (Red Hat
4.8.2-7)
  configuration: --prefix=/usr --bindir=/usr/bin
--datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg
--libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64
--optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
-fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches
-m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-frei0r
--enable-gnutls --enable-libass --enable-libcdio --enable-libcelt
--enable-libdc1394 --disable-indev=jack --enable-libfreetype
--enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv
--enable-libopenjpeg --enable-libopus --enable-libpulse
--enable-libschroedinger --enable-libsoxr --enable-libspeex
--enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx
--enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter
--enable-avresample --enable-postproc --enable-pthreads --disable-static
--enable-shared --enable-gpl --disable-debug --disable-stripping
--shlibdir=/usr/lib64 --enable-runtime-cpudetect
  libavutil      52. 48.101 / 52. 48.101
  libavcodec     55. 39.101 / 55. 39.101
  libavformat    55. 19.104 / 55. 19.104
  libavdevice    55.  5.100 / 55.  5.100
  libavfilter     3. 90.100 /  3. 90.100
  libavresample   1.  1.  0 /  1.  1.  0
  libswscale      2.  5.101 /  2.  5.101
  libswresample   0. 17.104 /  0. 17.104
  libpostproc    52.  3.100 / 52.  3.100
[concat @ 0x1305720] Estimating duration from bitrate, this may be
inaccurate
Input #0, concat, from 'concat-test':
  Duration: 00:00:00.00, start: 0.000000, bitrate: 38095 kb/s
    Stream #0:0: Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1920x1080 [SAR 1:1 DAR 16:9], 35326 kb/s, 50 fps, 50 tbr, 12800 tbn, 100
tbc
    Stream #0:1: Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp,
384 kb/s
Output #0, mp4, to 'out.mp4':
  Metadata:
    encoder         : Lavf55.19.104
    Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p,
1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 35326 kb/s, 50 fps, 12800 tbn,
12800 tbc
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo,
384 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=  104 fps=0.0 q=-1.0 Lsize=    5567kB time=00:00:02.04
bitrate=22325.9kbits/s    
video:5469kB audio:94kB subtitle:0 global headers:0kB muxing overhead
0.064527%






From kk_konrad at gmx.at  Mon Apr 28 15:56:30 2014
From: kk_konrad at gmx.at (Konrad Karl)
Date: Mon, 28 Apr 2014 15:56:30 +0200
Subject: [FFmpeg-user] mp4 concat demuxer problem - follow up to myself
In-Reply-To: <0LZhoi-1XJd372j4G-00lYCs@mail.gmx.com>
References: <0LZhoi-1XJd372j4G-00lYCs@mail.gmx.com>
Message-ID: <0MK0ur-1WdwhG3lQL-001S1V@mail.gmx.com>

.... hit send too fast..

fortunately the traditional way still works OK:

for i in concat-test-?.mp4;do ffmpeg -i $i -f mpegts  -c copy  -bsf h264_mp4toannexb -; \
                           done | ffmpeg -i - -c copy  -absf aac_adtstoasc -y out.mp4

Konrad

From rogerdpack2 at gmail.com  Mon Apr 28 18:27:31 2014
From: rogerdpack2 at gmail.com (Roger Pack)
Date: Mon, 28 Apr 2014 10:27:31 -0600
Subject: [FFmpeg-user] Need help with the optimal ffmpeg settings to
 publish video
In-Reply-To: <535B3C7D.2060700@xopnetworks.com>
References: <535B3C7D.2060700@xopnetworks.com>
Message-ID: <CAL1QdWfNvNvzeW9ftMfDhrv3kfAxL-y847zFyd_TydHvdDy=Wg@mail.gmail.com>

On 4/25/14, Yan Brenman <ybrenman at xopnetworks.com> wrote:
> Reaching out to all ffmpeg gurus,
>
> need any help I can get picking up the most optimal ffmpeg setting for
> publishing captured desktop/screen video.
> The type of application we are trying to implement is sort of
> desktop/screen sharing "webinar" used for the company training sessions.
> Presenter publishes streaming video of his/here desktop to the
> nginx-rtmp web server which then gets reflected to all clients connected
> to the nginx web server hosted website.
> For the streaming video publishing side we are planning to use ffmpeg
> (we tried other clients but ffmpeg seems to be providing much
> better quality) and for the client side it's going to be either ffplay
> or browser embedded flash player (if we get it working with ffmepg). The
> most obvious target platform for this is going to be Microsoft Windows.
> With H.264 compression, FLV format and "fast" preset we are getting
> pretty decent quality of the video, but the delay is around 10 seconds
> (which is obviously unacceptable). Even with reducing the quality we
> can't get delay lower than 5 seconds (and at that time quality
> becomes simply unacceptable).
> Clearly there is something we are not setting correctly, because I keep
> hearing about the delay bellow 1 seconds (unfortunately none of
> the examples provided with those posting ever works in reality).
> I would greatly appreciate any type of help/advise anybody can provide.
>
> Just for reference - here are the settings I am using right now:
>
> *ffmpeg -rtbufsize 300M -f dshow -i video="screen-capture-recorder" -c:v
> libx264 -preset fast -an -f flv rtmp://[server IP
> address]/live/screen_share <http://usn7.xopnetworks.com/live/screen_share>*

One thing you may be able to do is multicast through UDP to all your
receiving points.  If that's an option.

Anyway https://trac.ffmpeg.org/wiki/StreamingGuide#Latency may help.
I have never tried optimizing something with audio for low latency...

From dopelabs at dubstep.fm  Mon Apr 28 19:00:16 2014
From: dopelabs at dubstep.fm (DopeLabs)
Date: Mon, 28 Apr 2014 10:00:16 -0700
Subject: [FFmpeg-user] Need help with the optimal ffmpeg settings to
	publish video
In-Reply-To: <535B3C7D.2060700@xopnetworks.com>
References: <535B3C7D.2060700@xopnetworks.com>
Message-ID: <6E627E2A-9AB6-4C1D-9DFD-2EE67351AFAF@dubstep.fm>

i would start by looking at what the recommended settings are on sites like twitch/justin/ustream.tv/youtube.com/etc.. thats pretty much all these gamers are doing is capturing desktop and streaming.


generally speaking though

ffmpeg -f dshow -i video="screen-capture-recorder" -c:v libx264 -pix_fmt yuv420p -b:v 3500k -preset ultrafast -an -f flv rtmp://outurl.com/live


once you have established things look good you can start to scale down the bitrate or change the buffer size as needed until you reach a happy medium between bandwidth/cpu usage and the end quality of your output stream.






From cehoyos at ag.or.at  Mon Apr 28 22:57:00 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 20:57:00 +0000 (UTC)
Subject: [FFmpeg-user] mp4 concat demuxer problem
References: <0LZhoi-1XJd372j4G-00lYCs@mail.gmx.com>
Message-ID: <loom.20140428T225556-606@post.gmane.org>

Konrad Karl <kk_konrad <at> gmx.at> writes:

> I tried to concatenate a few movies (h264/aac mp4, 
> all produced by kdenlive/melt using the same rendering 
> settings) 

(The files have very different bitrates, are you sure 
they were produced with the same settings?)

> using the concat demuxer like this on Fedora 20 x86_64:
> 
> ffmpeg -f concat -i concat-test -c copy out.mp4
> 
> The output file fails to play after the first input file

I opened ticket #3600, thank you for the report!

Carl Eugen


From seandarcy2 at gmail.com  Mon Apr 28 23:23:56 2014
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Mon, 28 Apr 2014 17:23:56 -0400
Subject: [FFmpeg-user] wav concat woes
In-Reply-To: <CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
References: <ljk2c8$jju$1@ger.gmane.org>	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
	<CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
Message-ID: <ljmgtc$r3a$1@ger.gmane.org>

On 04/28/2014 12:49 AM, Tim Hiles wrote:
>>> On Fedora 20, trying to concatenate a set of wav files.
>>>
>>> cat 01_list.txt
>>> file '01/01.01_test.wav'
>>> file '01/01.02_test.wav'
>>> file '01/01.03_test.wav'
>>> file '01/01.04_test.wav'
>>>
>>> ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a aac_he
>>> -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out
>>> And I can run the command on each individually.
>>> ..
>>> ..
>>> Any help appreciated.
>>
>>
> I will preface my comment by saying I'm not sure if this breaks ffmpeg
> email list rules by suggesting this, so my apologies in advance if I'm not
> supposed to.
>
> I used ffmpeg for almost every video encoding and some audio encoding as
> well.  Occasionally I will need to use opensource command line software
> called sox.  I'm not endorsing, I'm not on their team, nor a friend of
> anyone on the team, etc.  Just mentioning another tool to use.
>
> According to the sox documentation you can use example below to concatenate.
>
> sox ?m music.mp3 voice.wav mixed.flac
>
>   I just tested it out using this:
>
>   c:\sox\sox.exe "WS520148.WAV" "WS520149.WAV" total.wav
>

Thanks for the suggestion. I'd still lie to know why -f concat didn't 
work in ffmpeg, but I did use sox as a workaround.

sox 01/01*wav  -t wav  - |
    ffmpeg -i - -vn -c:a libfdk_aac \
    -profile:a aac_he -b:a 48k -ac 1 -signaling implicit \
    01.m4a

Worked like a charm.

sean


From casey.bateman at hudl.com  Mon Apr 28 23:50:46 2014
From: casey.bateman at hudl.com (Casey Bateman)
Date: Mon, 28 Apr 2014 21:50:46 +0000
Subject: [FFmpeg-user] Mp4 Concat Demuxer Produces Incorrect Output
Message-ID: <7d4cd2b08c8a4441b75752635616c78e@CO1PR06MB206.namprd06.prod.outlook.com>

Hello,

I am in need of some assistance because of an issue I am seeing using the concat demuxer. I am trying to concatenate two .mp4 files. When I concatenate the two files together using the concat demuxer, the output file will display the first clip at normal speed, then the second clip exponentially faster. After inspecting the Ffmpeg output logs, the only thing that I can see different between these two files, is the tbn. Would the tbn being different on these files effect the output of the concat demuxer?

Concat.txt:
file 'input000.mp4'
file 'input001.mp4'

Command:
ffmpeg -f concat -i concat.txt -c  copy output.mp4

Ffprobe input000.mp4:
ffprobe version N-61821-gc05065a Copyright (c) 2007-2014 the FFmpeg developers
  built on Mar 26 2014 02:09:10 with gcc 4.8.2 (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-libvidstab --enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp
ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable
-zlib
  libavutil      52. 69.100 / 52. 69.100
  libavcodec     55. 54.100 / 55. 54.100
  libavformat    55. 35.101 / 55. 35.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'temp/input000.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf55.35.101
  Duration: 00:00:19.00, start: 0.000000, bitrate: 1138 kb/s
    Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yu
v420p, 852x480 [SAR 1:1 DAR 71:40], 1137 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc
(default)
    Metadata:
      handler_name    : VideoHandler

Ffprobe input001.mp4:
ffprobe version N-61821-gc05065a Copyright (c) 2007-2014 the FFmpeg developers
  built on Mar 26 2014 02:09:10 with gcc 4.8.2 (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-libvidstab --enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp
ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable
-zlib
  libavutil      52. 69.100 / 52. 69.100
  libavcodec     55. 54.100 / 55. 54.100
  libavformat    55. 35.101 / 55. 35.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'temp/input001.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 1970-01-01 00:00:00
    encoder         : Lavf52.78.3
  Duration: 00:00:19.37, start: 0.000000, bitrate: 1099 kb/s
    Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yu
v420p, 852x480 [SAR 1:1 DAR 71:40], 1097 kb/s, 30 fps, 30 tbr, 30 tbn, 60 tbc (d
efault)
    Metadata:
      creation_time   : 1970-01-01 00:00:00
      handler_name    : VideoHandler

Ffmpeg output:
ffmpeg version N-61821-gc05065a Copyright (c) 2000-2014 the FFmpeg developers
  built on Mar 26 2014 02:09:10 with gcc 4.8.2 (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-libvidstab --enable-libvo-aa
cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp
ack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable
-zlib
  libavutil      52. 69.100 / 52. 69.100
  libavcodec     55. 54.100 / 55. 54.100
  libavformat    55. 35.101 / 55. 35.101
  libavdevice    55. 11.100 / 55. 11.100
  libavfilter     4.  3.100 /  4.  3.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
[concat @ 0272e700] Estimating duration from bitrate, this may be inaccurate
Input #0, concat, from 'temp/concat.txt':
  Duration: 00:00:00.00, start: 0.000000, bitrate: 1230 kb/s
    Stream #0:0: Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p
, 852x480 [SAR 1:1 DAR 71:40], 1137 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc

frame= 1151 fps=0.0 q=-1.0 Lsize=    5239kB time=00:00:19.00 bitrate=2258.1kbits
/s
video:5233kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.108556%



Casey Bateman
Hudl | Software Developer
casey.bateman at hudl.com<mailto:casey.bateman at hudl.com> | 402.817.0060


From seandarcy2 at gmail.com  Mon Apr 28 23:53:15 2014
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Mon, 28 Apr 2014 17:53:15 -0400
Subject: [FFmpeg-user] mono or stereo? ffmpeg converts or phone?
In-Reply-To: <loom.20140428T090223-420@post.gmane.org>
References: <ljk9lv$97h$1@ger.gmane.org>
	<loom.20140428T090223-420@post.gmane.org>
Message-ID: <ljmikc$nhm$1@ger.gmane.org>

On 04/28/2014 03:04 AM, Carl Eugen Hoyos wrote:
> Sean Darcy <seandarcy2 <at> gmail.com> writes:
>
>> So for a bitrate of let's say 48000, am I better off
>> converting to mono in ffmpeg ( -ac 1 ) or leaving it
>> stereo and letting the phone convert it?
>
> You may be better off (and if playback is only mono, you
> don't loose anything) but please understand that you cannot
> double the quality / half the bitrate.
> The chosen encoder is much more important, if your phone
> does not support opus, make sure to choose libfdk-aac.
>
> Carl Eugen
>

I don't have an audio book reader that can read opus installed on my 
phone. I don't even know if there is one. I do use opus for voip: it's 
really amazing.

But back to the OP. I am using libfdk-aac. I understand that 24000 mono 
will not equal 48000 stereo.

My real question is: how does ffmpeg downmix 2 -> 1. I assume it doesn't 
just take one channel. Does it "average" in some way the two channels?

I also assume whatever ffmpeg does, it does it better the android code 
on a phone doing it in realtime.

sean



From kk_konrad at gmx.at  Tue Apr 29 00:30:42 2014
From: kk_konrad at gmx.at (Konrad Karl)
Date: Tue, 29 Apr 2014 00:30:42 +0200
Subject: [FFmpeg-user] mp4 concat demuxer problem
In-Reply-To: <loom.20140428T225556-606@post.gmane.org>
References: <0LZhoi-1XJd372j4G-00lYCs@mail.gmx.com>
	<loom.20140428T225556-606@post.gmane.org>
Message-ID: <0MBFgr-1WmuQb1cIv-00ACQ4@mail.gmx.com>

Hi Carl Eugen!

On Mon, Apr 28, 2014 at 08:57:00PM +0000, Carl Eugen Hoyos wrote:
> Konrad Karl <kk_konrad <at> gmx.at> writes:
> 
> > I tried to concatenate a few movies (h264/aac mp4, 
> > all produced by kdenlive/melt using the same rendering 
> > settings) 
> 
> (The files have very different bitrates, are you sure 
> they were produced with the same settings?)

Yes, I am quite sure because I deliberately used the very same 
settings for these movies (two-pass  25 Mbit). I have tried
with several of the seven movies (one for each day of my holiday
trip, all but the first fading from and into black) and all
combinations I tried including reversing file ordering
failed to play after the first input file.

The moving Vietnamese flag on the first sample produces a higher
than average bitrate and the rather low bit rate of the second sample
seems to be caused by the fade-from-black transition. 

Original file size / duration comes quite close to 25 MBit/s.

Concatenating each file with itself seems to work OK.

Konrad

.....
> I opened ticket #3600, thank you for the report!
> 
> Carl Eugen


From cehoyos at ag.or.at  Tue Apr 29 00:57:01 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 22:57:01 +0000 (UTC)
Subject: [FFmpeg-user] wav concat woes
References: <ljk2c8$jju$1@ger.gmane.org>	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
	<CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
	<ljmgtc$r3a$1@ger.gmane.org>
Message-ID: <loom.20140429T005624-989@post.gmane.org>

Sean Darcy <seandarcy2 <at> gmail.com> writes:

> I'd still lie to know why -f concat didn't 
> work in ffmpeg

Could you elaborate on what did not work?
I apparently misunderstood your report...

Carl Eugen


From cehoyos at ag.or.at  Tue Apr 29 01:03:55 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Mon, 28 Apr 2014 23:03:55 +0000 (UTC)
Subject: [FFmpeg-user] Mp4 Concat Demuxer Produces Incorrect Output
References: <7d4cd2b08c8a4441b75752635616c78e@CO1PR06MB206.namprd06.prod.outlook.com>
Message-ID: <loom.20140429T010301-212@post.gmane.org>

Casey Bateman <casey.bateman <at> hudl.com> writes:

> Would the tbn being different on these files 
> effect the output of the concat demuxer?

I would assume that different timebases may ruin 
timestamps, yes.
But I wonder if this cannot be fixed...

Carl Eugen


From asdfg1945 at gmail.com  Sun Apr 27 21:31:02 2014
From: asdfg1945 at gmail.com (Arachnia)
Date: Sun, 27 Apr 2014 12:31:02 -0700 (PDT)
Subject: [FFmpeg-user] duration errors for help
In-Reply-To: <1398405968411-4664924.post@n4.nabble.com>
References: <1398405968411-4664924.post@n4.nabble.com>
Message-ID: <1398627062570-4664955.post@n4.nabble.com>

Oh, nobody can help?



--
View this message in context: http://ffmpeg-users.933282.n4.nabble.com/duration-errors-for-help-tp4664924p4664955.html
Sent from the FFmpeg-users mailing list archive at Nabble.com.

From boguslaw.lotko at chello.at  Mon Apr 28 23:14:01 2014
From: boguslaw.lotko at chello.at (Bogdan Lotko)
Date: Mon, 28 Apr 2014 23:14:01 +0200
Subject: [FFmpeg-user] Poor quality of screencast 1920x1200 on Linux
Message-ID: <535EC499.6060909@chello.at>

Hello,

I've tried to make a screencast to demonstrate a running software.
I am working on SLES11 installed on Lenovo W500 laptop. The screen 
resolution is 1920x1200.
I've compiled and installed libx264 (/usr/local/lib) and ffmpeg with 
libx264 of course.

I've tried to use following commands:
ffmpeg -f x11grab -r 30 -s 1920x1200 -vb 2000 -i :0.0  -threads 0 
output.xxx
ffmpeg -f x11grab -s 1920x1200 -framerate 25 -vb 2000 -i :0.0 -vb 2000  
output.xxx
ffmpeg -framerate 30 -video_size 1920x1200 -f x11grab -vb 2000 -i :0.0 
-vcodec libx264 -crf 0 -preset ultrafast output.xxx

where xxx was:
  -  mkv
  - flv
  - mp4 (with -vcodec libx264 -crf 0 -preset ultrafast was not readable)
  - avi

I've also tried all above commands with the frame rate 25.

With some exceptions (last command and mp4 output file) I've got usable 
output files, but the quality was a big problem: blurred pixels - so 
that the texts were hardly readable, and
video seems to run too fast (the second is not critical)

My questions:
  - Is it at all possible to make the screencasts with so high screen 
resolution?
  - If yest how to do it?
  - I am absolutely new in this topics, so please write me  what kind of 
input do you need to be able to help me?

thank you in advance for your help,

with best regards

Bogdan
-- 

*Bogdan Lotko*

b.lotko at chello.at
Phone: +43 1 2852458
Mobile: +43 676 6615012

From seandarcy2 at gmail.com  Mon Apr 28 23:17:23 2014
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Mon, 28 Apr 2014 17:17:23 -0400
Subject: [FFmpeg-user] wav concat woes
In-Reply-To: <loom.20140428T092701-280@post.gmane.org>
References: <ljk2c8$jju$1@ger.gmane.org>	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
	<loom.20140428T092701-280@post.gmane.org>
Message-ID: <535EC563.2090809@gmail.com>

On 04/28/2014 03:28 AM, Carl Eugen Hoyos wrote:
> RDP <gliese849b <at> gmail.com> writes:
>
>> On Mon, Apr 28, 2014 at 12:03 AM, Sean Darcy wrote:
>>> trying to concatenate a set of wav files.
>
>> As far as I am aware, conjoining .wav files is not
>> that simple.
>
> This is why FFmpeg contains a concat demuxer and a
> concat filter, both are perfectly capable of
> concatenating wav files.
>
> I opened ticket #3598 for the excessive verboseness.
>
> Carl Eugen
>

Not sure I understand. Should my command line have worked? If not, what 
was I doing wrong? Or is it the case that the concat demuxer doesn't 
handle wav?

sean

From seandarcy2 at gmail.com  Tue Apr 29 02:37:25 2014
From: seandarcy2 at gmail.com (Sean Darcy)
Date: Mon, 28 Apr 2014 20:37:25 -0400
Subject: [FFmpeg-user] mono or stereo? ffmpeg converts or phone?
In-Reply-To: <ljmikc$nhm$1@ger.gmane.org>
References: <ljk9lv$97h$1@ger.gmane.org>	<loom.20140428T090223-420@post.gmane.org>
	<ljmikc$nhm$1@ger.gmane.org>
Message-ID: <ljms86$tk$1@ger.gmane.org>

On 04/28/2014 05:53 PM, Sean Darcy wrote:
> On 04/28/2014 03:04 AM, Carl Eugen Hoyos wrote:
>> Sean Darcy <seandarcy2 <at> gmail.com> writes:
>>
>>> So for a bitrate of let's say 48000, am I better off
>>> converting to mono in ffmpeg ( -ac 1 ) or leaving it
>>> stereo and letting the phone convert it?
>>
>> You may be better off (and if playback is only mono, you
>> don't loose anything) but please understand that you cannot
>> double the quality / half the bitrate.
>> The chosen encoder is much more important, if your phone
>> does not support opus, make sure to choose libfdk-aac.
>>
>> Carl Eugen
>>
>
> I don't have an audio book reader that can read opus installed on my
> phone. I don't even know if there is one. I do use opus for voip: it's
> really amazing.
>
> But back to the OP. I am using libfdk-aac. I understand that 24000 mono
> will not equal 48000 stereo.
>
> My real question is: how does ffmpeg downmix 2 -> 1. I assume it doesn't
> just take one channel. Does it "average" in some way the two channels?
>
> I also assume whatever ffmpeg does, it does it better the android code
> on a phone doing it in realtime.
>
> sean

More specifically. My concern is file size on the phone. It seems to me 
I'll always have smaller file sizes for the same quality (in mono) if I 
convert to mono first. Maybe not double the quality / half the bitrate, 
but for any given bitrate, half the size. If I have a 48000 bitrate with 
two channels, isn't that file size ~twice the file size of a mono 48000 
bitrate?

sean


From thorswill at hotmail.com  Tue Apr 29 02:43:11 2014
From: thorswill at hotmail.com (Tobie Horswill)
Date: Mon, 28 Apr 2014 20:43:11 -0400
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <535A5187.5030107@mail.de>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>,
	<535A5187.5030107@mail.de>
Message-ID: <BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>


Thilo,

I've checked the config.log file although I can't say I knew exactly what I was looking for... Didn't see anything explicitly wrong regarding AVFoundation but I did notice a few error messages about header files not being found such as 'features.h', 'newlib.h', '_mingw.h', etc. Relevant?

Here is a link to the file (I couldn't attach it, too large)
https://www.sugarsync.com/pf/D6573428_87559618_018747


Thanks,

--
TH

> Date: Fri, 25 Apr 2014 14:13:59 +0200
> From: thilo.borgmann at mail.de
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
>
> Am 24.04.14 19:52, schrieb Tobie Horswill:
>> I'm having difficulties getting the new AVfoundation input device working
>> under Mac OS X 10.7
>>
>> I've rebuilt from the latest source which includes the AVFoundation additions
>> with --enable-indev=avfoundation but I still get an : Unknown input format
>> 'avfoundation' whenever I try to use it.
>>
>> I gather this should work under OS X 10.7+ correct?
>
> Yes.
>
>> I unfortunately can't run
>> Mavericks on this computer.
>>
>> Any input is welcome.
>
> Check your config.log, it will most likely be a compiler/framework header issue.
>
> If you can't try another (or updated) compiler, you might be able to switch to
> the qtkit input device, still functional up to 10.9.
>
> If in doubt you should make your config.log available for us to see.
>
> -Thilo
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
 		 	   		  

From mrukant at gmail.com  Tue Apr 29 07:15:20 2014
From: mrukant at gmail.com (Mrukant Popat)
Date: Mon, 28 Apr 2014 22:15:20 -0700
Subject: [FFmpeg-user] Not able to get constant 25fps for H.264 HD Transport
	Stream after transcoding
Message-ID: <CADDQtht7VApc+T9yoiOtHGwkYQ05zy4inQPbM+Mh9tKFF0KA6A@mail.gmail.com>

Dear All,

Below command line generates a Constant bitrate TS of 5000Kbps , but the
video elementary stream inside does not show 25 fps when using MediaInfo.
Infact when I play it with mplayer, it plays with 50fps. However, VLC plays
with 25fps.

Here is the command line

ffmpeg.exe -loglevel debug -i input.ts -pix_fmt yuv420p -r 25.0 -c:s copy
-c:v libx264 -preset slow -tune film  -x264opts
tff:fps=25.0:bitrate=4500:ref=3:threads=6:no-scenecut
:keyint=50:vbv-bufsize=180:vbv-maxrate=4500:videoformat=pal -c:a libfaac
 -ab 192000 -ar 48000 -f mpegts -mpegts_pmt_start_pid 0x64 -map 0:0 -map
0:1 -map 0:2 -
map 0:3 -streamid 0:1000 -streamid 1:1001 -streamid 2:1002 -streamid 3:1005
-muxrate 5M output.ts > c:\commandoutput.txt 2>&1



Here is the link to detailed output to this command --->
https://www.dropbox.com/s/qbb754cxaozxjik/commandoutput.txt


I am looking to get video with 25 fps and also the value of 25 fps set in
VUI of SEI inside H.264 stream.


Thank you,
Mrukant Popat

From barsnick at gmx.net  Tue Apr 29 07:42:25 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Tue, 29 Apr 2014 07:42:25 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
	<535A5187.5030107@mail.de>
	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>
Message-ID: <20140429054225.GA2402@sunshine.barsnick.net>

On Mon, Apr 28, 2014 at 20:43:11 -0400, Tobie Horswill wrote:

> I've checked the config.log file although I can't say I knew exactly
> what I was looking for... Didn't see anything explicitly wrong
> regarding AVFoundation but I did notice a few error messages about

Oh, very much so, from line 8923. (I am missing some explicit message
from ./configure saying "device avfoundation failed" or the likes
though. But I don't know whether it should be expected.)

Moritz

From cehoyos at ag.or.at  Tue Apr 29 09:26:23 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 29 Apr 2014 07:26:23 +0000 (UTC)
Subject: [FFmpeg-user] Poor quality of screencast 1920x1200 on Linux
References: <535EC499.6060909@chello.at>
Message-ID: <loom.20140429T092402-407@post.gmane.org>

Bogdan Lotko <boguslaw.lotko <at> chello.at> writes:

> ffmpeg -f x11grab -r 30 -s 1920x1200 -vb 2000 -i :0.0 
> -threads 0 output.xxx

Complete, uncut console output missing.
(-threads 0 is the default)

> With some exceptions (last command and mp4 output file) 
> I've got usable output files, but the quality was a 
> big problem: blurred pixels

If you encode to pix_fmt yuv420p you get blurring, if 
you encode to rgb (or yuv444p) with x264, you get files 
that only FFmpeg (-based software) can decode.

Carl Eugen


From cehoyos at ag.or.at  Tue Apr 29 09:28:15 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 29 Apr 2014 07:28:15 +0000 (UTC)
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
	<535A5187.5030107@mail.de>
	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>
	<20140429054225.GA2402@sunshine.barsnick.net>
Message-ID: <loom.20140429T092649-354@post.gmane.org>

Moritz Barsnick <barsnick <at> gmx.net> writes:

> I am missing some explicit message from 
> ./configure saying "device avfoundation failed" 
> or the likes though.

That would be very annoying for users of Linux, 
*BSD, OSX 10.5 etc. ...

Carl Eugen


From dopelabs at dubstep.fm  Tue Apr 29 11:04:38 2014
From: dopelabs at dubstep.fm (DopeLabs)
Date: Tue, 29 Apr 2014 02:04:38 -0700
Subject: [FFmpeg-user] Poor quality of screencast 1920x1200 on Linux
In-Reply-To: <535EC499.6060909@chello.at>
References: <535EC499.6060909@chello.at>
Message-ID: <E18F78E2-8AED-45CD-8F14-AED0C61F865A@dubstep.fm>

arent the video bitrates specified in bits? should that not be 2000k ?

From barsnick at gmx.net  Tue Apr 29 11:06:33 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Tue, 29 Apr 2014 11:06:33 +0200
Subject: [FFmpeg-user] mono or stereo? ffmpeg converts or phone?
In-Reply-To: <ljms86$tk$1@ger.gmane.org>
References: <ljk9lv$97h$1@ger.gmane.org>
	<loom.20140428T090223-420@post.gmane.org>
	<ljmikc$nhm$1@ger.gmane.org> <ljms86$tk$1@ger.gmane.org>
Message-ID: <20140429090633.GA3679@sunshine.barsnick.net>

On Mon, Apr 28, 2014 at 20:37:25 -0400, Sean Darcy wrote:

> 48000 bitrate with two channels, isn't that file size ~twice the file
> size of a mono 48000 bitrate?

No, not with compressed formats, unless your two audio channels carry
totally uncorrelated content. I'm saying compression can take advantage
of redundancies between the two channels.

And I don't agree with this concern of yours:

> My real question is: how does ffmpeg downmix 2 -> 1. I assume it
> doesn't just take one channel. Does it "average" in some way the two
> channels?
>
> I also assume whatever ffmpeg does, it does it better the android
> code on a phone doing it in realtime.

Downmixing stereo to mono is usually just an average (yes, there are
other approaches), and both the complexity and the required compute
power are negligible.

Moritz

From barsnick at gmx.net  Tue Apr 29 12:25:24 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Tue, 29 Apr 2014 12:25:24 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <loom.20140429T092649-354@post.gmane.org>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
	<535A5187.5030107@mail.de>
	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>
	<20140429054225.GA2402@sunshine.barsnick.net>
	<loom.20140429T092649-354@post.gmane.org>
Message-ID: <20140429102524.GB3679@sunshine.barsnick.net>

On Tue, Apr 29, 2014 at 07:28:15 +0000, Carl Eugen Hoyos wrote:
> Moritz Barsnick <barsnick <at> gmx.net> writes:
> 
> > I am missing some explicit message from 
> > ./configure saying "device avfoundation failed" 
> > or the likes though.
> 
> That would be very annoying for users of Linux, 
> *BSD, OSX 10.5 etc. ...

Okay, so let me rephrase that:

a) A feature was explicitly requested (though I agree this is
debatable), and cannot be fulfilled. A note - in order to avoid the
word "error message" seems appropriate.

I just noticed that avfoundation is being checked either way, whether I
explicitly --enable it or not. Which is okay.

b) ./configure is testing for certain features. I see errors in the
log, but can't easily decode which feature was being tested and what
the result was. I'm thinking in terms of autoconf: "Checking for ...
not found."

That's all I was trying to say. :-)

Regards,
Moritz

From cehoyos at ag.or.at  Tue Apr 29 12:42:12 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Tue, 29 Apr 2014 10:42:12 +0000 (UTC)
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
	<535A5187.5030107@mail.de>
	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>
	<20140429054225.GA2402@sunshine.barsnick.net>
	<loom.20140429T092649-354@post.gmane.org>
	<20140429102524.GB3679@sunshine.barsnick.net>
Message-ID: <loom.20140429T123515-181@post.gmane.org>

Moritz Barsnick <barsnick <at> gmx.net> writes:

> a) A feature was explicitly requested

The "feature" (AVFoundation) is autodetected, you 
cannot "request" it. Only if you specify 
--disable-indevs (or --disable-everything) you 
may enable the specific device you need.

It is similar to "--enable-zlib" which is also 
an option that (normally) has no effect (and 
never triggers a configure error).

Given that this is the behaviour for many, many 
years, I am not convinced that changing it will 
make us many friends.

Please understand that FFmpeg's configure is 
not autoconf-based and is not supposed to 
work like an autoconf-base configure script.

Carl Eugen


From barsnick at gmx.net  Tue Apr 29 13:11:03 2014
From: barsnick at gmx.net (Moritz Barsnick)
Date: Tue, 29 Apr 2014 13:11:03 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <loom.20140429T123515-181@post.gmane.org>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>
	<535A5187.5030107@mail.de>
	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>
	<20140429054225.GA2402@sunshine.barsnick.net>
	<loom.20140429T092649-354@post.gmane.org>
	<20140429102524.GB3679@sunshine.barsnick.net>
	<loom.20140429T123515-181@post.gmane.org>
Message-ID: <20140429111103.GC3679@sunshine.barsnick.net>

On Tue, Apr 29, 2014 at 10:42:12 +0000, Carl Eugen Hoyos wrote:
> Given that this is the behaviour for many, many 
> years, I am not convinced that changing it will 
> make us many friends.

No no, I was in no way asking you to change it. :)

> Please understand that FFmpeg's configure is 
> not autoconf-based and is not supposed to 
> work like an autoconf-base configure script.

Well understood, and the reasoning behind it. It's only feedback that I
myself had slight trouble identifying whether the avfoundation indev
feature had succeeded or not.

Moritz

From casey.bateman at hudl.com  Tue Apr 29 16:23:28 2014
From: casey.bateman at hudl.com (Casey Bateman)
Date: Tue, 29 Apr 2014 14:23:28 +0000
Subject: [FFmpeg-user] Mp4 Concat Demuxer Produces Incorrect Output
In-Reply-To: <loom.20140429T010301-212@post.gmane.org>
References: <7d4cd2b08c8a4441b75752635616c78e@CO1PR06MB206.namprd06.prod.outlook.com>
	<loom.20140429T010301-212@post.gmane.org>
Message-ID: <403a1b3709c3418399f2b87c041b0845@CO1PR06MB206.namprd06.prod.outlook.com>

> I would assume that different timebases may ruin timestamps, yes.
> But I wonder if this cannot be fixed...

Are you proposing an Ffmpeg fix, or is there something (besides a re-encode) that I could do to get this working? 


Casey Bateman
Hudl | Software Developer
casey.bateman at hudl.com | 402.817.0060 

From thorswill at hotmail.com  Tue Apr 29 16:26:38 2014
From: thorswill at hotmail.com (Tobie Horswill)
Date: Tue, 29 Apr 2014 10:26:38 -0400
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <20140429054225.GA2402@sunshine.barsnick.net>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>,
	<535A5187.5030107@mail.de>,
	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>,
	<20140429054225.GA2402@sunshine.barsnick.net>
Message-ID: <BLU168-W9310BF54330CCF48D2EB9AA3460@phx.gbl>

Thank you Moritz for pointing this out and thank you Carl Eugen for explaining why I may have missed it in the first place!

So it would appear mmintrin.h, xmmintrin.h and emmintrin.h are each missing multiple prototypes.?

Any hints as to why this is happening? Not sure if it is relevant but I first tried compiling using MacPorts then switched to Brew which got me a bit further but apparently not far enough.

My ./config file seems somewhat similar to the one mentioned here : http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2011-April/110864.html
Would clearing the definition of ?__GNUC_STDC_INLINE__ be relevant in my case?

Tobie

----------------------------------------
> Date: Tue, 29 Apr 2014 07:42:25 +0200
> From: barsnick at gmx.net
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
>
> On Mon, Apr 28, 2014 at 20:43:11 -0400, Tobie Horswill wrote:
>
>> I've checked the config.log file although I can't say I knew exactly
>> what I was looking for... Didn't see anything explicitly wrong
>> regarding AVFoundation but I did notice a few error messages about
>
> Oh, very much so, from line 8923. (I am missing some explicit message
> from ./configure saying "device avfoundation failed" or the likes
> though. But I don't know whether it should be expected.)
>
> Moritz
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
 		 	   		  

From jshupert at pps-inc.com  Tue Apr 29 17:06:05 2014
From: jshupert at pps-inc.com (Jim Shupert)
Date: Tue, 29 Apr 2014 11:06:05 -0400
Subject: [FFmpeg-user] Mp4 Concat Demuxer Produces Incorrect Output
In-Reply-To: <7d4cd2b08c8a4441b75752635616c78e@CO1PR06MB206.namprd06.prod.outlook.com>
References: <7d4cd2b08c8a4441b75752635616c78e@CO1PR06MB206.namprd06.prod.outlook.com>
Message-ID: <535FBFDD.8000501@pps-inc.com>


  Would the tbn being different on these files effect the output of the concat demuxer?

I do not have a direct response to your question but I know that
one can have 2 files that are the same in every way except for 1
file a is drop frame time code
file b is Non drop time code

and those 2 files will NOT concat

in regards to your Q -my guess is - yes.  I think that explains what your seeing



From arnauda.contact at gmail.com  Tue Apr 29 15:01:49 2014
From: arnauda.contact at gmail.com (Arnaud A)
Date: Tue, 29 Apr 2014 15:01:49 +0200
Subject: [FFmpeg-user] RTMP to RTMP video track copy throwing error
Message-ID: <CAJUc2VC777dcbRCMeD+97j287Z6WJxR3FWzu6edqUh=txjq1oQ@mail.gmail.com>

Hi all,

I need to fecth a RTMP H.264/MP3 stream and push it to RTMP H264/AAC
stream. I would like to leave the video track untouched as I have several
streams to process and would like CPU usage to remain as low as possible
So far I have been using:
./ffmpeg -i rtmp://myserver/live/test -c:v copy -c:a libvo_aacenc -b:a 96k
-f flv rtmp://myserver/live-flash/test

Now one out of 10 tries it passes and the stream gets transcoded but most
of time I have the following error. The incoming stream is not transcoded
to H264 with x264 (it is with Flash Media Live Encoder) but if I specify
-c:v libx264 (+x264 options) instead of -c:v copy it works 100% of the
time. Any hints on what is happening? I tried on another machine and the
same RTMP feed but I got the exact same error.

Thanks
Arnaud

ffmpeg version 2.2.1-   http://johnvansickle.com/ffmpeg/    Copyright (c)
2000-2014 the FFmpeg developers
  built on Apr 11 2014 04:58:33 with gcc 4.8 (Debian 4.8.2-19)
  configuration: --enable-gpl --enable-version3 --disable-shared
--disable-debug --enable-runtime-cpudetect --enable-libmp3lame
--enable-libx264 --enable-libwebp --enable-libspeex --enable-libvorbis
--enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-gray
--enable-libopenjpeg --enable-libopus --disable-ffserver
  libavutil      52. 66.100 / 52. 66.100
  libavcodec     55. 52.102 / 55. 52.102
  libavformat    55. 33.100 / 55. 33.100
  libavdevice    55. 10.100 / 55. 10.100
  libavfilter     4.  2.100 /  4.  2.100
  libswscale      2.  5.102 /  2.  5.102
  libswresample   0. 18.100 /  0. 18.100
  libpostproc    52.  3.100 / 52.  3.100
[flv @ 0x401d520] Stream discovered after head already parsed
[flv @ 0x401d520] negative cts, previous timestamps might be wrong
    Last message repeated 61 times
Input #0, flv, from 'rtmp://87.106.183.100/live/test':
  Metadata:
    author          :
    copyright       :
    description     :
    keywords        :
    rating          :
    title           :
    presetname      : Custom
    creationdate    : Tue Apr 29 14:17:16 2014
                    :
    videodevice     : Logitech HD Webcam C525
    avclevel        : 40
    avcprofile      : 77
    videokeyframe_frequency: 5
    audiodevice     : Microphone (Jabra UC VOICE 550a)
    audiochannels   : 2
    audioinputvolume: 75
  Duration: N/A, start: 0.000000, bitrate: 847 kb/s
    Stream #0:0: Video: h264 (Main), yuv420p(tv), 640x360 [SAR 1:1 DAR
16:9], 716 kb/s, 15.17 fps, 48.17 tbr, 1k tbn, 30 tbc
    Stream #0:1: Audio: mp3, 44100 Hz, stereo, s16p, 131 kb/s
    Stream #0:2: Data: none
Output #0, flv, to 'rtmp://87.106.183.100/live-flash/test':
  Metadata:
    author          :
    copyright       :
    description     :
    keywords        :
    rating          :
    title           :
    presetname      : Custom
    creationdate    : Tue Apr 29 14:17:16 2014
                    :
    videodevice     : Logitech HD Webcam C525
    avclevel        : 40
    avcprofile      : 77
    videokeyframe_frequency: 5
    audiodevice     : Microphone (Jabra UC VOICE 550a)
    audiochannels   : 2
    audioinputvolume: 75
    encoder         : Lavf55.33.100
    Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 640x360 [SAR
1:1 DAR 16:9], q=2-31, 716 kb/s, 15.17 fps, 1k tbn, 1k tbc
    Stream #0:1: Audio: aac (libvo_aacenc) ([10][0][0][0] / 0x000A), 44100
Hz, stereo, s16, 96 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (mp3 -> libvo_aacenc)
Press [q] to stop, [?] for help
[flv @ 0x41b4680] pts (-4080) < dts (67) in stream 0
av_interleaved_write_frame(): Invalid argument
[libvo_aacenc @ 0x41b5640] Trying to remove 1024 samples, but the queue is
empty
[flv @ 0x41b4680] Encoder did not produce proper pts, making some up.
[libvo_aacenc @ 0x41b5640] Trying to remove 1024 samples, but the queue is
empty
[flv @ 0x41b4680] Failed to update header with correct duration.
[flv @ 0x41b4680] Failed to update header with correct filesize.
frame=    2 fps=0.0 q=-1.0 Lsize=      30kB time=00:00:00.04
bitrate=5383.3kbits/s
video:36kB audio:1kB subtitle:0 data:0 global headers:0kB muxing overhead
-16.655897%

From boguslaw.lotko at chello.at  Tue Apr 29 10:23:21 2014
From: boguslaw.lotko at chello.at (Bogdan Lotko)
Date: Tue, 29 Apr 2014 10:23:21 +0200
Subject: [FFmpeg-user] Poor quality of screencast 1920x1200 on Linux
In-Reply-To: <loom.20140429T092402-407@post.gmane.org>
References: <535EC499.6060909@chello.at>
	<loom.20140429T092402-407@post.gmane.org>
Message-ID: <535F6179.2070108@chello.at>

Hello Carl Eugen,

Thank you for the answer.

the console output looks kike this:

bogdan at SLES11:~> /opt/ffmpeg-2.2.1/bin/ffmpeg -f x11grab -s 1920x1200 
-framerate 30 -vb 2000k -i :0.0 output.avi
ffmpeg version 2.2.1 Copyright (c) 2000-2014 the FFmpeg developers
   built on Apr 26 2014 09:48:05 with gcc 4.3 (SUSE Linux)
   configuration: --prefix=/opt/ffmpeg-2.2.1 --disable-yasm 
--enable-x11grab --enable-gpl --enable-libx264
   libavutil      52. 66.100 / 52. 66.100
   libavcodec     55. 52.102 / 55. 52.102
   libavformat    55. 33.100 / 55. 33.100
   libavdevice    55. 10.100 / 55. 10.100
   libavfilter     4.  2.100 /  4.  2.100
   libswscale      2.  5.102 /  2.  5.102
   libswresample   0. 18.100 /  0. 18.100
   libpostproc    52.  3.100 / 52.  3.100
[x11grab @ 0x9233e60] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 
1920 height: 1200
[x11grab @ 0x9233e60] shared memory extension found
Input #0, x11grab, from ':0.0':
   Duration: N/A, start: 1398759630.081104, bitrate: N/A
     Stream #0:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1920x1200, 
-2147483 kb/s, 30 tbr, 1000k tbn, 30 tbc
Output #0, avi, to 'output.avi':
   Metadata:
     ISFT            : Lavf55.33.100
     Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 1920x1200, 
q=2-31, 200 kb/s, 30 tbn, 30 tbc
Stream mapping:
   Stream #0:0 -> #0:0 (rawvideo -> mpeg4)
Press [q] to stop, [?] for help
[swscaler @ 0x9222040] Warning: data is not aligned! This can lead to a 
speedloss
frame=  228 fps= 10 q=31.0 Lsize=    2849kB time=00:00:22.46 
bitrate=1038.8kbits/s
video:2828kB audio:0kB subtitle:0 data:0 global headers:0kB muxing 
overhead 0.758680%

As I wrote I'd like to use it for presentation so the video will be 
played on Windows.
My goal is just to produce a video with acceptable quality, so I'll be 
very pleased if you could give ma a kind of cook-book instruction how to 
do it.

Thank your for your help

regards,

Bogdan

On 04/29/2014 09:26 AM, Carl Eugen Hoyos wrote:
> Bogdan Lotko <boguslaw.lotko <at> chello.at> writes:
>
>> ffmpeg -f x11grab -r 30 -s 1920x1200 -vb 2000 -i :0.0
>> -threads 0 output.xxx
> Complete, uncut console output missing.
> (-threads 0 is the default)
>
>> With some exceptions (last command and mp4 output file)
>> I've got usable output files, but the quality was a
>> big problem: blurred pixels
> If you encode to pix_fmt yuv420p you get blurring, if
> you encode to rgb (or yuv444p) with x264, you get files
> that only FFmpeg (-based software) can decode.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


-- 

*Bogdan Lotko*

b.lotko at chello.at
Phone: +43 1 2852458
Mobile: +43 676 6615012

From boguslaw.lotko at chello.at  Tue Apr 29 12:06:16 2014
From: boguslaw.lotko at chello.at (Bogdan Lotko)
Date: Tue, 29 Apr 2014 12:06:16 +0200
Subject: [FFmpeg-user] Poor quality of screencast 1920x1200 on Linux
In-Reply-To: <E18F78E2-8AED-45CD-8F14-AED0C61F865A@dubstep.fm>
References: <535EC499.6060909@chello.at>
	<E18F78E2-8AED-45CD-8F14-AED0C61F865A@dubstep.fm>
Message-ID: <535F7998.9050904@chello.at>

Thank you,
Of course it was a mistake in the e-mail, in fact it was 2000k

Bogdan
On 04/29/2014 11:04 AM, DopeLabs wrote:
> arent the video bitrates specified in bits? should that not be 2000k ?
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>


-- 

*Bogdan Lotko*

b.lotko at chello.at
Phone: +43 1 2852458
Mobile: +43 676 6615012

From lou at lrcd.com  Tue Apr 29 17:50:04 2014
From: lou at lrcd.com (Lou)
Date: Tue, 29 Apr 2014 07:50:04 -0800
Subject: [FFmpeg-user] Need help with the optimal ffmpeg settings to
 publish video
In-Reply-To: <6E627E2A-9AB6-4C1D-9DFD-2EE67351AFAF@dubstep.fm>
References: <535B3C7D.2060700@xopnetworks.com>
	<6E627E2A-9AB6-4C1D-9DFD-2EE67351AFAF@dubstep.fm>
Message-ID: <1398786604.29451.111737025.11107B97@webmail.messagingengine.com>

On Mon, Apr 28, 2014, at 09:00 AM, DopeLabs wrote:
> i would start by looking at what the recommended settings are on sites
> like twitch/justin/ustream.tv/youtube.com/etc.. thats pretty much all
> these gamers are doing is capturing desktop and streaming.

Also see:
<http://trac.ffmpeg.org/wiki/EncodingForStreamingSites>

Although it does not address any issues regarding the delay mentioned
earlier.

From gliese849b at gmail.com  Tue Apr 29 19:11:51 2014
From: gliese849b at gmail.com (RDP)
Date: Tue, 29 Apr 2014 18:11:51 +0100
Subject: [FFmpeg-user] wav concat woes
In-Reply-To: <20140428094138.GB17030@sunshine.barsnick.net>
References: <ljk2c8$jju$1@ger.gmane.org>
	<CAEUFz7f3NTY1j171LLgVxruOjDmnufsHy-Ci_gY3gDu+3hGzrQ@mail.gmail.com>
	<CAKb6y5U40MCTw1xt6JYUqs-12PVGyYE5XyikaDHnHgBWgQ2=AA@mail.gmail.com>
	<CAEUFz7cHD8QEHhXbrk1nttsRyh1YAio43f0UWn2Tn9aOQqkskw@mail.gmail.com>
	<20140428094138.GB17030@sunshine.barsnick.net>
Message-ID: <CAEUFz7eb6OwfNbzLbzucc3D1s6OAf2nEheGvoPOJ8CS5CsosZg@mail.gmail.com>

On Mon, Apr 28, 2014 at 10:41 AM, Moritz Barsnick <barsnick at gmx.net> wrote:
> On Mon, Apr 28, 2014 at 07:24:14 +0100, RDP wrote:
>> >> > cat 01_list.txt
>> >> > file '01/01.01_test.wav'
>> >> > file '01/01.02_test.wav'
>> >> > file '01/01.03_test.wav'
>> >> > file '01/01.04_test.wav'
>> >> >
>> >> > ffmpeg  -f concat -i 01_list.txt -vn -c:a libfdk_aac    -profile:a aac_he
>> >> > -b:a 48k -ac 1 -signaling implicit    test2.m4a 2> ffmpeg.out
>> >> > And I can run the command on each individually.
>
>> I was specifically targetting the quoted use of the cat command here,
>> not ffmpeg.
>> cat does exactly what it says on the tin, i.e. joins files together.
>> cat does not diffentiate between types of file, other than text and
>> binary, and therefore has no knowledge of format of a wave file.
>
> Chris, where do you see the use of 'cat' to combine files in the
> original request? Sean used 'cat' only to show the content of the file
> handed to ffmpeg's concat muxer. It's all about ffmpeg here.

Quite right Moritz, I'm seeing things that aren't there!  Sean was indeed
referring to concatenation within ffmpeg.  My apologies.

What was in my mind I think, is that some file formats don't go down this
extra header route, which makes conjoining and splitting files real easy.

Regards,


Chris.

From thilo.borgmann at mail.de  Tue Apr 29 20:31:17 2014
From: thilo.borgmann at mail.de (Thilo Borgmann)
Date: Tue, 29 Apr 2014 20:31:17 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <BLU168-W9310BF54330CCF48D2EB9AA3460@phx.gbl>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>,
	<535A5187.5030107@mail.de>,
	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>,
	<20140429054225.GA2402@sunshine.barsnick.net>
	<BLU168-W9310BF54330CCF48D2EB9AA3460@phx.gbl>
Message-ID: <535FEFF5.5070009@mail.de>

Am 29.04.14 16:26, schrieb Tobie Horswill:
> Thank you Moritz for pointing this out and thank you Carl Eugen for
> explaining why I may have missed it in the first place!
> 
> So it would appear mmintrin.h, xmmintrin.h and emmintrin.h are each missing
> multiple prototypes.
> 
> Any hints as to why this is happening? Not sure if it is relevant but I first
> tried compiling using MacPorts then switched to Brew which got me a bit
> further but apparently not far enough.

It's an issue with the compiler and the Apple provided headers, so there is not
much for FFmpeg to do about.

> My ./config file seems somewhat similar to the one mentioned here :
> http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2011-April/110864.html Would
> clearing the definition of  __GNUC_STDC_INLINE__ be relevant in my case?

Have a look at TRAC, there is an avfoundation related issue with a suggestion
pointing in that direction. I never tested it and I'm not sure, if it will work.
It's annoying that Apple compilers fail there.

Sorry for my late reply.

-Thilo


From thilo.borgmann at mail.de  Tue Apr 29 20:33:11 2014
From: thilo.borgmann at mail.de (Thilo Borgmann)
Date: Tue, 29 Apr 2014 20:33:11 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <535A5BD5.90708@gmx.de>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>	<535A5187.5030107@mail.de>
	<535A5BD5.90708@gmx.de>
Message-ID: <535FF067.8030805@mail.de>

Am 25.04.14 14:57, schrieb neve_capricorn at gmx.de:
> hi thilo,
> 
> can i use av-foundation in ffmpeg to grab audio devices with the underlaying
> coreAudio?
> 
> currently i was able to capute from my sight camera videoframes, but can i also
> address an audio device for capturing audio frames under osx 10.9 with ffmpeg
> included qtkit or avfoundation?

I've a patch coming that covers basic audio capturing, yes.
Unfortunately I have not much time to spend on it right now.
I think it can be expected sometime in early Mai.

Sorry for my late reply.

-Thilo

From thilo.borgmann at mail.de  Tue Apr 29 20:35:30 2014
From: thilo.borgmann at mail.de (Thilo Borgmann)
Date: Tue, 29 Apr 2014 20:35:30 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <20140429111103.GC3679@sunshine.barsnick.net>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>	<535A5187.5030107@mail.de>	<BLU168-W93B4F7A21E0B75E5FC1BB0A3460@phx.gbl>	<20140429054225.GA2402@sunshine.barsnick.net>	<loom.20140429T092649-354@post.gmane.org>	<20140429102524.GB3679@sunshine.barsnick.net>	<loom.20140429T123515-181@post.gmane.org>
	<20140429111103.GC3679@sunshine.barsnick.net>
Message-ID: <535FF0F2.3060604@mail.de>

Am 29.04.14 13:11, schrieb Moritz Barsnick:
> On Tue, Apr 29, 2014 at 10:42:12 +0000, Carl Eugen Hoyos wrote:
>> Given that this is the behaviour for many, many 
>> years, I am not convinced that changing it will 
>> make us many friends.
> 
> No no, I was in no way asking you to change it. :)
> 
>> Please understand that FFmpeg's configure is 
>> not autoconf-based and is not supposed to 
>> work like an autoconf-base configure script.
> 
> Well understood, and the reasoning behind it. It's only feedback that I
> myself had slight trouble identifying whether the avfoundation indev
> feature had succeeded or not.

The feature itself is there but unfortunately it is unavailable for (too many)
OSX-compiler combinations. QTKit is still an alternative if you're not on mobile
devices and should work fine. I cannot promise any audio support soon, though.

-Thilo


From neve_capricorn at gmx.de  Tue Apr 29 23:21:11 2014
From: neve_capricorn at gmx.de (neve_capricorn at gmx.de)
Date: Tue, 29 Apr 2014 23:21:11 +0200
Subject: [FFmpeg-user] AVFoundation under Mac OS 10.7.5
In-Reply-To: <535FF067.8030805@mail.de>
References: <BLU168-W8D7BE34907646ADCED774A35B0@phx.gbl>	<535A5187.5030107@mail.de>	<535A5BD5.90708@gmx.de>
	<535FF067.8030805@mail.de>
Message-ID: <536017C7.5050203@gmx.de>

i fully understand..thanks for your hard work and spending your time on 
this.

tom

Am 29.04.14 20:33, schrieb Thilo Borgmann:
> Am 25.04.14 14:57, schrieb neve_capricorn at gmx.de:
>> hi thilo,
>>
>> can i use av-foundation in ffmpeg to grab audio devices with the underlaying
>> coreAudio?
>>
>> currently i was able to capute from my sight camera videoframes, but can i also
>> address an audio device for capturing audio frames under osx 10.9 with ffmpeg
>> included qtkit or avfoundation?
> I've a patch coming that covers basic audio capturing, yes.
> Unfortunately I have not much time to spend on it right now.
> I think it can be expected sometime in early Mai.
>
> Sorry for my late reply.
>
> -Thilo
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user


From cehoyos at ag.or.at  Wed Apr 30 09:57:04 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 30 Apr 2014 07:57:04 +0000 (UTC)
Subject: [FFmpeg-user] Poor quality of screencast 1920x1200 on Linux
References: <535EC499.6060909@chello.at>
	<loom.20140429T092402-407@post.gmane.org>
	<535F6179.2070108@chello.at>
Message-ID: <loom.20140430T094710-112@post.gmane.org>

Bogdan Lotko <boguslaw.lotko <at> chello.at> writes:

> ffmpeg -f x11grab -s 1920x1200 
> -framerate 30 -vb 2000k -i :0.0 output.avi

"q=31.0" in your output indicates that the bitrate is 
not sufficient / cannot be reached for some reason.
For maximum quality, remove "-vb 2000k" and use 
"-qscale 2 -mbd 2" instead.

> ffmpeg version 2.2.1 Copyright (c) 2000-2014 the FFmpeg developers

Mostly unrelated:
Please use current FFmpeg git head when asking for 
support on this mailing list.
(2.2.1 is new atm but this will change.)

> As I wrote I'd like to use it for presentation so 
> the video will be played on Windows.

Do you mean "on Windows with (vanilla) WMP" or "on 
Windows with vlc"?
For WMP either use -vcodec msmpeg4 (larger files) or 
"-vtag" - unfortunately I forgot the value, it should 
be either "MP4S" or "MP4V" (XVID should also work).

> My goal is just to produce a video with acceptable quality

Reading your console output, -qscale 2 should increase the 
quality but please understand that all "normal" video 
codecs only support yuv420p and are by definition not 
well suitable for screen capture.
There are alternatives (everything with "rle" and Flash 
screen video) that support RGB but I suspect none of them 
will play on vanilla WMP.
-vcodec libx264rgb is high-quality, high-performance but 
is only supported if the playback software is FFmpeg-based 
(vlc).

Please do not top-post here, it is considered rude.

Carl Eugen


From boguslaw.lotko at chello.at  Wed Apr 30 13:11:54 2014
From: boguslaw.lotko at chello.at (Bogdan Lotko)
Date: Wed, 30 Apr 2014 13:11:54 +0200
Subject: [FFmpeg-user] Poor quality of screencast 1920x1200 on Linux
In-Reply-To: <loom.20140430T094710-112@post.gmane.org>
References: <535EC499.6060909@chello.at>	<loom.20140429T092402-407@post.gmane.org>	<535F6179.2070108@chello.at>
	<loom.20140430T094710-112@post.gmane.org>
Message-ID: <5360DA7A.6030307@chello.at>

On 04/30/2014 09:57 AM, Carl Eugen Hoyos wrote:
> Bogdan Lotko <boguslaw.lotko <at> chello.at> writes:
>
>> ffmpeg -f x11grab -s 1920x1200
>> -framerate 30 -vb 2000k -i :0.0 output.avi
> "q=31.0" in your output indicates that the bitrate is
> not sufficient / cannot be reached for some reason.
> For maximum quality, remove "-vb 2000k" and use
> "-qscale 2 -mbd 2" instead.
>
>> ffmpeg version 2.2.1 Copyright (c) 2000-2014 the FFmpeg developers
> Mostly unrelated:
> Please use current FFmpeg git head when asking for
> support on this mailing list.
> (2.2.1 is new atm but this will change.)
>
>> As I wrote I'd like to use it for presentation so
>> the video will be played on Windows.
> Do you mean "on Windows with (vanilla) WMP" or "on
> Windows with vlc"?
> For WMP either use -vcodec msmpeg4 (larger files) or
> "-vtag" - unfortunately I forgot the value, it should
> be either "MP4S" or "MP4V" (XVID should also work).
>
>> My goal is just to produce a video with acceptable quality
> Reading your console output, -qscale 2 should increase the
> quality but please understand that all "normal" video
> codecs only support yuv420p and are by definition not
> well suitable for screen capture.
> There are alternatives (everything with "rle" and Flash
> screen video) that support RGB but I suspect none of them
> will play on vanilla WMP.
> -vcodec libx264rgb is high-quality, high-performance but
> is only supported if the playback software is FFmpeg-based
> (vlc).
>
> Please do not top-post here, it is considered rude.
>
> Carl Eugen
>
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
Thank you very much I'll test these possibilities.
Meanwhile I've got the acceptable results using the following settings:

ffmpeg -f x11grab -framerate 30 -video_size 1920x1080 -i :0.0 -vcodec 
libx264 -preset ultrafast -maxrate 2048k -bufsize 3072k -vf 
"format=yuv420p" -g 60  output.avi
(-video_size 1920x1200) does not work)

Unfortunately I have no time to for learning more about this great software.

Thank you for your help,

with best regards,

Bogdan

-- 

*Bogdan Lotko*

b.lotko at chello.at
Phone: +43 1 2852458
Mobile: +43 676 6615012

From kqt4at5v at gmail.com  Wed Apr 30 15:23:12 2014
From: kqt4at5v at gmail.com (kqt4at5v at gmail.com)
Date: Wed, 30 Apr 2014 08:23:12 -0500 (CDT)
Subject: [FFmpeg-user] video4linux2 framerate
Message-ID: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>

I know I will get flamed for referring to a specific distribution but I 
think it is irrelevant. Previously using ffmpeg package in Ubuntu 10.04 I 
used "ffmpeg -f video4linux2 -s 640x480 -r 1 -i /dev/${CAMERA} -an ${CAPTURE_DIR}/\%05d.jpg"
to get 1 frame per second. I upgraded to Ubuntu 12.04 and It is broken.

ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
   built on Feb  6 2014 20:59:08 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.
[video4linux2 @ 0x8997aa0] ioctl set time per frame(1/1) failed
[video4linux2 @ 0x8997aa0] ioctl(VIDIOC_DQBUF): Invalid argument
[video4linux2 @ 0x8997aa0] decoding for stream 0 failed
[video4linux2 @ 0x8997aa0] Estimating duration from bitrate, this may be inaccurate
/dev/shop: could not find codec parameters

I tried avconv and got the same error. I compiled the latest tarball from 
ffmpeg.org, ffmpeg-2.2.1.tar.bz2,

ffmpeg version 2.2.1 Copyright (c) 2000-2014 the FFmpeg developers  built on Apr 29 2014 13:33:50 with gcc 4.6 (Ubuntu/Linaro 
4.6.3-1ubuntu5)
   configuration: --prefix=/home/rray/ffmpeg-2.2.1
   libavutil      52. 66.100 / 52. 66.100
   libavcodec     55. 52.102 / 55. 52.102
   libavformat    55. 33.100 / 55. 33.100
   libavdevice    55. 10.100 / 55. 10.100
   libavfilter     4.  2.100 /  4.  2.100
   libswscale      2.  5.102 /  2.  5.102
   libswresample   0. 18.100 /  0. 18.100
[video4linux2,v4l2 @ 0xb073dc0] The driver does not allow to change time per frame
Input #0, video4linux2,v4l2, from '/dev/shop':
   Duration: N/A, start: 1398864022.798506, bitrate: 92160 kb/s
     Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480, 
92160 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
[swscaler @ 0xb062040] deprecated pixel format used, make sure you did set range correctly
Output #0, image2, to '/backup_1/camera/20140430/shop_082021/%05d.jpg':
   Metadata:
     encoder         : Lavf55.33.100
     Stream #0:0: Video: mjpeg, yuvj420p, 640x480, q=2-31, 200 kb/s, 90k tbn, 1 tbc
Stream mapping:
   Stream #0:0 -> #0:0 (rawvideo -> mjpeg)
Press [q] to stop, [?] for help
frame=   22 fps= 15 q=24.8 Lsize=N/A time=00:00:22.00 bitrate=N/A
video:1407kB audio:0kB subtitle:0 data:0 global headers:0kB muxing overhead -100.001526%


It records but runs at 15 fps, I guess this is why "The driver does not 
allow to change time per frame". So my question is how can I record 1 fps?

Thanks
Richard


From t.rapp at noa-audio.com  Wed Apr 30 15:35:48 2014
From: t.rapp at noa-audio.com (Tobias Rapp)
Date: Wed, 30 Apr 2014 15:35:48 +0200
Subject: [FFmpeg-user] video4linux2 framerate
In-Reply-To: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
References: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
Message-ID: <20140430153548.28d38bdb29c96dd7e14bd89a@noa-audio.com>

kqt4at5v at gmail.com wrote:
> It records but runs at 15 fps, I guess this is why "The driver does not 
> allow to change time per frame". So my question is how can I record 1 fps?

Have you tried specifying the frame rate as an output option (after
"-i /dev/${CAMERA}")?

Regards,
Tobias


From cehoyos at ag.or.at  Wed Apr 30 16:17:23 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 30 Apr 2014 14:17:23 +0000 (UTC)
Subject: [FFmpeg-user] video4linux2 framerate
References: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
Message-ID: <loom.20140430T161416-5@post.gmane.org>

 <kqt4at5v <at> gmail.com> writes:

> ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright 
> (c) 2000-2013 the Libav developers

This is not FFmpeg but an intentionally broken version with 
several hundred known, user-reported bugs that are not 
reproducible with FFmpeg, some of them security-relevant. 
Please understand that we cannot support it here.
Read the following to get more information:
http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html

> It records but runs at 15 fps, I guess this is why "The driver does not 
> allow to change time per frame". So my question is how can I record 1 fps?

As explained, move the "-r 1" behind the input option ("-i") to 
specify an output frame rate.

Carl Eugen


From kqt4at5v at gmail.com  Wed Apr 30 17:07:17 2014
From: kqt4at5v at gmail.com (kqt4at5v at gmail.com)
Date: Wed, 30 Apr 2014 10:07:17 -0500 (CDT)
Subject: [FFmpeg-user] video4linux2 framerate
In-Reply-To: <20140430153548.28d38bdb29c96dd7e14bd89a@noa-audio.com>
References: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
	<20140430153548.28d38bdb29c96dd7e14bd89a@noa-audio.com>
Message-ID: <alpine.DEB.2.02.1404301002160.21234@tehzcl2.tehzcl-arg>

On Wed, 30 Apr 2014, Tobias Rapp wrote:

> kqt4at5v at gmail.com wrote:
>> It records but runs at 15 fps, I guess this is why "The driver does not
>> allow to change time per frame". So my question is how can I record 1 fps?
>
> Have you tried specifying the frame rate as an output option (after
> "-i /dev/${CAMERA}")?
>

That works. Is this a fixed bug or just a change to the driver?
I see in top that ffmpeg is using more resources. Is that because it is 
receiving 15 fps and writing 1? Is there a better, less resources, way to 
capture 1 fps?

Thanks
Richard

From olivier.pis.langlois at transport.alstom.com  Wed Apr 30 17:20:35 2014
From: olivier.pis.langlois at transport.alstom.com (LANGLOIS Olivier PIS -EXT)
Date: Wed, 30 Apr 2014 15:20:35 +0000
Subject: [FFmpeg-user] video4linux2 framerate
In-Reply-To: <alpine.DEB.2.02.1404301002160.21234@tehzcl2.tehzcl-arg>
References: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
	<20140430153548.28d38bdb29c96dd7e14bd89a@noa-audio.com>
	<alpine.DEB.2.02.1404301002160.21234@tehzcl2.tehzcl-arg>
Message-ID: <89F8E4011FD7234ABFE48744E300CBC206C2F7DC@041-DB3MPN1-091.041d.mgd.msft.net>

> >
> > Have you tried specifying the frame rate as an output option (after
> > "-i /dev/${CAMERA}")?
> >
>
> That works. Is this a fixed bug or just a change to the driver?
> I see in top that ffmpeg is using more resources. Is that because it is receiving
> 15 fps and writing 1? Is there a better, less resources, way to capture 1 fps?
>

Supported fps by webcams are just a few limited discreet values (ie: 15, 30, 60). To explore your HW capabilities, I suggest that you use the Qt app qv4l2 bundled in the v4l-utils package from http://linuxtv.org/


Please ignore the confidentiality notice below.
It is automatically added without my consent.


________________________________
CONFIDENTIALITY : This e-mail and any attachments are confidential and may be privileged. If you are not a named recipient, please notify the sender immediately and do not disclose the contents to another person, use it for any purpose or store or copy the information in any medium.

From u at pkh.me  Wed Apr 30 17:36:06 2014
From: u at pkh.me (=?utf-8?B?Q2zDqW1lbnQgQsWTc2No?=)
Date: Wed, 30 Apr 2014 17:36:06 +0200
Subject: [FFmpeg-user] video4linux2 framerate
In-Reply-To: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
References: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
Message-ID: <20140430153605.GG1191@leki>

On Wed, Apr 30, 2014 at 08:23:12AM -0500, kqt4at5v at gmail.com wrote:
> I know I will get flamed for referring to a specific distribution but I
> think it is irrelevant. Previously using ffmpeg package in Ubuntu 10.04 I
> used "ffmpeg -f video4linux2 -s 640x480 -r 1 -i /dev/${CAMERA} -an
> ${CAPTURE_DIR}/\%05d.jpg"
> to get 1 frame per second. I upgraded to Ubuntu 12.04 and It is broken.
> 
> ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
>   built on Feb  6 2014 20:59:08 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.
> [video4linux2 @ 0x8997aa0] ioctl set time per frame(1/1) failed
> [video4linux2 @ 0x8997aa0] ioctl(VIDIOC_DQBUF): Invalid argument
> [video4linux2 @ 0x8997aa0] decoding for stream 0 failed
> [video4linux2 @ 0x8997aa0] Estimating duration from bitrate, this may be inaccurate
> /dev/shop: could not find codec parameters
> 
> I tried avconv and got the same error. I compiled the latest tarball from
> ffmpeg.org, ffmpeg-2.2.1.tar.bz2,
> 
> ffmpeg version 2.2.1 Copyright (c) 2000-2014 the FFmpeg developers  built on
> Apr 29 2014 13:33:50 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
>   configuration: --prefix=/home/rray/ffmpeg-2.2.1
>   libavutil      52. 66.100 / 52. 66.100
>   libavcodec     55. 52.102 / 55. 52.102
>   libavformat    55. 33.100 / 55. 33.100
>   libavdevice    55. 10.100 / 55. 10.100
>   libavfilter     4.  2.100 /  4.  2.100
>   libswscale      2.  5.102 /  2.  5.102
>   libswresample   0. 18.100 /  0. 18.100
> [video4linux2,v4l2 @ 0xb073dc0] The driver does not allow to change time per frame
> Input #0, video4linux2,v4l2, from '/dev/shop':
>   Duration: N/A, start: 1398864022.798506, bitrate: 92160 kb/s
>     Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 640x480,
> 92160 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
> [swscaler @ 0xb062040] deprecated pixel format used, make sure you did set range correctly
> Output #0, image2, to '/backup_1/camera/20140430/shop_082021/%05d.jpg':
>   Metadata:
>     encoder         : Lavf55.33.100
>     Stream #0:0: Video: mjpeg, yuvj420p, 640x480, q=2-31, 200 kb/s, 90k tbn, 1 tbc
> Stream mapping:
>   Stream #0:0 -> #0:0 (rawvideo -> mjpeg)
> Press [q] to stop, [?] for help
> frame=   22 fps= 15 q=24.8 Lsize=N/A time=00:00:22.00 bitrate=N/A
> video:1407kB audio:0kB subtitle:0 data:0 global headers:0kB muxing overhead -100.001526%
> 
> 
> It records but runs at 15 fps, I guess this is why "The driver does not
> allow to change time per frame". So my question is how can I record 1 fps?
> 

The v4l2 device input option you are probably looking for is -framerate.

Try ffmpeg -f v4l2 -framerate 1 -i ...

Also, -s is probably not doing what you expect it to do: use the
-video_size v4l2 input option.

-- 
Cl?ment B.
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From cehoyos at ag.or.at  Wed Apr 30 18:14:02 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 30 Apr 2014 16:14:02 +0000 (UTC)
Subject: [FFmpeg-user] video4linux2 framerate
References: <alpine.DEB.2.02.1404300805030.21234@tehzcl2.tehzcl-arg>
	<20140430153605.GG1191@leki>
Message-ID: <loom.20140430T181120-130@post.gmane.org>

Cl?ment B?sch <u <at> pkh.me> writes:

> The v4l2 device input option you are probably 
> looking for is -framerate.

Input option -r maps to -framerate.

> Try ffmpeg -f v4l2 -framerate 1 -i ...
> 
> Also, -s is probably not doing what you expect 
> it to do: use the -video_size v4l2 input option.

Input option -s maps to -video_size.

I still believe that it is very useful if the 
same input and output options have the same 
name.

Carl Eugen


From pstevens at evertz.com  Wed Apr 30 19:26:58 2014
From: pstevens at evertz.com (Peter Stevens)
Date: Wed, 30 Apr 2014 13:26:58 -0400
Subject: [FFmpeg-user] combinding seek with the select filter
Message-ID: <C70A636B101FD44999B82525C3E92AFA03A13F8A@otis.burlington.evertz.tv>

I'm looking to extract a particular frame from an input file as fast as
possible. I can get this to work with "-ss" by providing the appropriate
timestamp, but I would much rather provide the specific frame number I
am interested in. So far I've attempted to combined the fast seek option
with the "select" filter but this fails. I believe this is because the
frame numbering starts at zero after the "seek" has completed. Take for
instance, the below mp4 example where frame 143857 has a pts_timestamp
of 6000.041708

 

ffmpeg -ss  6000 -i pphinferb_timecode.mp4 -vf 'select=eq(n\,143834)'
-vframes 1 -y test.jpeg

 

ffmpeg accepts the command as being valid, but throws the following
error after a bit .. "Output file is empty, nothing was encoded (check
-ss / -t / -frames parameters if used)"

 

Any ideas?

Peter

 


From battistel at gmail.com  Wed Apr 30 19:50:52 2014
From: battistel at gmail.com (Massimo Battistel)
Date: Wed, 30 Apr 2014 19:50:52 +0200
Subject: [FFmpeg-user] Possible issue with asf muxer
In-Reply-To: <loom.20140423T172750-634@post.gmane.org>
References: <CAHss8oGqhCV52HWQ-8CU4JfNbCCbHkgFvsbXXZgCm0-RquHcEA@mail.gmail.com>
	<loom.20140423T172750-634@post.gmane.org>
Message-ID: <CAHss8oHizA6h7X1pEt-h-K0X7mt=-q94JJFQvADp=QY5u=CgfQ@mail.gmail.com>

>
> This should be fixed in current FFmpeg git head,
> thank you for the report!
>
>
thanks a lot!
Now It works much better.

MB

From rogerdpack2 at gmail.com  Wed Apr 30 23:06:17 2014
From: rogerdpack2 at gmail.com (Roger Pack)
Date: Wed, 30 Apr 2014 15:06:17 -0600
Subject: [FFmpeg-user] dedupe
Message-ID: <CAL1QdWfZBtBXh0NdYJYcPqB7LYOjQ8MceXLdiPpkO80XfCSBaQ@mail.gmail.com>

Is there any way to dedupe incoming video frames?
Just wondering.
Thanks!
-roger-

From cehoyos at ag.or.at  Wed Apr 30 23:29:36 2014
From: cehoyos at ag.or.at (Carl Eugen Hoyos)
Date: Wed, 30 Apr 2014 21:29:36 +0000 (UTC)
Subject: [FFmpeg-user] dedupe
References: <CAL1QdWfZBtBXh0NdYJYcPqB7LYOjQ8MceXLdiPpkO80XfCSBaQ@mail.gmail.com>
Message-ID: <loom.20140430T232859-88@post.gmane.org>

Roger Pack <rogerdpack2 <at> gmail.com> writes:

> Is there any way to dedupe incoming video frames?

(I don't understand but:)
Are you searching for mpdecimate?

Carl Eugen