[FFmpeg-user] FFmpeg accuracy issues.

[Shlomi Shalem]

Hello everybody,

Recently, I've been trying to use FFmpeg for an application which requires
a VERY accurate manipulation when it comes to the time parameter
(milliseconds resolution). Unfortunately, I was surprised to find out that
FFmpeg's manipulation functionalities return some inaccurate results.

Here is the output of '*ffmpeg*':

*ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers*
*  built on Jul 25 2012 19:55:05 with gcc 4.2.1 (Apple Inc. build 5664)*
*  configuration: --enable-gpl --enable-shared --enable-pthreads
--enable-libx264 --enable-libmp3lame*
*  libavutil      51. 54.100 / 51. 54.100*
*  libavcodec     54. 23.100 / 54. 23.100*
*  libavformat    54.  6.100 / 54.  6.100*
*  libavdevice    54.  0.100 / 54.  0.100*
*  libavfilter     2. 77.100 /  2. 77.100*
*  libswscale      2.  1.100 /  2.  1.100*
*  libswresample   0. 15.100 /  0. 15.100*
*  libpostproc    52.  0.100 / 52.  0.100*

Now, let's assume I want to rip the audio track of 'foo.mov'. Here is the
relevant output of '*ffmpeg -i foo.mov*':

*Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'foo.mov':*
*  Metadata:*
*    major_brand     : qt  *
*    minor_version   : 0*
*    compatible_brands: qt  *
*    creation_time   : 2012-07-24 23:16:08*
*  Duration: 00:00:40.38, start: 0.000000, bitrate: 805 kb/s*
*    Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p,
480x360, 733 kb/s, 24.46 fps, 29.97 tbr, 600 tbn, 1200 tbc*
*    Metadata:*
*      rotate          : 90*
*      creation_time   : 2012-07-24 23:16:08*
*      handler_name    : Core Media Data Handler*
*    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, s16,
63 kb/s*
*    Metadata:*
*      creation_time   : 2012-07-24 23:16:08*
*      handler_name    : Core Media Data Handler*
*
*
As you probably noticed, the video file duration is 00:00:40.38. Using the
following command, I ripped it's audio track:

'*ffmpeg -i foo.mov foo.wav*'

Output:

*Output #0, wav, to 'foo.wav':*
*  Metadata:*
*    major_brand     : qt  *
*    minor_version   : 0*
*    compatible_brands: qt  *
*    creation_time   : 2012-07-24 23:16:08*
*    encoder         : Lavf54.6.100*
*    Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz,
mono, s16, 705 kb/s*
*    Metadata:*
*      creation_time   : 2012-07-24 23:16:08*
*      handler_name    : Core Media Data Handler*
*Stream mapping:*
*  Stream #0:1 -> #0:0 (aac -> pcm_s16le)*
*Press [q] to stop, [?] for help*
*size=    3482kB time=00:00:40.42 bitrate= 705.6kbits/s    *
*video:0kB audio:3482kB global headers:0kB muxing overhead 0.001290%*
*
*
As you can see, the output file is longer than the file in the input.

Another example is audio (and video) file trimming:
Let's assume I would like to use ffmpeg for audio file trimming. I used the
next command:

'*ffmpeg -t 00:00:10.000 -i foo.wav trimmed_foo.wav -ss 00:00:25.000*'

Output:

*[wav @ 0x10180e800] max_analyze_duration 5000000 reached at 5015510*
*Guessed Channel Layout for  Input Stream #0.0 : mono*
*Input #0, wav, from 'foo.wav':*
*  Duration: 00:00:40.42, bitrate: 705 kb/s*
*    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono,
s16, 705 kb/s*
*Output #0, wav, to 'trimmed_foo.wav':*
*  Metadata:*
*    encoder         : Lavf54.6.100*
*    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono,
s16, 705 kb/s*
*Stream mapping:*
*  Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)*
*Press [q] to stop, [?] for help*
*size=     864kB time=00:00:10.03 bitrate= 705.6kbits/s    *
*video:0kB audio:864kB global headers:0kB muxing overhead 0.005199%*
*
*
Again, the output file is 30 milliseconds longer than I expected.
I tried, for a long time, to research the issue without any success.
I you guys have any idea about how to solve this problem, I
will appreciate a lot!

Thanks,
Shlomi.