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I want to extract the audio stream from many .MOV files with ffmpeg.
The goal is to have output files with small file size and perfectly equal sound to the input file. Currently I'm using this ffmpeg command:
ffmpeg -i input.mov -vn -ab 92k -acodec aac -strict -2 output.m4a
My problem is that re-encoding pcm_s24le to AAC creates an output file where some milliseconds at the beginning are missing.
Is there a workaround or a way resolve this? Like padding the output for 50ms
I/O information:
ffmpeg version 2.5 Copyright (c) 2000-2014 the FFmpeg developers
built on Dec 20 2014 09:33:55 with Apple LLVM version 6.0 (clang-600.0.56) (based on LLVM 3.5svn)
configuration: --prefix=/usr/local/Cellar/ffmpeg/2.5 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-libvo-aacenc --enable-libfdk-aac --enable-nonfree --enable-vda
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 13.100 / 56. 13.100
libavformat 56. 15.102 / 56. 15.102
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
libavresample 2. 1. 0 / 2. 1. 0
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '6.2.mov':
Metadata:
major_brand : qt
minor_version : 0
compatible_brands: qt
creation_time : 2014-12-03 10:12:08
Duration: 00:00:01.58, start: 0.000000, bitrate: 42754 kb/s
Stream #0:0(und): Video: prores (apcn / 0x6E637061), yuv422p10le(bt709), 960x540, 40383 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 24k tbn, 24k tbc (default)
Metadata:
creation_time : 2014-12-03 10:12:08
handler_name : Core Media Data Handler
encoder : Apple ProRes 422
timecode : 00:00:00:00
Stream #0:1(und): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, stereo, s32 (24 bit), 2304 kb/s (default)
Metadata:
creation_time : 2014-12-03 10:12:08
handler_name : Core Media Data Handler
Stream #0:2(und): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
Metadata:
creation_time : 2014-12-03 10:12:08
handler_name : Core Media Data Handler
timecode : 00:00:00:00
File 'try.m4a' already exists. Overwrite ? [y/N] y
Output #0, ipod, to 'try.m4a':
Metadata:
major_brand : qt
minor_version : 0
compatible_brands: qt
encoder : Lavf56.15.102
Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp (24 bit), 92 kb/s (default)
Metadata:
creation_time : 2014-12-03 10:12:08
handler_name : Core Media Data Handler
encoder : Lavc56.13.100 aac
Stream mapping:
Stream #0:1 -> #0:0 (pcm_s24le (native) -> aac (native))
Press [q] to stop, [?] for help
size= 19kB time=00:00:01.60 bitrate= 97.9kbits/s
video:0kB audio:18kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 5.684210%
Marco
Posted 2014-12-21T09:15:05.303
Reputation: 121
Some have recently complained about timing/synch issues with the aac encoder in ffmpeg. You might want to try a different encoder such as the libfdk_aac. This requires ffmpeg to be built with --enable-libfdk_aac and also --enable-nonfree. – Rajib – 2014-12-21T14:11:30.043
Unfortunately libfdk_aac is also causing timing errors. Possibly an ffmpeg issue? – Rajib – 2014-12-21T15:20:11.440
could depend to AAC codec? https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
– Marco – 2014-12-21T20:54:33.807No- its happening with libmp3lame as well. – Rajib – 2014-12-22T01:45:15.643
with libmp3lame the output have 47ms of silent at the beginning, the rest of the file it's correct – Marco – 2014-12-22T07:48:55.873
Yes 44ms with libfdk_aac. :( Maybe we should report a bug. – Rajib – 2014-12-22T07:49:54.793
1http://lame.sourceforge.net/tech-FAQ.txt – Marco – 2014-12-22T08:03:35.510
Maybe it's normal that encoders add some sample frame at the beginning of each encoded audio file – Marco – 2014-12-22T08:05:16.387