To splice audio only in FFmpeg ?

[shantiq]

a while back (http://ubuntuforums.org/showthread.php?t=1782270&highlight=splice+audio+shantiq)i worked out how to splice most of the audio formats; and it worked fine on aac with cat [as does for mp3]

but trying this recently on files made with new codecs [libaacplus fdk_aac] it says it does not find the headers [i think]


so i really want to nail this with ffmpeg once and for all

i tried a few things but got nowhere fast

so what would the syntax be? just audio and in this case aac/m4a predominently thanx in advance

[evilsoup]

You might be able to use the concat protocol - I've had no problems using this with just audio files.



ffmpeg -i "concat:input1.m4a|input2.m4a|input3.m4a" -c copy output.m4a


If that doesn't work, you could try using the concat demuxer (currently requires you to compile yourself, from git; the jon-severinsson PPA isn't quite up to date).

First, create a plaintext file containing the following:



file 'input1.m4a'
file 'input2.m4a'
file 'input3.m4a'


Call it 'inputs.txt', and then use this command:



ffmpeg -f concat -i inputs.txt -c copy output.m4a

[shantiq]

thanx esoup for reply


had already found and tried both of those and sadly they both yield same result on m4a



Output #0, ipod, to 'output.m4a':
Metadata:
date : 2010
major_brand : mp42
minor_version : 0
............................
Stream #0:0: Video: mjpeg, yuvj420p, 500x500 [SAR 96:96 DAR 1:1], q=2-31, 90k tbn, 90k tbc
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, 37 channels (FL+FR), 63 kb/s
Metadata:
creation_time : 2012-10-06 09:14:24
handler_name : Sound Media Handler
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Stream #0:2 -> #0:1 (copy)
Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted


======
tested on flac it still does not work for me and does what cat does; which is to tell you it has worked and then you find the output file only contains the first audio; but for flac i have sox which does work for splicing


puzzling business........ there has to be a way...

[evilsoup]

I just tested on some files I have... the concat protocol doesn't work on M4As, for me, but the demuxer works perfectly well. Neither works for -c:a copy for flac inputs, but if I use -c:a flac with the demuxer I can losslessly join them up.

Please put up the actual command you are using, and the full terminal output. I suspect that your problem is that you're using mjpeg video. I have no idea if that is supported in the MP4 container format (the error there may indicate 'no'), and besides you wrote that you wanted to concatenate audio only, so try it with the audio on its own.

[shantiq]

hi again


from the first command complete terminal output is



ffmpeg -i "concat:1.m4a|2.m4a" -c copy output.m4a
ffmpeg version git-2012-12-22-e749b5d Copyright (c) 2000-2012 the FFmpeg developers
built on Dec 22 2012 20:45:07 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 --enable-libaacplus
libavutil 52. 12.100 / 52. 12.100
libavcodec 54. 81.100 / 54. 81.100
libavformat 54. 49.102 / 54. 49.102
libavdevice 54. 3.102 / 54. 3.102
libavfilter 3. 29.101 / 3. 29.101
libswscale 2. 1.103 / 2. 1.103
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 2.100 / 52. 2.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x31f7760] stream 0, timescale not set
Last message repeated 1 times
[aac @ 0x32096a0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x32096a0] channel element 2.2 is not allocated
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x31f7760] max_analyze_duration 5000000 reached at 5015510
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'concat:1.m4a|2.m4a':
Metadata:


[metadata info]
:
:
Duration: 00:03:11.33, start: 0.000000, bitrate: 147 kb/s
Chapter #0.0: start 0.105941, end 191.332422
Metadata:
title : Heartbeat
Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 63 kb/s
Metadata:
creation_time : 2012-10-06 09:13:36
handler_name : Sound Media Handler
Stream #0:1: Video: mjpeg, yuvj420p, 500x500 [SAR 96:96 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
Stream #0:2(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, 37 channels (FL+FR), fltp, 63 kb/s
Metadata:
creation_time : 2012-10-06 09:14:24
handler_name : Sound Media Handler
Stream #0:3: Video: mjpeg, yuvj420p, 500x500 [SAR 96:96 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
[ipod @ 0x3284800] track 0: could not find tag, codec not currently supported in container
Output #0, ipod, to 'output.m4a':
Metadata:

[metadata info]


Stream #0:0: Video: mjpeg, yuvj420p, 500x500 [SAR 96:96 DAR 1:1], q=2-31, 90k tbn, 90k tbc
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, 37 channels (FL+FR), 63 kb/s
Metadata:
creation_time : 2012-10-06 09:14:24
handler_name : Sound Media Handler
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Stream #0:2 -> #0:1 (copy)
Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted



and second



ffmpeg -f concat -i inputs.txt -c copy output.m4a
ffmpeg version git-2012-12-22-e749b5d Copyright (c) 2000-2012 the FFmpeg developers
built on Dec 22 2012 20:45:07 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 --enable-libaacplus
libavutil 52. 12.100 / 52. 12.100
libavcodec 54. 81.100 / 54. 81.100
libavformat 54. 49.102 / 54. 49.102
libavdevice 54. 3.102 / 54. 3.102
libavfilter 3. 29.101 / 3. 29.101
libswscale 2. 1.103 / 2. 1.103
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 2.100 / 52. 2.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x32462c0] stream 0, timescale not set
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x32462c0] max_analyze_duration 5000000 reached at 5015510
[concat @ 0x323d7a0] max_analyze_duration 5000000 reached at 5015510
[concat @ 0x323d7a0] Estimating duration from bitrate, this may be inaccurate
Input #0, concat, from 'inputs.txt':
Duration: 00:00:00.00, start: 0.000000, bitrate: 63 kb/s
Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 63 kb/s
Stream #0:1: Video: mjpeg, yuvj420p, 500x500 [SAR 96:96 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
[ipod @ 0x328f740] track 0: could not find tag, codec not currently supported in container
Output #0, ipod, to 'output.m4a':
Metadata:
encoder : Lavf54.49.102
Stream #0:0: Video: mjpeg, yuvj420p, 500x500 [SAR 96:96 DAR 1:1], q=2-31, 90k tbn, 90k tbc
Stream #0:1: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, 63 kb/s
Stream mapping:
Stream #0:1 -> #0:0 (copy)
Stream #0:0 -> #0:1 (copy)
Could not write header for output file #0 (incorrect codec parameters ?): Operation not permitted




in both cases a video stream is discussed altho there isnt one.... probably where the kink appears when trying to splice... if extension changed for aac instead same result tho

if i work from libfaac aac no problem then with either cat or ffmpeg but not what i seek :::]]