After FFmpeg concat, second audio is not playing

eskimoroll · 08-12-2014, 01:04 PM

Hello all I am a new ffmpeg user.

After a concat of two mp4 files (each file has both a video and audio stream), the second audio segment is dropped.

Quote:

$ ffprobe temp5.ts
ffprobe version 2.3.git-1ace957 Copyright (c) 2007-2014 the FFmpeg developers
built on Jul 26 2014 20:25:06 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-x11grab --enable-libpulse --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr
libavutil 52. 92.101 / 52. 92.101
libavcodec 55. 69.100 / 55. 69.100
libavformat 55. 49.100 / 55. 49.100
libavdevice 55. 13.102 / 55. 13.102
libavfilter 4. 11.102 / 4. 11.102
libavresample 1. 3. 0 / 1. 3. 0
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
Input #0, mpegts, from 'temp5.ts':
Duration: 00:00:45.70, start: 1.466667, bitrate: 37885 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 30 fps, 30 tbr, 90k tbn, 60 tbc
Stream #0:1[0x101]: Audio: mp3 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 128 kb/s

Quote:

ffprobe ../static/ubi-credits.ts
ffprobe version 2.3.git-1ace957 Copyright (c) 2007-2014 the FFmpeg developers
built on Jul 26 2014 20:25:06 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-x11grab --enable-libpulse --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr
libavutil 52. 92.101 / 52. 92.101
libavcodec 55. 69.100 / 55. 69.100
libavformat 55. 49.100 / 55. 49.100
libavdevice 55. 13.102 / 55. 13.102
libavfilter 4. 11.102 / 4. 11.102
libavresample 1. 3. 0 / 1. 3. 0
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
Input #0, mpegts, from '../static/ubi-credits.ts':
Duration: 00:00:03.02, start: 1.443644, bitrate: 2460 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 30 fps, 30 tbr, 90k tbn, 60 tbc
Stream #0:1[0x101]: Audio: mp3 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 128 kb/s

Quote:

ffmpeg -y -f concat -i ts-file-list.txt -c copy output.mp4 </dev/null

Quote:

file '/home/mark/Videos/czar/tmp/temp5.ts'
file '/home/mark/Videos/czar/static/ubi-credits.ts'

Can anyone tell me what I am doing wrong, please?

szboardstretcher · 08-12-2014, 01:07 PM

Why do you have that dev/null there?

eskimoroll · 08-12-2014, 02:04 PM

The dev/null was a workaround I found.

This code is a snippet of a batch program that loops through ++files. I do not need the dev/null in a single execution, but without it I was getting errors executing it inside a loop.

Would that have an effect on the 'concat' call?

szboardstretcher · 08-12-2014, 02:06 PM

If you run your command without the dev/null bit, from the command line, does it still remove the audio from the second part?

The command looks fine. I want to know whether it is actually your command/program somehow, or your script. Lets eliminate the script first.

eskimoroll · 08-12-2014, 02:18 PM

I removed the dev/null and re-executed with the same result.

I just discovered: The second audio segment plays fine in VLC but still fails with Totem...which leads me to believe that it is not standard somehow.

Here is the info from the generated file:

Quote:

$ ffprobe output.mp4
ffprobe version 2.3.git-1ace957 Copyright (c) 2007-2014 the FFmpeg developers
built on Jul 26 2014 20:25:06 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-x11grab --enable-libpulse --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr
libavutil 52. 92.101 / 52. 92.101
libavcodec 55. 69.100 / 55. 69.100
libavformat 55. 49.100 / 55. 49.100
libavdevice 55. 13.102 / 55. 13.102
libavfilter 4. 11.102 / 4. 11.102
libavresample 1. 3. 0 / 1. 3. 0
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf55.49.100
Duration: 00:00:48.72, start: 0.000000, bitrate: 33082 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 32959 kb/s, 30.01 fps, 30 tbr, 90k tbn, 60 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: mp3 (mp4a / 0x6134706D), 48000 Hz, stereo, s16p, 127 kb/s (default)
Metadata:
handler_name : SoundHandler

Edit: I tried playing output.mp4 in Kaffeine and the sound drops in the middle of the first audio segment for the rest of the clip...no more sound at all.