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• ffmpeg : set the language of an audio stream

  22 juillet 2023, par Steven Van Ingelgem

  I have 3 files : video.mp4, audionl.mp4 & audioeng.mp4.
I tried adding them all together in 1 output.mp4 file like this :

  



  ffmpeg -y -i video.mp4 -i audionl.mp4 -i audioeng.mp4  -map 0:v -map 1:a -map 2:a -metadata:s:a:0 language=nl -metadata:s:a:0 title="NL" -metadata:s:a:1 language=eng  -metadata:s:a:1 title="ENG" -acodec copy -bsf:a aac_adtstoasc -vcodec copy  -movflags faststart output.mp4 2>&1


  



  This is the output I see from ffmpeg :

  



  ffmpeg version 2.6.9 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 4.9.2 (Debian 4.9.2-10)
  configuration: --prefix=/usr --extra-cflags='-g -O2 -fstack-protector-strong -Wformat -Werror=format-security ' --extra-ldflags='-Wl,-z,relro' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-nonfree --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-libvpx --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-librtmp --enable-avfilter --enable-libfreetype --enable-libvo-aacenc --disable-decoder=amrnb --enable-libvo-amrwbenc --enable-libaacplus --libdir=/usr/lib/x86_64-linux-gnu --disable-vda --enable-libbluray --enable-libcdio --enable-gnutls --enable-frei0r --enable-openssl --enable-libass --enable-libopus --enable-fontconfig --enable-libpulse --disable-mips32r2 --disable-mipsdspr1 --disable-mipsdspr2 --enable-libvidstab --enable-libzvbi --enable-avresample --disable-htmlpages --disable-podpages --enable-libutvideo --enable-libfdk-aac --enable-libx265 --enable-libiec61883 --enable-vaapi --enable-libdc1394 --disable-altivec --shlibdir=/usr/lib/x86_64-linux-gnu
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  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 'video.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 0
    compatible_brands: isomavc1mp42dash
    creation_time   : 2017-06-04 06:35:06
  Duration: 00:20:52.80, start: 0.000000, bitrate: 1172 kb/s
    Stream #0:0(eng): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 1168 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc (default)
    Metadata:
      creation_time   : 2017-06-04 06:35:06
      handler_name    : VideoHandler
      encoder         : VideoHandler
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'audionl.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 0
    compatible_brands: isomM4A mp42dash
    creation_time   : 2017-06-04 06:35:06
  Duration: 00:20:52.80, start: 0.000000, bitrate: 96 kb/s
    Stream #1:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 93 kb/s (default)
    Metadata:
      creation_time   : 2017-06-04 06:35:06
      handler_name    : SoundHandler
Input #2, mov,mp4,m4a,3gp,3g2,mj2, from 'audioeng.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 0
    compatible_brands: isomM4A mp42dash
    creation_time   : 2017-06-04 06:35:06
  Duration: 00:20:52.78, start: 0.000000, bitrate: 64 kb/s
    Stream #2:0(eng): Audio: aac (HE-AAC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 62 kb/s (default)
    Metadata:
      creation_time   : 2017-06-04 06:35:06
      handler_name    : SoundHandler
Output #0, mp4, to 'output.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 0
    compatible_brands: isomavc1mp42dash
    encoder         : Lavf56.25.101
    Stream #0:0(eng): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 640x360 [SAR 1:1 DAR 16:9], q=2-31, 1168 kb/s, 25 fps, 25 tbr, 90k tbn, 90k tbc (default)
    Metadata:
      creation_time   : 2017-06-04 06:35:06
      handler_name    : VideoHandler
      encoder         : VideoHandler
    Stream #0:1(nl): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, 93 kb/s (default)
    Metadata:
      creation_time   : 2017-06-04 06:35:06
      handler_name    : SoundHandler
      title           : NL
    Stream #0:2(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, 62 kb/s (default)
    Metadata:
      creation_time   : 2017-06-04 06:35:06
      handler_name    : SoundHandler
      title           : ENG
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (copy)
  Stream #2:0 -> #0:2 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0xbb1320] Starting second pass: moving the moov atom to the beginning of the file
frame=31320 fps=20993 q=-1.0 Lsize=  204217kB time=00:20:52.79 bitrate=1335.4kbits/s
video:178734kB audio:23867kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.797284%


  



  As you can see in there : "Stream #0:0(eng)", "Stream #0:1(nl)" & "Stream #0:2(eng)".
So that looks fine.

  



  But when I open it in VLC, I see under Audio tracks :
NL - [English]
ENG - [English]

  



  So I went back to check on ffmpeg with "ffmpeg -i output.mp4" :

  



  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf56.25.101
  Duration: 00:20:52.80, start: 0.000000, bitrate: 1335 kb/s
    Stream #0:0(eng): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 1168 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 93 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
    Stream #0:2(eng): Audio: aac (HE-AAC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 62 kb/s (default)
    Metadata:
      handler_name    : SoundHandler


  



  And here they are English again...

  



  Afterwards I tried this :

  



  ffmpeg -y -i audionl.mp4 -c:a copy -metadata:s:a:0 language=nl output.mp4


  



  But also that failed to set the correct language on the output file...
I'm at a loss here and hope someone can help me set these languages correctly.

  



  Thanks a lot !

  


• Where i made mistake - FFMPEG (LINUX) PROBLEM - basic

  22 mars 2019, par Fejor

  I just start learing ffmpeg, i have code (like bellow), but it’s doing nothing. Where i make mistake ? What wrong with that ?
  Sory for my language mistakes, i am not realy fluent in english

  fmpeg -i videoplayback.mp4 -filter_complex "[1:v]trim=start=0:end=1,setpts=PTS-STARTPTS,scale=480x360,setsar=sar=16/9[intro1];
  
  [1:v]trim=start=1:end=123.39,setpts=PTS-STARTPTS,scale=480x360,setsar=sar=16/9[main1];
  
  [1:v]trim=start=123.39:end=124.39,setpts=PTS-STARTPTS,scale=480x360,setsar=sar=16/9[end1];
  
  [intro1]format=pix_fmts=yuva420p, fade=t=in:st=0:d=1:alpha=1[intro1];
  
  [end1]format=pix_fmts=yuva420p, fade=t=in:st=0:d=1:alpha=1[end1];
  
  [intro1][main1][end1][output];
  
  [a:1][audio]; -vcodec libx264 -map "[output]" -map"[audio]" "output.mp4"

• {OpenAL(+FFmpeg)} How to queue variable size buffer due to ogg format ?

  11 février 2014, par user3293833

  (First of all, I may feel sorry about my poor English as it's not my native language.)

  I use FFmpeg to decode some audio file and play it with OpenAL by "stream"(i.e."queue" and "unqueue" function of OpenAL).

  When I use my program to play .ogg file, I find that it has a variable nb_samples.(due to ogg has variable bit rate ??) There are 128 B, 512 B and 1024 B of nb_samples. As a results, I must call alDeleteBuffers and alGenBuffers before I use alBufferSamplesSOFT(similar to alBufferData) because it would fail to call alBufferSamplesSOFT without recreate the buffer.
  Notes : alBufferSamplesSOFT is provided by OpenAL Soft. You can just see it as alBufferData.

  Nevertheless, I think it's foolish and inefficient if I do this. Is there is some smart method ? I paste the part of code :

  while (av_read_frame(...) == 0){
  
    avcodec_decode_audio4(...);
  
    swr_convert(...);  // to convert PCM format from FLTP to FLT
  
    alDeleteBuffers(1, theBuffers[?]);
  
    alGenBuffers(1, theBuffers[?]);
  
    alBufferSamplesSOFT(...);  // put those data into OpenAL buffer
  
  }

  if I don't do this, It would failed to update the OpenAL buffer. Is there any method to create a variable size buffer or a big size buffer ? Or is there any method to change the size of buffer ?

  Thanks for you guys.