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• FFmpeg : Choosing audio streams by language

  16 octobre 2017, par jdauthre

  I am trying to transcode tv streams but with only the english audio stream included. I have tried using the -map 0:m:language:eng stream specifier, but I get :

  "Automatic encoder selection failed for output stream #0:3. Default encoder for format mpegts (codec none) is probably disabled. Please choose an encoder manually.
  
  Error selecting an encoder for stream 0:3"

  This is despite including an encoder. I have tried all sorts of variations on this theme without success.

  Full output for one attempt is below :

  ffmpeg -i http://192.168.1.74:8001/1:0:1:189E:7FD:2:11A0000:0:0:0: -ignore_unknown -map 0:a -map 0:m:language:eng -map 0:v -acodec aac -vcodec libx264 -b:v 1100000 -t 00:00:30 "somethin.ts" 2>output.txt
  
  
  
      Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [mpegts @ 03db7b60] Could not find codec parameters for stream 17 (Unknown: none ([11][0][0][0] / 0x000B)): unknown codec
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [mpegts @ 03db7b60] Could not find codec parameters for stream 18 (Unknown: none ([11][0][0][0] / 0x000B)): unknown codec
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  Input #0, mpegts, from 'http://192.168.1.74:8001/1:0:1:189E:7FD:2:11A0000:0:0:0:':
  
    Duration: N/A, start: 23690.732933, bitrate: N/A
  
    Program 6321 
  
    Program 6322 
  
    Program 6338 
  
    Program 6301 
  
    Program 6302 
  
      Stream #0:0[0x13ec]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, top first), 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
  
      Stream #0:1[0x13ee](NAR): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 224 kb/s
  
      Stream #0:2[0x13ed](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16p, 224 kb/s
  
      Stream #0:3[0x13ef](eng,eng): Subtitle: dvb_teletext ([6][0][0][0] / 0x0006)
  
      Stream #0:4[0x13f0](eng): Subtitle: dvb_subtitle ([6][0][0][0] / 0x0006)
  
      Stream #0:5[0xf04]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:6[0xf03]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:7[0xf02]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:8[0xf01]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:9[0xf00]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:10[0x92a]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:11[0x913]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:12[0x912]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:13[0x911]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:14[0x919]: Unknown: none ([5][0][0][0] / 0x0005)
  
      Stream #0:15[0xf09]: Unknown: none ([11][0][0][0] / 0x000B)
  
      Stream #0:16[0xf08]: Unknown: none ([11][0][0][0] / 0x000B)
  
      Stream #0:17[0xf07]: Unknown: none ([11][0][0][0] / 0x000B)
  
      Stream #0:18[0xf06]: Unknown: none ([11][0][0][0] / 0x000B)
  
    Program 6318 
  
    Program 6390 
  
    Program 6391 
  
    Program 6351 
  
    Program 6361 
  
    Program 6306 
  
    Program 6341 
  
  Automatic encoder selection failed for output stream #0:3. Default encoder for format mpegts (codec none) is probably disabled. Please choose an encoder manually.
  
  Error selecting an encoder for stream 0:3

  Any ideas on how to do this. I cant specify streams by number as I want to use it for lots of tv streams and the order is often different.

  Thanks

• converting eac3 to aac with ffmpeg

  30 novembre 2023, par vespino

  I have a number of video files stored on my Synology of which the audio doesn't play on when opening them with DS File. This is because DS File does not support eac3. I would like to convert those files to aac using ffmpeg, but when doing so, all audio is lost.

  


  This is the file info :

  


  Input #0, matroska,webm, from &#x27;<file>.mkv&#x27;:&#xA;  Metadata:&#xA;    encoder         : libebml v1.4.2 &#x2B; libmatroska v1.6.4&#xA;  Duration: 00:48:43.42, start: 0.000000, bitrate: 6566 kb/s&#xA;    Stream #0:0: Video: h264 (High), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)&#xA;    Stream #0:1(eng): Audio: eac3, 48000 Hz, 5.1 (default)&#xA;    Stream #0:2(eng): Subtitle: subrip&#xA;    Metadata:&#xA;      title           : English [SDH]&#xA;    Stream #0:3(ara): Subtitle: subrip&#xA;    Metadata:&#xA;      title           : Arabic&#xA;    Stream #0:4(chi): Subtitle: subrip&#xA;</file>

  


  This is my attempt :

  


  ffmpeg -i <file>.mkv -map 0:v -map 0:a:0 -map 0:s -c copy -c:a aac -b:a 640k output.mkv&#xA;</file>

  


  Subtitles are kept, but audio is completely removed. I would like to automate the process if possible.

  


  This is my ffmpeg configuration :

  


  ffmpeg version 4.1.8 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 8.5.0 (GCC)
  configuration: --prefix=/usr --incdir='${prefix}/include/ffmpeg' --arch=i686 --target-os=linux --cross-prefix=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu- --enable-cross-compile --enable-optimizations --enable-pic --enable-gpl --enable-shared --disable-static --disable-stripping --enable-version3 --enable-encoders --enable-pthreads --disable-protocols --disable-protocol=rtp --enable-protocol=file --enable-protocol=pipe --disable-muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-ffplay --disable-ffprobe --disable-doc --disable-devices --disable-bzlib --disable-altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-vaapi --disable-cuvid --disable-nvenc --disable-decoder=aac --disable-decoder=aac_fixed --disable-encoder=aac --disable-decoder=amrnb --disable-decoder=ac3 --disable-decoder=ac3_fixed --disable-encoder=zmbv --disable-encoder=dca --disable-decoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed --disable-encoder=eac3 --disable-decoder=eac3 --disable-encoder=truehd --disable-decoder=truehd --disable-encoder=hevc_vaapi --disable-decoder=hevc --disable-muxer=hevc --disable-demuxer=hevc --disable-parser=hevc --disable-bsf=hevc_mp4toannexb --x86asmexe=yasm --cc=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu-wrap-gcc --enable-yasm --enable-libx264 --enable-encoder=libx264
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100


  


  Synology does not support eac3 out of the box. So is this even possible ?

  


• Android + ffmpeg + AudioTrack produces bad audio output

  12 septembre 2014, par Goddchen

  here is what I am trying to do : use an AudioRecord and "pipe" the output of AudioRecord.read(byte[],...) to an ffmpeg process’ stdin that will convert to a 3gp (AAC) file.

  The ffmpeg call is as follows :

          ProcessBuilder processBuilder = new ProcessBuilder(BINARY.getAbsolutePath(),
  
                  "-y",
  
                  "-ar", "44100", "-c:a", "pcm_s16le", "-ac", "1","-f","s16le",
  
                  "-i", "-",
  
                  "-strict", "-2", "-c:a", "aac",
  
                  outFile.getAbsolutePath());

  The AudioRecord is setup as follows :

   AudioRecord record = new AudioRecord(/*AudioSource.VOICE_RECOGNITION,*/ AudioSource.MIC,
  
              SAMPLING_RATE,
  
              AudioFormat.CHANNEL_IN_MONO,
  
              AudioFormat.ENCODING_PCM_16BIT,
  
              bufferSize);

  SAMPLING_RATE = 44100 and bufferSize is the one returned by AudioRecord.getMinBufferSize(...)

  I am writing the data to ffmpeg like this :

  try {
  
                          IOUtils.write(data, getFFmpegHelper().getCurrentProcessOutputStream());
  
                      } catch (Exception e) {
  
                          Log.e(Application.LOG_TAG, "Error writing data to ffmpeg process", e);
  
                          //TODO notify user, stop the recording, etc...
  
                      }

  So far so good, the ffmpeg runs and created a proper 3gp file. But the audio in the file is totally off. It seems "choppy" (not sure if this is the correct english word ;) ) and also the pace is wrong, is plays too fast.

  Check out this sample : http://goddchen.de/android/tmp/tmp.3gp

  This is the output of the ffmpeg process :

      [s16le @ 0x23634d0] Estimating duration from bitrate, this may be inaccurate
  
      Guessed Channel Layout for  Input Stream #0.0 : mono
  
      Input #0, s16le, from 'pipe:':
  
      Duration: N/A, start: 0.000000, bitrate: 705 kb/s
  
      Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
  
      [aformat @ 0x2363100] auto-inserting filter 'auto-inserted resampler 0' between the filter 'src' and the filter 'aformat'
  
      [aresample @ 0x235b0a0] chl:mono fmt:s16 r:44100Hz -> chl:mono fmt:flt r:44100Hz
  
      Output #0, 3gp, to '/data/data/com.test.audio/files/tmp.3gp':
  
      Metadata:
  
      encoder         : Lavf54.6.100
  
      Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, flt, 128 kb/s
  
      Stream mapping:
  
      Stream #0:0 -> #0:0 (pcm_s16le -> aac)
  
      size=       3kB time=00:00:00.18 bitrate= 132.5kbits/s    
  
  size=       8kB time=00:00:00.55 bitrate= 120.9kbits/s    
  
  size=      12kB time=00:00:00.83 bitrate= 121.8kbits/s    
  
  size=      16kB time=00:00:01.04 bitrate= 122.8kbits/s    
  
  size=      20kB time=00:00:01.32 bitrate= 122.5kbits/s    
  
  size=      23kB time=00:00:01.53 bitrate= 121.6kbits/s    
  
  size=      27kB time=00:00:01.81 bitrate= 121.0kbits/s    
  
  size=      31kB time=00:00:02.11 bitrate= 120.7kbits/s    
  
  size=      35kB time=00:00:02.32 bitrate= 123.4kbits/s
  
      video:0kB audio:34kB global headers:0kB muxing overhead 3.031610%