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• Using ffmpeg libraries to decode wav audio as PCM samples

  25 septembre 2016, par Lorenzo Monni

  I’m using the ffmpeg libraries to process audio files.

  I need to decode .wav audio files to make some operations having their samples in an understandable format, i.e. decimal numbers comprised between [-1,1] as a normal audio waveform.

  I have written the code for decoding and it’s apparently working well, but when I see the decoded samples it seems something in the sample numbers translation went bad. I paste here only the part of code where I translate the samples from the audio frame in PCM 16 bits :

   while(av_read_frame(pFormatCtx, &apkt)>=0) {
  
      if(apkt.stream_index==audioStream->index)
  
      {
  
  
  
          // Try to decode the packet into a frame
  
          int frameFinished = 0;
  
          avcodec_decode_audio4(aCodecCtx, aFrame, &frameFinished, &apkt);
  
          int data_size = av_samples_get_buffer_size(&plane_size, aCodecCtx->channels,
  
                                                          aFrame->nb_samples,
  
                                                          aCodecCtx->sample_fmt, 1);
  
          // Some frames rely on multiple packets, so we have to make sure the frame is finished before
  
          // we can use it
  
          if (frameFinished)
  
          {
  
              for(int a=0;a < plane_size/sizeof(int16_t);a++)
  
              {
  
                  fprintf(fd,"%d\n",(int16_t*)aFrame->data[a]);
  
  
  
              }
  
          }
  
      }
  
  
  
      av_free_packet(&apkt);
  
  // Free the packet that was allocated by av_read_frame
  
  
  
  }

  Additional information and issues :

  • the sample_fmt in my allocated AVCodecContext is "AV_SAMPLE_FMT_S16" so the samples numbers should be 16bit signed binaries, I guess if translated in decimal format numbers comprised between -32768 and 32767 (I don’t remember how the problem of disparity between positives and negatives is solved). However when I decode them in int16_t I see much higher numbers that seem to fall in the 32bit signed format (but the file is in 16bit anyway). E.g., the max of my decoded audio (after int16_t translation) is 2044951012 ;

  • My .wav file has two channels, but I can’t access both two, if I use extended_data of the audio frame struct pointing to the second channel (index 1) the execution returns a segmentation fault. The same happens with the data pointer. I’m able to recover only one channel, from data[0].

  Here is how my audio file decoded with the aforementioned code and saved in a txt looks like :
  

  Here is how the trend of the signal should look like :

  

  If I play my decoded signal the sound shows some similarities with the original audio file, but with a lot of destructing artifacts in it.

  Final remarks : ffmpeg documentation and past questions of Stackoverflow are not working well to solve this problem.

• How to restream m3u8 with ffmpeg

  23 mars 2020, par Nk nk nk

  I use nginx rtmp and followig command :

  ffmpeg -fflags +igndts -hide_banner -i https://ch.iptvmate.net/ec6e5689ffd6f9690102640bddd2f9e7.m3u8 -c copy -f hls -hls_time 4 -hls_flags append_list+delete_segments -hls_list_size 6 -hls_segment_filename 'hls/ch2/file%03d.ts' hls/ch2/playlist.m3u8

  It streams but not live, it has recorded one period of the channel and this link streams this period again and again. Is the command correct ? I just created the folder called ch2 and it is.

• When rTorrent finishes, run FFmpeg and convert audio track in mkv to ac-3 5.1

  11 septembre 2021, par miniHessel

  What is the best approach for this ? I saw the following answer, is that still a valid approach ? If so, how do you execute that when a torrent finishes ? I mostly want to convert dts-hd either 5.1 or 7.1. Sometimes also Atmos.