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• Using ffmpeg libraries to decode wav audio as PCM samples and display waveform

  26 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.

• 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.

• FFmpeg : Create a Video Slideshow from PNG Images for MS PowerPoint

  20 septembre 2016, par Klaidonis

  I am using FFmpeg on Windows 7 to create a video from PNG image sequence for Microsoft PowerPoint.

  The best results I have achieved so far, is by using the following command :

  ffmpeg -framerate 10 -start_number 3 -i .\folder\name_%d.png -q:v 0 -c:v libx264 -pix_fmt yuv420p output-video.avi

  It seems perfect, 50 images of total size 30 MB are converted into 200 KB video with no loss in the quality. Placing it in PowerPoint also seems right, but there is a slight color shift (yellow appears darker and possibly more orangish). By using some other conversion options, I obtained a video in PowerPoint where the first image of the video (like album cover art) is exactly as the original but the rest of the video plays with the mentioned color shift.

  When I play this file in VLC, it’s good. Although, if in the settings "Use hardware YUV->RGB conversions" is enabled, colors appear a bit washed out, and white color is a bit gray.

  I also tried to convert the images to a GIF file and at first it seems good but outer edges and numbers from the left and top side are blurred, and the background has turned from white to a bit gray color, although, white segments not on the background are white. The output size is 18 MB. I ended up with a way better GIF 600 KB by converting from the first video file ; just it is slightly more dotted, and the background again is grayish.

  ffmpeg -i output-video.avi output-video.gif

  Could someone help ?