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• De-quantising audio with ffmpeg

  11 mai 2016, par Henry W.

  I am using FFmpeg library to decode and (potentially) modify some audio.

  I managed to use the following functions to iterate through all frames of the audio file :

  avformat_open_input // Obtains formatContext
  
  avformat_find_stream_info
  
  av_find_best_stream // The argument AVMEDIA_TYPE_AUDIO is fed in to find the audio stream
  
  avcodec_open2 // Obtains codecContext
  
  av_init_packet
  
  
  
  // The following is used to loop through the frames
  
  av_read_frame
  
  avcodec_decode_audio4

  In the end, I have these three values available on each iteration

  int dataSize; // return value of avcodec_decode_audio4
  
  AVFrame* frame;
  
  AVCodecContext* codecContext; // Codec context of the best stream

  I supposed that a loop like this can be used to iterate over all samples :

  for (int i = 0; i < frame->nb_samples; ++i)
  
  {
  
      // Bytes/Sample is known to be 4
  
      // Extracts audio from Channel 1. There are in total 2 channels.
  
      int* sample = (int*)frame->data[0] + dataSize * i;
  
      // Now *sample is accessible
  
  }

  However, when I plotted the data using gnuplot, I did not get a waveform as expected, and some of the values reached the the limit of 32 bits integers : (The audio stream is not silent in the first few seconds)

  I suppose that some form of quantisation is going on to prevent the data from being interpreted mathematically. What should I do to de-quantise this ?

• Enabling mp4/mpeg4/avc support for Qt5 WebEngine on Linux

  12 avril 2016, par Thomas233

  i installed Qt 5.4.1 x64 on LUbuntu and created an app which uses the new QtWebEngine.

  I`m trying to display a html5 page with that component which is using the tag.
  All works fine except if I try to playback a mp4 video. The video area remains black. It works if I use other video types like webm/ogg as source.

  I know this is due to license restrictions, so that mp4 is deactivated by default in Ubuntu/Linux for Qt.

  What is needed in Qt to activate it to allow mp4 playback and on what do I have pay attention in case of license terms (I read that statically linking the library is allowed ?) ?

  I`ve already tried to copy over the x64 distribution of libffmpegsuo.so which is included in Chrome (2,2Mb) over to the Qt directory to /plugins/webengine/ and replaced that one that was already there (1,1 Mb) but it had no effect.
  In Chrome playback works fine btw.

  If you need more details like paths etc. please tell me.

  Thanks !

• FFMPEG Audio decode and draw waveform

  7 avril 2016, par Haris

  I am trying to decode the audio and draw the waveform using ffmpeg, and the input audio data is AV_SAMPLE_FMT_S16P, basically I am following the tutorial here, and the audio is playing fine with libao. Now I need to plot the waveform using decoded data, currently I am writing left and right channel to separate csv file and plotting on excel. But the waveform is something different from the waveform shown in Audacity using the same audio clip. When I analyzed the value written on csv most of the values are close to maximum of uint16_t(65535), but there are some other lower values, but majority is high peak.

  Here is the source code,

      const char* input_filename="/home/user/Music/Clip.mp3";
  
      av_register_all();
  
      AVFormatContext* container=avformat_alloc_context();
  
      if(avformat_open_input(&container,input_filename,NULL,NULL)<0){
  
          endApp("Could not open file");
  
      }
  
  
  
      if(avformat_find_stream_info(container, NULL)<0){
  
          endApp("Could not find file info");
  
      }
  
  
  
      av_dump_format(container,0,input_filename,false);
  
  
  
      int stream_id=-1;
  
      int i;
  
      for(i=0;inb_streams;i++){
  
          if(container->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO){
  
              stream_id=i;
  
              break;
  
          }
  
      }
  
      if(stream_id==-1){
  
          endApp("Could not find Audio Stream");
  
      }
  
  
  
      AVDictionary *metadata=container->metadata;
  
  
  
      AVCodecContext *ctx=container->streams[stream_id]->codec;
  
      AVCodec *codec=avcodec_find_decoder(ctx->codec_id);
  
  
  
      if(codec==NULL){
  
          endApp("cannot find codec!");
  
      }
  
  
  
      if(avcodec_open2(ctx,codec,NULL)<0){
  
          endApp("Codec cannot be found");
  
      }
  
  
  
  
  
  
  
      AVPacket packet;
  
      av_init_packet(&packet);
  
  
  
      //AVFrame *frame=avcodec_alloc_frame();
  
      AVFrame *frame=av_frame_alloc();
  
  
  
      int buffer_size=AVCODEC_MAX_AUDIO_FRAME_SIZE+ FF_INPUT_BUFFER_PADDING_SIZE;
  
  
  
      // MSVC can't do variable size allocations on stack, ohgodwhy
  
      uint8_t *buffer = new uint8_t[buffer_size];
  
      packet.data=buffer;
  
      packet.size =buffer_size;
  
  
  
      int frameFinished=0;
  
  
  
      int plane_size;
  
  
  
      ofstream fileCh1,fileCh2;
  
      fileCh1.open ("ch1.csv");
  
      fileCh2.open ("ch2.csv");
  
  
  
      AVSampleFormat sfmt=ctx->sample_fmt;
  
  
  
      while(av_read_frame(container,&packet)>=0)
  
      {
  
  
  
          if(packet.stream_index==stream_id){
  
              int len=avcodec_decode_audio4(ctx,frame,&frameFinished,&packet);
  
              int data_size = av_samples_get_buffer_size(&plane_size, ctx->channels,
  
                                                  frame->nb_samples,
  
                                                  ctx->sample_fmt, 1);
  
  
  
  
  
              if(frameFinished){
  
                  int write_p=0;
  
                  // QTime t;
  
                  switch (sfmt){
  
  
  
                      case AV_SAMPLE_FMT_S16P:
  
  
  
                          for (int nb=0;nbsizeof(uint16_t);nb++){
  
                              for (int ch = 0; ch < ctx->channels; ch++) {
  
                                  if(ch==0)
  
                                      fileCh1 <<((uint16_t *) frame->extended_data[ch])[nb]<<"\n";
  
                                  else if(ch==1)
  
                                      fileCh2 <<((uint16_t *) frame->extended_data[ch])[nb]<<"\n";
  
                              }
  
                          }
  
  
  
                          break;
  
  
  
                  }
  
              } else {
  
                  DBG("frame failed");
  
              }
  
          }
  
  
  
  
  
          av_free_packet(&packet);
  
      }
  
      fileCh1.close();
  
      fileCh2.close();
  
      avcodec_close(ctx);
  
      avformat_close_input(&container);
  
      delete buffer;
  
      return 0;

  Edit :

  I have attached the waveform image draw using opencv, here I scaled the sample value to 0-255 range, and took value 127 as 0(Y-axis). Now for each sample draw line from (x,127) to (x,sample value) where x=1,2,3,...