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• avfilter/af_dynaudnorm : do not clip audio

  4 janvier 2020, par Paul B Mahol

  avfilter/af_dynaudnorm : do not clip audio
  Clipping can happen when smoothed gain is higher than maximum
  
  allowed gain factor for current frame and peak value option is
  
  set to enough low value.
  

  • [DH] libavfilter/af_dynaudnorm.c

• 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,...

  

• af_hdcd : Report PE as being intermittent or permanent

  24 juillet 2016, par Burt P

  af_hdcd : Report PE as being intermittent or permanent
  The Peak Extend feature could be enabled permanently or only
  
  when needed. This is now reported.
  Signed-off-by : Burt P <pburt0@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavfilter/af_hdcd.c