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• PTS and DTS calculation for video and audio frames

  12 juin 2022, par Kaidul

  I am receiving video H264 encoded data and audio G.711 PCM encoded data from two different threads to mux / write into mov multimedia container.

  



  The writer function signatures are like :

  



  bool WriteAudio(const unsigned char *pEncodedData, size_t iLength);
bool WriteVideo(const unsigned char *pEncodedData, size_t iLength, bool const bIFrame);


  



  And the function for adding audio and video streams looks like :

  



  AVStream* AudioVideoRecorder::AddMediaStream(enum AVCodecID codecID) {
    Log("Adding stream: %s.", avcodec_get_name(codecID));
    AVCodecContext* pCodecCtx;
    AVStream* pStream;

    /* find the encoder */
    AVCodec* codec = avcodec_find_encoder(codecID);
    if (!codec) {
        LogErr("Could not find encoder for %s", avcodec_get_name(codecID));
        return NULL;
    }

    pStream = avformat_new_stream(m_pFormatCtx, codec);
    if (!pStream) {
        LogErr("Could not allocate stream.");
        return NULL;
    }
    pStream->id = m_pFormatCtx->nb_streams - 1;
    pStream->time_base = (AVRational){1, VIDEO_FRAME_RATE};
    pCodecCtx = pStream->codec;


    switch(codec->type) {
    case AVMEDIA_TYPE_VIDEO:
        pCodecCtx->codec_id = codecID;
        pCodecCtx->bit_rate = VIDEO_BIT_RATE;
        pCodecCtx->width = PICTURE_WIDTH;
        pCodecCtx->height = PICTURE_HEIGHT;
        pCodecCtx->gop_size = VIDEO_FRAME_RATE;
        pCodecCtx->pix_fmt = PIX_FMT_YUV420P;
        m_pVideoStream = pStream;
        break;

    case AVMEDIA_TYPE_AUDIO:
        pCodecCtx->codec_id = codecID;
        pCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
        pCodecCtx->bit_rate = 64000;
        pCodecCtx->sample_rate = 8000;
        pCodecCtx->channels = 1;
        m_pAudioStream = pStream;
        break;

    default:
        break;
    }

    /* Some formats want stream headers to be separate. */
    if (m_pOutputFmt->flags & AVFMT_GLOBALHEADER)
        m_pFormatCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;

    return pStream;
}


  



  Inside WriteAudio(..) and WriteVideo(..) functions, I am creating AVPakcet using av_init_packet(...) and set pEncodedData and iLength as packet.data and packet.size. I printed packet.pts and packet.dts and its equivalent to AV_NOPTS_VALUE.

  



  Now, how do I calculate the PTS, DTS, and packet duration (packet.dts, packet.pts and packet.duration) correctly for both audio and video data so that I can sync audio & video and play it properly ? I saw many examples on the internet, but none of them are making sense to me. I am new with ffmpeg, and my conception may not be correct in some context. I want to do it in the appropriate way.

  



  Thanks in advance !

  



  EDIT : In my video streams, there is no B frame. So, I think PTS and DTS can be kept the same here.

  


• FFmpeg - confused with the concept of audio frame size and its calculation

  12 mars 2019, par Jinx

  I’m learning video and audio codecs with FFmpeg. I’m strgulling to understand frame size and some other concepts.

  Frame size
  This is the size in bytes of each frame. This can vary a lot : if each sample
  is 8 bits, and we’re handling mono sound, the frame size is one byte.
  Similarly in 6 channel audio with 64 bit floating point samples, the frame size is 48 bytes (PCM Terminology and Concepts)

  As described above, if each sample is 8 bits and there are 6 channels, then the frame size will be 48 bytes. The result from my code was 96 (16 bits * 6 channels). On the other side, the result from the call stream->codecpar->frame_size was 1024. Why were they different ? Is the frame size 1024 for 6 channels or just each channel ?

  main.cpp :

  else if (stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
  
      std::cout << "audio sample rate: " << stream->codecpar->sample_rate << std::endl;
  
      std::cout << "audio bits: " << stream->codecpar->bits_per_coded_sample << std::endl;
  
      std::cout << "audio channels: " << stream->codecpar->channels << std::endl;
  
      std::cout << "audio frame size: " << stream->codecpar->frame_size << std::endl; 
  
      std::cout << "audio frame size: (16bits * 6 channels) = " << stream->codecpar->channels * stream->codecpar->bits_per_coded_sample << std::endl; 
  
      audio_stream_index = i;
  
  }

  console :

  audio sample rate: 48000
  
  audio bits: 16
  
  audio channels: 6
  
  audio frame size: 1024
  
  audio frame size: (16bits * 6 channels) = 96

• vp56data : Move all shared enum/struct declarations to common header

  14 août 2013, par Diego Biurrun

  vp56data : Move all shared enum/struct declarations to common header
  

  • [DH] libavcodec/vp56.h
  • [DH] libavcodec/vp56data.h
  • [DH] libavcodec/vp6data.h
  • [DH] libavcodec/vp8.h