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• ffmpeg Format settings, Matrix bt709

  18 mai 2016, par user1722669

  Does anyone know how I can achieve the following color space (bt.709) via FFmpeg ?
  Here is what I have now in my files...

  

  As you can see there is Format settings, Matrix as default, how can I set it like this :

  • Format settings (Matrix) : Custom or Standard
  • Component Color primaries : BT.709
  • Transfer characteristics : BT.709
  • Matrix coefficients : BT.709

  Thank you

• What is the best jpeg encoder/decoder ? [on hold]

  27 mai 2014, par user3678446

  So I made a server and I was playing with data transfer. And i wanted to make a program like TeamViewer(just for fun not planning to achieve teamViewer performance). And I was sending png screenshots. After a few tests I realized that png encoding takes way too long.
  So i decided to change all that part to jpeg. And I would want to know what is the best library to use. I heard in some other question that libavcodec in the ffmpeg project is really good. It seems pretty complicated especially cause I don’t have a source of learning it and I will try to learn it no my own. And I don’t want to put that much effort(if it’s that much effort) to learn it when i could learn the best one.

  So the question is what is the best library for jpeg encoding and decoding. Or maybe if it’s there something better for something like teamviewer let me know.

• why the output of mp3 decode sounds so delayed ?(with ffmpeg mp3lame lib)

  11 mars 2014, par user3401739

  i'm recording sound and encoding to mp3 with ffmpeg lib. then decode the mp3 data right away, play the decode data, but it's sounds so delayed.
  here are the codes :
  the function encode first parameter accepts the raw pcm data, len = 44100.

  encode parameters :

  cntx_->channels = 1;
  
  cntx_->sample_rate = 44100;
  
  cntx_->sample_fmt = 6;
  
  cntx_->channel_layout =  AV_CH_LAYOUT_MONO;
  
  cntx_->bit_rate = 8000;
  
  err_ = avcodec_open2(cntx_, codec_, NULL);
  
  
  
  vector<unsigned char="char">       encode(unsigned char* encode_data, unsigned int len)
  
  {
  
      vector<unsigned char="char"> ret;
  
      AVPacket avpkt;
  
      av_init_packet(&amp;avpkt);
  
  
  
      unsigned int len_encoded = 0; 
  
      int data_left = len / 2;
  
      int miss_c = 0, i = 0;
  
      while (data_left > 0)
  
      {
  
          int sz = data_left > cntx_->frame_size ? cntx_->frame_size : data_left;
  
          mp3_frame_->nb_samples = sz;
  
          mp3_frame_->format = cntx_->sample_fmt;
  
          mp3_frame_->channel_layout = cntx_->channel_layout;
  
  
  
          int needed_size = av_samples_get_buffer_size(NULL, 1,
  
              mp3_frame_->nb_samples, cntx_->sample_fmt, 1);
  
  
  
          int r = avcodec_fill_audio_frame(mp3_frame_, 1, cntx_->sample_fmt, encode_data + len_encoded, needed_size, 0);
  
  
  
          int gotted = -1;
  
  
  
          r = avcodec_encode_audio2(cntx_, &amp;avpkt, mp3_frame_, &amp;gotted);
  
          if (gotted){
  
              i++;
  
              ret.insert(ret.end(), avpkt.data, avpkt.data + avpkt.size);
  
          }
  
          else if (gotted == 0){
  
              miss_c++;
  
          }
  
          len_encoded += needed_size;
  
          data_left -= sz;
  
          av_free_packet(&amp;avpkt);
  
      }
  
      return ret;
  
  }
  
  
  
  std::vector<unsigned char="char">  decode(unsigned char* data, unsigned int len)
  
  {
  
      std::vector<unsigned char="char"> ret;
  
  
  
      AVPacket avpkt;
  
      av_init_packet(&amp;avpkt);
  
      avpkt.data = data;
  
      avpkt.size = len;
  
  
  
      AVFrame* pframe = av_frame_alloc();
  
      while (avpkt.size > 0){
  
          int goted = -1;av_frame_unref(pframe);
  
          int used = avcodec_decode_audio4(cntx_, pframe, &amp;goted, &amp;avpkt);
  
          if (goted){
  
              ret.insert(ret.end(), pframe->data[0], pframe->data[0] + pframe->linesize[0]);
  
              avpkt.data += used;
  
              avpkt.size -= used;
  
              avpkt.dts = avpkt.pts = AV_NOPTS_VALUE; 
  
          }
  
          else if (goted == 0){
  
              avpkt.data += used;
  
              avpkt.size -= used;
  
              avpkt.dts = avpkt.pts = AV_NOPTS_VALUE; 
  
          }
  
          else if(goted &lt; 0){
  
              break;
  
          }
  
      }
  
      av_frame_free(&amp;pframe);
  
      return ret;
  
  }
  
  </unsigned></unsigned></unsigned></unsigned>

  Suppose it's the 100th call to encode(data, len), this "frame" would appear in 150th or later in the decode call, the latency is not acceptable. It seems the mp3lame encoder would keep the sample data for later use, but not my desire.
  I don't know what is going wrong. Thank you for any information.

  today i debug the code again and post some detail :

  encode : each pcm sample frame len = 23040 ,which is 10 times of mp3 frame size, each time call encode only output 9 frames, this output cause decode output 20736 samples, 1 frame(2304 bytes) is lost, and the sound is noisy.

  if the mp3 or mp2 encode is not suitable for real time voice transfer, which encoder should i choose ?