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• ffmpeg forcing the usage of nvenc instead of libx264 c++

  3 octobre 2016, par tankyx

  The code below works, but it loads the nvenc encoder instead of the libx264 encoder, which I need for 0 latency streaming.

  this->pCodec = avcodec_find_encoder(AV_CODEC_ID_H264);
  
  if (this->pCodec == NULL)
  
      throw myExceptions("Error: Can't initialize the encoder. FfmpegEncoder.cpp l:9\n");
  
  
  
  this->pCodecCtx = avcodec_alloc_context3(this->pCodec);
  
  
  
  //Alloc output context
  
  if (avformat_alloc_output_context2(&outFormatCtx, NULL, "rtsp", url) < 0)
  
      throw myExceptions("Error: Can't alloc stream output. FfmpegEncoder.cpp l:17\n");

  How can I force the usage of x264 ?

• ffmpeg forcing the usage of nvenc instead of libx264 c++

  3 octobre 2016, par tankyx

  The code below works, but it loads the nvenc encoder instead of the libx264 encoder, which I need for 0 latency streaming.

  this->pCodec = avcodec_find_encoder(AV_CODEC_ID_H264);
  
  if (this->pCodec == NULL)
  
      throw myExceptions("Error: Can't initialize the encoder. FfmpegEncoder.cpp l:9\n");
  
  
  
  this->pCodecCtx = avcodec_alloc_context3(this->pCodec);
  
  
  
  //Alloc output context
  
  if (avformat_alloc_output_context2(&outFormatCtx, NULL, "rtsp", url) < 0)
  
      throw myExceptions("Error: Can't alloc stream output. FfmpegEncoder.cpp l:17\n");

  How can I force the usage of x264 ?

• how do i create a stereo mp3 file with latest version of ffmpeg ?

  17 juin 2016, par Sean

  I’m updating my code from the older version of ffmpeg (53) to the newer (54/55). Code that did work has now been deprecated or removed so i’m having problems updating it.

  Previously I could create a stereo MP3 file using a sample format called :

  SAMPLE_FMT_S16

  That matched up perfectly with my source stream. This has now been replace with

  AV_SAMPLE_FMT_S16

  Which works fine for mono recordings but when I try to create a stereo MP3 file it bugs out at avcodec_open2 with :

  "Specified sample_fmt is not supported."

  Through trial and error I’ve found that using

  AV_SAMPLE_FMT_S16P

  ...is accepted by avcodec_open2 but when I get through and create the MP3 file the sound is very distorted - it sounds about 2 octaves lower than usual with a massive hum in the background - here’s an example recording :

  http://hosting.ispyconnect.com/example.mp3

  I’ve been told by the ffmpeg guys that this is because I now need to manually deinterleave my byte stream before calling :

  avcodec_fill_audio_frame

  How do I do that ? I’ve tried using the swrescale library without success and i’ve tried manually feeding in L/R data into avcodec_fill_audio_frame but the results i’m getting are sounding exactly the same as without interleaving.

  Here is my code for encoding :

  void add_audio_sample( AudioWriterPrivateData^ data, BYTE* soundBuffer, int soundBufferSize)
  
  {
  
      libffmpeg::AVCodecContext* c = data->AudioStream->codec;
  
      memcpy(data->AudioBuffer + data->AudioBufferSizeCurrent,  soundBuffer, soundBufferSize);
  
      data->AudioBufferSizeCurrent += soundBufferSize;
  
      uint8_t* pSoundBuffer = (uint8_t *)data->AudioBuffer;
  
      DWORD nCurrentSize    = data->AudioBufferSizeCurrent;
  
  
  
      libffmpeg::AVFrame *frame;
  
  
  
      int got_packet;
  
      int ret;
  
      int size = libffmpeg::av_samples_get_buffer_size(NULL, c->channels,
  
                                                data->AudioInputSampleSize,
  
                                                c->sample_fmt, 1);
  
  
  
      while( nCurrentSize >= size)    {
  
  
  
          frame=libffmpeg::avcodec_alloc_frame();
  
          libffmpeg::avcodec_get_frame_defaults(frame);
  
  
  
          frame->nb_samples = data->AudioInputSampleSize;
  
  
  
          ret = libffmpeg::avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, pSoundBuffer, size, 1);
  
          if (ret<0)
  
          {
  
              throw gcnew System::IO::IOException("error filling audio");
  
          }
  
          //audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
  
  
  
          libffmpeg::AVPacket pkt = { 0 };
  
          libffmpeg::av_init_packet(&pkt);
  
  
  
          ret = libffmpeg::avcodec_encode_audio2(c, &pkt, frame, &got_packet);
  
  
  
          if (ret<0)
  
                  throw gcnew System::IO::IOException("error encoding audio");
  
          if (got_packet) {
  
              pkt.stream_index = data->AudioStream->index;
  
  
  
              if (pkt.pts != AV_NOPTS_VALUE)
  
                  pkt.pts = libffmpeg::av_rescale_q(pkt.pts, c->time_base, c->time_base);
  
              if (pkt.duration > 0)
  
                  pkt.duration = av_rescale_q(pkt.duration, c->time_base, c->time_base);
  
  
  
              pkt.flags |= AV_PKT_FLAG_KEY;
  
  
  
              if (libffmpeg::av_interleaved_write_frame(data->FormatContext, &pkt) != 0)
  
                      throw gcnew System::IO::IOException("unable to write audio frame.");
  
  
  
  
  
          }
  
          nCurrentSize -= size;  
  
          pSoundBuffer += size;   
  
      }
  
      memcpy(data->AudioBuffer, data->AudioBuffer + data->AudioBufferSizeCurrent - nCurrentSize, nCurrentSize);
  
      data->AudioBufferSizeCurrent = nCurrentSize; 
  
  
  
  }

  Would love to hear any ideas - I’ve been trying to get this working for 3 days now :(