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• Anomalie #2550 (Fermé) : Lien email sur les annnonces n’est pas bon

  5 mars 2012, par cedric -

  Corrigé par http://zone.spip.org/trac/spip-zone/changeset/59089

• lavc/hevcdec : Put slice address checking after hwaccel decode slice

  24 juin 2024, par Fei Wang

  lavc/hevcdec : Put slice address checking after hwaccel decode slice
  Slice address tab only been updated in software decode slice data.
  Fixes hwaccel decoding after d725c737fe2a19091b481d4d115fd939e0a674b2.
  Signed-off-by : Fei Wang <fei.w.wang@intel.com>
  

  • [DH] libavcodec/hevc/hevcdec.c

• LibAV - what approach to take for realtime audio and video capture ?

  26 juillet 2012, par pollux

  I'm using libav to encode raw RGB24 frames to h264 and muxing it to flv. This works
  all fine and I've streamed for more then 48 hours w/o any problems ! My next step
  is to add audio to the stream. I'll be capturing live audio and I want to encode it
  in real time using speex, mp3 or nelly moser.

  Background info

  I'm new to digital audio and therefore I might be doing things wrong. But basically my application gets a "float" buffer with interleaved audio. This "audioIn" function gets called by the application framework I'm using. The buffer contains 256 samples per channel,
  and I have 2 channels. Because I might be mixing terminology, this is how I use the
  data :

  // input = array with audio samples
  
  // bufferSize = 256
  
  // nChannels = 2
  
  void audioIn(float * input, int bufferSize, int nChannels) {
  
      // convert from float to S16
  
          short* buf = new signed short[bufferSize * 2];
  
      for(int i = 0; i < bufferSize; ++i) {  // loop over all samples
  
          int dx = i * 2;
  
          buf[dx + 0] = (float)input[dx + 0] * numeric_limits<short>::max();  // convert frame  of the first channel
  
          buf[dx + 1] = (float)input[dx + 1] * numeric_limits<short>::max();  // convert frame  of the second channel
  
      }
  
  
  
          // add this to the libav wrapper. 
  
      av.addAudioFrame((unsigned char*)buf, bufferSize, nChannels);
  
  
  
      delete[] buf;
  
  }
  
  </short></short>

  Now that I have a buffer, where each sample is 16 bits, I pass this short* buffer, to my
  wrapper av.addAudioFrame() function. In this function I create a buffer, before I encode
  the audio. From what I read, the AVCodecContext of the audio encoder sets the frame_size. This frame_size must match the number of samples in the buffer when calling avcodec_encode_audio2(). Why I think this, is because of what is documented here.

  Then, especially the line :
  If it is not set, frame->nb_samples must be equal to avctx->frame_size for all frames except the last.*(Please correct me here if I'm wrong about this).

  After encoding I call av_interleaved_write_frame() to actually write the frame.
  When I use mp3 as codec my application runs for about 1-2 minutes and then my server, which is receiving the video/audio stream (flv, tcp), disconnects with a message "Frame too large: 14485504". This message is generated because the rtmp-server is getting a frame which is way to big. And this is probably due to the fact I'm not interleaving correctly with libav.

  Questions :

  • There quite some bits I'm not sure of, even when going through the source code of libav and therefore I hope if someone has an working example of encoding audio which comes from a buffer which which comes from "outside" libav (i.e. your own application). i.e. how do you create a buffer which is large enough for the encoder ? How do you make the "realtime" streaming work when you need to wait on this buffer to fill up ?

  • As I wrote above I need to keep track of a buffer before I can encode. Does someone else has some code which does this ? I'm using AVAudioFifo now. The functions which encodes the audio and fills/read the buffer is here too : https://gist.github.com/62f717bbaa69ac7196be

  • I compiled with —enable-debug=3 and disable optimizations, but I'm not seeing any
    debug information. How can I make libav more verbose ?

  Thanks !