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• 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 !

• Using LibX265 with FFMPEG to encode YUV to MKV

  11 mars 2014, par Jonathan Lunberg

  I am trying to use libx265 to encode my raw YUV file to a MKV file for the purposes of testing. I am new to video compression and hope I'm making a simple mistake that someone can correct. The current command I'm using is :

  ffmpeg -s 1280:720 -i input.yuv -c:v libx265 -b:v 1000k -c:a copy output.mkv 

  The error that I'm getting reads :

  error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  I have tried searching for the parameters I need to use but cannot fine anything which works. Any help is greatly appreciated

• FFmpeg converted mp4 file fails to load in Quicktime

  21 décembre 2015, par wonea

  I’ve been using FFmpeg to convert a movie I need to play in MP4, however in Quicktime the following error is presented ;

  

  (Error -2041 : an invalid sample description was found in the movie (output.mp4)).

  I used the following FFmpeg command parameters to convert the file ;

  C :\Temp\EOBTemp>ffmpeg -i input.mp4 -y
  -acodec libmp3lame -ab 96k -vcodec libx264 -vpre lossless_slow -crf 22
  -threads 0 output.mp4

  Now this file plays absolutely fine, in Windows Media Player, and VideoLAN (FFmpeg based, so no surprise here. I using the latest build from HawkEye’s FFmpeg Windows Builds
  (FFmpeg git-a304071 32-bit Static (Latest)).

  I really hope this isn’t a AAC problem, as I’ve been trying to get FFmpeg to use the libfaac.dll library (in the same folder as the FFmpeg.exe) with the command ;

  -acodec libfaac

  Help ! I’m at a loss !