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• can i mux h264 stream into mp4(aac) through libavformat without libx264 ?

  6 septembre 2011, par Faller

  i just have h.264 encoded video stream and i want to make mp4 file.

  /* find output format for mp4 */

  m_pOutputFormat= av_guess_format("mp4", NULL, NULL) ;

  if (!m_pOutputFormat) return FALSE ; // could not find suitable output format(mp4).

  on this code, i get mpeg for video codec not h264, i think that's because i build ffmpeg without libx264. (and i dont want to build ffmpeg with libx264 for license)

  m_pOutputFormat->video_codec= CODEC_ID_H264 ;

  when i change its video_codec to CODEC_ID_H264, it works fine some player(kmplayer). but it's not work on ipod, QuickTime.

  this code maybe wrong because it could not change codec_tag value(this variable has const property).

  1. how can i get other result for av_guess_format("mp4",NULL,NULL) without recompile libav+libx264 ?

  2. how can i make mp4 file properly ?

• Interact with ffmpeg from a .NET program ?

  18 septembre 2011, par Shimmy

  I'm trying to create a .NET wrapper for media-file conversion using ffmepg, here is what I've tried :

  static void Main(string[] args)
  
  {
  
    if (File.Exists("sample.mp3")) File.Delete("sample.mp3");
  
  
  
    string result;
  
  
  
    using (Process p = new Process())
  
    {
  
      p.StartInfo.FileName = "ffmpeg";
  
      p.StartInfo.Arguments = "-i sample.wma sample.mp3";
  
  
  
      p.StartInfo.UseShellExecute = false;
  
      p.StartInfo.RedirectStandardOutput = true;
  
  
  
      p.Start();
  
  
  
      //result is assigned with an empty string!
  
      result = p.StandardOutput.ReadToEnd();
  
  
  
      p.WaitForExit();
  
    }
  
  }

  What actually happens is the content of the ffmpeg program is printed out to the Console app, but the result variable is an empty string. I want to control the conversion progress interactively, so the user doesn't even have to know I'm using ffmpeg, but he still knows the conversion progress' details and what percentage etc. the app is up to.

  Basically I would also be happy with a .NET wrapper for a P/Invoke to conversion function ONLY (I am not interested in a whole external library, unless I can extract the PI function from it).

  Anyone with experience in ffmpeg & .NET ?

  Update
  Please view my further question, how to write input to a running ffmpeg process.

• iPhone ffmpeg dev using libav to decode from AMR to ACC codec

  10 octobre 2011, par VictorT

  It seems to be that, the AMR support of AudioQueue has been disappeared since iOS 4.3 was released. I can't use audio frame received from RSTP server with old way :

  audioFormat.mFormatID = kAudioFormatAMR; 
  
  int err = AudioQueueNewOutput(&audioFormat, MyAudioQueueOutputCallback, self, NULL, kCFRunLoopCommonModes, 0, &audioQueue);

  As a result I received an error in last string.

  Maybe someone know how to decode AMR AVPacket into raw buffer and encode it with AAC or MP3 using LIBAV ?

  I've tried to use

  avcodec_decode_audio3

  It works and I can get raw buffer but when I'm trying to encode it with

  avcodec_encode_audio

  I get 0 as result

  This is my method to encode buffer :

  - (AVPacket) encodeRawFrame:(const short *) in_buffer withSize:(unsigned int) in_buf_byte_size
  
  {
  
      AVPacket res;
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int count, out_size, outbuf_size, frame_byte_size;
  
      uint8_t *outbuf;
  
  
  
      avcodec_init();
  
      avcodec_register_all();
  
  
  
      printf("Audio encoding\n");
  
  
  
      codec = avcodec_find_encoder(CODEC_ID_AAC);
  
      if (!codec) {
  
          fprintf(stderr, "codec not found\n");
  
          return res;
  
      }
  
  
  
      c= avcodec_alloc_context();
  
  
  
      c->bit_rate = 64000;
  
      c->sample_rate = 24000;
  
      c->channels = 2;
  
  
  
      if (avcodec_open(c, codec) < 0) 
  
      {
  
          fprintf(stderr, "could not open codec\n");
  
      }
  
      else
  
      {
  
          frame_byte_size=c->frame_size*2*2;
  
          count = in_buf_byte_size/frame_byte_size;
  
  
  
          fprintf(stderr, "Number of frames: %d\n", count);
  
  
  
          outbuf_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
          outbuf = (uint8_t*) malloc(outbuf_size);
  
  
  
          out_size = avcodec_encode_audio(c, outbuf, outbuf_size, &in_buffer[frame_byte_size*i]);
  
          if(out_size >= 0)
  
          {
  
              res.size = outbuf_size;
  
              res.data = malloc(outbuf_size);                
  
          }
  
  
  
          free(outbuf);
  
      }
  
  
  
  
  
      avcodec_close(c);
  
      av_free(c);
  
      return res;
  
  }

  After encoding "out_size" is always 0 and result buffer is empty.

  Thanks.