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• how to decode wmp3 video using libavcodec ? (ffmpeg)

  13 mars 2012, par Renan Elias

  I'm developing an application that reads a live tv stream from the internet and I need to play it on my ipad application.

  I've compiled the ffmpeg into my ipad application to use the libavcodec lib, but I've not been able to use it in this lib...

  I know how to get the stream packets, read if it is an audio or video packet, but I don't know how to use the lib to convert the original packet codecs to h264 and mp3 codec...

  I need to convert an stream packet wmv3 to h264 and save it on a file.

  My code is below...

  AVFormatContext* pFormatCtx = avformat_alloc_context();
  
  int             i, videoStream, audioStream;
  
  AVCodecContext  *pCodecCtx;
  
  AVCodecContext *aCodecCtx;
  
  AVCodec         *pCodec;
  
  AVCodec         *aCodec;
  
  AVFrame         *pFrame; 
  
  AVFrame         *pFrameRGB;
  
  AVPacket        packet;
  
  int             frameFinished;
  
  int             numBytes;
  
  uint8_t         *buffer;
  
  
  
  static struct SwsContext *img_convert_ctx;
  
  
  
  
  
  // Register all formats and codecs
  
  av_register_all();
  
  avcodec_register_all();
  
  avformat_network_init();
  
  
  
  // Open video file
  
  if(avformat_open_input(&pFormatCtx, [objURL cStringUsingEncoding:NSASCIIStringEncoding] ,NULL,NULL) != 0){
  
      return -1;
  
  }
  
  
  
  // Retrieve stream information
  
  if(avformat_find_stream_info(pFormatCtx, NULL)<0)
  
      return -1; // Couldn't find stream information
  
  
  
  // Dump information about file onto standard error
  
  av_dump_format(pFormatCtx, 0, [objURL cStringUsingEncoding:NSASCIIStringEncoding], 0);
  
  
  
  // Find the first video stream
  
  videoStream=-1;
  
  audioStream=-1;
  
  for(i=0; i < pFormatCtx->nb_streams; i++) {
  
      if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_VIDEO && videoStream < 0) {
  
          videoStream=i;
  
      }
  
      if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO && audioStream < 0) {
  
          audioStream=i;
  
      }
  
  }
  
  if(videoStream==-1)
  
      return -1; // Didn't find a video stream
  
  if(audioStream==-1)
  
      return -1;
  
  
  
  // Get a pointer to the codec context for the video stream
  
  pCodecCtx=pFormatCtx->streams[videoStream]->codec;
  
  
  
  // Find the decoder for the video stream
  
  pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
  
  if(pCodec==NULL) {
  
      fprintf(stderr, "Unsupported codec!\n");
  
      return -1; // Codec not found
  
  }
  
  
  
  // Open codec
  
  if(avcodec_open2(pCodecCtx, pCodec, NULL)<0)
  
      return -1; // Could not open codec
  
  
  
  // Get a pointer to the codec context for the audio stream
  
  aCodecCtx=pFormatCtx->streams[audioStream]->codec;
  
  
  
  // Find the decoder for the audio stream
  
  aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
  
  if(!aCodec) {
  
      fprintf(stderr, "Unsupported codec!\n");
  
      return -1; // Codec not found
  
  }
  
  
  
  // Open codec
  
  if(avcodec_open2(aCodecCtx, aCodec, NULL)<0)
  
      return -1; // Could not open codec
  
  
  
  // Allocate video frame
  
  pFrame=avcodec_alloc_frame();
  
  
  
  // Allocate an AVFrame structure
  
  pFrameRGB=avcodec_alloc_frame();
  
  if(pFrameRGB==NULL)
  
      return -1;
  
  
  
  // Determine required buffer size and allocate buffer
  
  numBytes=avpicture_get_size(PIX_FMT_RGB24, pCodecCtx->width,
  
                              pCodecCtx->height);
  
  buffer=(uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
  
  
  
  // Assign appropriate parts of buffer to image planes in pFrameRGB
  
  // Note that pFrameRGB is an AVFrame, but AVFrame is a superset
  
  // of AVPicture
  
  avpicture_fill((AVPicture *)pFrameRGB, buffer, PIX_FMT_RGB24,
  
                 pCodecCtx->width, pCodecCtx->height);
  
  
  
  // Read frames and save first five frames to disk
  
  i=0;
  
  while(av_read_frame(pFormatCtx, &packet)>=0) {
  
      // Is this a packet from the video stream?
  
      if(packet.stream_index==audioStream) {
  
          NSLog(@"Audio.. i'll solve the video first...");
  
      } else if(packet.stream_index==videoStream) {
  
  
  
          /// HOW CONVERT THIS VIDEO PACKET TO H264 and save on a file? :(
  
      }
  
  
  
      // Free the packet that was allocated by av_read_frame
  
      av_free_packet(&packet);
  
  }
  
  
  
  // Free the RGB image
  
  av_free(buffer);
  
  av_free(pFrameRGB);
  
  
  
  // Free the YUV frame
  
  av_free(pFrame);
  
  
  
  // Close the codec
  
  avcodec_close(pCodecCtx);
  
  
  
  // Close the video file
  
  avformat_close_input(&pFormatCtx);
  
  
  
  return 0;

• ffmpeg - Incompatible sample format '(null)'

  17 juin 2012, par abrahab

  I convert videos from different sources with ffmpeg to mp4 with libx264 codec. After conversation time to time I got incorrect format error when trying to load already converted video with ffmpeg : Incompatible sample format '(null)' for codec 'aac', auto-selecting format 's16'

  Seems, it is audio stream problem with aac codec ? How to fix or how to convert videos to mp4 to be sure that I will not get the following error ? Maybe somehow need to specify audio format at the ffmpeg convert command ?

  ps. Video must work at Iphone/Ipad.
  pps. the main problem, that such videos with the following error can not be pseudo-streamed with nginx (got 500 Error)
  ppps. and sorry for my bad english, please, correct my text if need. thanks.

  Some additional info from file :

    Duration: 00:00:10.30, start: 0.000000, bitrate: 614 kb/s
  
      Stream #0.0(rus): Video: h264 (High), yuv420p, 1024x576, 567 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00
  
      Stream #0.1(rus): Audio: aac, 22050 Hz, stereo, s16, 57 kb/s
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00

• avcodec_decode_audio4 decoded data different on device and simulator

  5 décembre 2013, par Jona

  I'm having issues where the decoded audio sounds punchy/garbled simply can't tell what is being heard when decoded on an iPhone/iPad device. However, when decoding the audio on simulator the audio is perfect.

  I do realize emulator runs on a different CPU architecture i386 while actual device is ARMv7 so the issue could be something around that.

  I have printed the compressed data on device and simulator before being decoded and it matches. Then I printed the decoded data for device and simulator and they are different. That leads me to think something is happening with the decoder. I can see that it is not a byte swap issue since the data is just completely different.

  Any ideas on what I could or look into ? I don't have this issue on Android...

  First decoded audio frame.

  DEVICE :

  0x00, 0x00, 0xff, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x70, 0x08, 0x00, 0x00, 0x00, 0x00, 0x03, 0x00, 0xda, 0x0a, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,

  EMULATOR :

  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,

  ---

  UPDATE 1
  I was able to resolve the issue by disabling asm with --disable-asm. Something must not be working correctly with the ARM asm files.