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• FFmpeg - MJPEG decoding - getting different values

  27 décembre 2016, par ahmadh

  I have a set of JPEG frames which I am muxing into an avi, which gives me a mjpeg video. This is the command I run on the console :

  ffmpeg -y -start_number 0 -i %06d.JPEG -codec copy vid.avi

  When I try to demux the video using ffmpeg C api, I get frames which are slightly different in values. Demuxing code looks something like this :

  AVFormatContext* fmt_ctx = NULL;
  
  AVCodecContext* cdc_ctx = NULL;
  
  AVCodec* vid_cdc = NULL;
  
  int ret;
  
  unsigned int height, width;
  
  
  
  ....
  
  // read_nframes is the number of frames to read
  
  output_arr = new unsigned char [height * width * 3 * 
  
                                  sizeof(unsigned char) * read_nframes];
  
  
  
  avcodec_open2(cdc_ctx, vid_cdc, NULL);
  
  
  
  int num_bytes;
  
  uint8_t* buffer = NULL;
  
  const AVPixelFormat out_format = AV_PIX_FMT_RGB24;
  
  
  
  num_bytes = av_image_get_buffer_size(out_format, width, height, 1);
  
  buffer = (uint8_t*)av_malloc(num_bytes * sizeof(uint8_t));
  
  
  
  AVFrame* vid_frame = NULL;
  
  vid_frame = av_frame_alloc();
  
  AVFrame* conv_frame = NULL;
  
  conv_frame = av_frame_alloc();
  
  
  
  av_image_fill_arrays(conv_frame->data, conv_frame->linesize, buffer, 
  
                       out_format, width, height, 1);
  
  
  
  struct SwsContext *sws_ctx = NULL;
  
  sws_ctx = sws_getContext(width, height, cdc_ctx->pix_fmt,
  
                           width, height, out_format,
  
                           SWS_BILINEAR, NULL,NULL,NULL);
  
  
  
  int frame_num = 0;
  
  AVPacket vid_pckt;
  
  while (av_read_frame(fmt_ctx, &vid_pckt) >=0) {
  
      ret = avcodec_send_packet(cdc_ctx, &vid_pckt);
  
      if (ret < 0)
  
          break;
  
  
  
      ret = avcodec_receive_frame(cdc_ctx, vid_frame);
  
      if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF)
  
          break;
  
      if (ret >= 0) {
  
          // convert image from native format to planar GBR
  
          sws_scale(sws_ctx, vid_frame->data, 
  
                    vid_frame->linesize, 0, vid_frame->height, 
  
                    conv_frame->data, conv_frame->linesize);
  
  
  
          unsigned char* r_ptr = output_arr + 
  
              (height * width * sizeof(unsigned char) * 3 * frame_num);
  
          unsigned char* g_ptr = r_ptr + (height * width * sizeof(unsigned char));
  
          unsigned char* b_ptr = g_ptr + (height * width * sizeof(unsigned char));
  
          unsigned int pxl_i = 0;
  
  
  
          for (unsigned int r = 0; r < height; ++r) {
  
              uint8_t* avframe_r = conv_frame->data[0] + r*conv_frame->linesize[0];
  
              for (unsigned int c = 0; c < width; ++c) {
  
                  r_ptr[pxl_i] = avframe_r[0];
  
                  g_ptr[pxl_i]   = avframe_r[1];
  
                  b_ptr[pxl_i]   = avframe_r[2];
  
                  avframe_r += 3;
  
                  ++pxl_i;
  
              }
  
          }
  
  
  
          ++frame_num;
  
  
  
          if (frame_num >= read_nframes)
  
              break;
  
      }
  
  }
  
  
  
  ...

  In my experience around two-thirds of the pixel values are different, each by +-1 (in a range of [0,255]). I am wondering is it due to some decoding scheme FFmpeg uses for reading JPEG frames ? I tried encoding and decoding png frames, and it works perfectly fine.

  In short my goal is to get the same pixel by pixel values for each JPEG frame as I would I have gotten if I was reading the JPEG images directly. Here is the stand-alone code I used. It includes cmake files to build code, and a couple of jpeg frames with the converted avi file to test this problem. (give —filetype png to test the png decoding).

• Revision d9b62160a0 : Implements several heuristics to prune mode search Skips mode searches for intr

  3 juillet 2013, par Deb Mukherjee

  Changed Paths :
   Modify /vp9/encoder/vp9_encodeframe.c

  
   Modify /vp9/encoder/vp9_onyx_if.c

  
   Modify /vp9/encoder/vp9_onyx_int.h

  
   Modify /vp9/encoder/vp9_rdopt.c

  
  
  Implements several heuristics to prune mode search

  Skips mode searches for intra and compound inter modes depending
  on the best mode so far and the reference frames. The various
  heuristics to be used are selected by bits from a flag. The
  previous direction based intra mode search pruning is also absorbed
  in this framework.

  Specifically the flags and their impact are :

  1) FLAG_SKIP_INTRA_BESTINTER (skip intra mode search for oblique
  directional modes and TM_PRED if the best so far is
  an inter mode)
  derfraw300 : -0.15%, 10% speedup

  2) FLAG_SKIP_INTRA_DIRMISMATCH (skip D27, D63, D117 and D153
  mode search if the best so far is not one of the closest
  hor/vert/diagonal directions.
  derfraw300 : -0.05%, about 9% speedup

  3) FLAG_SKIP_COMP_BESTINTRA (skip compound prediction mode
  search if the best so far is an intra mode)
  derfraw300 : -0.06%, about 7-8% speedup

  4) FLAG_SKIP_COMP_REFMISMATCH (skip compound prediction search
  if the best single ref inter mode does not have the same ref
  as one of the two references being tested in the compound mode)
  derfraw300 : -0.56%, about 10% speedup

  Change-Id : I1a736cd29b36325489e7af9f32698d6394b2c495

• when audioqueue play lpcm decoded from ffmpeg, the elapsed time of audio queue exceeds the duraion of the media

  15 août 2012, par zhzhy

  When play the lpcm data decoded from ffmpeg with audioqueue, the elapsed time got by AudioQueueGetCurrentTime exceeds the duration of media. But when decode the same media with AVFoundation framework, the elapsed time equals duration of the media, and so when read media by ffmpeg with no decoded, then send the compressed media data to audioqueue, the elapsed time also equals duration of the media. The AudioStreamBasicDescription set as following :

      asbd.mSampleRate = 44100;
  
      asbd.mFormatID = kAudioFormatLinearPCM;
  
      asbd.mFormatFlags = kAudioFormatFlagsCanonical;
  
      asbd.mBytesPerPacket = 4;
  
      asbd.mFramesPerPacket = 1;
  
      asbd.mBytesPerFrame = 4;
  
      asbd.mChannelsPerFrame = 2;
  
      asbd.mBitsPerChannel = 16;
  
      asbd.mReserved = 0;

  When playing with data decoded from AVFoundation, the setting of AudioStreamBasicDescription is the same as above. By my test found that AudioTimeStamp.mSampleTime got by AudioQueueGetCurrentTime is different between ffmpeg and AVFoundation, the value of ffmpeg is greater than AVFoundation. So I want to know how this happen, and how to fix it ?
  Thanks !