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• lavu/frame : add QP side data

  1er mars 2018, par wm4

  lavu/frame : add QP side data
  This adds a way for an API user to transfer QP data and metadata without
  
  having to keep the reference to AVFrame, and without having to
  
  explicitly care about QP APIs. It might also provide a way to finally
  
  remove the deprecated QP related fields. In the end, the QP table should
  
  be handled in a very similar way to e.g. AV_FRAME_DATA_MOTION_VECTORS.
  There are two side data types, because I didn't care about having to
  
  repack the QP data so the table and the metadata are in a single
  
  AVBufferRef. Otherwise it would have either required a copy on decoding
  
  (extra slowdown for something as obscure as the QP data), or would have
  
  required making intrusive changes to the codecs which support export of
  
  this data.
  The new side data types are added under deprecation guards, because I
  
  don't intend to change the status of the QP export as being deprecated
  
  (as it was before this patch too).
  

  • [DH] doc/APIchanges
  • [DH] libavutil/frame.c
  • [DH] libavutil/frame.h
  • [DH] libavutil/version.h
  • [DH] tests/ref/fate/exif-image-embedded
  • [DH] tests/ref/fate/exif-image-jpg

• pipe:0 : could not find codec parameters

  10 avril 2018, par Harish

  I am trying to store streaming video as a segments of 15 min,it is working fine with gst-launch-1.0 my command is

  gst-launch-1.0 -e --gst-debug-level=2 v4l2src device=/dev/video0 ! vspmfilter outbuf-alloc=true ! \
  
  video/x-raw,width=640,height=480,framerate=15/2 ! omxh264enc target-bitrate=10485760 num-p-frames=0 ! \
  
  mpegtsmux name="muxer" ! fdsink fd=1 \
  
  alsasrc device=hw:0,0 latency-time=30000 buffer-time=800000 ! audioconvert ! audio/x-raw,format=S16LE,channel=2,rate=44100 ! \
  
  faac bitrate=128000 ! muxer. \
  
  | /usr/bin/ffmpeg -v info -i pipe:0 -map 0 -c copy -f segment \
  
  -reset_timestamps 1 -segment_atclocktime 1 -segment_time 900 \
  
  -segment_list_type csv -segment_list_size 0 \
  
  -segment_list /home/root/ContinuousCapture/CamFiles.txt
  
  -segment_list_entry_prefix /home/root/ContinuousCapture/ \
  
  -segment_format mpegts -strftime 1 /home/root/ContinuousCapture/video_%g%m%d_%H%M%S.mpegts

  I changed same command to gstd,my command is

  gstd-client pipeline_create p v4l2src device=/dev/video0 ! ${textoverlay} text=${dev0} ! ${clockoverlay} ! vspmfilter outbuf-alloc=true ! \
  
  video/x-raw,width=640,height=480,framerate=15/2 ! omxh264enc target-bitrate=10485760 num-p-frames=0 \
  
  mpegtsmux name="muxer" ! fdsink fd=1 \
  
  alsasrc device=hw:0,0 latency-time=30000 buffer-time=800000 ! audioconvert ! audio/x-raw,format=S16LE,channel=2,rate=44100 ! \
  
  faac bitrate=128000 ! muxer. \ 
  
  gstd-client pipeline_play p | /usr/bin/ffmpeg  -v info -i pipe:0 -map 0 -c copy -f segment \
  
  -reset_timestamps 1 -segment_atclocktime 1 -segment_time 900 \
  
  -segment_list_type csv -segment_list_size 0 \
  
  -segment_list /home/root/ContinuousCapture/CamFiles.txt -segment_list_entry_prefix /home/root/ContinuousCapture/ \
  
  -segment_format mpegts -strftime 1 /home/root/ContinuousCapture/video_%g%m%d_%H%M%S.mpegts

  its giving me the following error

  libdir=/usr/lib --shlibdir=/usr/lib --datadir=/usr/share/ffmpeg --disable-mipsdsp --disable-mipsdspr2 --cpu=generic --pkg-config=pkg-config --enable-avcodec --enable-avdevice --enable-avfilter --enable-avformat --disable-avresample --enable-bzlib --enable-gpl --disable-libgsm --disable-indev=jack --disable-libvorbis --disable-libmp3lame --disable-openssl --enable-postproc --disable-libschroedinger --disable-sdl2 --disable-libspeex --enable-swresample --enable-swscale --enable-libtheora --disable-vaapi --disable-vdpau --disable-libvpx --enable-libx264 --enable-outdev=xv
  
  libavutil      55. 58.100 / 55. 58.100
  
  libavcodec     57. 89.100 / 57. 89.100
  
  libavformat    57. 71.100 / 57. 71.100
  
  libavdevice    57.  6.100 / 57.  6.100
  
  libavfilter     6. 82.100 /  6. 82.100
  
  libswscale      4.  6.100 /  4.  6.100
  
  libswresample   2.  7.100 /  2.  7.100
  
  libpostproc    54.  5.100 / 54.  5.100
  
  [mpegts @ 0x49450] Format mpegts detected only with low score of 2, misdetection possible!
  
  [mpegts @ 0x49450] Could not detect TS packet size, defaulting to non-FEC/DVHS
  
  pipe:0: could not find codec parameters

  Thank you

• Live555 truncates encoded data of FFMpeg

  22 novembre 2019, par Harshil Makwana

  I am trying to stream H264 based data using Live555 over RTSP.

  I am capturing data using V4L2, and then encodes it using FFMPEG and then passing data to Live555’s DeviceSource file, in that I using H264VideoStreamFramer class,

  Below is my codec settings to configure AVCodecContext of encoder,

  codec = avcodec_find_encoder_by_name(CODEC_NAME);
  
  if (!codec) {
  
      cerr << "Codec " << codec_name << " not found\n";
  
      exit(1);
  
  }
  
  
  
  c = avcodec_alloc_context3(codec);
  
  if (!c) {
  
      cerr << "Could not allocate video codec context\n";
  
      exit(1);
  
  }
  
  
  
  pkt = av_packet_alloc();
  
  if (!pkt)
  
      exit(1);
  
  
  
  /* put sample parameters */
  
  c->bit_rate = 400000;
  
  /* resolution must be a multiple of two */
  
  c->width = PIC_HEIGHT;
  
  c->height = PIC_WIDTH;
  
  /* frames per second */
  
  c->time_base = (AVRational){1, FPS};
  
  c->framerate = (AVRational){FPS, 1};
  
  c->gop_size = 10;
  
  c->max_b_frames = 1;
  
  c->pix_fmt = AV_PIX_FMT_YUV420P;
  
  c->rtp_payload_size = 30000;
  
  if (codec->id == AV_CODEC_ID_H264)
  
      av_opt_set(c->priv_data, "preset", "fast", 0);
  
  av_opt_set_int(c->priv_data, "slice-max-size", 30000, 0);
  
  /* open it */
  
  ret = avcodec_open2(c, codec, NULL);
  
  if (ret < 0) {
  
      cerr << "Could not open codec\n";
  
      exit(1);
  
  }

  And I am getting encoded data using avcodec_receive_packet() function. which will return AVPacket.

  And I am passing AVPacket’s data into DeviceSource file below is code snippet of my Live555 code :

  void DeviceSource::deliverFrame() {
  
      if (!isCurrentlyAwaitingData()) return; // we're not ready for the data yet
  
  
  
      u_int8_t* newFrameDataStart = (u_int8_t*) pkt->data;
  
      unsigned newFrameSize = pkt->size; //%%% TO BE WRITTEN %%%
  
      // Deliver the data here:
  
      if (newFrameSize > fMaxSize) { // Condition becomes true many times
  
          fFrameSize = fMaxSize;
  
          fNumTruncatedBytes = newFrameSize - fMaxSize;
  
      } else {
  
          fFrameSize = newFrameSize;
  
      }
  
      gettimeofday(&fPresentationTime, NULL); // If you have a more accurate time - e.g., from an encoder - then use that instead.
  
      // If the device is *not* a 'live source' (e.g., it comes instead from a file or buffer), then set "fDurationInMicroseconds" here.
  
      memmove(fTo, newFrameDataStart, fFrameSize);
  
  }

  But here, sometimes my packet’s size is getting more than fMaxSize value and as per LIVE555 logic it will truncate frame data, so that sometimes I am getting bad frames on my VLC,

  From Live555 forum, I get to know that encoder should not send packet whose size is more than fMaxSize value, so my question is :

  How to restrict encoder to limit size of packet ?

  Thanks in Advance,

  Harshil