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• What could be the cause of these http-livestream artefacts in google chrome ?

  21 avril 2021, par NGauthier

  Here is the http-livestream setup : The server is running ffmpeg with the DASH protocol and h264 encoding. The client is using Dash.js. Resolution is fixed to 1920x1080, with 24 bit depth, and 60hz.

  


  The artefacting (image below) is only present when the last row of the video is within chrome viewport (so it disapears if the page is scrolled up). It manifests itself as stretching of the center row of pixels downwards, and appears to only affect some color channels.

  


  I have attempted changing the bitrate, and cutting the last row from the source, thinking the issue could be on the server side, without any impact. The fact that the issue depends on the position in the viewport makes me suspect a glitch in chrome itself.

  


  I have also attempted to force hardware decoding off in chrome :\flags and it does not solve the issue.

  


  Please submit your hypothesis on what could be the cause of this issue. Thanks.

  


  

  


  ---

  


  Update #1

  


  Here is the ffmpeg command line and logs :

  


  export DISPLAY=:0 && ffmpeg -f x11grab -framerate 60 -video_size 1920x1080 -i :0.0+0,0 -draw_mouse 0 -f dash -utc_timing_url https://time.akamai.com/?iso -streaming 1 -seg_duration 2 -frag_duration 0.033 -fflags nobuffer -fflags flush_packets -c:v h264 -preset ultrafast data/stream.mpd


  


  And the logs :

  


  ffmpeg version 4.2.4-1ubuntu0.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
  configuration: --prefix=/usr --extra-version=1ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[x11grab @ 0x561ca34b9980] Stream #0: not enough frames to estimate rate; consider increasing probesize
Input #0, x11grab, from ':0.0+0,0':
  Duration: N/A, start: 1618941693.853256, bitrate: N/A
    Stream #0:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1920x1080, 60 fps, 1000k tbr, 1000k tbn, 1000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
Press [q] to stop, [?] for help
[libx264 @ 0x561ca34c5300] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2 AVX512
[libx264 @ 0x561ca34c5300] profile High 4:4:4 Predictive, level 4.2, 4:4:4 8-bit
[libx264 @ 0x561ca34c5300] 264 - core 155 r2917 0a84d98 - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=6 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
[dash @ 0x561ca34c3740] No bit rate set for stream 0
[dash @ 0x561ca34c3740] Opening 'data/init-stream0.m4s' for writing
Output #0, dash, to 'data/stream.mpd':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #0:0: Video: h264 (libx264), yuv444p, 1920x1080, q=-1--1, 60 fps, 15360 tbn, 60 tbc
    Metadata:
      encoder         : Lavc58.54.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
[dash @ 0x561ca34c3740] Opening 'data/chunk-stream0-00001.m4s.tmp' for writing
frame=   34 fps=0.0 q=15.0 size=N/A time=00:00:00.43 bitrate=N/A dup=5 drop=0 speed=0.836x    
frame=   65 fps= 64 q=15.0 size=N/A time=00:00:00.95 bitrate=N/A dup=5 drop=0 speed=0.929x    
frame=   96 fps= 62 q=15.0 size=N/A time=00:00:01.46 bitrate=N/A dup=5 drop=2 speed=0.955x    
frame=  126 fps= 62 q=15.0 size=N/A time=00:00:01.96 bitrate=N/A dup=5 drop=3 speed=0.962x    
frame=  157 fps= 62 q=15.0 size=N/A time=00:00:02.48 bitrate=N/A dup=5 drop=3 speed=0.973x    
frame=  188 fps= 61 q=15.0 size=N/A time=00:00:03.00 bitrate=N/A dup=5 drop=3 speed=0.98x    
frame=  217 fps= 61 q=15.0 size=N/A time=00:00:03.48 bitrate=N/A dup=5 drop=3 speed=0.977x    
frame=  247 fps= 61 q=15.0 size=N/A time=00:00:03.98 bitrate=N/A dup=6 drop=3 speed=0.976x    
[dash @ 0x561ca34c3740] Opening 'data/stream.mpd.tmp' for writing
[dash @ 0x561ca34c3740] Opening 'data/chunk-stream0-00002.m4s.tmp' for writing
frame=  279 fps= 61 q=15.0 size=N/A t


  


• Google Speech API + Go - Transcribing Audio Stream of Unknown Length

  14 février 2018, par Josh

  I have an rtmp stream of a video call and I want to transcribe it. I have created 2 services in Go and I’m getting results but it’s not very accurate and a lot of data seems to get lost.

  Let me explain.

  I have a transcode service, I use ffmpeg to transcode the video to Linear16 audio and place the output bytes onto a PubSub queue for a transcribe service to handle. Obviously there is a limit to the size of the PubSub message, and I want to start transcribing before the end of the video call. So, I chunk the transcoded data into 3 second clips (not fixed length, just seems about right) and put them onto the queue.

  The data is transcoded quite simply :

  var stdout Buffer
  
  
  
  cmd := exec.Command("ffmpeg", "-i", url, "-f", "s16le", "-acodec", "pcm_s16le", "-ar", "16000", "-ac", "1", "-")
  
  cmd.Stdout = &stdout
  
  
  
  if err := cmd.Start(); err != nil {
  
      log.Fatal(err)
  
  }
  
  
  
  ticker := time.NewTicker(3 * time.Second)
  
  
  
  for {
  
      select {
  
      case <-ticker.C:
  
          bytesConverted := stdout.Len()
  
          log.Infof("Converted %d bytes", bytesConverted)
  
  
  
          // Send the data we converted, even if there are no bytes.
  
          topic.Publish(ctx, &pubsub.Message{
  
              Data: stdout.Bytes(),
  
          })
  
  
  
          stdout.Reset()
  
      }
  
  }

  The transcribe service pulls messages from the queue at a rate of 1 every 3 seconds, helping to process the audio data at about the same rate as it’s being created. There are limits on the Speech API stream, it can’t be longer than 60 seconds so I stop the old stream and start a new one every 30 seconds so we never hit the limit, no matter how long the video call lasts for.

  This is how I’m transcribing it :

  stream := prepareNewStream()
  
  clipLengthTicker := time.NewTicker(30 * time.Second)
  
  chunkLengthTicker := time.NewTicker(3 * time.Second)
  
  
  
  cctx, cancel := context.WithCancel(context.TODO())
  
  err := subscription.Receive(cctx, func(ctx context.Context, msg *pubsub.Message) {
  
  
  
      select {
  
      case <-clipLengthTicker.C:
  
          log.Infof("Clip length reached.")
  
          log.Infof("Closing stream and starting over")
  
  
  
          err := stream.CloseSend()
  
          if err != nil {
  
              log.Fatalf("Could not close stream: %v", err)
  
          }
  
  
  
          go getResult(stream)
  
          stream = prepareNewStream()
  
  
  
      case <-chunkLengthTicker.C:
  
          log.Infof("Chunk length reached.")
  
  
  
          bytesConverted := len(msg.Data)
  
  
  
          log.Infof("Received %d bytes\n", bytesConverted)
  
  
  
          if bytesConverted > 0 {
  
              if err := stream.Send(&speechpb.StreamingRecognizeRequest{
  
                  StreamingRequest: &speechpb.StreamingRecognizeRequest_AudioContent{
  
                      AudioContent: transcodedChunk.Data,
  
                  },
  
              }); err != nil {
  
                  resp, _ := stream.Recv()
  
                  log.Errorf("Could not send audio: %v", resp.GetError())
  
              }
  
          }
  
  
  
          msg.Ack()
  
      }
  
  })

  I think the problem is that my 3 second chunks don’t necessarily line up with starts and end of phrases or sentences so I suspect that the Speech API is a recurrent neural network which has been trained on full sentences rather than individual words. So starting a clip in the middle of a sentence loses some data because it can’t figure out the first few words up to the natural end of a phrase. Also, I lose some data in changing from an old stream to a new stream. There’s some context lost. I guess overlapping clips might help with this.

  I have a couple of questions :

  1) Does this architecture seem appropriate for my constraints (unknown length of audio stream, etc.) ?

  2) What can I do to improve accuracy and minimise lost data ?

  (Note I’ve simplified the examples for readability. Point out if anything doesn’t make sense because I’ve been heavy handed in cutting the examples down.)

• ffmpeg with Axis P1347 returns 400 Bad Request, but Axis 1357 works

  3 mai 2016, par steampowered

  I have two cameras : an Axis P1347 and an Axis P1357. ffmpeg gets a 400 Bad Request on the P1347 but everything works fine with the P1357.

  I am able to successfully stream rtsp video using vlc from an Axis P1347 Camera using the following url :

  rtsp://10.8.3.85:554/axis-media/media.amp?videocodec=h264&audio=1 

  However, this same url in ffmpeg gives the following for the Axis P1347 Camera :

  root@ubuntu4-virtual-machine:/home/ubuntu4# ffmpeg -re -v verbose  -i "rtsp://10.8.3.85:554/axis-media/media.amp?videocodec=h264&audio=1"
  
  ffmpeg version git-2016-05-02-9fcb59c Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
  
    configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3
  
    libavutil      55. 23.100 / 55. 23.100
  
    libavcodec     57. 38.100 / 57. 38.100
  
    libavformat    57. 35.100 / 57. 35.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 44.100 /  6. 44.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  rtsp://10.8.3.85:554/axis-media/media.amp?videocodec=h264&audio=1: Server returned 400 Bad Request

  The same ffmpeg command works great with the nearly identical Axis P1357 Camera :

  root@ubuntu4-virtual-machine:/home/ubuntu4# ffmpeg -re -rtsp_transport tcp -i "rtsp://10.8.3.90:554/axis-media/media.amp?videocodec=h264&audio=1"
  
  ffmpeg version git-2016-05-02-9fcb59c Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
  
    configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-version3
  
    libavutil      55. 23.100 / 55. 23.100
  
    libavcodec     57. 38.100 / 57. 38.100
  
    libavformat    57. 35.100 / 57. 35.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 44.100 /  6. 44.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, rtsp, from 'rtsp://10.8.3.90:554/axis-media/media.amp?videocodec=h264&audio=1':
  
    Metadata:
  
      title           : Media Presentation
  
    Duration: N/A, start: 0.083300, bitrate: N/A
  
      Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709), 2592x1944 [SAR 1:1 DAR 4:3], 12 tbr, 90k tbn
  
      Stream #0:1: Audio: aac (LC), 16000 Hz, mono, fltp

  Obviously rtsp is turned on and working if vlc can display video, correct ? So why does VLC work with the camera, but not ffmpeg ? Note ffmpeg is installed and works correctly with a similar camera.