Recherche avancée

Sur d’autres sites (5405)

• FFMPEG - How to wait until all blobs are written before finishing ffmpeg process when getting them from media recorder API

  7 novembre 2020, par Caio Nakai

  I'm using media recorder API to record a video from user's screen and sending the blobs through web socket to a nodejs server. The nodejs server is using the blobs to create a webm video file, the video is being created fine but with a delay, after the user clicks on the stop recording button it stops the media recorder api, however the server didn't finish the processing of all blobs (at least that's what I think it's happening) and then when I check the video file generated the last few seconds of the recording are missing I wonder if there's an way to solve this. Any help is appreciated :)

  


  This is the front-end code that sends the blobs to the nodejs server

  


    const startScreenCapture = async () => {
  try {
    let screenStream;
    videoElem = document.getElementById("myscreen");
    screenStream = await navigator.mediaDevices.getDisplayMedia(
      displayMediaOptions
    );

    const recorderOptions = {
      mimeType: "video/webm;codecs=vp9",
      videoBitsPerSecond: 3 * 1024 * 1024,
    };

    screenMediaRecorder = new MediaRecorder(screenStream, recorderOptions);
    screenMediaRecorder.start(1); // 1000 - the number of milliseconds to record into each Blob
    screenMediaRecorder.ondataavailable = (event) => {
      console.debug("Got blob data:", event.data);
      console.log("Camera stream: ", event.data);
      if (event.data && event.data.size > 0) {
        socket.emit("screen_stream", event.data);
      }
    };

    videoElem.srcObject = screenStream;
    // console.log("Screen stream", screenStream);
    // socket.emit("screen_stream", screenStream);
  } catch (err) {
    console.error("Error: " + err);
  }
};

const stopCapture = (evt) => {
  let tracks = videoElem.srcObject.getTracks();

  tracks.forEach((track) => track.stop());
  videoElem.srcObject = null;
  screenMediaRecorder.stop();
  socket.emit("stop_screen");
  socket.close();
};


  


  This is the nodejs back-end that handle the blobs and generates the videofile

  


    const ffmpeg2 = child_process.spawn("ffmpeg", [
    "-i",
    "-",
    "-c:v",
    "copy",
    "-c:a",
    "copy",
    "screen.webm",
  ]);


  socket.on("screen_stream", (msg) => {
    console.log("Writing screen blob! ");
    ffmpeg2.stdin.write(msg);
  });

  socket.on("stop_screen", () => {
    console.log("Stop recording..");
  });


  


• ffmpeg rtmp live stream to Azure Media Services

  18 mars 2015, par Pilskalns

  Somehow I got publishing Azure Media Services to work, except there is huge frame drop (almost every frame) with those settings :

  ffmpeg -i "http://stream01/hls/live/playlist-mid.m3u8" -strict -2 -c:a aac -b:a 192k
  
  -ar 44100 -r 25 -g 60 -keyint_min 60 -b:v 2000k -c:v libx264 -bufsize 2000k
  
  -maxrate 2000k -f flv "rtmp://publishurl.working.is/mystream1"

  + I can’t play URL from Azure.

  When I publish to local nginx-rtmp server, frame drop stays and despite it I can play it. So, I believe that frame drop is ffmpeg related not Azure. Processor is not overloaded, available bandwidth is excellent.

  When I try original settings from this URL (at very bottom), I this get additionaly this ffmpeg error Past duration 0.719994 too large

  Here’s more from output :

  c:\ffmpeg\bin>ffmpeg -i "http://stream01/hls/live/playlist-mid.m3u8" -strict -2 -c:a aac -b:a 192k -ar 44100 -r 25 -g 60 -keyint_min 60 -b:v 2000k -c:
  
  v libx264 -bufsize 1000k -maxrate 2000k -f flv "rtmp://********.channel.mediaservices.windows.net:1935/live/9590f0be/mystream1"
  
  ffmpeg version N-69972-g6c91afe Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.9.2 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnu
  
  tls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-
  
  libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-l
  
  ibrtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --
  
  enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --ena
  
  ble-libxvid --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      54. 19.100 / 54. 19.100
  
    libavcodec     56. 25.100 / 56. 25.100
  
    libavformat    56. 23.100 / 56. 23.100
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 11.100 /  5. 11.100
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, hls,applehttp, from 'http://stream01/hls/live/playlist-mid.m3u8':
  
    Duration: N/A, start: 3314.605633, bitrate: N/A
  
    Program 0
  
      Metadata:
  
        variant_bitrate : 1500000
  
      Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv), 854x480 [SAR 1:1 DAR 427:240], 50 fps, 50 tbr, 90k tbn, 100 tbc
  
      Metadata:
  
        variant_bitrate : 1500000
  
      Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 216 kb/s
  
      Metadata:
  
        variant_bitrate : 1500000
  
  [libx264 @ 0000000000306280] using SAR=1/1
  
  [libx264 @ 0000000000306280] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
  
  [libx264 @ 0000000000306280] profile High, level 3.0
  
  [libx264 @ 0000000000306280] 264 - core 144 r2525 40bb568 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options:
  
   cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 dead
  
  zone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_in
  
  tra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=60 keyint_min=31 scenecut=40 intra_refresh=0 rc_lookahea
  
  d=40 rc=cbr mbtree=1 bitrate=2000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=2000 vbv_bufsize=1000 nal_hrd=none filler=0 ip_ratio=1.
  
  40 aq=1:1.00
  
  Output #0, flv, to 'rtmp://*******.channel.mediaservices.windows.net:1935/live/37b4e25/mystream1':
  
    Metadata:
  
      encoder         : Lavf56.23.100
  
      Stream #0:0: Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 854x480 [SAR 1:1 DAR 427:240], q=-1--1, 2000 kb/s, 25 fps, 1k tbn, 25 tbc
  
      Metadata:
  
        variant_bitrate : 1500000
  
        encoder         : Lavc56.25.100 libx264
  
      Stream #0:1: Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        variant_bitrate : 1500000
  
        encoder         : Lavc56.25.100 aac
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  
    Stream #0:1 -> #0:1 (aac (native) -> aac (native))
  
  Press [q] to stop, [?] for help
  
  [flv @ 0000000000305820] Failed to update header with correct duration.1.4kbits/s dup=0 drop=10697
  
  [flv @ 0000000000305820] Failed to update header with correct filesize.
  
  frame=10714 fps= 24 q=-1.0 Lsize=  100583kB time=00:07:11.08 bitrate=1911.4kbits/s dup=0 drop=10701
  
  video:90024kB audio:10042kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.516478%
  
  [libx264 @ 0000000000306280] frame I:190   Avg QP:10.58  size: 72438
  
  [libx264 @ 0000000000306280] frame P:3406  Avg QP:14.32  size: 19296
  
  [libx264 @ 0000000000306280] frame B:7118  Avg QP:19.55  size:  1784
  
  [libx264 @ 0000000000306280] consecutive B-frames:  8.8%  4.4% 10.5% 76.3%
  
  [libx264 @ 0000000000306280] mb I  I16..4: 23.0% 20.2% 56.8%
  
  [libx264 @ 0000000000306280] mb P  I16..4:  2.7%  4.5%  5.2%  P16..4: 41.9% 11.6%  8.7%  0.0%  0.0%    skip:25.4%
  
  [libx264 @ 0000000000306280] mb B  I16..4:  0.1%  0.1%  0.2%  B16..8: 26.5%  2.2%  0.5%  direct: 3.2%  skip:67.1%  L0:51.0% L1:38.2% BI:10.8%
  
  [libx264 @ 0000000000306280] 8x8 transform intra:31.3% inter:36.6%
  
  [libx264 @ 0000000000306280] coded y,uvDC,uvAC intra: 80.5% 73.5% 51.1% inter: 16.1% 18.8% 5.8%
  
  [libx264 @ 0000000000306280] i16 v,h,dc,p: 21% 29% 27% 23%
  
  [libx264 @ 0000000000306280] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 32% 19%  4%  4%  4%  5%  4%  7%
  
  [libx264 @ 0000000000306280] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 27% 30% 15%  4%  5%  5%  5%  4%  5%
  
  [libx264 @ 0000000000306280] i8c dc,h,v,p: 38% 30% 24%  8%
  
  [libx264 @ 0000000000306280] Weighted P-Frames: Y:1.8% UV:0.4%
  
  [libx264 @ 0000000000306280] ref P L0: 78.0%  8.5% 10.1%  3.3%  0.0%
  
  [libx264 @ 0000000000306280] ref B L0: 91.3%  7.9%  0.8%
  
  [libx264 @ 0000000000306280] ref B L1: 95.4%  4.6%
  
  [libx264 @ 0000000000306280] kb/s:1716.97
  
  Received signal 2: terminating.

• python how to combine video file and media file without ffmpeg [on hold]

  27 juillet 2017, par daniel yanovsky

  Hello I am working on a program that is recording the screen. And I like to add an audio for the video.
  How can I do it without using ffmpeg.
  I want my program to do so.