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• Play video using mse (media source extension) in google chrome

  23 août 2019, par liyuqihxc

  I’m working on a project that convert rtsp stream (ffmpeg) and play it on the web page (signalr + mse).

  So far it works pretty much as I expected on the latest version of edge and firefox, but not chrome.

  here’s the code

  public class WebmMediaStreamContext
  
  {
  
      private Process _ffProcess;
  
      private readonly string _cmd;
  
      private byte[] _initSegment;
  
      private Task _readMediaStreamTask;
  
      private CancellationTokenSource _cancellationTokenSource;
  
  
  
      private const string _CmdTemplate = "-i {0} -c:v libvpx -tile-columns 4 -frame-parallel 1 -keyint_min 90 -g 90 -f webm -dash 1 pipe:";
  
  
  
      public static readonly byte[] ClusterStart = { 0x1F, 0x43, 0xB6, 0x75, 0x01, 0x00, 0x00, 0x00 };
  
  
  
      public event EventHandler<clusterreadyeventargs> ClusterReadyEvent;
  
  
  
      public WebmMediaStreamContext(string rtspFeed)
  
      {
  
          _cmd = string.Format(_CmdTemplate, rtspFeed);
  
      }
  
  
  
      public async Task StartConverting()
  
      {
  
          if (_ffProcess != null)
  
              throw new InvalidOperationException();
  
  
  
          _ffProcess = new Process();
  
          _ffProcess.StartInfo = new ProcessStartInfo
  
          {
  
              FileName = "ffmpeg/ffmpeg.exe",
  
              Arguments = _cmd,
  
              UseShellExecute = false,
  
              CreateNoWindow = true,
  
              RedirectStandardOutput = true
  
          };
  
          _ffProcess.Start();
  
  
  
          _initSegment = await ParseInitSegmentAndStartReadMediaStream();
  
      }
  
  
  
      public byte[] GetInitSegment()
  
      {
  
          return _initSegment;
  
      }
  
  
  
      // Find the first cluster， and everything before it is the InitSegment
  
      private async Task ParseInitSegmentAndStartReadMediaStream()
  
      {
  
          Memory<byte> buffer = new byte[10 * 1024];
  
          int length = 0;
  
          while (length != buffer.Length)
  
          {
  
              length += await _ffProcess.StandardOutput.BaseStream.ReadAsync(buffer.Slice(length));
  
              int cluster = buffer.Span.IndexOf(ClusterStart);
  
              if (cluster >= 0)
  
              {
  
                  _cancellationTokenSource = new CancellationTokenSource();
  
                  _readMediaStreamTask = new Task(() => ReadMediaStreamProc(buffer.Slice(cluster, length - cluster).ToArray(), _cancellationTokenSource.Token), _cancellationTokenSource.Token, TaskCreationOptions.LongRunning);
  
                  _readMediaStreamTask.Start();
  
                  return buffer.Slice(0, cluster).ToArray();
  
              }
  
          }
  
  
  
          throw new InvalidOperationException();
  
      }
  
  
  
      private void ReadMoreBytes(Span<byte> buffer)
  
      {
  
          int size = buffer.Length;
  
          while (size > 0)
  
          {
  
              int len = _ffProcess.StandardOutput.BaseStream.Read(buffer.Slice(buffer.Length - size));
  
              size -= len;
  
          }
  
      }
  
  
  
      // Parse every single cluster and fire ClusterReadyEvent
  
      private void ReadMediaStreamProc(byte[] bytesRead, CancellationToken cancel)
  
      {
  
          Span<byte> buffer = new byte[5 * 1024 * 1024];
  
          bytesRead.CopyTo(buffer);
  
          int bufferEmptyIndex = bytesRead.Length;
  
  
  
          do
  
          {
  
              if (bufferEmptyIndex &lt; ClusterStart.Length + 4)
  
              {
  
                  ReadMoreBytes(buffer.Slice(bufferEmptyIndex, 1024));
  
                  bufferEmptyIndex += 1024;
  
              }
  
  
  
              int clusterDataSize = BitConverter.ToInt32(
  
                  buffer.Slice(ClusterStart.Length, 4)
  
                  .ToArray()
  
                  .Reverse()
  
                  .ToArray()
  
              );
  
              int clusterSize = ClusterStart.Length + 4 + clusterDataSize;
  
              if (clusterSize > buffer.Length)
  
              {
  
                  byte[] newBuffer = new byte[clusterSize];
  
                  buffer.Slice(0, bufferEmptyIndex).CopyTo(newBuffer);
  
                  buffer = newBuffer;
  
              }
  
  
  
              if (bufferEmptyIndex &lt; clusterSize)
  
              {
  
                  ReadMoreBytes(buffer.Slice(bufferEmptyIndex, clusterSize - bufferEmptyIndex));
  
                  bufferEmptyIndex = clusterSize;
  
              }
  
  
  
              ClusterReadyEvent?.Invoke(this, new ClusterReadyEventArgs(buffer.Slice(0, bufferEmptyIndex).ToArray()));
  
  
  
              bufferEmptyIndex = 0;
  
          } while (!cancel.IsCancellationRequested);
  
      }
  
  }
  
  </byte></byte></byte></clusterreadyeventargs>

  I use ffmpeg to convert the rtsp stream to vp8 WEBM byte stream and parse it to "Init Segment" (ebml head、info、tracks...) and "Media Segment" (cluster)， then send it to browser via signalR

  $(function () {
  
  
  
      var mediaSource = new MediaSource();
  
      var mimeCodec = 'video/webm; codecs="vp8"';
  
  
  
      var video = document.getElementById('video');
  
  
  
      mediaSource.addEventListener('sourceopen', callback, false);
  
      function callback(e) {
  
          var sourceBuffer = mediaSource.addSourceBuffer(mimeCodec);
  
          var queue = [];
  
  
  
          sourceBuffer.addEventListener('updateend', function () {
  
              if (queue.length === 0) {
  
                  return;
  
              }
  
  
  
              var base64 = queue[0];
  
              if (base64.length === 0) {
  
                  mediaSource.endOfStream();
  
                  queue.shift();
  
                  return;
  
              } else {
  
                  var buffer = new Uint8Array(atob(base64).split("").map(function (c) {
  
                      return c.charCodeAt(0);
  
                  }));
  
                  sourceBuffer.appendBuffer(buffer);
  
                  queue.shift();
  
              }
  
          }, false);
  
  
  
          var connection = new signalR.HubConnectionBuilder()
  
              .withUrl("/signalr-video")
  
              .configureLogging(signalR.LogLevel.Information)
  
              .build();
  
          connection.start().then(function () {
  
              connection.stream("InitVideoReceive")
  
                  .subscribe({
  
                      next: function(item) {
  
                          if (queue.length === 0 && !!!sourceBuffer.updating) {
  
                              var buffer = new Uint8Array(atob(item).split("").map(function (c) {
  
                                  return c.charCodeAt(0);
  
                              }));
  
                              sourceBuffer.appendBuffer(buffer);
  
                              console.log(blockindex++ + " : " + buffer.byteLength);
  
                          } else {
  
                              queue.push(item);
  
                          }
  
                      },
  
                      complete: function () {
  
                          queue.push('');
  
                      },
  
                      error: function (err) {
  
                          console.error(err);
  
                      }
  
                  });
  
          });
  
      }
  
      video.src = window.URL.createObjectURL(mediaSource);
  
  })

  chrome just play the video for 3 5 seconds and then stop for buffering, even though there are plenty of cluster transfered and inserted into SourceBuffer.

  here’s the information in chrome ://media-internals/

  Player Properties :

  render_id: 217
  
  player_id: 1
  
  origin_url: http://localhost:52531/
  
  frame_url: http://localhost:52531/
  
  frame_title: Home Page
  
  url: blob:http://localhost:52531/dcb25d89-9830-40a5-ba88-33c13b5c03eb
  
  info: Selected FFmpegVideoDecoder for video decoding, config: codec: vp8 format: 1 profile: vp8 coded size: [1280,720] visible rect: [0,0,1280,720] natural size: [1280,720] has extra data? false encryption scheme: Unencrypted rotation: 0°
  
  pipeline_state: kSuspended
  
  found_video_stream: true
  
  video_codec_name: vp8
  
  video_dds: false
  
  video_decoder: FFmpegVideoDecoder
  
  duration: unknown
  
  height: 720
  
  width: 1280
  
  video_buffering_state: BUFFERING_HAVE_NOTHING
  
  for_suspended_start: false
  
  pipeline_buffering_state: BUFFERING_HAVE_NOTHING
  
  event: PAUSE

  Log

  Timestamp       Property            Value
  
  00:00:00 00     origin_url          http://localhost:52531/
  
  00:00:00 00     frame_url           http://localhost:52531/
  
  00:00:00 00     frame_title         Home Page
  
  00:00:00 00     url                 blob:http://localhost:52531/dcb25d89-9830-40a5-ba88-33c13b5c03eb
  
  00:00:00 00     info                ChunkDemuxer: buffering by DTS
  
  00:00:00 35     pipeline_state      kStarting
  
  00:00:15 213    found_video_stream  true
  
  00:00:15 213    video_codec_name    vp8
  
  00:00:15 216    video_dds           false
  
  00:00:15 216    video_decoder       FFmpegVideoDecoder
  
  00:00:15 216    info                Selected FFmpegVideoDecoder for video decoding, config: codec: vp8 format: 1 profile: vp8 coded size: [1280,720] visible rect: [0,0,1280,720] natural size: [1280,720] has extra data? false encryption scheme: Unencrypted rotation: 0°
  
  00:00:15 216    pipeline_state      kPlaying
  
  00:00:15 213    duration            unknown
  
  00:00:16 661    height              720
  
  00:00:16 661    width               1280
  
  00:00:16 665    video_buffering_state       BUFFERING_HAVE_ENOUGH
  
  00:00:16 665    for_suspended_start         false
  
  00:00:16 665    pipeline_buffering_state    BUFFERING_HAVE_ENOUGH
  
  00:00:16 667    pipeline_state      kSuspending
  
  00:00:16 670    pipeline_state      kSuspended
  
  00:00:52 759    info                Effective playback rate changed from 0 to 1
  
  00:00:52 759    event               PLAY
  
  00:00:52 759    pipeline_state      kResuming
  
  00:00:52 760    video_dds           false
  
  00:00:52 760    video_decoder       FFmpegVideoDecoder
  
  00:00:52 760    info                Selected FFmpegVideoDecoder for video decoding, config: codec: vp8 format: 1 profile: vp8 coded size: [1280,720] visible rect: [0,0,1280,720] natural size: [1280,720] has extra data? false encryption scheme: Unencrypted rotation: 0°
  
  00:00:52 760    pipeline_state      kPlaying
  
  00:00:52 793    height              720
  
  00:00:52 793    width               1280
  
  00:00:52 798    video_buffering_state       BUFFERING_HAVE_ENOUGH
  
  00:00:52 798    for_suspended_start         false
  
  00:00:52 798    pipeline_buffering_state    BUFFERING_HAVE_ENOUGH
  
  00:00:56 278    video_buffering_state       BUFFERING_HAVE_NOTHING
  
  00:00:56 295    for_suspended_start         false
  
  00:00:56 295    pipeline_buffering_state    BUFFERING_HAVE_NOTHING
  
  00:01:20 717    event               PAUSE
  
  00:01:33 538    event               PLAY
  
  00:01:35 94     event               PAUSE
  
  00:01:55 561    pipeline_state      kSuspending
  
  00:01:55 563    pipeline_state      kSuspended

  Can someone tell me what’s wrong with my code, or dose chrome require some magic configuration to work ?

  Thanks 

  Please excuse my english :)

• will Adult script pro work with google app engine as server ? [on hold]

  1er février 2016, par niko nørre

  i want to user Google app engine as a server for my website and app.
  but a programmer needs to set the script up for me.
  the script is called Adult Script Pro and it need FFMpeg and more to Work.

  do google app engine provide me with ssh root access ?

  The requirements are the following :

  Apache, Lighttpd or Nginx Server (with rewrite support)

  MySQL 5.x

  PHP >= 5.2.x (mod_php or CGI/FastCGI)

  GD2 Support

  MySQL Support

  CURL Support

  SimpleXML Support

  FTP Support

  PCRE with UTF8/Unicode Properties

  PHP CLI >= 5.2.x (see above for support)

  FFMpeg => 0.11.5 (with support for lame, x264, theora, vpx, xvid, faac, faad2, amr, webm, jpeg, png, gif and freetype)

  Will the site Work whid Google app engine as server ??

  Pleas help thanks
  Nikolaj

• libavformat/libavfilter transcoded audio is choppy

  9 mai 2016, par JohnnyD

  I am attempting to transcode an mp4 file into a standard format. The video seems ok but the audio is choppy (and out of sync with the video).

  My test input file has the following properties :

  Stream #0:1(eng), 0, 1/48000: Audio: aac (mp4a / 0x6134706D), 48000 Hz, 2 channels, 129 kb/s (default)

  and I’m outputing to :

  Stream #0:1, 0, 1/44100: Audio: aac (libfaac) (Main), 44100 Hz, stereo, s16, 128 kb/s

  When I construct the audio filter graph I get the following debug output :

  [in @ 0x103954380] Setting 'time_base' to value '1/48000'
  
  [in @ 0x103954380] Setting 'sample_rate' to value '48000'
  
  [in @ 0x103954380] Setting 'sample_fmt' to value 'fltp'
  
  [in @ 0x103954380] Setting 'channel_layout' to value '0x3'
  
  [in @ 0x103954380] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3
  
  [format @ 0x10390b3e0] Setting 'sample_fmts' to value 's16'
  
  [format @ 0x10390b3e0] Setting 'sample_rates' to value '44100'
  
  [format @ 0x10390b3e0] Setting 'channel_layouts' to value '0x3'
  
  [format @ 0x10390b3e0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'in' and the filter 'format'
  
  [AVFilterGraph @ 0x101f21b80] query_formats: 3 queried, 3 merged, 3 already done, 0 delayed
  
  [auto-inserted resampler 0 @ 0x103952bc0] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz

  This looks correct to me but I’m getting a lot of the following messages when I process the file...

  [libfaac @ 0x102063a00] Trying to remove 80 more samples than there are in the queue

  ...and the audio is choppy. Also I’m seeing that the sample format is the same as the original file (from ffprobe) :

  Stream #0:1(und): Audio: aac (Main) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 127 kb/s (default)

  i.e. it hasn’t done the conversion from AV_SAMPLE_FMT_FLT to AV_SAMPLE_FMT_S16.

  I’m wondering if the bitrate is the cause of the problem but I can’t see any way to transform the input bitrate to the output bitrate. Any thoughts ?