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• How can I correctly provide a mock webcam video to Chrome ?

  15 décembre 2022, par doppelgreener

  I'm trying to run end-to-end testing in Chrome for a product that requires a webcam feed halfway through to operate. From what I understand this means providing a fake webcam video to Chrome using the --use-file-for-fake-video-capture="/path/to/video.y4m" command line argument. It will then use that as a webcam video.

  



  However, no matter what y4m file I provide, I get the following error from Chrome running under these conditions :

  



  DOMException: Could not start video source
{
  code: 0,
  message: "Could not start video source",
  name: "NotReadableError"
}


  



  Notably I can provide an audio file just fine using --use-file-for-fake-audio-capture and Chrome will work with it well. The video has been my sticking point.

  



  This error comes out of the following straightforward mediaDevices request :

  



  navigator.mediaDevices.getUserMedia({ video: true, audio: true })
  .then(data => {
    // do stuff
  })
  .catch(err => {
    // oh no!
  });


  



  (This always hits the “oh no !” branch when a video file is provided.)

  



  What I've tried so far

  



  I've been running Chrome with the following command line arguments (newlines added for readability), and I'm using a Mac hence the open command :

  





  open -a "Google Chrome" --args
  --disable-gpu
  --use-fake-device-for-media-stream
  --use-file-for-fake-video-capture="~/Documents/mock/webcam.y4m"
  --use-file-for-fake-audio-capture="~/Documents/mock/microphone.wav"


  



  webcam.y4m and microphone.wav were generated from a video file I recorded.

  



  I first recorded a twenty-second mp4 video using my browser's MediaRecorder, downloaded the result, and converted it using the following command line commands :

  



  ffmpeg -y -i original.mp4 -f wav -vn microphone.wav
ffmpeg -y -i original.mp4 webcam.y4m


  



  When this didn't work, I tried the same using a twenty-second movie file I recorded in Quicktime :

  



  ffmpeg -y -i original.mov -f wav -vn microphone.wav
ffmpeg -y -i original.mov webcam.y4m


  



  When that also failed, I went straight to the Chromium file that explains fake video capture, went to the example y4m file list it provided, and downloaded the grandma file and provided that as a command line argument to Chrome instead :

  



  open -a "Google Chrome" --args
  --disable-gpu
  --use-fake-device-for-media-stream
  --use-file-for-fake-video-capture="~/Documents/mock/grandma_qcif.y4m"
  --use-file-for-fake-audio-capture="~/Documents/mock/microphone.wav"


  



  Chrome provides me with the exact same error in all of these situations.

  



  The only time Chrome doesn't error out with that mediaDevices request is when I omit the video completely :

  



  open -a "Google Chrome" --args
  --disable-gpu
  --use-fake-device-for-media-stream
  --use-file-for-fake-audio-capture="~/Documents/mock/microphone.wav"


  



  Accounting for C420mpeg2

  



  TestRTC suggests Chrome will “crash” if I give it a C420mpeg2 file, and recommends that simply replacing the metadata fixes the issue. Indeed the video file I generate from ffmpeg gives me the following header :

  



  YUV4MPEG2 W1280 H720 F30:1 Ip A1:1 C420mpeg2 XYSCSS=420MPEG2


  



  Chrome doesn't actually crash when run with this file, I just get the error above. If I edit the video file to the following header though per TestRTC's recommendations I get the same situation :

  



  YUV4MPEG2 W1280 H720 F30:1 Ip A1:1 C420 XYSCSS=420MPEG2


  



  The video file still gives me the above error in these conditions.

  



  What can/should I do ?

  



  How should I be providing a video file to Chrome for this command line argument ?

  



  How should I be recording or creating the video file ?

  



  How should I convert it to y4m ?

  


• Problems piping ffmpeg to flac encoder

  19 mai 2019, par Sebastian Olsen

  I need to encode a flac file with seektables, ffmpeg’s flac encoder does not include seektables, so I need to use the flac CLI. I’m trying to make it possible to convert any arbitrary audio file to a seekable flac file by first piping it through ffmpeg, then to the flac encoder.

  export const transcodeToFlac: AudioTranscoder<{}> = ({
  
    source,
  
    destination
  
  }) => {
  
    return new Promise((resolve, reject) => {
  
      let totalSize = 0
  
  
  
      const { stdout: ffmpegOutput, stderr: ffmpegError } = spawn("ffmpeg", [
  
        "-i",
  
        source,
  
        "-f",
  
        "wav",
  
        "pipe:1"
  
      ])
  
  
  
      const { stdout: flacOutput, stdin: flacInput, stderr: flacError } = spawn(
  
        "flac",
  
        ["-"]
  
      )
  
  
  
      flacOutput.on("data", (buffer: Buffer) => {
  
        totalSize += buffer.byteLength
  
      })
  
  
  
      ffmpegError.on("data", error => {
  
        console.log(error.toString())
  
      })
  
  
  
      flacError.on("data", error => {
  
        console.log(error.toString())
  
      })
  
  
  
      //stream.on("error", reject)
  
  
  
      destination.on("finish", () => {
  
        resolve({
  
          mime: "audio/flac",
  
          size: totalSize,
  
          codec: "flac",
  
          bitdepth: 16,
  
          ext: "flac"
  
        })
  
      })
  
  
  
      ffmpegOutput.pipe(flacInput)
  
      flacOutput.pipe(destination)
  
    })
  
  }

  While this code works, the resulting flac file is not correct. The source audio is of duration 06:14, but the flac file is of duration 06:45:47. Encoding the flac manually without piping ffmpeg to it works fine, but I cannot do that in a server environment where I need to utilize streams.

  Here’s what the flac encoder outputs when transcoding :

  flac 1.3.2
  
  Copyright (C) 2000-2009  Josh Coalson, 2011-2016  Xiph.Org Foundation
  
  flac comes with ABSOLUTELY NO WARRANTY.  This is free software, and you are
  
  welcome to redistribute it under certain conditions.  Type `flac' for details.
  
  
  
  -: WARNING: skipping unknown chunk 'LIST' (use --keep-foreign-metadata to keep)
  
  -: WARNING, cannot write back seekpoints when encoding to stdout
  
  -: 0% complete, ratio=0.357
  
  0% complete, ratio=0.432
  
  0% complete, ratio=0.482
  
  0% complete, ratio=0.527
  
  0% complete, ratio=0.541
  
  1% complete, ratio=0.554
  
  1% complete, ratio=0.563
  
  1% complete, ratio=0.571
  
  size=   36297kB time=00:03:30.70 bitrate=1411.2kbits/s speed= 421x
  
  1% complete, ratio=0.572
  
  1% complete, ratio=0.570
  
  1% complete, ratio=0.577
  
  1% complete, ratio=0.583
  
  1% complete, ratio=0.584
  
  1% complete, ratio=0.590
  
  1% complete, ratio=0.592
  
  size=   64512kB time=00:06:14.49 bitrate=1411.2kbits/s speed= 421x
  
  video:0kB audio:64512kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead:
  
  0.000185%
  
  
  
  -: WARNING: unexpected EOF; expected 1073741823 samples, got 16510976 samples
  
  2% complete, ratio=0.579

• PyAV : force new framerate while remuxing stream ?

  7 juin 2019, par ToxicFrog

  I have a Python program that receives a sequence of H264 video frames over the network, which I want to display and, optionally, record. The camera records at 30FPS and sends frames as fast as it can, which isn’t consistently 30FPS due to changing network conditions ; sometimes it falls behind and then catches up, and rarely it drops frames entirely.

  The "display" part is easy ; I don’t need to care about timing or stream metadata, just display the frames as fast as they arrive :

  input = av.open(get_video_stream())
  
  for packet in input.demux(video=0):
  
    for frame in packet.decode():
  
      # A bunch of numpy and pygame code here to convert the frame to RGB
  
      # row-major and blit it to the screen

  The "record" part looks like it should be easy :

  input = av.open(get_video_stream())
  
  output = av.open(filename, 'w')
  
  output.add_stream(template=input.streams[0])
  
  for packet in input.demux(video=0):
  
    for frame in packet.decode():
  
      # ...display code...
  
    packet.stream = output.streams[0]
  
    output.mux_one(packet)
  
  output.close()

  And indeed this produces a valid MP4 file containing all the frames, and if I play it back with mplayer -fps 30 it works fine. But that -fps 30 is absolutely required :

  $ ffprobe output.mp4
  
  Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 960x720,
  
                    1277664 kb/s, 12800 fps, 12800 tbr, 12800 tbn, 25600 tbc (default)

  Note that 12,800 frames/second. It should look something like this (produced by calling mencoder -fps 30 and piping the frames into it) :

  $ ffprobe mencoder_test.mp4
  
  Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 960x720,
  
                    2998 kb/s, 30 fps, 30 tbr, 90k tbn, 180k tbc (default)

  Inspecting the packets and frames I get from the input stream, I see :

  stream: time_base=1/1200000
  
  codec: framerate=25 time_base=1/50
  
  packet: dts=None pts=None duration=48000 time_base=1/1200000
  
  frame: dst=None pts=None time=None time_base=1/1200000

  So, the packets and frames don’t have timestamps at all ; they have a time_base which doesn’t match either the timebase that ends up in the final file or the actual framerate of the camera ; the codec has a framrate and timebase that doesn’t match the final file, the camera framerate, or the other video stream metadata !

  The PyAV documentation is all but entirely absent when it comes to issues of timing and framerate, but I have tried manually setting various combinations of stream, packet, and frame time_base, dts, and pts with no success. I can always remux the recorded videos again to get the correct framerate, but I’d rather write video files that are correct in the first place.

  So, how do I get pyAV to remux the video in a way that produces an output that is correctly marked as 30fps ?