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• ffmpeg stream chrome kiosk mode ubuntu 16.04 server

  15 février 2021, par Raul

  I have a weird out-of-sync issue while using ffmpeg to stream to youtube live a chrome browser from an ub untu 16.04 server.

  



  Issue : output video streamed to youtube has audio/video out of sync, sometimes with as much as 3s

  



  Current flow :

  



  1) start pulseaudio - we using something like this to start it :

  



  pulseaudio --start -vvv --disallow-exit --log-target=syslog --high-priority --exit-idle-time=-1 --daemonize


  



  2) start Xvfb

  



  Xvfb :0 -ac -screen 0 1920x1080x24


  



  3) start chrome linux in kiosk mode

  



  google-chrome --kiosk --disable-gpu --incognito --no-first-run --disable-java --disable-plugins --disable-translate --disk-cache-size=$((1024 * 1024)) --disk-cache-dir=/tmp/chrome/ --user-data-dir=/tmp/chrome/ --force-device-scale-factor=1 --window-size=1920,1080 --window-position=0,0 LOCATION_URL


  



  4) start ffmpeg

  



  ffmpeg -y \
  -thread_queue_size 8192 -rtbufsize 250M -f x11grab -video_size 1920x1080 -framerate 24 -i :0 \
  -thread_queue_size 8192 -channel_layout stereo -f alsa -i pulse \
  -c:v libx264 -pix_fmt yuv420p -c:v libx264 -g 48 -crf 24 -filter:v fps=24 -preset ultrafast -tune zerolatency \
  -c:a aac -strict -2 -channel_layout stereo -ab 96k -ac 2 -flags +global_header \
  -f flv YOUTUBE_LIVE_STREAMING_RTMP


  



  Note : this is running on an amazon ec2 instance, meaning there is no soundcard, so alsa and pulseaudio are creating a dummy audio card. However, the latency does not come from there. Logs :

  



  Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Adjust latency mode enabled, configuring sink latency to half of overall latency.
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Requested latency=23.22 ms, Received latency=23.22 ms
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Final latency 69.66 ms = 23.22 ms + 2*11.61 ms + 23.22 ms


  



  At this point, here's what we observed :

  



  


  1. if we start ffmpeg exactly after issuing the command to start chrome, we see the DTS errors from ffmpeg. Audio is out of sync with the video and has delay of 3-5seconds AHEAD. We also noticed the out of sync remains the same for the full duration of the stream

  


  2. if we start ffmpeg after around 10seconds, audio and video are almost in sync. We then manually added a -itsoffset -0.125 to the ffmpeg command and everything is perfect.

  


  



  Questions :

  



  

  1. Why would ffmpeg have so much lag if it's started right after chrome ?

  


  2. Is starting the ffmpeg after 10s or X seconds the expected behavior ? That is, is this because the system needs to wait for audio/video signals to be "ready" or something ?

  


  3. Is there a way to 100% calculate or know when Chrome is fully ready and start ffmpeg ? We found sometimes it takes 5s, sometimes 10. Depends on the URL we load.

  


  4. Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything. And a restart is required to "re-balance" the audio/video inputs and get them back in sync.

  


  5. Can pulseaudio be the problem in this scenario ?

  


  



  Thank you

  



  UPDATE Dec 20

  



  We were able to do some tricks to force chrome to start the audio on page load, and that will force connect to pulseaudio. Doing that, plus adding a 3s delay for ffmpeg to start, there is no more delay when ffmpeg starts.
However, our app is a webRTC app, and we have a STRANGER thing happening : if we start the page with no webcam/audio, once the webcam/audio is enabled, ffmpeg (while showing no errors) has a delay of 2s or so. While keep talking, in about max 30s, that delay is GONE.

  



  So the new questions are :

  



  

  1. Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything.

  


  2. What could cause the initial audio/video out of sync issue and then catching up ?

  


  


• avcodec/mp3 : fix skipping zeros

  30 septembre 2015, par wm4

  avcodec/mp3 : fix skipping zeros
  Commits 43bc5cf9 and c5371f77 add code for skipping initial zeros in mp3
  
  packets. This code forgot to report to the user that data was skipped at
  
  all.
  Since audio codecs allow partial packet decoding, the user application
  
  has to rely on the return value. It will remove the data reported as
  
  consumed by the decoder, and feed it to the decoder again. This resulted
  
  in the mp3 frame after the zero region to be decoded over and over
  
  again, until the zero region was finally skipped by the application.
  Fix this by including the amount of skipped bytes to the number of
  
  consumed bytes returned by the decode call.
  Fixes trac ticket #4890.
  

  • [DH] libavcodec/mpegaudiodec_template.c

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  7 juin 2018, par Matomo Core Team — Community, Development

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