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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/me_cmp : remove ff_me_cmp_init_static()

  3 février 2018, par Muhammad Faiz

  avcodec/me_cmp : remove ff_me_cmp_init_static()
  Precalculate and constify ff_square_tab.
  Reviewed-by : James Almer <jamrial@gmail.com>
  
  Signed-off-by : Muhammad Faiz <mfcc64@gmail.com>
  

  • [DH] libavcodec/me_cmp.c
  • [DH] libavcodec/me_cmp.h
  • [DH] libavcodec/mpegvideo_enc.c
  • [DH] libavcodec/mpegvideoencdsp.c
  • [DH] libavcodec/snowenc.c
  • [DH] libavcodec/utils.c

• swresample/resample : optimize exact_rational=on:linear_interp=on case

  24 novembre 2016, par Muhammad Faiz

  swresample/resample : optimize exact_rational=on:linear_interp=on case
  separate dsp.resample to dsp.resample_common and dsp.resample_linear
  
  and choose to call faster resample_common even when linear_interp=on
  
  when c->frac and c->dst_incr_mod are both zero
  speed up resampling when exact_rational and linear_interp are both
  
  enabled because exact_rational force c->frac and c->dst_incr_mod to
  
  be zero when soft compensation does not happen
  benchmark on exact_rational=on:linear_interp=on
  
          old     new
  
  real    8.432s  5.097s
  
  user    7.679s  4.989s
  
  sys     0.125s  0.107s
  Reviewed-by : Michael Niedermayer <michael@niedermayer.cc>
  
  Signed-off-by : Muhammad Faiz <mfcc64@gmail.com>
  

  • [DH] libswresample/arm/resample_init.c
  • [DH] libswresample/resample.c
  • [DH] libswresample/resample.h
  • [DH] libswresample/resample_dsp.c
  • [DH] libswresample/x86/resample_init.c