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• Newly coded libvpx 1.8.1 webm MSE can't be played in chrome, firefox is OK

  25 septembre 2019, par d3im

  Error of Chrome :

  Error Message:  CHUNK_DEMUXER_ERROR_APPEND_FAILED: Got a block with negative timecode offset -14

  I make webm video using latest ffmpeg 4.2.1 with libvpx 1.8.1 :

  1-st pass :

  ffmpeg -y -i input.mp4 -c:v libvpx-vp9 -b:v 800k -pix_fmt yuv420p -vf scale=-1:720 -tile-columns 2 -quality good -speed 4 -max-intra-rate 0 -lag-in-frames 25 -row-mt 1 -pass 1 -an -threads 0 -f webm /dev/null

  2-nd pass :

  ffmpeg -i input.mp4 -c:v libvpx-vp9 -b:v 800k -pix_fmt yuv420p -vf scale=-1:720 -tile-columns 2 -quality good -speed 0 -max-intra-rate 0 -auto-alt-ref 1 -lag-in-frames 25 -row-mt 1 -pass 2 -c:a libopus -b:a 64k -frame_duration 60 output.webm

  Then I use mse-tools (https://github.com/acolwell/mse-tools) to remux output.webm. When I use dump tool from this package it shows me negative offsets which are source of problem in Chrome - It can’t append chunk with negative offset.

• 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 ?

  


  


• Media player get stuck in the middle of a buffered range on Chrome

  29 septembre 2019, par Feng Yu

  WHAT IS MY PROBLEM ?

  My website’s live streaming player use hls.js. From my server’s stat, there is many case where player get stuck in the middle of a buffered range.

  Here is my server raw stat log(removed some useless params) :

  tm=2019-09-27 12:04:41`bufferLevel=8.447303999999974`currentTime=158.4`buffered=[6.024,166.832]`readyState=4`ua=Mozilla/5.0 (Windows NT 6.2; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/53.0.2785.116 Safari/537.36 QBCore/3.53.1153.400 QQBrowser/9.0.2524.400 Tencent AppMarket/4.8 GameCenter

  currentTime is got by HTMLMediaElement.currentTime and buffered is got by HTMLMediaElement.buffered :

  currentTime=158.4
  
  buffered=[6.024,166.832]
  
  readyState=4

  From W3c :

  If HTMLMediaElement.buffered contains a TimeRange that includes the current playback position and enough data to ensure uninterrupted playback :

  1. Set the HTMLMediaElement.readyState attribute to HAVE_ENOUGH_DATA.
  2. Playback may resume at this point if it was previously suspended by a transition to HAVE_CURRENT_DATA.

  In this case, 613.3 is in the middle of [469.277,677.612], video should be progressing, but it is not.

  Hls.js will periodly check currentTime has progressed every 100ms. if currentTime has not progressed for 1000ms, then hls.js will trigger STALL event and I will send a stall stat to server.

  I cannot reproduce this problem on my side, it only appears on my server stat.

  WHAT I’VE TRIED

  shaka player has a module detect this case(https://www.ellealcatrase.eu/player2/docs/api/lib_media_stall_detector.js.html), Its comment shows that :

  Some platforms/browsers can get stuck in the middle of a
  buffered range (e.g. when seeking in a background tab). Detect when
  we get stuck so that the player can respond.

  but I cannot reproduce when my browser is in a background tab.