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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 ?

  


  


• FFMPEG "Segmentation fault" with stream source and watermark

  4 mars 2020, par Magikey

  I use release : 4.2.2 (static) from "https://johnvansickle.com/ffmpeg/"

  Final code will be on "Amazon AWS lambda"

  Goal : use a url stream and add watermak

  Link to video : ​https://feoval.fr/519.mp4

  Link to Watermak : ​https://feoval.fr/watermark.png

  ./ffmpeg -i "https://feoval.fr/519.mp4" -i "./watermark.png" -filter_complex "overlay=W-w-10:H-h-10:format=rgb" -f "mp4" -movflags "frag_keyframe+empty_moov" -pix_fmt "yuv420p" test.mp4

  return "Segmentation fault"

  I have the same error on my computer and on AWS Lambda server

  ./ffmpeg -i "https://feoval.fr/519.mp4" -f "mp4" -movflags "frag_keyframe+empty_moov" -pix_fmt "yuv420p" test.mp4

  work (but not watermak)

  ./ffmpeg -i "./519.mp4" -i "./watermark.png" -filter_complex "overlay=W-w-10:H-h-10:format=rgb" -f "mp4" -movflags "frag_keyframe+empty_moov" -pix_fmt "yuv420p" test.mp4

  work (but not with stream)

  Thanks you very much !

  Logs for the first case who return "Segmentation fault" :

  ...
  
  Stream mapping:
  
  Stream #0:0 (h264) -> overlay:main (graph 0)
  
  Stream #1:0 (png) -> overlay:overlay (graph 0)
  
  overlay (graph 0) -> Stream #0:0 (libx264)
  
  Stream #0:1 -> #0:1 (aac (native) -> aac (native))
  
  Press [q] to stop, ? for help
  
  [libx264 @ 0x742e480] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 0x742e480] profile High, level 3.1, 4:2:0, 8-bit
  
  [libx264 @ 0x742e480] 264 - core 159 r2991 1771b55 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - ​http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  
  Output #0, mp4, to 'test.mp4':
  
  Metadata:
  
  major_brand : mp42
  
  minor_version : 1
  
  compatible_brands: isommp41mp42
  
  encoder : Lavf58.29.100
  
  Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 480x848, q=-1--1, 30 fps, 15360 tbn, 30 tbc (default)
  
  Metadata:
  
  encoder : Lavc58.54.100 libx264
  
  Side data:
  
  cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
  
  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
  
  Metadata:
  
  creation_time : 2020-01-13T08:54:26.000000Z
  
  handler_name : Core Media Audio
  
  encoder : Lavc58.54.100 aac
  
  Segmentation fault (core dumped)

• Convert a file using FFMPEG and upload to AWS S3 Nodejs

  6 mai 2020, par codernoob8

  Hey everyone so quick question I want to allow a user to upload a WebM file and convert it using FFmpeg to mp4. I am using Nodejs for the backend and already have a route that uploads files to Amazon S3 file storage. But let's say I wanted to send that file and not store it anywhere but convert it to mp4 from the request itself is that possible ? If not is it possible to take an s3 file URL and convert it to mp4 ? Can anybody point me in the right direction as to what is possible and the best way to do this ?

  



  basically all I want to do is

  



  const objectUrl = createObjectURL(Blob);
ffmpeg -i objectURL S3OutputLocation


  



  or

  



  ffmpeg -i myS3InputLocation myS3OutputLocation