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• whether it takes long time for convert flv to MP4 using ffmpeg from server

  29 décembre 2016, par Firman Firdaus

  I want to convert a flv file to mp4. I use with a basic command like this :

  ffmpeg -i /var/www/html/rawmedia/d2cb9f152f27d9beb4a15d61e177fa22.flv /var/www/html/media/new.mp4

  The command above has result 0 B mp4.
  I read for My reference for this in this link,
  there is a very simple and basic I think.
  So I try the command above for testing php vibe command like this :

  ffmpeg -i /var/www/html/rawmedia/d2cb9f152f27d9beb4a15d61e177fa22.flv -vcodec libx264 -s 640x360 -threads 4 -movflags faststart /var/www/html/media/test4.mp4

  I got that command from PHPVIBE for convert other format file except mp4.
  So I got a result just the same as like first command that still 0 B mp4.
  But the Output of the second command give me like this

  Incompatible sample format 's16' for codec 'aac', auto-selecting format 'flt'
  
  [libx264 @ 0xcebde0] using cpu capabilities: none!
  
  [libx264 @ 0xcebde0] profile High, level 3.0
  
  [libx264 @ 0xcebde0] 264 - core 120 r2151 a3f4407 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - 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=4 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
  
  [NULL @ 0xd5f1a0] Codec is experimental but experimental codecs are not enabled, see -strict -2
  
  Output #0, mp4, to '/var/www/html/media/test4.mp4':
  
    Metadata:
  
      starttime       : 0
  
      totalduration   : 57
  
      totaldatarate   : 353
  
      bytelength      : 2535814
  
      canseekontime   : true
  
      sourcedata      : B0AFCE7F5HH1428394559963956
  
      purl            :
  
      pmsg            :
  
      Stream #0:0: Video: h264, yuv420p, 640x360, q=-1--1, 90k tbn, 30 tbc
  
      Stream #0:1: Audio: none, 22050 Hz, stereo, flt, 128 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (flv -> libx264)
  
    Stream #0:1 -> #0:1 (mp3 -> aac)
  
  Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height

  Output give me some information that test.mp4 have bytelength      : 2535814.
  But I still have 0B mp4 converted result until now. Is there taking a time or I was wrong with the command. But especially my first command very basic and still get zero byte result.

• HTML5 Progressive Streaming — no follow-up range requests

  20 septembre 2023, par user2333829

  I'm working on an embedded device that is recording video on the fly. I'd like to stream that to an HTML5 video element, using our own custom server. I have this almost working and would like some help.

  



  So far as I can tell, I've got libav / ffmpeg doing their job right. I encoded an mp4 in RAM with the moov atom at the start of the file. I've written this file to disk and it plays everywhere it should.

  



  The problem, I think, lies with how I'm responding to HTTP range requests. When I try to do a live stream, I get an initial range request from the browser / player (currently tried Chrome, Firefox, and VLC) for bytes:0-. I responded with some initial bytes. The browser / player actually plays this fine, but never asks again. So the live stream doesn't work, just the first 3 seconds or whatever.

  



  I've looked at the RFC spec of partial content, and my understanding is I'm doing what I should be... Clearly I'm not though. Here is an example of a request / response with Chrome as the requester :

  



  ---

  



  
get /live.mp4 HTTP/1.1
host: localhost:1235
connection: keep-alive
accept-encoding: identity;q=1, *;q=0
user-agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_13_2) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/64.0.3282.167 Safari/537.36
accept: */*
dnt: 1
accept-language: en-GB,en-US;q=0.9,en;q=0.8
range: bytes=0-


  



  
HTTP/1.1 206 Partial Content
Accept-Ranges: bytes
Content-Type: video/mp4
Content-Length: 182400
Content-Range: bytes 0-182399/*


  



  ---

  



  Again, with that request / response pair, Chrome plays the first 182400 bytes but never makes a second request. I thought having the '*' in Content-Range would make this happen...

  


• 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 ?