Sur d’autres sites (6232)

• need help on solving my ffmpeg command line

  3 avril 2019, par DRMTV

  i create a small bash script to encode 1080p video , the video will be added with watermark at bottom left and i need to add a black padding on top and bottom

  i tried several way but still no luck , i tried -vf and yes it worked but cant use padding and watermark together , and suggest use filter_complex

  if i use this code directly without bash script it work

  time ffmpeg -hide_banner -i transformers.mp4 -i transformers.ass -loop 1 -i watermark.png -loop 1 -i logo.png -map 0:0 -map 0:1 -filter_complex "[0:0]scale=(iw*sar)*min(1920/(iw*sar)\,800/ih):ih*min(1920/(iw*sar)\,800/ih), pad=1920:800:(1920-iw*min(1920/iw\,800/ih))/2:(800-ih*min(1920/iw\,800/ih))/2;ass=transformers.ass[FID1];[FID1][2:v]overlay=10:${WATERMARKPOSITION}:repeatlast=0:enable='between(t,300,600)'[FID3];[3:v]fade=in:st=1200:d=1.6:alpha=1,fade=out:st=107998:d=1.6:alpha=1[FID6];[FID3][FID6]overlay=10:5:repeatlast=0:enable='between(t,1200,187922)'" -c:v libx264 -minrate 1800k -maxrate 1800k -bufsize 3600k -profile:v high -c:a aac -b:a 128k -profile:a aac_main -movflags faststart -strict -2 -f mp4 -y "transformers.mp4"

  but when i include it with my bash script i got this error ,

  [libx264 @ 0x2a063e0] height not divisible by 2 (300x39)
  
  Output #0, mp4, to '/movie/Paddy/output/Transformers.Age.of.Extinction.2014.1080p.BluRay.H264.AAC-RARBG.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      comment         : Transformers.Age.of.Extinction.2014.1080p.BluRay.H264.AAC-RARBG
  
      encoder         : Lavf57.71.100
  
      title           : Transformers Age of Extinction 2014 1080p BluRay H264 AAC-RARBG - Visit us @ Juraganfilm.COM
  
      Stream #0:0: Video: h264 (libx264), yuv420p, 1920x800 [SAR 1:1 DAR 12:5], q=-1--1, max. 2300 kb/s, 23.98 fps, 23.98 tbn, 23.98 tbc
  
      Metadata:
  
        encoder         : Lavc56.60.100 libx264
  
      Stream #0:1: Video: h264, none, q=2-31, 128 kb/s, SAR 1:1 DAR 0:0, 25 fps
  
      Metadata:
  
        encoder         : Lavc56.60.100 libx264
  
      Stream #0:2(eng): Audio: aac, 0 channels, 128 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-12-19 07:58:39
  
        handler_name    : SoundHandler
  
        encoder         : Lavc56.60.100 aac
  
  Stream mapping:
  
    Stream #0:0 (h264) -> scale (graph 0)
  
    Stream #0:0 (h264) -> overlay:overlay (graph 0)
  
    Stream #2:0 (png) -> ass (graph 0)
  
    Stream #3:0 (png) -> fade (graph 0)
  
    pad (graph 0) -> Stream #0:0 (libx264)
  
    overlay (graph 0) -> Stream #0:1 (libx264)
  
    Stream #0:1 -> #0:2 (aac (native) -> aac (native))
  
  Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height

  and below is the script i have been working on

  output="/movie/output"
  
  
  
  FILESDIR=`find $PWD -type f -regex ".*\.\(mp4\|mkv\|avi\)" | sed 's@.*/@@' | sort -n`
  
  for video in $FILESDIR
  
  do
  
  
  
  MOVIETITLE=${video%.*}
  
  INFOVID=${MOVIETITLE//./ }
  
  BITRATE="${HEIGHT}"
  
  WIDTH=$(ffprobe -v error -select_streams v:0 -show_entries stream=width,height -of csv=s=x:p=0 ${video} 2>&1 | sed -e 's|\[.*||g' | sed ':a;N;$!ba;s/\n//g' | sed -e 's|x.*||g')
  
  HEIGHT=$(ffprobe -v error -select_streams v:0 -show_entries stream=width,height -of csv=s=x:p=0 ${video} | sed -e 's/.*x//')
  
  WATERMARKPOSITION=$(expr $HEIGHT - 50)
  
  VIDEOMAP=$(ffmpeg -i $video 2>&1 | grep "Stream #" | grep Video | sed -e "s|.*\#||g" | sed -e "s|: Video.*||g" | sed -e "s|(.*||g")
  
  AUDIOMAP=$(ffmpeg -i $video 2>&1 | grep "Stream #" | grep Audio | sed -e "s|.*\#||g" | sed -e "s|: Audio.*||g" | sed -e "s|(.*||g")
  
  MAXRATE=$(expr $BITRATE + 500)
  
  BUFFSIZE=$(expr $MAXRATE \* 2)
  
  
  
  time ffmpeg -hide_banner -i $video -i $MOVIETITLE.ass -loop 1 -i $WATERMARK -loop 1 -i $LOGO -map ${VIDEOMAP} -map ${AUDIOMAP} -filter_complex "[${VIDEOMAP}]scale=(iw*sar)*min(${WIDTH}/(iw*sar)\,${HEIGHT}/ih):ih*min(${WIDTH}/(iw*sar)\,${HEIGHT}/ih), pad=${WIDTH}:${HEIGHT}:(${WIDTH}-iw*min(${WIDTH}/iw\,${HEIGHT}/ih))/2:(${HEIGHT}-ih*min(${WIDTH}/iw\,${HEIGHT}/ih))/2;ass=$MOVIETITLE.ass[FID1];[FID1][2:v]overlay=10:${WATERMARKPOSITION}:repeatlast=0:enable='between(t,300,600)'[FID3];[3:v]fade=in:st=1200:d=1.6:alpha=1,fade=out:st=107998:d=1.6:alpha=1[FID6];[FID3][FID6]overlay=10:5:repeatlast=0:enable='between(t,1200,187922)'" -c:v libx264 -minrate ${BITRATE}k -maxrate ${MAXRATE}k -bufsize ${BUFFSIZE}k -profile:v high -c:a aac -b:a 128k -profile:a aac_main -movflags faststart -strict -2 -f mp4 -y "${output}/$MOVIETITLE.mp4"
  
  done

  has been working all day and still i cant make it to work.

  can someone guide me which part is wrong ?

• 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 stream chrome kiosk mode ubuntu 16.04 server

  21 décembre 2016, 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 ?