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• ffmpeg transcode to hls and multiple aac audio tracks

  23 août 2023, par Aquarius

  I'm trying to transcode h264 mp4 file to HLS format with multiple AAC audio tracks. On Windows browsers everythins works like a shine. There's an option to switch audio track

  


  

  


  It doesn't work on Android devices or TV. On Android Opera browser video works with sound, but screen stuck while seeking and there's no option to switch audio track. On TV browser video works, it doesn't stuck when seeking but also no option to switch audio track

  


  Using video.js player on site. Below code for transcode and produced manifest file

  


  ffmpeg -i "file.mp4" -i PL-6ch.aac -i EN-6ch.aac -map 0:v -map 1:a -map 2:a -c:v copy -c:a:0 aac -ac:a:0 6 -af channelmap=channel_layout=5.1 -c:a:1 aac -ac:a:1 6 -af channelmap=channel_layout=5.1 -tag:v:0 h264 -b:v:0 2500k -tag:a:0 aac -b:a:0 384k -tag:a:1 aac -b:a:1 384k -f hls -hls_time 10 -hls_playlist_type vod -hls_list_size 0 -master_pl_name file.m3u8 -var_stream_map "v:0,agroup:AAC a:0,agroup:AAC,language:pl a:1,agroup:AAC,language:en" file_%v_.m3u8


  


  #EXTM3U
#EXT-X-VERSION:3
#EXT-X-INDEPENDENT-SEGMENTS
#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="AAC",NAME="Polish",AUTOSELECT=YES,DEFAULT=YES,LANGUAGE="pl",URI="file_1_.m3u8"
#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="AAC",NAME="English",AUTOSELECT=YES,DEFAULT=NO,LANGUAGE="en",URI="file_2_.m3u8"
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=3172400,RESOLUTION=1920x872,CODECS="avc1.640029,mp4a.40.2",AUDIO="AAC" 
file_0_.m3u8

#EXT-X-STREAM-INF:BANDWIDTH=422400,CODECS="mp4a.40.2",AUDIO="AAC"
file_1_.m3u8

#EXT-X-STREAM-INF:BANDWIDTH=422400,CODECS="mp4a.40.2",AUDIO="AAC"
file_2_.m3u8


  


  Any clues what's wrong ?

  


• dash.js : four channels audio streaming

  19 juillet 2016, par Carlos Chacon

  Does dash.js supports 4 channels in the audio stream ?

  I’m using mp4 container : video : h254 and audio : AAC.

  For the case that the AAC audio stream is 2 channels it works fine, both video and audio play correctly. But 4 channels does NOT work.

  I’m using FFMPEG to create the files and WOWZA to setup the DASH streaming.

  For the case that the AAC audio stream is 4 channels I get the following error :

   Video can't be played because the file is corrupt.

  Command line to generate the 2-channels file :

  ffmpeg -i video_in.mp4 -i audio_in_4ch.wav -c:v copy -c:a aac -ac 2 output_2channels.mp4

  Command line to generate the 4-channels file :

  ffmpeg -i video_in.mp4 -i audio_in_4ch.wav -c:v copy -c:a aac -ac 4 output_4channels.mp4

  This is the fmmpeg -i information printed for each file :

  2 Channels output file details :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output_2channels.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.37.101
  
    Duration: 00:00:17.59, start: 0.000000, bitrate: 901 kb/s
  
      Stream #0:0(eng): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1024x512 [SAR 1:1 DAR 2:1], 894 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 129 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler

  4 Channels output file details :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output_4channels.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.37.101
  
    Duration: 00:00:17.59, start: 0.000000, bitrate: 1038 kb/s
  
      Stream #0:0(eng): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1024x512 [SAR 1:1 DAR 2:1], 894 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 4.0, fltp, 265 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler

  4 Channels output file mdp file :

  <?xml version="1.0" encoding="UTF-8"?>
  
  <mpd xmlns="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-live:2011" type="static" publishtime="2016-07-19T05:56:28Z" mediapresentationduration="PT17.585S" minbuffertime="PT1.5S">
  
  <programinformation>
  
      
  
  </programinformation>
  
  <location>http://192.168.0.103:1935/vod/_definst_/mp4:output_4channels.mp4/manifest_w882219731.mpd</location>
  
  <period start="PT0.0S">
  
      <baseurl>http://192.168.0.103:1935/vod/_definst_/mp4:output_4channels.mp4/</baseurl>
  
      <adaptationset mimetype="video/mp4" width="1024" height="512" par="1024:512" framerate="30000/1001" segmentalignment="true" startwithsap="1" subsegmentalignment="true" subsegmentstartswithsap="1">
  
          <segmenttemplate presentationtimeoffset="0" timescale="90000" media="chunk_ctvideo_rid$RepresentationID$_cs$Time$_w882219731_mpd.m4s" initialization="chunk_ctvideo_rid$RepresentationID$_cinit_w882219731_mpd.m4s">
  
              <segmenttimeline>
  
                  <s t="0" d="1582650"></s>
  
              </segmenttimeline>
  
          </segmenttemplate>
  
          <representation codecs="avc1.42c01e" sar="1:1" bandwidth="894760"></representation>
  
      </adaptationset>
  
      <adaptationset mimetype="audio/mp4" lang="eng" segmentalignment="true" startwithsap="1" subsegmentalignment="true" subsegmentstartswithsap="1">
  
          <audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>
  
          <segmenttemplate presentationtimeoffset="0" timescale="48000" media="chunk_ctaudio_rid$RepresentationID$_cs$Time$_w882219731_mpd.m4s" initialization="chunk_ctaudio_rid$RepresentationID$_cinit_w882219731_mpd.m4s">
  
              <segmenttimeline>
  
                  <s t="0" d="844080"></s>
  
              </segmenttimeline>
  
          </segmenttemplate>
  
          <representation codecs="mp4a.40.2" audiosamplingrate="48000" bandwidth="265851">
  
          </representation>
  
      </adaptationset>
  
  </period>
  
  </mpd>

  HTML Player Code :

  <code class="echappe-js">&lt;script src=&quot;http://cdn.dashjs.org/latest/dash.all.min.js&quot;&gt;&lt;/script&gt;
  
         
  
  

• ffmpeg audio and video gradually go out of sync when recorded with shell script and make

  6 août 2020, par cgmil

  I wrote the following shell script for making streamcasts for recording streams using ffmpeg :

  


  [ -z $1 ] && { echo "No recording device selected!" 2>&1 && exit ; } || DEV=$1
VIDFILE=video/bvid.mkv
AUDFILE=audio/baud.wav
DIM=$(xdpyinfo | grep dimensions: | awk '{print $2;}')
(sleep 10 && echo "Streaming...") &
ffmpeg -nostdin -loglevel panic -f x11grab -s $DIM -i $DISPLAY.0 -c:v libx264 "$VIDFILE" &
ffmpeg -nostdin -loglevel panic -f alsa -i hw:$DEV -c:a libmp3lame  "$AUDFILE" &
sleep 1
[ -f $VIDFILE ] && (echo "Created video" 1>&2) || (echo "No video!" 1>&2)
sleep 1
[ -f $AUDFILE ] && (echo "Created audio" 1>&2) || (echo "No audio!" 1>&2)

wait
trap "killall background -s 2" SIGINT


  


  This produces a video and audio file that can then be used to make a stream via make using the command make stream. Here is the Makefile (note that there is a 3-minute long audio file and a file with images that is used to make an opening splash screen) :

  


  FULLAUDIO=audio/full.wav
AUDSTART=84
TITLE="Stats Aside:\\nInterpreting\\nProbability"
SIZEW=1920
SIZEH=1080

audio/clip.wav : $(FULLAUDIO)
        ffmpeg -ss $(AUDSTART) -i $< -t 6 -af "afade=t=in:st=0:d=3:curve=qsin,afade=t=out:st=3:d=3:curve=qsin" -c:a libmp3lame $@

img/thumb.png : img/bg.png img/ico.png
        convert -stroke white -fill white -background black -transparent "#000000" \
                 -font "URWBookman-Light" -gravity Center -size 928x1080 \
                label:$(TITLE) png:- | composite -gravity west -geometry +64+0 - \
                img/bg.png $@
        convert img/ico.png -transparent "#ffffff" png:- | composite \
                 -gravity southeast -geometry +32+32 - $@ $@
        convert $@ -resize $(SIZEW)x$(SIZEH) $@

video/intro.mp4 : img/thumb.png audio/clip.wav
        ffmpeg -i audio/clip.wav -loop 1 -t 6 -i img/thumb.png -vf "fade=t=in:st=0:d=1,fade=t=out:st=5:d=1:c=white,scale=$(SIZEW):$(SIZEH)" -c:v libx264 -c:a libmp3lame $@

ProbabilityInterpretation.mp4 : video/bvid.mkv video/intro.mp4 audio/baud.wav
        ffmpeg -i video/intro.mp4 -ss 11 -i video/bvid.mkv \
                 -ss 10 -i audio/baud.wav \
                 -filter_complex \
                "[1]fade=t=in:st=0:d=1:c=white,scale=$(SIZEW):$(SIZEH)[bvid];\
                 [2]afftdn=nr=20[baud];\
                 [0:v][0:a][bvid][baud]concat=n=2:v=1:a=1[v][a]" -map "[v]" -map "[a]" \
                 -c:v libx264 -c:a libmp3lame $@

stream : ProbabilityInterpretation.mp4

clean:
        -rm ProbabilityInterpretation.mp4
        -rm video/intro.mp4
        -rm audio/clip.wav
        -rm img/thumb.png

veryclean:
        make clean
        -rm video/bvid.mkv
        -rm audio/baud.wav



  


  Because of lag between my webcam and what appears on screen, I do intentionally start the video and audio at different times so that they can eventually become in sync. However, the end result is a video that starts with the video and audio synced together, but eventually become out of sync, with the video lagging well behind the audio. Here is a link to the video with the end result ; it's long, but notice that the video starts in sync and ends out-of-sync. I should probably also note that the operating system is living in a virtual machine (VirtualBox).

  


  Why is this happening and what can I do to prevent or correct ?