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• Ffmpeg - How can I create HLS multiple language streams, in multiple qualities ?

  28 avril 2022, par Daniel Ellis

  Preface

  


  I'm working on converting videos from 4k to multiple qualities with multiple languages but am having issues with the multiple languages overlaying, sometimes losing quality and sometimes being out of sync. (this is less of a problem in the German audio, as this is voice over anyhow)

  


  We as a team are complete noobs in terms of Video / Audio + HLS — I'm a front end developer who has no experience of this so apologies if my question is poorly phrased

  


  ---

  


  Videos

  


  I have the video in a 4k format and have removed the original sound as I have English and German audio files that need to be overlayed. I am then taking these files and throwing them together into a .ts file like this :

  


  $ ffmpeg -i ep03-ns-4k.mp4 -i nkit-ep3-de-output.m4a -i nkit-ep3-en-output.m4a \
> -thread 0 -muxdelay 0 -y \
> -map 0:v -map 1 -map 2  -movflags +faststart -refs 1 \
> -vcodec libx264 -acodec aac -profile:v baseline -level 30 -ar 44100 -ab 64k -f mpegts out.ts 


  


  This outputs a 4k out.ts video, with both audio tracks playing.

  


  The hard part

  


  This is where I'm finding it tricky, I now need to convert this single file into multiple quality levels (480, 720, 1080, 1920) and I attempt this with the following command :

  


  ffmpeg -hide_banner -y -i out.ts \
-crf 20 -sc_threshold 0 -g 48 -keyint_min 48 -ar 48000 \
-map 0:v:0 -map 0:v:0 -map 0:v:0 -map 0:v:0 \
-c:v:0 h264 -profile:v:0 main -filter:v:0 "scale=w=848:h=480:force_original_aspect_ratio=decrease" -b:v:0 1400k -maxrate:v:0 1498k -bufsize:v:0 2100k \
-c:v:1 h264 -profile:v:1 main -filter:v:1 "scale=w=1280:h=720:force_original_aspect_ratio=decrease" -b:v:1 2800k -maxrate:v:1 2996k -bufsize:v:1 4200k \
-c:v:2 h264 -profile:v:2 main -filter:v:2 "scale=w=1920:h=1080:force_original_aspect_ratio=decrease" -b:v:2 5600k -maxrate:v:2 5992k -bufsize:v:2 8400k \
-c:v:3 h264 -profile:v:3 main -filter:v:3 "scale=w=3840:h=1920:force_original_aspect_ratio=decrease" -b:v:3 11200k -maxrate:v:3 11984k -bufsize:v:3 16800k \
-var_stream_map "v:0 v:1 v:2 v:3" \
-master_pl_name master.m3u8 \
-f hls -hls_time 4 -hls_playlist_type vod -hls_list_size 0 \
-hls_segment_filename "%v/episode-%03d.ts" "%v/episode.m3u8"


  


  This creates the required qualities, but I'm now at a loss of how this might work with the audio

  


  Audio

  


  For the audio I run this command :

  


  ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:0 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-de/audio-de.m3u8 -segment_format mpegts audio-de/audio-de_%d.aac
ffmpeg -i out.ts -threads 0 -muxdelay 0 -y -map 0:a:1 -codec copy -f segment -segment_time 4 -segment_list_size 0 -segment_list audio-en/audio-en.m3u8 -segment_format mpegts audio-en/audio-en_%d.aac



  


  This creates the required audio segments.

  


  The question

  


  I realise this is quite an ask, but is there anything wrong with our inputs ? Is there a way that this can be done a bit more streamlined ?

  


  Any answers are greatly appreciated.

  


• Convert DTS to AC3 but only if there is no AC3 track already present in container

  24 mars, par Domagoj

  I have sound system that does not support DTS only AC3. I'm automating the process using bash that detects when movie was added to folder, downloads subtitles and converts audio track to AC3 using this command (one part of it) :

  


  ffmpeg -i "{{episode}}" -map 0:v -map 0:a:0 -map 0:a -map 0:s -c:v copy -c:a copy -c:s copy -c:a:0 ac3 -b:a:0 640k "{{directory}}"/{{episode_name}}temp2.mkv

  


  This works without issue and I end up with a .mkv file that contains original DTS audio track and newly created AC3 audio track. The issue is that some files already contain both AC3 and DTS tracks and in those cases I end up with two AC3 tracks and one DTS track. Another issue is that this command is triggered every time there is update to subtitles. So it's possible that the command will execute multiple times in a period of a few days and the container will have X number of the AC3 tracks.

  


  I need a way to detect if file already contains AC3 track before I initiate command from above, but I'm not sure what the command would be. Any help is appreciated !

  


• Feed output of one filter to the input of one filter multiple times with ffmpeg

  27 août 2021, par kentcdodds

  I have the following ffmpeg commands to create a podcast episode :

  


  # remove all silence at start and end of the audio files
ffmpeg -i call.mp3 -af silenceremove=1:0:-50dB call1.mp3
ffmpeg -i response.mp3 -af silenceremove=1:0:-50dB response1.mp3

# remove silence longer than 1 second anywhere within the audio files
ffmpeg -i call1.mp3 -af silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB call2.mp3
ffmpeg -i response1.mp3 -af silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB response2.mp3

# normalize audio files
ffmpeg -i call2.mp3 -af loudnorm=I=-16:LRA=11:TP=0.0 call3.mp3
ffmpeg -i response2.mp3 -af loudnorm=I=-16:LRA=11:TP=0.0 response3.mp3

# cross fade audio files with intro/interstitial/outro
ffmpeg -i intro.mp3 -i call3.mp3 -i interstitial.mp3 -i response3.mp3 -i outro.mp3
  -filter_complex "[0][1]acrossfade=d=1:c2=nofade[a01];
                   [a01][2]acrossfade=d=1:c1=nofade[a02];
                   [a02][3]acrossfade=d=1:c2=nofade[a03];
                   [a03][4]acrossfade=d=1:c1=nofade"
  output.mp3


  


  While this "works" fine, I can't help but feel like it would be more efficient to do this all in one ffmpeg command. Based on what I found online this should be possible, but I don't understand the syntax well enough to know how to make it work. Here's what I tried :

  


  ffmpeg -i intro.mp3 -i call.mp3 -i interstitial.mp3 -i response.mp3 -i outro.mp3
       -af [1]silenceremove=1:0:-50dB[trimmedCall]
       -af [3]silenceremove=1:0:-50dB[trimmedResponse]
       -af [trimmedCall]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceCall]
       -af [trimmedResponse]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceResponse]
       -af [noSilenceCall]loudnorm=I=-16:LRA=11:TP=0.0[call]
       -af [noSilenceResponse]loudnorm=I=-16:LRA=11:TP=0.0[response]
      -filter_complex "[0][call]acrossfade=d=1:c2=nofade[a01];
                       [a01][2]acrossfade=d=1:c1=nofade[a02];
                       [a02][response]acrossfade=d=1:c2=nofade[a03];
                       [a03][4]acrossfade=d=1:c1=nofade"
  output.mp3


  


  But I have a feeling I have a fundamental misunderstanding of this because I got this error which I don't understand :

  


  Stream specifier 'call' in filtergraph description 
[0][call]acrossfade=d=1:c2=nofade[a01];
[a01][2]acrossfade=d=1:c1=nofade[a02];
[a02][response]acrossfade=d=1:c2=nofade[a03];
[a03][4]acrossfade=d=1:c1=nofade
       matches no streams.


  


  For added context, I'm running all these commands through @ffmpeg/ffmpeg so that last command actually looks like this (in JavaScript) :

  


  await ffmpeg.run(
  '-i', 'intro.mp3',
  '-i', 'call.mp3',
  '-i', 'interstitial.mp3',
  '-i', 'response.mp3',
  '-i', 'outro.mp3',
  '-af', '[1]silenceremove=1:0:-50dB[trimmedCall]',
  '-af', '[3]silenceremove=1:0:-50dB[trimmedResponse]',
  '-af', '[trimmedCall]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceCall]',
  '-af', '[trimmedResponse]silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-50dB[noSilenceResponse]',
  '-af', '[noSilenceCall]loudnorm=I=-16:LRA=11:TP=0.0[call]',
  '-af', '[noSilenceResponse]loudnorm=I=-16:LRA=11:TP=0.0[response]',
  '-filter_complex', `
[0][call]acrossfade=d=1:c2=nofade[a01];
[a01][2]acrossfade=d=1:c1=nofade[a02];
[a02][response]acrossfade=d=1:c2=nofade[a03];
[a03][4]acrossfade=d=1:c1=nofade
  `,
  'output.mp3',
)