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• ffmpeg replace central channel with ffmpeg

  24 mai 2020, par Rafal B

  I'm trying to replace central channel in the movie (Movie-A) with another audio stream from another movie (Movie-B).
I was able to extract stereo stream from Movie-B :

  



  ffmpeg.exe" -i "Movie-B.mkv"

Stream #0:1(pol): Audio: ac3, 48000 Hz, stereo, fltp, 320 kb/s (default)


  



  and convert it to mono stream :

  



  ffmpeg.exe" -i "Movie-B.mkv" -vn -ab 320k -ac 1 "Movie-B-Audio.ac3"
Output #0, ac3, to 'Movie-B-Audio.ac3':
Stream #0:0(pol): Audio: ac3, 48000 Hz, mono, fltp, 320 kb/s (default)


  



  For Movie-A :

  



  ffmpeg.exe" -i "Movie-B.mkv"
Stream #0:1(eng): Audio: aac (LC), 48000 Hz, 5.1, fltp (default)


  



  so Layout is 5.1 (FL+FR+FC+LFE+BL+BR), not 5.1(side) (FL+FR+FC+SL+SR)

  



  Media Info shows :

  



  Audio
ID: 2
Format: AAC
Format info: Advanced Audio Codec Low Complexity
Commercial name: AAC
Codec ID: A_AAC-2
Bit rate: 426 kb/s
Channel(s): 6 channels
Channel positions: Front: L C R, Side: L R, LFE
Channel layout: C L R Ls Rs LFE


  



  I have extracted all 6 channels into one stream with containing 6 channels :

  



  ffmpeg.exe" -i "Movie-A.mkv" -vn -ab 320k -ac 6 "Movie-A-Audio.ac3"
Output #0, ac3, to 'Movie-A-Audio.ac3':
Stream #0:0(eng): Audio: ac3, 48000 Hz, 5.1, fltp, 320 kb/s (default)


  



  But at this stage I'm getting different Layout in Movie-A-Audio.ac3 :

  



  Audio
Format                                   : AC-3
Format/Info                              : Audio Coding 3
Commercial name                          : Dolby Digital
Duration                                 : 2 h 16 min
Bit rate mode                            : Constant
Bit rate                                 : 320 kb/s
Channel(s)                               : 6 channels
Channel layout                           : L R C LFE Ls Rs


  



  In source Movie-A.mkv it is : Channel layout : C L R Ls Rs LFE

  



  In output Movie-A-Audio.ac3 it is : Channel layout : L R C LFE Ls Rs

  



  Then I tried to extract those 6 channels into separate mono-stream files :

  



  ffmpeg.exe" -i "Movie-A-Audio.ac3" -filter_complex "channelsplit=channel_layout=5.1[FL][FR][FC][LFE][BL][BR]" -ab 320k -ac 1 -map "[FL]" extr_front_left.ac3 -ab 320k -ac 1 -map "[FR]" extr_front_right.ac3 -ab 320k -ac 1 -map "[FC]" extr_front_center.ac3 -ab 320k -ac 1 -map "[LFE]" extr_subwoofer.ac3 -ab 320k -ac 1 -map "[BL]" extr_back_left.ac3 -ab 320k -ac 1 -map "[BR]" extr_back_right.ac3


  



  With result :

  



  Stream mapping:
  Stream #0:0 (ac3) -> channelsplit
  channelsplit:FL -> Stream #0:0 (ac3)
  channelsplit:FR -> Stream #1:0 (ac3)
  channelsplit:FC -> Stream #2:0 (ac3)
  channelsplit:LFE -> Stream #3:0 (ac3)
  channelsplit:BL -> Stream #4:0 (ac3)
  channelsplit:BR -> Stream #5:0 (ac3)
Press [q] to stop, [?] for help
Output #0, ac3, to 'extr_front_left.ac3':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #0:0: Audio: ac3, 48000 Hz, mono, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 ac3
Output #1, ac3, to 'extr_front_right.ac3':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #1:0: Audio: ac3, 48000 Hz, mono, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 ac3
Output #2, ac3, to 'extr_front_center.ac3':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #2:0: Audio: ac3, 48000 Hz, mono, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 ac3
Output #3, ac3, to 'extr_subwoofer.ac3':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #3:0: Audio: ac3, 48000 Hz, mono, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 ac3
Output #4, ac3, to 'extr_back_left.ac3':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #4:0: Audio: ac3, 48000 Hz, mono, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 ac3
Output #5, ac3, to 'extr_back_right.ac3':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #5:0: Audio: ac3, 48000 Hz, mono, fltp, 320 kb/s
    Metadata:
      encoder         : Lavc58.54.100 ac3


  



  When I'm checking channels layouts of those files extracted I see each of them is having Channel Layout marked with Central, for instant front_left.ac3 :

  



  Audio
Format                                   : AC-3
Format/Info                              : Audio Coding 3
Commercial name                          : Dolby Digital
Duration                                 : 2 h 16 min
Bit rate mode                            : Constant
Bit rate                                 : 320 kb/s
Channel(s)                               : 1 channel
Channel layout                           : C
Sampling rate                            : 48.0 kHz


  



  Then I try to join needed channels :

  



  ffmpeg.exe" -i extr_front_left.ac3 -i extr_front_right.ac3 -i Movie-B-Audio.ac3 -i extr_back_left.ac3 -i extr_back_right.ac3 -i extr_subwoofer.ac3 -filter_complex "[0:a][1:a][2:a][3:a][4:a][5:a]join=inputs=6:channel_layout=5.1[a]" -map "[a]" "final-output.ac3


  



  But it is completely messed up : central channel (voice) is heard in front right speaker, in back right speaker is silence (or maybe it is subwoofer ?)

  



  Target is easy :

  



  

  1. convert source stereo central audio channel in Movie-B.mkv to mono channel
and

  


  2. insert it as central channel into Movie-A.mkv that have 5.1 Audio stream with layout C L R Ls Rs LFE

  


  



  So far I was able to complete 1. but I have a problem with mappings.
Can someone try to guide me how to do that ?

  


• Low latency routing of rtp input to rtsp output with ffmpeg on a server

  29 mai 2020, par guillefix

  I want to be able to do some simple low-latency screen share. I know peer-to-peer would be the lowest latency, but it seems using an intermediate server is a lot easier to setup. I have found this awesome little library, which sets up an RTSP server, which I'm running on my digital ocean server. I then :

  



  

  1. Set up OBS to stream using RTP to the server on port 8558 with libx264 encoding.

  


  2. Run ffmpeg -re -stream_loop -1 -i rtp://127.0.0.1:8558 -c:v libx264 -c:a aac -f rtsp rtsp://localhost:8554/mystream on the server.

  


  3. Open rtsp://<server ip="ip">:8554/mystream</server> on VLC.

  


  



  However, the latency seems to be quite high. With my crappy internet it must have been like half a minute. A friend with better internet saw it fluctuating between 4-15 seconds. Furthermore, there seems to be a lot of artifacts on the video (problems with encoding ? I'm not sure why these happen ?)

  



  I attach below my OBS settings, and an example of artifacts.

  



  My question is : is there some settings on OBS and on ffmpeg that would allow this to have as low end-to-end latency as possible, while not having too many bad artifacts ? I'm not very well versed on video encoding and streaming, so this is all quite new to me. I'm willing to learn !

  



  

  



  

  


• Is there any way that I can speed up the ffmpeg processing time

  20 mai 2020, par Ahmed Al-Rayan

  I am facing a problem with the processing process. I use a real joint server in a digital hosting package of $ 10 and use cloud service from Amazon s3. The problem is when uploading a video, whatever the size of the video, whether its size is 1 megabyte or 2 Giga. After the upload process, the processing process starts to upload, there is no problem But when the processing process takes a very long time so that I cannot complete it, what is the solution to that, is there a problem for me or is this process normal ?
 I use laravel-ffmpeg and through laravel queue I am cutting the video into several qualities I will attach the code to you below.

  



  public function handle()
{
    //180p
    $lowBitrate1 = (new X264('aac'))->setKiloBitrate(613);
    //270p
    $lowBitrate2 = (new X264('aac'))->setKiloBitrate(906);
    //360p
    $midBitrate1 = (new X264('aac'))->setKiloBitrate(1687);
    //540p
    $midBitrate2 = (new X264('aac'))->setKiloBitrate(2227);
    //720p
    $highBitrate1 = (new X264('aac'))->setKiloBitrate(4300);
    //1080
    $highBitrate2 = (new X264('aac'))->setKiloBitrate(7917);

FFMpeg::fromDisk('s3')
    ->open($this->movie->path)
    ->exportForHLS()
    ->onProgress(function ($percent) {
        $this->movie->update([
            'percent' => $percent
        ]);
    })
    ->setSegmentLength(10)// optional
    ->addFormat($lowBitrate1)
    ->addFormat($lowBitrate2)
    ->addFormat($midBitrate1)
    ->addFormat($midBitrate2)
    ->addFormat($highBitrate1)
    ->addFormat($highBitrate2)
    ->toDisk('s3')
    ->save("public/Movies/{$this->movie->id}/{$this->movie->id}.m3u8");
}//end of handle