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• SDL2 won't play with more than 6 audio channels

  13 juin 2020, par Hiko Haieto

  I am trying to stream (raw) video and audio from a capture device as part of my home media setup (with my pc acting similarly to a receiver in a typical home theatre setup), but the biggest problem I haven't been able to get past is that I haven't been able to get ffplay (using SDL2 as its audio backend) to work with all 8 channels in 7.1 streams - two simply get dropped, despite it recognising 8 channel input or me specifying a 7.1 layout.

  



  I have been able to confirm that all 8 channels are present in the source by first using ffmpeg to save the output of a speaker test to a file and playing that back with both mplayer (which works) and ffplay (which doesn't). I also wrote some minimal code to play the audio directly through SDL's API with the same result, so it's not the fault of ffplay. I might simply use mplayer if it weren't for the fact that piping output from ffmpeg adds too much latency for real-time use. I am using libSDL version 2.0.12 and ffplay 4.2.3, both of which are the latest at the time of writing and are ostensibly supposed to support 7.1 audio.

  



  Using output recorded from speaker-test -c 8, I am using the following to play it back in mplayer :

  



  mplayer -channels 8 -rawaudio channels=8 -format s16le -demuxer rawaudio speaker-test.pcm


  



  and the following to play it back in ffplay :

  



  ffplay -f s16le -ac 8 -af 'channelmap=channel_layout=7.1' speaker-test.pcm


  



  No matter what I try, the two side channels get dropped. I couldn't figure out how to play raw pcm in SDL, so I repeated the same tests with wav output and used the following code to play it back :

  



  #include <sdl2></sdl2>SDL.h>&#xA;&#xA;int main(int argc, char **argv) {&#xA;    SDL_Init(SDL_INIT_AUDIO);&#xA;    SDL_AudioSpec wavSpec;&#xA;    Uint32 wavLength;&#xA;    Uint8 *wavBuffer;&#xA;    SDL_LoadWAV("speaker-test.wav", &amp;wavSpec, &amp;wavBuffer, &amp;wavLength);&#xA;    SDL_AudioDeviceID deviceID = SDL_OpenAudioDevice(NULL, 0, &amp;wavSpec, NULL, 0);&#xA;    SDL_QueueAudio(deviceID, wavBuffer, wavLength);&#xA;    SDL_PauseAudioDevice(deviceID, 0);&#xA;    SDL_Delay(30000);&#xA;    SDL_CloseAudioDevice(deviceID);&#xA;    SDL_FreeWAV(wavBuffer);&#xA;    SDL_Quit();&#xA;    return 0;&#xA;}&#xA;

  



  The above code exhibits the same behaviour of dropping the two additional side channels, despite it being the latest version of SDL that should have supported 7.1 for many releases now. Why might this be happening, and how might I fix it ?

  


• FFmpeg - convert 2 audio tracks from video to 5.1 audio, (play video with different languages to different devices) [closed]

  10 février, par Sandre

  How to watch a movie with one language playing through the speakers and another through the headphones ?

  


  Disclaimer : I know nothing about audio conversion, and don't want to study ffmpeg. I spend a few hours searching how to do, actually much more than I want. I found a bunch of questions from different people and not a single working solution, so I made a clunky but working solution. If someone helps me make it more elegant, I'll be happy. If my question just gets downvoted like most ffmpeg newbie questions, it probably deserves it. And I hope my question can help people who want enjoy video with 2 different languages.

  


  A clumsy but working solution.

  


  


  1. Setup Aggregate Audio Device to play 2 channels of 5.1 through speakers and 2 through bluetooth headphones. (On screenshot Audio MIDI Setup for MacOS)


     


  


  2. Use ffmpeg to convert 2 audio tracks into 5.1 audio.

     


  


  3. Play video with new external audio track.

     


  


  


  # print list of channels
ffprobe INPUT.mkv 2>&1 >/dev/null | grep Stream

--- sample output ---
    Stream #0:0(eng): Video: h264 (High), yuv420p(progressive), 1280x544, SAR 1:1 DAR 40:17, 23.98 fps, 23.98 tbr, 1k tbn (default)
    Stream #0:1(rus): Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s (default)
    Stream #0:2(eng): Audio: ac3, 48000 Hz, 5.1(side), fltp, 640 kb/s

# extract audio
ffmpeg -i INPUT.mkv -map 0:1 -acodec copy ru.ac3
ffmpeg -i INPUT.mkv -map 0:2 -acodec copy en.ac3

# extract only front_center channel from 5.1 - speach
ffmpeg -i en.ac3 -filter_complex "channelsplit=channel_layout=5.1:channels=FC[center]" -map "[center]" en_front_center.wav
ffmpeg -i ru.ac3 -filter_complex "channelsplit=channel_layout=5.1:channels=FC[center]" -map "[center]" ru_front_center.wav

# join to 5.1
ffmpeg -i en_front_center.wav -i ru_front_center.wav -filter_complex "[0:a][0:a][0:a][0:a][1:a][1:a]join=inputs=6:channel_layout=5.1[a]" -map "[a]" output.wav


  


  Is it possible to avoid re-encoding the audio and copying the same channel many times to reduce the file size ?

  


• FFMpeg - Adding Timecode track messes with video output

  19 octobre 2022, par Muindor

  I'm using FFMpeg to setup some automation.

  


  I want to pull in a PNG, repeat it for 5 seconds, then add on an mov file to the end. I have that working great. The problem comes when I add a timecode to the output. The output video remains the correct length, but the PNG only repeats for 24 frames instead of 120, then the mov gets added, then black frames fill the rest.

  


  This is the command I'm using. Without the timecode flag it works perfectly.

  


  ./ffmpeg  -loop 1 -t 5 -r 24 -i 'image.png' -i 'end.mov' -filter_complex "[0]setdar=16/9[a];[1]setdar=16/9[b];[a][b]concat=n=2:v=1" -vsync 2  -r 24 -vcodec prores -timecode 00:59:53:00 output.mov


  


  Just a note, I don't want any any audio track.

  


  Thanks in advance !