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• FFMPEG : Generate 7.1 channel audio file with the longest time of input file

  10 juin 2020, par Anthony

  I want to use ffmpeg to generate 7.1 channel audio file from 8 different audio files.
But I found the output file's duration is decided by the input file with shortest duration.
I didn't find any parameter to auto-pad the shorter audio file or choose the longest duration as final duration.

  



  I already have seen the offlical document as below.
https://ffmpeg.org/ffmpeg-all.html
https://trac.ffmpeg.org/wiki/AudioChannelManipulation
But nothing is helpful.

  



  This is the command I use right now :

  



  ffmpeg -i fl.wav -i fr.wav -i fc.wav -i lfe.wav -i bl.wav -i bl.wav -i sl.wav -i sr.wav -filter_complex "[0:a][1:a][2:a][3:a][4:a][5:a][6:a][7:a]join=inputs=8:channel_layout=7.1[a]" -map "[a]" output.wav

  


• Python : Passing complex (ffmpeg) arguments to Popen

  25 juin 2016, par xaccrocheur

  This ffmpeg Popen invocation works :

  command = ['ffmpeg', '-y',
  
             '-i', filename,
  
             '-filter_complex', 'showwavespic',
  
             '-colorkey', 'red',
  
             '-frames:v', '1',
  
             '-s', '800:30',
  
             '-vsync', '2',
  
             '/tmp/waveform.png']
  
  process = sp.Popen( command, stdin=sp.PIPE, stderr=sp.PIPE)
  
  process.wait()

  But I need to use ’compand, showwavespic’ and this comma seems to be blocking the execution. I also need to pass all sorts of strange characters, like columns and, well, all that you can find in a CLI invocation.

  How can I pass complex arguments ?

• Transcode HLS Segments individually using FFMPEG

  27 mai 2013, par rayh

  I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

  Here is an example ffmpeg command line :

  ffmpeg -threads 1 -nostdin -loglevel verbose \
  
     -nostdin -y -i input.ts -c:a libfdk_aac \
  
     -ac 2 -b:a 64k -y -metadata -vn output.ts

  Inspecting an example sound file shows that there is a gap at the end of the audio :

  

  And the start of the file looks suspiciously attenuated (although this may not be an issue) :

  

  My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

  Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?

  ** UPDATE 1 **

  Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)

  
  

  ** UPDATED 2 **

  So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :

  
  

  I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

  ** UPDATE 3 **

  According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

  For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.