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• How to split audio file into equal-length segments with ffmpeg ?

  11 mars, par GPWR

  I want to split an audio file into several equal-length segments using FFmpeg. I want to specify the general segment duration (no overlap), and I want FFmpeg to render as many segments as it takes to go over the whole audio file (in other words, the number of segments to be rendered is unspecified).
Also, since I am not very experienced with FFmpeg (I only use it to make simple file conversions with few arguments), I would like a description of the code you should use to do this, rather than just a piece of code that I won't necessarily understand, if possible.
Thank you in advance.

  


  P.S. Here's the context for why I'm trying to do this :
I would like to sample a song into single-bar loops automatically, instead of having to chop them manually using a DAW. All I want to do is align the first beat of the song to the beat grid in my DAW, and then export that audio file and use it to generate one-bar loops in FFmpeg.

  


  In the future, I will try to do something like a batch command in which one can specify the tempo and key signature, and it will generate the loops using FFmpeg automatically (as long as the loop is aligned to the beat grid, as I've mentioned earlier). 😀

  


• Audacity vocal removal failed when ffmpeg-conversion was involved

  10 mars 2018, par fyang

  I downloaded some songs coded with FLAC, and Audacity could remove the vocals quite well.

  When I downloaded songs coded with ALAC, I must use ffmpeg to convert them to some other forms because Audacity didn’t recognise .m4a files.

  I used the command ffmpeg -i "song 01.m4a" -f flac "song 01.flac". Now Audacity could load the song, but its vocal removal failed to remove the vocals.

  I tried again with this command in order to be precise, ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac", and vocal removal did not work either.

  I tried to do it manually by splitting, inverting and changing both channels to mono, but the vocals were still there.

  I think the problem lies with the ffmpeg conversion step. Is there any fix ? Thanks !

  Below is the result of the conversion :

  ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac"
  
  ffmpeg version N-90143-gb6652f5100 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libmfx --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
  
    libavutil      56.  7.101 / 56.  7.101
  
    libavcodec     58. 12.102 / 58. 12.102
  
    libavformat    58.  9.100 / 58.  9.100
  
    libavdevice    58.  2.100 / 58.  2.100
  
    libavfilter     7. 12.100 /  7. 12.100
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0000019f2b258000] stream 0, timescale not set
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'song 01.m4a':
  
    Metadata:
  
      major_brand     : M4A
  
      minor_version   : 0
  
      compatible_brands: M4A mp42isom
  
      creation_time   : 2009-12-27T00:15:23.000000Z
  
      track           : 1/10
  
      genre           : 
  
      album           : 
  
      artist          : 
  
      comment         : ExactAudioCopy v0.95b4
  
      DISCID          : 
  
      iTunNORM        :  00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
  
      title           : song 01
  
      encoder         : iTunes 9.0.2.25
  
      date            : 2005
  
      album_artist    : 
  
      lyrics          : 
  
    Duration: 00:08:10.84, start: 0.000000, bitrate: 921 kb/s
  
      Stream #0:0(und): Audio: alac (alac / 0x63616C61), 44100 Hz, stereo, s16p, 920 kb/s (default)
  
      Metadata:
  
        creation_time   : 2009-12-27T00:15:23.000000Z
  
      Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 300x300 [SAR 100:100 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (alac (native) -> flac (native))
  
  Press [q] to stop, [?] for help
  
  [Parsed_pan_0 @ 0000019f2b2a6fc0] Pure channel mapping detected: 0 1
  
  Output #0, flac, to 'song 01.flac':
  
    Metadata:
  
      major_brand     : M4A
  
      minor_version   : 0
  
      compatible_brands: M4A mp42isom
  
      lyrics          : 
  
      TRACKNUMBER     : 1/10
  
      genre           : 
  
      album           : 
  
      artist          : 
  
      DESCRIPTION     : ExactAudioCopy v0.95b4
  
      DISCID          : 
  
      iTunNORM        :  00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
  
      title           : song 01
  
      ALBUMARTIST     : 
  
      date            : 2005
  
      encoder         : Lavf58.9.100
  
      Stream #0:0(und): Audio: flac, 44100 Hz, stereo, s16, 128 kb/s (default)
  
      Metadata:
  
        creation_time   : 2009-12-27T00:15:23.000000Z
  
        encoder         : Lavc58.12.102 flac
  
  size=   54518kB time=00:08:10.84 bitrate= 909.9kbits/s speed=  35x
  
  video:0kB audio:54508kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.018294%

• Change audio speed using ffmpeg node js

  26 juillet 2022, par Getwrong

  how do i change the speed of the song output using this function :

  



  function addAudio()
{
      console.log("adding audio...");

      ffmpeg()
      .videoCodec('libx264')
      .format('mp4')
      .outputFormat('mp4')
      .input(song)
      .input(video)
      .output(output1)
      .on('end', () => {                    
          resolve(output1);
      }).on('error', (_err) => {
          reject(_err);
      }).run();
}


  



  https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20video

  



  this shows command line input but not sure how to use it in js function ?