I have been trying to get sidechaining to work with FFMPEG, but my complex
filter skills are lacking.
I'm running two aif files into ffmpeg, hoping the voice would "duck" the
background music when the voice-over is speaking. But I get an error with
this command-line (on Windows, with latest FFMPEG binary):
ffmpeg.exe -i temp_music.aif -i temp_voice.aif -filter_complex
"[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress=threshold=0.1:ratio=5[bg];
[bg][mix]amerge[final]" -map [final] final.mp3
Everything decodes fine:
ffmpeg version N-87353-g183fd30 Copyright (c) 2000-2017 the FFmpeg
developers
built with gcc 7.2.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-cuda
--enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx
--enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig
--enable-frei0r --enable-gnutls --enable-iconv --enable-libass
--enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype
--enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
--enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib
libavutil 55. 76.100 / 55. 76.100
libavcodec 57.106.101 / 57.106.101
libavformat 57. 82.101 / 57. 82.101
libavdevice 57. 8.101 / 57. 8.101
libavfilter 6.105.100 / 6.105.100
libswscale 4. 7.103 / 4. 7.103
libswresample 2. 8.100 / 2. 8.100
libpostproc 54. 6.100 / 54. 6.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, aiff, from 'temp_music.aif':
Duration: 00:01:26.68, start: 0.000000, bitrate: 1536 kb/s
Stream #0:0: Audio: pcm_s16be, 48000 Hz, stereo, s16, 1536 kb/s
Guessed Channel Layout for Input Stream #1.0 : stereo
Input #1, aiff, from 'temp_voice.aif':
Duration: 00:01:26.68, start: 0.000000, bitrate: 1536 kb/s
Stream #1:0: Audio: pcm_s16be, 48000 Hz, stereo, s16, 1536 kb/s
Output with label 'final' does not exist in any defined filter graph, or
was already used elsewhere.
But then I receive an error:
Output with label 'final' does not exist in any defined filter graph, or
was already used elsewhere.
Anybody knows why?
ᐧ
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