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?
This looks to be some type of character encoding issue.
When I execute your command as copy-pasted, I get the same error. But if I remove the final label and retype it, it works. For now, since you have only one stream to export, you can just leave it unlabelled and ffmpeg will automatically map it.
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.mp3
Related
I'm trying to decode 2 Dolby-E audio streams simultaneously.
ffmpeg -non_pcm_mode copy -i C:\vlc-record.ts -map 0:3 -map 0:4 -c:a pcm_s24le -f s24le - | ffplay -
Using -map 0:3 it works for one at a time but when I add the second one -map 0:4 it only process the first one and there is just one stream on my dolby_e output pipe.
Is there a way to create a two streams multicanal non-pcm output from 2 Dolbys PCM streams ?
I tried to find a solution with named pipe but it seems like windows command line doesn't accept this. I also tried to merge the PCM streams but then it says Invalid data found when processing input.
I can't make it with 2 pass because I'll be using it with udp live video streams.
ffmpeg version 2022-03-07-git-e645a1ddb9-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
built with gcc 11.2.0 (Rev7, Built by MSYS2 project)
configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
libavutil 57. 22.100 / 57. 22.100
libavcodec 59. 22.100 / 59. 22.100
libavformat 59. 17.103 / 59. 17.103
libavdevice 59. 5.100 / 59. 5.100
libavfilter 8. 27.100 / 8. 27.100
libswscale 6. 5.100 / 6. 5.100
libswresample 4. 4.100 / 4. 4.100
libpostproc 56. 4.100 / 56. 4.100
ffplay version 2022-03-07-git-e645a1ddb9-full_build-www.gyan.dev Copyright (c) 2003-2022 the FFmpeg developers
built with gcc 11.2.0 (Rev7, Built by MSYS2 project)
configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
libavutil 57. 22.100 / 57. 22.100
libavcodec 59. 22.100 / 59. 22.100
libavformat 59. 17.103 / 59. 17.103
libavdevice 59. 5.100 / 59. 5.100
libavfilter 8. 27.100 / 8. 27.100
libswscale 6. 5.100 / 6. 5.100
libswresample 4. 4.100 / 4. 4.100
libpostproc 56. 4.100 / 56. 4.100
Input #0, mpegts, from 'C:\vlc-record.ts':
Duration: 00:00:15.76, start: 42530.445511, bitrate: 31505 kb/s
Program 1
Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn
Stream #0:1: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 256 kb/s
Stream #0:2: Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 256 kb/s
Stream #0:3: Audio: s302m (BSSD / 0x44535342), stereo, s32 (20 bit), 2304 kb/s
Stream #0:4: Audio: s302m (BSSD / 0x44535342), stereo, s32 (20 bit), 2304 kb/s
Stream mapping:
Stream #0:3 -> #0:0 (s302m (native) -> pcm_s24le (native))
Press [q] to stop, [?] for help
Output #0, s24le, to 'pipe:':
Metadata:
encoder : Lavf59.17.103
Stream #0:0: Audio: pcm_s24le, 48000 Hz, stereo, s32 (20 bit), 2304 kb/s
Metadata:
encoder : Lavc59.22.100 pcm_s24le
[dolby_e # 0000026b16da0640] Stream has 2 programs (configuration 0), channels will be output in native order.
Input #0, s337m, from 'pipe:':
Duration: N/A, bitrate: N/A
Stream #0:0: Audio: dolby_e, 44800 Hz, 7.1, fltp
[dolby_e # 0000026b16d7ca80] Stream has 2 programs (configuration 0), channels will be output in native order.
1.25 M-A: 0.000 fd= 0 aq= 292KB vq= 0KB sq= 0B f=0/0
av_interleaved_write_frame(): Broken pipe
Error writing trailer of pipe:: Broken pipe
size= 1451kB time=00:00:05.20 bitrate=2286.3kbits/s speed=3.07x
video:0kB audio:1462kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Error closing file pipe:: Broken pipe
Conversion failed!
While concatenating audio files with
ffmpeg -safe 0 -f concat
-segment_time_metadata 1
-i {m3u_file_name}
-vf select=concatdec_select
-af aselect=concatdec_select,aresample=async=1 {ogg_file_name}
I get the following output
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, concat, from 'fm3u.m3u':
Duration: N/A, start: -0.006500, bitrate: N/A
Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
Stream mapping:
Stream #0:0 -> #0:0 (opus (native) -> vorbis (libvorbis))
Press [q] to stop, [?] for help
Output #0, ogg, to 'fout.ogg':
Metadata:
encoder : Lavf57.83.100
Stream #0:0: Audio: vorbis (libvorbis), 48000 Hz, mono, fltp
Metadata:
encoder : Lavc57.107.100 libvorbis
[opus # 0x558b2c245400] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus # 0x558b2c245400] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus # 0x558b2c245400] Error parsing the packet header.
...
...
Error while decoding stream #0:0: Invalid data found when processing input
[opus # 0x558b2c245400] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
size= 47kB time=00:00:10.84 bitrate= 35.2kbits/s speed=4.38x
video:0kB audio:42kB subtitle:0kB other streams:0kB global headers:3kB muxing overhead: 10.251080%
/fout.ogg
The final audio sounds good.
ffprobe on the 2 participating files looks like
# ffprobe <FILE>
ffprobe version 3.4.8-0ubuntu0.2 Copyright (c) 2007-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, ogg, from 'f1.ogg':
Duration: 00:00:01.08, start: 0.000000, bitrate: 34 kb/s
Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
# ffprobe <FILE>
ffprobe version 3.4.8-0ubuntu0.2 Copyright (c) 2007-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, ogg, from 'f2.ogg':
Duration: 00:00:05.00, start: 0.000000, bitrate: 6 kb/s
Stream #0:0: Audio: vorbis, 48000 Hz, mono, fltp, 80 kb/s
Metadata:
ENCODER : Lavc57.107.100 libvorbis
What is the reason of the error?
As explained in this answer
"file-level concatenation (-f concat) requires all files in the listing to have the exact same codec parameters. "
In this case, the input files have codecs opus and vorbis.
To resolve the issue, I recreated some files (silence files) so they all have the same CODEC as the files I am mixing them with.
In particular, instead of creating them using:
ffmpeg -f lavfi -i anullsrc=r=48000:cl=mono -t 5.8 -c:a libvorbis silence-5.8.ogg
I use now (note libopus)
ffmpeg -f lavfi -i anullsrc=r=48000:cl=mono -t 5.8 -c:a libopus silence-5.8.ogg
And the "concat with ffmpeg" works without any problem.
I also tried
ffmpeg -f lavfi -i anullsrc=r=48000:cl=mono -t 5.8 -c:a opus silence-5.8.ogg
But that returns with
[opus # 0x5607f1765900] The encoder 'opus' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it.
[opus # 0x5607f1765900] Alternatively use the non experimental encoder 'libopus'.
So, libopus works just fine (no need for opus)
I am creating a program that takes an audio file, breaks it up into 30-second audio files, and sends those files to Google's Speech Recognition API to transcribe the audio. I'm using FFmpeg to break up the original audio file and make the 30-second files. It was working for about 20 minutes and I didn't change anything but now it's giving me an error saying that no such file or directory exists.
Here is the error it gives me:
C:\Users\hmkur\Desktop\Python\Transcribing_Audio_GoogleAPI_Python>ffmpeg -i source/valve.flac -f segment -segment_time 30 -c copy parts/out%01d.flac
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9.1.1 (GCC) 20190807
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --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-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'source/valve.flac':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2mp41
encoder : Lavf58.29.100
Duration: 00:11:42.54, start: 0.000000, bitrate: 129 kb/s
Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : ISO Media file produced by Google Inc.
[segment # 000001f8e84bd100] Opening 'parts/out0.flac' for writing
[segment # 000001f8e84bd100] Failed to open segment 'parts/out0.flac'
Could not write header for output file #0 (incorrect codec parameters ?): No such file or directory
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Last message repeated 1 times
Both the folder "source" with "valve.flac" exists as well as the folder "parts" and the files "out0.flac", etc. I was originally trying in the wav file format, switched to flac and it worked for about 20 minutes then stopped working.
Here is the command I used and the output when I try and parse the .wav file into 30-second clips:
C:\Users\hmkur>ffmpeg -i C:\Users\hmkur\Desktop\Python\Transcribing_Audio\source\valve.wav -f segment -segment_time 30 -c copy parts/out%09d.wav
ffmpeg version git-2019-11-05-c54268c Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9.2.1 (GCC) 20191010
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --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-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
libavutil 56. 35.101 / 56. 35.101
libavcodec 58. 60.100 / 58. 60.100
libavformat 58. 34.101 / 58. 34.101
libavdevice 58. 9.100 / 58. 9.100
libavfilter 7. 66.100 / 7. 66.100
libswscale 5. 6.100 / 5. 6.100
libswresample 3. 6.100 / 3. 6.100
libpostproc 55. 6.100 / 55. 6.100
[wav # 0000023133098a00] Discarding ID3 tags because more suitable tags were found.
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'C:\Users\hmkur\Desktop\Python\Transcribing_Audio\source\valve.wav':
Metadata:
title : valve
encoder : Lavf58.20.100 (libsndfile-1.0.24)
Duration: 00:06:47.20, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
[segment # 00000231334d18c0] Opening 'parts/out000000000.wav' for writing
[segment # 00000231334d18c0] Failed to open segment 'parts/out000000000.wav'
Could not write header for output file #0 (incorrect codec parameters ?): No such file or directory
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Last message repeated 1 times
Here is the error I am getting, I took a 7.1 AAC file, and remixed one of the tracks, and now I am trying to export back to the original format.
ffmpeg -i - -strict experimental -c:a aac -ac 8 -b:a 240k "F:\Something.aac"
ffmpeg version 2.2.2 Copyright (c) 2000-2014 the FFmpeg developers
built on May 22 2014 19:56:44 with gcc 4.8.2 (GCC)
configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-
libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
libavutil 52. 66.100 / 52. 66.100
libavcodec 55. 52.102 / 55. 52.102
libavformat 55. 33.100 / 55. 33.100
libavdevice 55. 10.100 / 55. 10.100
libavfilter 4. 2.100 / 4. 2.100
libswscale 2. 5.102 / 2. 5.102
libswresample 0. 18.100 / 0. 18.100
libpostproc 52. 3.100 / 52. 3.100
Guessed Channel Layout for Input Stream #0.0 : 7.1
Input #0, wav, from 'pipe:':
Duration: N/A, bitrate: 5644 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 7.1, s16, 5644 kb/s
[aac # 02567320] Unsupported number of channels: 8
Output #0, adts, to 'F:\Something.aac':
Stream #0:0: Audio: aac, 44100 Hz, 7.1, fltp, 240 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le -> aac)
Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Conversion failed!
Your ffmpeg is too old.
You can get a new build for Windows at Zeranoe. Users of other OSs can find links to builds at the FFmpeg Download page.