Opened 7 years ago
Closed 19 months ago
#6638 closed defect (fixed)
aresample does not work for unknown channel layouts
Reported by: | CoRoNe | Owned by: | |
---|---|---|---|
Priority: | normal | Component: | avfilter |
Version: | git-master | Keywords: | aresample |
Cc: | Blocked By: | ||
Blocking: | Reproduced by developer: | yes | |
Analyzed by developer: | no |
Description
FFmpeg used:
ffmpeg version N-86763-g341f012-Reino Copyright (c) 2000-2017 the FFmpeg developers built with gcc 7.1.0 (GCC) configuration: --arch=x86 --target-os=mingw32 --cross-prefix=/cygdrive/m/ffmpeg-windows-build-helpers-master/native_build/windows/ffmpeg_local_builds/sandbox/cross_compilers/mingw-w64-i686/bin/i686-w64-mingw32- --pkg-config=pkg-config --pkg-config-flags=--static --extra-version=Reino --enable-gray --enable-version3 --disable-debug --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-w32threads --disable-schannel --enable-fontconfig --enable-gmp --enable-gnutls --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfdk-aac --enable-libflite --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libzimg --enable-libzvbi --enable-gpl --enable-avisynth --enable-frei0r --enable-filter=frei0r --enable-librubberband --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-avresample --extra-cflags='-march=pentium3' --extra-cflags=-O2 --extra-cflags='-mfpmath=sse' --extra-cflags=-msse --enable-static --disable-shared --prefix=/cygdrive/m/ffmpeg-windows-build-helpers-master/native_build/windows/ffmpeg_local_builds/sandbox/cross_compilers/mingw-w64-i686/i686-w64-mingw32 libavutil 55. 67.100 / 55. 67.100 libavcodec 57.100.104 / 57.100.104 libavformat 57. 75.100 / 57. 75.100 libavdevice 57. 7.100 / 57. 7.100 libavfilter 6. 95.100 / 6. 95.100 libavresample 3. 6. 0 / 3. 6. 0 libswscale 4. 7.101 / 4. 7.101 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100
The last couple of days I've been experimenting with the loudnorm-filter, but only to find out that something fishy is going on with the aresample-filter:
ffmpeg -v 40 -i input.wav -af "loudnorm,aresample=44100" -f null NUL [graph_0_in_0_0 @ 04bc07a0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3 [Parsed_loudnorm_0 @ 04b29540] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_loudnorm_0' [Parsed_aresample_1 @ 04bc0360] Cannot select channel layout for the link between filters Parsed_aresample_1 and format_out_0_0. Error reinitializing filters! Failed to inject frame into filter network: Invalid argument Error while processing the decoded data for stream #0:0
You'd explicitly have to use "aformat=channel_layouts=stereo" or "-ac 2", or it will fail.
ffmpeg -v 40 -i input.wav -af "loudnorm,aresample=44100,aformat=channel_layouts=stereo" -f null NUL [graph_0_in_0_0 @ 04bc04a0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3 [Parsed_loudnorm_0 @ 04b29560] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_loudnorm_0' [auto_resampler_0 @ 04bd2400] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:dbl r:192000Hz [Parsed_aresample_1 @ 04b754c0] ch:2 chl:stereo fmt:dbl r:192000Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz Output #0, null, to 'NUL': Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
For the video I'm trying to encode I use the adelay-filter next. Strange thing is that somehow in this case "aformat=channel_layouts=stereo" isn't needed:
ffmpeg -v 40 -i input.wav -af "loudnorm,aresample=44100,adelay=1000|1000" -f null NUL [graph_0_in_0_0 @ 04bb0b40] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3 [Parsed_loudnorm_0 @ 04b29520] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_loudnorm_0' [format_out_0_0 @ 04bd22c0] auto-inserting filter 'auto_resampler_1' between the filter 'Parsed_adelay_2' and the filter 'format_out_0_0' [auto_resampler_0 @ 04ba0a00] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:dbl r:192000Hz [Parsed_aresample_1 @ 04bc0320] ch:2 chl:stereo fmt:dbl r:192000Hz -> ch:2 chl:stereo fmt:dblp r:44100Hz [auto_resampler_1 @ 04d39ee0] ch:2 chl:stereo fmt:dblp r:44100Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz Output #0, null, to 'NUL': Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
But it gets weirder. The following encoders (that I've tested) seem to accept the adelay-filter's output:
ffmpeg -v 40 -i input.wav -af "loudnorm,aresample=44100,adelay=1000|1000" -c:a {libfdk_aac,libmp3lame,libtwolame,ac3,flac,alac,tta} -f null NUL [graph_0_in_0_0 @ 04bc0220] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3 [graph_0_in_0_0 @ 04bc0520] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3 [Parsed_loudnorm_0 @ 04baf760] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_loudnorm_0' [format_out_0_0 @ 04ba0ba0] auto-inserting filter 'auto_resampler_1' between the filter 'Parsed_adelay_2' and the filter 'format_out_0_0' [auto_resampler_0 @ 04b91760] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:dbl r:192000Hz [Parsed_aresample_1 @ 04b29580] ch:2 chl:stereo fmt:dbl r:192000Hz -> ch:2 chl:stereo fmt:dblp r:44100Hz [auto_resampler_1 @ 04d39e40] ch:2 chl:stereo fmt:dblp r:44100Hz -> ch:2 chl:stereo fmt:s16 r:44100Hz Output #0, null, to 'NUL': Stream #0:0: Audio: aac (libfdk_aac), 44100 Hz, stereo, s16, delay 2048, 128 kb/s Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp, delay 1105 Stream #0:0: Audio: mp2 (libtwolame), 44100 Hz, stereo, fltp, delay 481, 384 kb/s Stream #0:0: Audio: ac3, 44100 Hz, stereo, fltp, delay 256, 192 kb/s Stream #0:0: Audio: flac, 44100 Hz, stereo, s32 (24 bit), 128 kb/s Stream #0:0: Audio: alac, 44100 Hz, stereo, s32p (24 bit), 128 kb/s Stream #0:0: Audio: tta, 44100 Hz, stereo, s32 (24 bit), 128 kb/s
But the following encoders (that I've tested) don't:
ffmpeg -v 40 -i input.wav -af "loudnorm,aresample=44100,adelay=1000|1000" -c:a {libopus,libvorbis,aac,wmav2,wavpack} -f null NUL [graph_0_in_0_0 @ 04bc0520] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3 [Parsed_loudnorm_0 @ 04b29580] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_loudnorm_0' [format_out_0_0 @ 04ba0ba0] auto-inserting filter 'auto_resampler_1' between the filter 'Parsed_adelay_2' and the filter 'format_out_0_0' [Parsed_aresample_1 @ 04cdb420] Cannot select channel layout for the link between filters Parsed_aresample_1 and Parsed_adelay_2. Error reinitializing filters! Failed to inject frame into filter network: Invalid argument Error while processing the decoded data for stream #0:0
They again need "aformat=channel_layouts=stereo" or "-ac 2".
ffmpeg -v 40 -i input.wav -af "loudnorm,aresample=44100,aformat=channel_layouts=stereo,adelay=1000|1000" -c:a {libopus,libvorbis,aac,wmav2,wavpack} -f null NUL [graph_0_in_0_0 @ 04baf7e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3 [Parsed_loudnorm_0 @ 04b29620] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_loudnorm_0' [format_out_0_0 @ 04b91f40] auto-inserting filter 'auto_resampler_1' between the filter 'Parsed_adelay_3' and the filter 'format_out_0_0' [auto_resampler_0 @ 04b83820] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:dbl r:192000Hz [Parsed_aresample_1 @ 04b906a0] ch:2 chl:stereo fmt:dbl r:192000Hz -> ch:2 chl:stereo fmt:dblp r:44100Hz [auto_resampler_1 @ 04bc0040] ch:2 chl:stereo fmt:dblp r:44100Hz -> ch:2 chl:stereo fmt:flt r:48000Hz Output #0, null, to 'NUL': Stream #0:0: Audio: opus (libopus), 48000 Hz, stereo, flt, delay 312, 96 kb/s Stream #0:0: Audio: vorbis (libvorbis), 44100 Hz, stereo, fltp Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, delay 1024, 128 kb/s Stream #0:0: Audio: wmav2, 44100 Hz, stereo, fltp, delay 2048, 128 kb/s Stream #0:0: Audio: wavpack, 44100 Hz, stereo, fltp, 128 kb/s
So it seems (if my judgement is correct) that not only aresample, but also adelay doesn't work for unknown channel layouts.
I guess this issue is quite similar to https://trac.ffmpeg.org/ticket/5218.
Change History (7)
comment:1 by , 7 years ago
comment:2 by , 7 years ago
Version: | unspecified → git-master |
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Today I've had another look at this. (FFmpeg used: ffmpeg-N-90853-g3914a76)
ffmpeg -v 40 -f lavfi -i sine -af "loudnorm" -f null NUL [...] [graph_0_in_0_0 @ 05b35e00] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x4 [Parsed_loudnorm_0 @ 05b35880] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_loudnorm_0' [format_out_0_0 @ 05b36dc0] auto-inserting filter 'auto_resampler_1' between the filter 'Parsed_loudnorm_0' and the filter 'format_out_0_0' [auto_resampler_0 @ 05b37ac0] ch:1 chl:mono fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:dbl r:192000Hz [auto_resampler_1 @ 05b37600] ch:1 chl:mono fmt:dbl r:192000Hz -> ch:1 chl:mono fmt:s16 r:192000Hz
To mimic the same output (sample-format and sample-rate wise):
ffmpeg -v 40 -f lavfi -i sine -af "aformat=dbl:192000" -f null NUL [...] [graph_0_in_0_0 @ 05b36040] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x4 [Parsed_aformat_0 @ 05b358c0] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_aformat_0' [format_out_0_0 @ 05b36dc0] auto-inserting filter 'auto_resampler_1' between the filter 'Parsed_aformat_0' and the filter 'format_out_0_0' [auto_resampler_0 @ 05b37900] ch:1 chl:mono fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:dbl r:192000Hz [auto_resampler_1 @ 05b37440] ch:1 chl:mono fmt:dbl r:192000Hz -> ch:1 chl:mono fmt:s16 r:192000Hz
Adding aresample
to the filter chain shows the same error as with the loudnorm
filter:
ffmpeg -v 40 -f lavfi -i sine -af "aformat=dbl:192000,aresample=44100" -f null NUL [...] [graph_0_in_0_0 @ 05b4ab40] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x4 [Parsed_aformat_0 @ 05b358c0] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_aformat_0' [Parsed_aresample_1 @ 05b360c0] Cannot select channel layout for the link between filters Parsed_aresample_1 and format_out_0_0. Error reinitializing filters! Failed to inject frame into filter network: Invalid argument Error while processing the decoded data for stream #0:0
This can be fixed with ocl=mono
:
ffmpeg -v 40 -f lavfi -i sine -af "aformat=dbl:192000,aresample=44100:ocl=mono" -f null NUL [...] [graph_0_in_0_0 @ 05b4ac40] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x4 [Parsed_aformat_0 @ 05b358c0] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_aformat_0' [auto_resampler_0 @ 05b4c980] ch:1 chl:mono fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:dbl r:192000Hz [Parsed_aresample_1 @ 05b360c0] ch:1 chl:mono fmt:dbl r:192000Hz -> ch:1 chl:mono fmt:s16 r:44100Hz
...but the strange thing is that it appears the channel layout only gets lost with this particular format (double, 192000Hz), because ocl=mono
isn't needed when I only change the sample-format or the sample-rate separately:
ffmpeg -v 40 -f lavfi -i sine -af "aformat=sample_rates=192000,aresample=44100" -f null NUL [...] [graph_0_in_0_0 @ 05b4aa00] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x4 [Parsed_aformat_0 @ 05b358c0] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_aformat_0' [auto_resampler_0 @ 05b4c800] ch:1 chl:mono fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:s16 r:192000Hz [Parsed_aresample_1 @ 05b35e80] ch:1 chl:mono fmt:s16 r:192000Hz -> ch:1 chl:mono fmt:s16 r:44100Hz
ffmpeg -v 40 -f lavfi -i sine -af "aformat=sample_fmts=dbl,aresample=44100" -f null NUL [...] [graph_0_in_0_0 @ 05b4a9c0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x4 [Parsed_aformat_0 @ 05b358c0] auto-inserting filter 'auto_resampler_0' between the filter 'graph_0_in_0_0' and the filter 'Parsed_aformat_0' [auto_resampler_0 @ 05b4c780] ch:1 chl:mono fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:dbl r:44100Hz [Parsed_aresample_1 @ 05b35e80] ch:1 chl:mono fmt:dbl r:44100Hz -> ch:1 chl:mono fmt:s16 r:44100Hz
comment:4 by , 6 years ago
Similar behavior in last commit (Nov 2018):
[root@localhost 2]# /usr/local/ffmpeg-master-nvenc-fdk-aac/bin/ffmpeg -buildconf ffmpeg version N-92394-g75625c5 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-16) configuration: --prefix=/usr/local/ffmpeg-master-nvenc-fdk-aac --pkg-config-flags=--static --extra-cflags=-I/usr/src/ffmpeg_build/include --extra-cflags=-I/usr/local/cuda-9.2/include/ --extra-ldflags=-L/usr/local/cuda-9.2/lib64/ --extra-ldflags=-L/usr/src/ffmpeg_build/lib --extra-libs=-lpthread --extra-libs=-lm --enable-gpl --enable-libx264 --enable-nonfree --enable-cuda --enable-cuvid --enable-nvenc --enable-libnpp --enable-libfdk_aac libavutil 56. 23.101 / 56. 23.101 libavcodec 58. 39.100 / 58. 39.100 libavformat 58. 22.100 / 58. 22.100 libavdevice 58. 6.100 / 58. 6.100 libavfilter 7. 43.100 / 7. 43.100 libswscale 5. 4.100 / 5. 4.100 libswresample 3. 4.100 / 3. 4.100 libpostproc 55. 4.100 / 55. 4.100 configuration: --prefix=/usr/local/ffmpeg-master-nvenc-fdk-aac --pkg-config-flags=--static --extra-cflags=-I/usr/src/ffmpeg_build/include --extra-cflags=-I/usr/local/cuda-9.2/include/ --extra-ldflags=-L/usr/local/cuda-9.2/lib64/ --extra-ldflags=-L/usr/src/ffmpeg_build/lib --extra-libs=-lpthread --extra-libs=-lm --enable-gpl --enable-libx264 --enable-nonfree --enable-cuda --enable-cuvid --enable-nvenc --enable-libnpp --enable-libfdk_aac
# MPEGTS test
/usr/local/ffmpeg-master-nvenc-fdk-aac/bin/ffmpeg -loglevel debug \ -hwaccel cuvid -c:v h264_cuvid \ -y -i /home/trz/content/profile-abr_progresive_hp_r30_hls/avengers60fps.mp4 \ -filter_complex "[v:0]split=2[s0][s1];[0:a]asplit=2[ao1][ao2]" \ -map [s0] -map [ao1] -c:v h264_nvenc -preset hp -b:v 3000k -rc cbr -c:a libfdk_aac -b:a 96k -ar 48.0k -ac 2 \ -f mpegts 3000.ts \ -map [s1] -map [ao2] -c:v h264_nvenc -preset hp -b:v 1000k -rc cbr -c:a libfdk_aac -b:a 96k -ar 48.0k -ac 2 \ -f mpegts 1000.ts 2> out1.log
[root@localhost 2]# mediainfo 1000.ts General ID : 1 (0x1) Complete name : 1000.ts Format : MPEG-TS File size : 104 MiB Duration : 11 min 20 s Overall bit rate mode : Variable Overall bit rate : 1 277 kb/s
[root@localhost 2]# mediainfo 3000.ts General ID : 1 (0x1) Complete name : 3000.ts Format : MPEG-TS File size : 271 MiB Duration : 11 min 20 s Overall bit rate mode : Variable Overall bit rate : 3 337 kb/s
# HLS test
/usr/local/ffmpeg-master-nvenc-fdk-aac/bin/ffmpeg -loglevel debug \ -hwaccel cuvid -c:v h264_cuvid \ -y -i /home/trz/content/profile-abr_progresive_hp_r30_hls/avengers60fps.mp4 \ -filter_complex "[v:0]split=2[s0][s1];[0:a]asplit=2[ao1][ao2]" \ -map [s0] -map [ao1] -c:v h264_nvenc -preset hp -b:v 3000k -rc cbr -c:a libfdk_aac -b:a 96k -ar 48.0k -ac 2 \ -map [s1] -map [ao2] -c:v h264_nvenc -preset hp -b:v 1000k -rc cbr -c:a libfdk_aac -b:a 96k -ar 48.0k -ac 2 \ -f hls \ -var_stream_map "v:0,a:0 v:1,a:1" -hls_list_size 0 -hls_time 9 \ -master_pl_name master.m3u8 vs%v/manifest.m3u8 2> out2.log
[root@localhost 2]# cat master.m3u8 #EXTM3U #EXT-X-VERSION:3 #EXT-X-STREAM-INF:BANDWIDTH=1205600,RESOLUTION=1920x1080,CODECS="avc1.4d402a,mp4a.40.2" vs0/manifest.m3u8 #EXT-X-STREAM-INF:BANDWIDTH=1205600,RESOLUTION=1920x1080,CODECS="avc1.4d402a,mp4a.40.2" vs1/manifest.m3u8
[root@localhost 2]# mediainfo vs0/manifest15.ts General ID : 1 (0x1) Complete name : vs0/manifest15.ts CompleteName_Last : vs0/manifest75.ts Format : MPEG-TS File size : 79.2 MiB Duration : 9 min 3 s Overall bit rate mode : Variable Overall bit rate : 1 222 kb/s
[root@localhost 2]# mediainfo vs1/manifest15.ts General ID : 1 (0x1) Complete name : vs1/manifest15.ts CompleteName_Last : vs1/manifest75.ts Format : MPEG-TS File size : 79.2 MiB Duration : 9 min 3 s Overall bit rate mode : Variable Overall bit rate : 1 222 kb/s
comment:5 by , 6 years ago
Keywords: | unknown channel layouts removed |
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comment:6 by , 19 months ago
Reproduced by developer: | set |
---|---|
Status: | new → open |
Fix on the way to merge.
comment:7 by , 19 months ago
Resolution: | → fixed |
---|---|
Status: | open → closed |
adelay works with unknown channel layouts, you will see that if you run it solely for filtering audio.