#5121 closed defect (fixed)
Opus initial padding incorrectly written to mkv for sample rates != 48kHz
| Reported by: | pszemus | Owned by: | |
|---|---|---|---|
| Priority: | normal | Component: | avformat |
| Version: | git-master | Keywords: | mkv opus |
| Cc: | Blocked By: | ||
| Blocking: | Reproduced by developer: | no | |
| Analyzed by developer: | no |
Description
Summary of the bug:
As reported in https://bugzilla.mozilla.org/show_bug.cgi?id=1227153 trying to create a 24kHz opus@webm file produces a file that has 48kHz written to an opus header (?). What's strange is that MediaInfo shows this sample rate correctly.
How to reproduce:
$ ffmpeg -i http://www.soundsnap.com/audio/play/72210%3A1448291371%3Ab4326304c013a3a61b3b87d83158c91b -c:a libopus -b:a 64k -ar 24000 /tmp/tmp.webm
ffmpeg version N-77704-g68eb208 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
configuration: --prefix=/home/pszemus/ffmpeg/build --enable-pic --enable-pthreads --enable-libmp3lame --enable-version3 --enable-libfaac --enable-gpl --enable-nonfree --enable-libvpx --enable-libvorbis --enable-libopus --enable-libx264 --enable-libwebp --enable-protocol=https --enable-openssl --extra-cflags=-I/opt/WP/common.libs/include --extra-ldflags='-L/opt/WP/common.libs/lib -ldl'
libavutil 55. 12.100 / 55. 12.100
libavcodec 57. 21.100 / 57. 21.100
libavformat 57. 21.100 / 57. 21.100
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 23.100 / 6. 23.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
[mp3 @ 0x2ae63a0] Skipping 0 bytes of junk at 0.
[mp3 @ 0x2ae63a0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'http://www.soundsnap.com/audio/play/72210%3A1448291371%3Ab4326304c013a3a61b3b87d83158c91b':
Metadata:
title : PleaseRefesh2015_1
date : 2015
track : 1
Duration: 00:00:04.13, start: 0.000000, bitrate: 192 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
Output #0, webm, to '/tmp/tmp.webm':
Metadata:
title : PleaseRefesh2015_1
date : 2015
track : 1
encoder : Lavf57.21.100
Stream #0:0: Audio: opus (libopus), 24000 Hz, stereo, s16, 64 kb/s
Metadata:
encoder : Lavc57.21.100 libopus
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (native) -> opus (libopus))
Press [q] to stop, [?] for help
size= 37kB time=00:00:04.13 bitrate= 73.7kbits/s speed=15.4x
video:0kB audio:35kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 5.376523%
$ ffmpeg -i /tmp/tmp.webm
ffmpeg version N-77704-g68eb208 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
configuration: --prefix=/home/pszemus/ffmpeg/build --enable-pic --enable-pthreads --enable-libmp3lame --enable-version3 --enable-libfaac --enable-gpl --enable-nonfree --enable-libvpx --enable-libvorbis --enable-libopus --enable-libx264 --enable-libwebp --enable-protocol=https --enable-openssl --extra-cflags=-I/opt/WP/common.libs/include --extra-ldflags='-L/opt/WP/common.libs/lib -ldl'
libavutil 55. 12.100 / 55. 12.100
libavcodec 57. 21.100 / 57. 21.100
libavformat 57. 21.100 / 57. 21.100
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 23.100 / 6. 23.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Input #0, matroska,webm, from '/tmp/tmp.webm':
Metadata:
title : PleaseRefesh2015_1
encoder : Lavf57.21.100
Duration: 00:00:04.14, start: 0.007000, bitrate: 73 kb/s
Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp (default)
At least one output file must be specified
$ mediainfo /tmp/tmp.webm General Unique ID : 23699742163318188445154506414781705045 (0x11D46780E7425B3B392AC806ACE92B55) Complete name : /tmp/tmp.webm Format : WebM Format version : Version 4 / Version 2 File size : 37.2 KiB Duration : 4s 135ms Overall bit rate : 73.6 Kbps Track name : PleaseRefesh2015_1 Writing application : Lavf57.21.100 Writing library : Lavf57.21.100 Audio ID : 1 Format : Opus Codec ID : A_OPUS Duration : 4s 135ms Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 24.0 KHz Bit depth : 16 bits Compression mode : Lossy Default : Yes Forced : No
Change History (8)
comment:1 by , 8 years ago
comment:2 by , 8 years ago
No, I want the ffmpeg to create with and report the correct sample rate value.
Let's create an opus file like this:
$ ffmpeg -i http://www.soundsnap.com/audio/play/72210%3A1448291371%3Ab4326304c013a3a61b3b87d83158c91b -c:a libopus -b:a 64k -ar 24000 /tmp/tmp.opus
ffmpeg version N-77704-g68eb208 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
configuration: --prefix=/home/pszemus/ffmpeg/build --enable-pic --enable-pthreads --enable-libmp3lame --enable-version3 --enable-libfaac --enable-gpl --enable-nonfree --enable-libvpx --enable-libvorbis --enable-libopus --enable-libx264 --enable-libwebp --enable-protocol=https --enable-openssl --extra-cflags=-I/opt/WP/common.libs/include --extra-ldflags='-L/opt/WP/common.libs/lib -ldl'
libavutil 55. 12.100 / 55. 12.100
libavcodec 57. 21.100 / 57. 21.100
libavformat 57. 21.100 / 57. 21.100
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 23.100 / 6. 23.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
[mp3 @ 0x3c683a0] Skipping 0 bytes of junk at 0.
[mp3 @ 0x3c683a0] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'http://www.soundsnap.com/audio/play/72210%3A1448291371%3Ab4326304c013a3a61b3b87d83158c91b':
Metadata:
title : PleaseRefesh2015_1
date : 2015
track : 1
Duration: 00:00:04.13, start: 0.000000, bitrate: 192 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
Output #0, opus, to '/tmp/tmp.opus':
Metadata:
title : PleaseRefesh2015_1
date : 2015
track : 1
encoder : Lavf57.21.100
Stream #0:0: Audio: opus (libopus), 24000 Hz, stereo, s16, 64 kb/s
Metadata:
encoder : Lavc57.21.100 libopus
title : PleaseRefesh2015_1
date : 2015
TRACKNUMBER : 1
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (native) -> opus (libopus))
Press [q] to stop, [?] for help
size= 36kB time=00:00:04.13 bitrate= 71.0kbits/s speed=15.3x
video:0kB audio:35kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.533776%
Now compare the output form FFmpeg and opusinfo (from opus-tools) analysis:
$ ffmpeg -i /tmp/tmp.opus
ffmpeg version N-77704-g68eb208 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
configuration: --prefix=/home/pszemus/ffmpeg/build --enable-pic --enable-pthreads --enable-libmp3lame --enable-version3 --enable-libfaac --enable-gpl --enable-nonfree --enable-libvpx --enable-libvorbis --enable-libopus --enable-libx264 --enable-libwebp --enable-protocol=https --enable-openssl --extra-cflags=-I/opt/WP/common.libs/include --extra-ldflags='-L/opt/WP/common.libs/lib -ldl'
libavutil 55. 12.100 / 55. 12.100
libavcodec 57. 21.100 / 57. 21.100
libavformat 57. 21.100 / 57. 21.100
libavdevice 57. 0.100 / 57. 0.100
libavfilter 6. 23.100 / 6. 23.100
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Input #0, ogg, from '/tmp/tmp.opus':
Duration: 00:00:04.13, start: 0.000000, bitrate: 70 kb/s
Stream #0:0: Audio: opus, 48000 Hz, stereo, fltp
Metadata:
ENCODER : Lavc57.21.100 libopus
TITLE : PleaseRefesh2015_1
DATE : 2015
track : 1
At least one output file must be specified
$ opusinfo /tmp/tmp.opus
Processing file "/tmp/tmp.opus"...
New logical stream (#1, serial: 281a7d51): type opus
Encoded with Lavf57.21.100
User comments section follows...
encoder=Lavc57.21.100 libopus
title=PleaseRefesh2015_1
date=2015
TRACKNUMBER=1
Opus stream 1:
Pre-skip: 156
Playback gain: 0 dB
Channels: 2
Original sample rate: 24000Hz
Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min)
Page duration: 1000.0ms (max), 828.0ms (avg), 140.0ms (min)
Total data length: 36674 bytes (overhead: 1.51%)
Playback length: 0m:04.130s
Average bitrate: 71.03 kb/s, w/o overhead: 69.96 kb/s
Logical stream 1 ended
Where this inconsistency come from? Why does FFmpeg and Firefox report 48kHz sample rate when MediaInfo and opusinfo show 24kHz?
comment:3 by , 8 years ago
Because Opus files always have a sample rate of 48kHz but the original sample rate is also written to the bitstream (but not used for decoding) so opusinfo is able to show it for informational purpose.
I sent a patch that may fix the issue described by Timothy Terriberry, this is not related to the shown sample rate.
comment:4 by , 8 years ago
You seem to be right as I read more about opus.
Also, browsing through forums, I've found a term:
YAMRI: Yet Another Mediainfo Related Issue.
comment:5 by , 8 years ago
| Component: | undetermined → avformat |
|---|---|
| Keywords: | mkv opus added |
| Resolution: | → fixed |
| Status: | new → closed |
| Summary: | Wrong sample rate read from / written to opus webm file → Opus initial padding incorrectly written to mkv for sample rates != 48kHz |
| Version: | unspecified → git-master |
comment:6 by , 8 years ago
Should be fixed in c3c22bee6362737cf290929b7f31df9fb88da983 - thank you for the report and the testing!
comment:7 by , 8 years ago
YAMRI: Yet Another Mediainfo Related Issue.
By curiosity, why do you consider that it is a MediaInfo related issue?
Looks like MediaInfo does its job: it shows the sampling rate indicated in the Matroska header (FYI, MediaInfo does not parse Opus so it relies on the Matroska header, and displays the exact content from the Matroska header, this is the expected result, even if Opus is parsed: when Opus is supported by MediaInfo, MediaInfo will report both sampling rates, the one from Matroska header and the one form Opus, and Opus is always 48 kHz if I understand well so MediaInfo will display the discrepancy between Matroska header and Opus 48 kHz only by design).
YAFRI (Yet Another FFmpeg Related Issue) and fixed in FFmpeg instead?
comment:8 by , 8 years ago
The issue that FFmpeg only displays the actual Opus sample rate but not the sample rate used for encoding is not fixed yet.
But his is completely unrelated to the muxing issue that was fixed.
I thought this is just how the opus codec works, but reading the Mozilla bug report maybe you just want MediaCodec not to show the original sample rate?