Opened 9 years ago
Last modified 4 years ago
#4178 open defect
Opus audio in MKV container
| Reported by: | agressiv | Owned by: | Vignesh Venkatasubramanian |
|---|---|---|---|
| Priority: | important | Component: | avformat |
| Version: | git-master | Keywords: | mkv opus regression |
| Cc: | moritz@bunkus.org, Michael Niedermayer | Blocked By: | |
| Blocking: | Reproduced by developer: | yes | |
| Analyzed by developer: | yes |
Description
Summary of the bug:
Opus Audio in MKV/WEBM/MKA containers is has crackling and lots of clicks/artifacts when played in ffplay, mplayer or using shared libraries (e.g. Kodi)
How to reproduce:
Mux Opus Audio into MKV. Play it with ffplay, mplayer or anything leveraging ffmpeg shared libraries.
MPC-HC has zero problems playing these files at all. I've taken the first minute of a bad MKA (for sizes' sake) and have attached it. If you extract the Opus, ffplay will play it fine.
Attachments (12)
Change History (32)
by , 9 years ago
| Attachment: | Star Trek (2009).mka added |
|---|
comment:1 by , 9 years ago
| Keywords: | regression added; libopus matroska crackling removed |
|---|---|
| Priority: | normal → important |
| Reproduced by developer: | set |
| Status: | new → open |
| Version: | 2.5 → git-master |
Regression since 7b0a839b
For future tickets: Please always test current FFmpeg git head and please provide complete, uncut output of the failing command to make the tickets valid.
$ ffmpeg -i Star\ Trek\ \(2009\).mka out.wav
ffmpeg version N-68356-gd43d5c5 Copyright (c) 2000-2014 the FFmpeg developers
built on Dec 10 2014 10:07:42 with gcc 4.7 (SUSE Linux)
configuration: --enable-gpl
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 14.100 / 56. 14.100
libavformat 56. 15.103 / 56. 15.103
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
Input #0, matroska,webm, from 'Star Trek (2009).mka':
Metadata:
title : StarTrek2009
ENCODER : Lavf56.11.100
Duration: 00:01:00.00, start: 0.007000, bitrate: 300 kb/s
Chapter #0:0: start 0.007000, end 60.000000
Metadata:
title : Chapter 01
Stream #0:0(eng): Audio: opus, 48000 Hz, 5.1, fltp (default)
Metadata:
LANGUAGE : eng
BPS : 253416
BPS-eng : 253416
DURATION : 02:06:50.260000000
DURATION-eng : 02:06:50.260000000
NUMBER_OF_FRAMES: 380514
NUMBER_OF_FRAMES-eng: 380514
NUMBER_OF_BYTES : 241070856
NUMBER_OF_BYTES-eng: 241070856
_STATISTICS_WRITING_APP: mkvmerge v7.3.0 ('Nouages') 64bit built on Oct 22 2014 18:53:34
_STATISTICS_WRITING_APP-eng: mkvmerge v7.3.0 ('Nouages') 64bit built on Oct 22 2014 18:53:34
_STATISTICS_WRITING_DATE_UTC: 2014-11-23 21:24:11
_STATISTICS_WRITING_DATE_UTC-eng: 2014-11-23 21:24:11
_STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
_STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
Output #0, wav, to 'out.wav':
Metadata:
INAM : StarTrek2009
ISFT : Lavf56.15.103
Chapter #0:0: start 0.014000, end 60.007000
Metadata:
title : Chapter 01
Stream #0:0(eng): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1, s16, 4608 kb/s (default)
Metadata:
LANGUAGE : eng
BPS : 253416
BPS-eng : 253416
DURATION : 02:06:50.260000000
DURATION-eng : 02:06:50.260000000
NUMBER_OF_FRAMES: 380514
NUMBER_OF_FRAMES-eng: 380514
NUMBER_OF_BYTES : 241070856
NUMBER_OF_BYTES-eng: 241070856
_STATISTICS_WRITING_APP: mkvmerge v7.3.0 ('Nouages') 64bit built on Oct 22 2014 18:53:34
_STATISTICS_WRITING_APP-eng: mkvmerge v7.3.0 ('Nouages') 64bit built on Oct 22 2014 18:53:34
_STATISTICS_WRITING_DATE_UTC: 2014-11-23 21:24:11
_STATISTICS_WRITING_DATE_UTC-eng: 2014-11-23 21:24:11
_STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
_STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
encoder : Lavc56.14.100 pcm_s16le
Stream mapping:
Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size= 21676kB time=00:01:00.00 bitrate=2959.5kbits/s
video:0kB audio:21676kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000559%
Also reproducible with the libopus decoder, the following produces a playable file:
$ ffmpeg -i Star\ Trek\ \(2009\).mka -acodec copy out.opus
comment:2 by , 9 years ago
Author of change 7b0a839b (the regression) here.
I looked into the attached sample file, it seems like it has DiscardPadding elements in frames where it shouldn't. DiscardPadding is supposed to be present only in the very last frame of the file. Whereas the attached sample file has DiscardPadding element scattered all over the file.
Could you please tell if this file was created using ffmpeg? If so, can you please tell the exact command line you used to encode this file?
If this file was created by some other tool, then the file is invalid and the crackling artifacts are intended behavior as per the DiscardPadding element.
comment:3 by , 9 years ago
| Analyzed by developer: | set |
|---|---|
| Owner: | set to |
comment:4 by , 9 years ago
The source mkv was about 4gb, I used ffmpeg to trim it to an mka (taking out the video stream and limiting it to the first 20 seconds) in order to facilitate uploading as part of this ticket.
However, the source mkv file (which plays fine in MPC-HC) was created with mkvmerge.exe (however the audio was encoded with ffmpeg).
So, it seems that ffplay (and the DLL libraries) don't like mkv files created by mkvmerge with Opus in the container, if I am getting that right. (And I tried at least 5-6 versions of mkvmerge at the time) MPC-HC is able to "discard" the "DiscardPadding?" elements, ironically.
When I get back in town this weekend I'll completely demux the source file, and use ffmpeg as the muxer instead of mkvmerge.
Could mkvmerge really be non-compliant, or would you think it's just a bug with how mkvmerge handles Opus Muxes? I'm not a developer but I can certainly open a bug with mkvtoolnix if need be. Any added details you can provide would be helpful.
comment:5 by , 9 years ago
I'm the author of the MKV demuxer in MPC-HC, and Discard Padding is indeed currently ignored, but mostly by accident and not by design (or rather, due to a missing feature, the DirectShow filter concept used by MPC-HC doesn't make communicating side-data from demux to decoder quite trivial).
If the sample contains DiscardPadding in frames that shouldn't have them, then I would argue that the file is muxed wrong. mkvmerge is not fool proof, especially when it comes to new things like Opus, so it might be worth inquiring with its author (or checking the code).
comment:6 by , 9 years ago
To help analyse the source, where did the Opus audio comes from originally before muxing it to MKV? Was it encoded into .ogg/.opus? It might as well be a bug in the Ogg/Opus reader in mkvmerge, or even a bug in the .opus muxer in ffmpeg (if thats what encoded the audio).
comment:7 by , 9 years ago
The Opus file was created by ffmpeg from a variety of sources, usually DTS-HD or TrueHD. I also have some coming from opusenc piped from eac3to to as well, but I haven't used that in a while - however I could do some more experimentation there as well.
comment:8 by , 9 years ago
The MKVToolNix' author here. For each Opus packet read from an Ogg file mkvmerge decodes the Opus TOC and calculates the expected decoded duration (and the expected end timestamp as the sum of the durations of all prior Opus packets). This duration is compared with the Ogg page's granulepos which signals when the containing packets are supposed to _end_. If there is a difference between what the Opus TOC and the Ogg page's granulepos say then this difference is written as a DiscardPadding element.
I've done some quick tests. For files created by opusenc there's exactly one Ogg page/Opus packet for which this is the case: the very last one in the file. The same is true for files created with ffmpeg 2.5 (from the Arch Linux repositories). The resulting Matroska files contain exactly one DiscardPadding element.
I've also given today's ffmpeg git a try (revision a30fd828abf4830d15a1bd7935d08477961f6628). That version does not show any irregularities either; mkvmerge only finds a single Ogg packet with such a difference.
Therefore I'm interested how the Ogg/Opus files the ticket's submitter has used were created. The amount of DiscardPadding elements in the sample file attached to this ticket suggests that pretty much all Ogg pages have granulepos values that differ from what the Opus TOC suggests.
I'd also like to get my hands on such an Ogg/Opus file if possible. The first couple of MB should be enough.
comment:9 by , 9 years ago
| Cc: | added |
|---|
by , 9 years ago
by , 9 years ago
by , 9 years ago
by , 9 years ago
by , 9 years ago
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by , 9 years ago
follow-up: 12 comment:10 by , 9 years ago
Ok, here is the explanation for the above files:
st6.mka/opus
Original Audio is encoded from TrueHD inside Source.mkv. Source.mkv is about 32gb.
ffmpeg.exe -i Source.mkv -acodec libopus -b:a 256k Audio.opus
Source.mkv contains extremely high-bitrate Video. I shrink that down with x264-x64.exe, and re-mux the file with mkvmerge.exe, which produces st6.mkv.
mkvmerge.exe -o st6.mkv -d 0 -A -S st6.264 --language 0:eng --sync 0:0 -a 0 -D -S Audio.opus --chapters st6.xml
I then use ffmpeg to extract 20 seconds of audio, with the following command:
ffmpeg.exe -i st6.mkv -acodec copy -ss 00:00:00.0 -c copy -t 00:00:20.0 st6.opus ffmpeg.exe -i st6.mkv -acodec copy -ss 00:00:00.0 -c copy -t 00:00:20.0 st6.mka
The MKA file crackles at two points, whereas the Opus file does not.
st7.mka/opus
I use ffmpeg to extract opus from the original mkv encoding I did several months ago.
ffmpeg.exe -i st7.mkv -acodec copy -ss 00:00:00.0 -c copy -t 00:00:20.0 st7.opus ffmpeg.exe -i st7.mkv -acodec copy -ss 00:00:00.0 -c copy -t 00:00:20.0 st7.mka
st7.mka is obviously very similar to the original attachment; st7.opus is extracted from the mkv rather than the mka.
The original Opus file which created st7.mkv is long gone.
I tried lots of experimentation to see if I could get inconsistent results, such as using multiple computers, stressing out the disk when doing the audio encoding and muxing - nothing would cause it to behave inconsistently.
st8.mka/opus (Not Uploaded)
I encoded these directly from the 32gb original. There were no artifacts. Note that mkvmerge is never in the equation either.
ffmpeg.exe -i Source.mkv -acodec libopus -b:a 256k -ss 00:00:00.0 -t 00:00:20.0 st8.mka ffmpeg.exe -i Source.mkv -acodec libopus -b:a 256k -ss 00:00:00.0 -t 00:00:20.0 st8.opus
These were all with the old ffmpeg version I have, ffmpeg version N-67289-g7f24e1e from October.
st9.mka/opus
I encoded these just like st6.mka/opus, with mkvmerge, except with the latest ffmpeg binary, ffmpeg version N-69190-g6c559a0. This actually sounds worse, more artifacts, but not as bad as st7. Opus file plays ok.
st10.mka/opus
When I mux with ffmpeg instead of mkvmerge, I am not getting any problems:
ffmpeg.exe -i st10.264 -i Audio.opus -c copy -map 0:v:0 -map 1:a:0 -shortest st10.mkv ffmpeg.exe -i st10.mkv -acodec copy -ss 00:00:00.0 -c copy -t 00:00:20.0 st10.opus ffmpeg.exe -i st10.mkv -acodec copy -ss 00:00:00.0 -c copy -t 00:00:20.0 st10.mka
So, this still points in the direction of mkvmerge, but since I do not really know how to analyze these files the way you guys can, I won't jump to any other conclusions.
I've spent many hours on this now, and I'm sorry about all of the attachments, but I am very detail-oriented. If there is still something else I can do to help narrow this down, let me know.
comment:11 by , 9 years ago
| Component: | undetermined → avformat |
|---|---|
| Priority: | important → wish |
| Type: | defect → enhancement |
comment:12 by , 9 years ago
Replying to agressiv:
Thanks for all the samples. I am sure they would be helpful.
So, this still points in the direction of mkvmerge, but since I do not really know how to analyze these files the way you guys can, I won't jump to any other conclusions.
Remember that there is always a possibility that there are two bugs in FFmpeg, one in the muxer and one in the demuxer, each covering up for the other. Such problems have been seen in the past where some files are only decodable by FFmpeg.
comment:13 by , 9 years ago
This sounds like an issue in ffmpeg's Ogg or Opus output code. If I follow the steps you've described then I do see several more Ogg packets for which the expected granulepos doesn't match the Ogg page's actual granulepos.
What I've done:
- Merged a WAV file with mkvmerge, e.g.
mkvmerge -o source.mka source.wav - Encoded that to Ogg/Opus with ffmpeg, e.g.
ffmpeg -i source.mka source.ogg - Merged that Ogg/Opus file to an .mka and turned on debugging output during this conversion:
mkvmerge --debug opus -o result.mka source.ogg
The debug output contains a couple of lines which report a difference between the actual granulepos and the expected one. Such lines look like this:
Debug> src/input/r_ogm.cpp:1134: Opus discard padding calculated 00:00:03.020000000 Ogg timestamp 00:00:03.019979166 diff 00:00:00.000020834 samples 1 (Ogg page's granulepos 144959)
(Note that the latest MKVToolNix release, v7.5.0, doesn't output the Ogg granulepos; that's a change I've just pushed to the MKVToolNix repo.)
Like I said: mkvmerge calculates the expected timestamp as the sum of the durations of all the Opus packets encountered so far by decoding their TOCs. This timestamp is calculated with the actual timestamp derived from the Ogg page's granulepos according to the official formula: timestamp_in_ns = 1_000_000_000 * granulepos / 48000 (note that pre-skip is not used in this calculation as the pre-skip is kept as a separate header value in Matroska).
Having said that this looks like a rounding error in the program (ffmpeg) creating the Ogg stream. That granulepos for the Ogg packet shown above is most likely supposed to be 144960 instead of 144959 as 144960 would result in a time stamp of exactly 3.02s – which would match the timestamp calculated by summing all the packet's durations by decoding the packet duration in the TOC.
comment:15 by , 9 years ago
| Priority: | wish → important |
|---|---|
| Type: | enhancement → defect |
comment:16 by , 8 years ago
| Cc: | added |
|---|
comment:17 by , 7 years ago
Good morning, I can confirm that this bug is still around.
I can reproduce this problem only with 7.1 tracks (now <=5.1 at least for me seems to work) and if and only if i'm encoding directly from an mkv container. Example:
ffmpeg -i "myfile.mkv" -af "channelmap=channel_layout=7.1" -c:a libopus -b:a 384K -mapping_family 1 out.opus
Now that out.opus plays well with ffplay but if i mux it with indifferently ffmpeg or mkvtoolnix, audio is crackling and with artifacts.
Instead if i do this (let's suppose myfile has a DTS stream):
ffmpeg -i "myfile.mkv" -c:a copy out.dts ffmpeg -i "out.dts" -af "channelmap=channel_layout=7.1" -c:a libopus -b:a 384K -mapping_family 1 out.opus
That ogg/opus stream works fine now, also if i mux it in a mkv container.
Another thing that works is this:
Let's suppose i'm encoding as described in the first example ( that produce corrupted out.opus)
and put that track in a mkv container and so i'm getting artifacts:
ffmpeg -i "myfile.mkv" -af "channelmap=channel_layout=7.1" -c:a libopus -b:a 384K -mapping_family 1 out.opus ffmpeg -i out.opus -c copy muxed.mkv
Now if do this:
ffmpeg -i muxed.mkv -c:a copy out2.opus ffmpeg -i out2.opus -c copy muxed2.mkv
So if i demux it and remux, it seems that ffmpeg fix something and muxed2.mkv is now working well.
comment:18 by , 6 years ago
I can reliably create such files with ffmpeg and have a theory on why this is happening. The ultrashort answer is: Bad things can happen if the timestamps that the libopus encoder receives aren't perfect.
Before I come to the long answer, let me add that I used the current git-master to produce the framehash logs that you will see. In more detail
ffmpeg version N-90800-g8592ae1a1e Copyright (c) 2000-2018 the FFmpeg developers built with gcc 7.3.0 (Rev1, Built by MSYS2 project) configuration: --disable-static --enable-shared --disable-amf --disable-cuda --disable-cuvid --disable-d3d11va --disable-nvenc --disable-ffnvcodec --disable-debug --enable-libopus --enable-libbluray --enable-libmfx --enable-libsoxr --enable-libwavpack --enable-gpl --enable-openssl --enable-avisynth --enable-libfdk-aac --enable-libzvbi --disable-encoder=dca --disable-encoder=nellymoser --disable-encoder=real_144 --disable-encoder=truehd --disable-encoder=vorbis --disable-encoder=sonic --disable-encoder=sonicls --disable-encoder=amv --disable-encoder=asv1 --disable-encoder=asv2 --disable-encoder=flashsv --disable-encoder=flashsv2 --disable-encoder=roqvideo --disable-encoder=svq1 --disable-encoder=zmbv --disable-encoder=zlib --disable-encoder=snow --disable-encoder=cinepak --disable-encoder=a64multi --disable-encoder=a64multi5 --disable-encoder=h261 --disable-encoder=h263 --disable-encoder=h263p --disable-encoder=wmv7 --disable-encoder=wmav1 --disable-encoder=wmav2 --disable-encoder=wmv8 --enable-nonfree --shlibdir=/local64/bin-video libavutil 56. 15.100 / 56. 15.100 libavcodec 58. 19.100 / 58. 19.100 libavformat 58. 13.100 / 58. 13.100 libavdevice 58. 4.100 / 58. 4.100 libavfilter 7. 19.100 / 7. 19.100 libswscale 5. 2.100 / 5. 2.100 libswresample 3. 2.100 / 3. 2.100 libpostproc 55. 2.100 / 55. 2.100
So the other logs won't have the version field.
a) First some information about granule positions: The granule positions in ogg pages indicate the position in the stream after decoding all packets which are completely within that page. They are restricted as follows:
i) All pages with completed packets except the first and the last MUST have a granule position equal to the number of samples contained in packets that complete on that page plus the granule position of the most recent page with completed packets. (From section 4 of RFC 7845.)
ii) If a page has the 'end of stream' flag set, then instead of the above the difference between the number of samples contained in the packets that complete on that page and the difference between the granule position of said page and the most recent page with completed packets indicates how many samples should be trimmed away at the end; if there was no earlier page with completed packets, then one should work as if the granule position of the most recent earlier page with completed packets were zero (in this case one also has to apply the preskip). (This is 4.4 of RFC 7845.)
iii) If the first page with completed packets isn't also the last page (then ii) applies) then it must have a granule position that is >= the sum of the number of samples contained in packages that complete on that page (the pre-skip is ignored in calculating the sum) so that there are no negative granule positions when working backwards. The granule position may be larger than the sum (useful for synchronization with other streams in the same multiplex); if the sum is larger then the stream is completely invalid (yes, the whole stream, not only the first page or the samples which would have negative granule positions). (This is 4.5 of RFC 7845.)
b) The easiest way to produce malformed files is by using a negative -itsoffset:
ffmpeg -itsoffset -0.5 -i test.dts -c:a libopus offset.-0.5.opus
opusinfo (a part of the opus-tools package from the creators of the opus codec) complains about this file:
Processing file "offset.-0.5.opus"... New logical stream (#1, serial: 7d2420f4): type opus Encoded with Lavf58.13.100 User comments section follows... encoder=Lavc58.19.100 libopus WARNING: Samples with negative granpos in stream 1 Opus stream 1: Pre-skip: 312 Playback gain: 0 dB Channels: 2 Original sample rate: 48000Hz Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min) Page duration: 1000.0ms (max), 968.4ms (avg), 20.0ms (min) Total data length: 386932 bytes (overhead: 0.811%) Playback length: 0m:30.005s Average bitrate: 103.2 kb/s, w/o overhead: 102.3 kb/s Logical stream 1 ended
Let's use the framehash muxer to see what the timestamps are when they leave the encoder:
ffmpeg -itsoffset -0.5 -i fl.dts -c:a libopus -f framehash -hash crc32 - ... #format: frame checksums #version: 2 #hash: CRC32 #extradata 0, 19, ea5d642a #software: Lavf58.13.100 #tb 0: 1/48000 #media_type 0: audio #codec_id 0: opus #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo #stream#, dts, pts, duration, size, hash 0, -24312, -24312, 960, 425, 996864ad 0, -23352, -23352, 960, 241, 1fcc1d4d 0, -22392, -22392, 960, 228, 05f1dd79 0, -21432, -21432, 960, 225, e56a7998 0, -20472, -20472, 960, 224, a12a261d 0, -19512, -19512, 960, 226, 27020d0e 0, -18552, -18552, 960, 249, ab31aeb9 0, -17592, -17592, 960, 241, 44e9b2e4 0, -16632, -16632, 960, 241, 8d5dbc65 0, -15672, -15672, 960, 253, 54c603d7 0, -14712, -14712, 960, 256, f9acaea3 0, -13752, -13752, 960, 254, 308a7027 0, -12792, -12792, 960, 262, 297c12b8 0, -11832, -11832, 960, 271, 86b889ca 0, -10872, -10872, 960, 266, 07e95927 0, -9912, -9912, 960, 271, eedd9414 0, -8952, -8952, 960, 275, 856a747f 0, -7992, -7992, 960, 275, 48c4343e 0, -7032, -7032, 960, 281, a54c56ad 0, -6072, -6072, 960, 275, a5ede609 0, -5112, -5112, 960, 271, f5795567 0, -4152, -4152, 960, 270, cb1f8e24 0, -3192, -3192, 960, 282, 9c81d325 0, -2232, -2232, 960, 287, c4bec144 0, -1272, -1272, 960, 276, 6978978a 0, -312, -312, 960, 280, 928ce969 0, 648, 648, 960, 288, a8b67809 0, 1608, 1608, 960, 289, d08817ee 0, 2568, 2568, 960, 281, 785f7424 0, 3528, 3528, 960, 271, 01a150fd 0, 4488, 4488, 960, 279, 1d1a6926 0, 5448, 5448, 960, 299, 4ad6192a 0, 6408, 6408, 960, 401, 1ba1ba43 0, 7368, 7368, 960, 297, 722e745b 0, 8328, 8328, 960, 399, bb637945 0, 9288, 9288, 960, 296, b746197e 0, 10248, 10248, 960, 276, 44dde335 0, 11208, 11208, 960, 279, 3ffcb2f5 0, 12168, 12168, 960, 293, 481af07f 0, 13128, 13128, 960, 286, fbc2d89c 0, 14088, 14088, 960, 278, 2983e9a8 0, 15048, 15048, 960, 283, 6a8c6b1b 0, 16008, 16008, 960, 285, b7b3a531 0, 16968, 16968, 960, 285, 5ee67d70 0, 17928, 17928, 960, 266, f3ad421b 0, 18888, 18888, 960, 261, bea0961e 0, 19848, 19848, 960, 272, d463ae16 0, 20808, 20808, 960, 383, 3b03279b 0, 21768, 21768, 960, 278, 9401e990 0, 22728, 22728, 960, 270, abfad09a 0, 23688, 23688, 960, 295, 1da5ee48 0, 24648, 24648, 960, 266, 21c45f34 0, 25608, 25608, 960, 256, 5003d43c 0, 26568, 26568, 960, 274, dae2fa79 0, 27528, 27528, 960, 268, 80b438cb
-0.5s are 24000 samples and the remaining difference of 312 samples are due to libopus' pre-skip of 312 samples. If one analyzes the generated file directly one sees that the first page contains 50 packets with 960 samples each, i.e. 48000 samples (of which the first 312 are invalid), but the granule position of the first page shows 24000; if one simply calculated backwards, this means that the first packet would have started at -24000 which is against a) iii) above. Needless to say that the first page doesn't have the 'end of stream' flag set.
Given the fact that according to the spec the whole stream has to be treated as invalid it is actually strange that opusinfo emits only a warning and not an error. The reference decoder, too, doesn't treat the stream as invalid: Instead it treats the first page as if it has end trimming although it doesn't have the 'end of stream' flag. In our sample this means that from the first page with 48000 samples the last 24000 samples are stripped away because of end trimming and the first 312 because of pre-skip. And indeed comparing the input file with the output of the reference decoder shows that they are essentially the same for the first 23688 samples and then totally different.
This shows that the ogg muxer should automatically shift the granule positions to make them (both those implied and those explicitly written) non-negative. (But there is a problem here: In ogg, the relationsship between a timestamp and the granule position is codec-dependant and it needn't be a linear relationsship like for opus so shifting other tracks might be complicated.)
c) If one uses an itsoffset larger than the page_duration of the ogg muxer, opusinfo complains even more: "Negative or zero granulepos (-14400) on Opus stream outside of headers. This file was created by a buggy encoder"
d) Here is another way to come into a situation like b), but without using itsoffset. It has to do with odd behaviour (I'd call it a bug) of the native opus decoder. Instead of stripping the pre-skip away like the reference decoder does, it simply gives them negative timestamps. Here is a part of framehash's output of what this looks like for a non-defective file:
#format: frame checksums #version: 2 #hash: CRC32 #software: Lavf58.13.100 #tb 0: 1/48000 #media_type 0: audio #codec_id 0: pcm_s16le #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo #stream#, dts, pts, duration, size, hash 0, -312, -312, 960, 3840, 1908c39f 0, 648, 648, 960, 3840, 239ecff4 0, 1608, 1608, 960, 3840, c5dd9714 0, 2568, 2568, 960, 3840, 1173d416 0, 3528, 3528, 960, 3840, e4e9ca53 0, 4488, 4488, 960, 3840, dbc3e9f0 0, 5448, 5448, 960, 3840, 2187b445 0, 6408, 6408, 960, 3840, 25180cb2 0, 7368, 7368, 960, 3840, 788bf31b 0, 8328, 8328, 960, 3840, 1c3b1f55 0, 9288, 9288, 960, 3840, a67eae2f 0, 10248, 10248, 960, 3840, 17cc83a0
So if one uses an ordinary opus file as input, decodes it with the native decoder (the default decoder) and encodes this with libopus, one is in the very same situation as b). If one uses the libopus decoder, the result is fine.
This behaviour of the native decoder is actually at the heart of #4692.
This was exactly the situation which made me realize what's going on. See here.
e) I can also explain anthontex's observation with the exception of the part where he claims that streams <=5.1 seem to work. This time the root cause is lacing (that is used by default by mkvmerge for e.g. dts/dca tracks) probably coupled with strange timestamp rounding. Notice that test.dts is actually a stereo dts track.
The timestamps from the dts file are good:
ffmpeg -i test.dts -f framehash -hash crc32 - ... #format: frame checksums #version: 2 #hash: CRC32 #software: Lavf58.13.100 #tb 0: 1/48000 #media_type 0: audio #codec_id 0: pcm_s16le #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo #stream#, dts, pts, duration, size, hash 0, 0, 0, 512, 2048, 52bbda48 0, 512, 512, 512, 2048, 2b037e9f 0, 1024, 1024, 512, 2048, f69e3985 0, 1536, 1536, 512, 2048, 04f27523 0, 2048, 2048, 512, 2048, 0c9b0963 0, 2560, 2560, 512, 2048, de6e37eb 0, 3072, 3072, 512, 2048, 2230f372 0, 3584, 3584, 512, 2048, b4275a94 0, 4096, 4096, 512, 2048, e2efc7d5 0, 4608, 4608, 512, 2048, e6ff0c6f 0, 5120, 5120, 512, 2048, 43d5c355 0, 5632, 5632, 512, 2048, f689afdb 0, 6144, 6144, 512, 2048, 7ce06f4f 0, 6656, 6656, 512, 2048, d639e9c7 0, 7168, 7168, 512, 2048, 87aee60f 0, 7680, 7680, 512, 2048, 6e32d1e1 0, 8192, 8192, 512, 2048, 99b53229 0, 8704, 8704, 512, 2048, 46803053 0, 9216, 9216, 512, 2048, 4e4143b5 0, 9728, 9728, 512, 2048, 2116fa38 ...
If one remuxes test.dts with mkvmerge and specifies a timecode/timestamp-scale factor of 1000000 (for files who don't have a video track, mkvmerge by default uses a timecode/timestamp-scale factor that is small enough so that 1 tick of the timebase is less than the time of one sample so that timecodes/timestamps in the file are actually sample accurate; if there is a video track, it defaults to 1000000 (i.e. 1ms precision)), the timestamps aren't good any more (the merged file is called "Test.Laced.Big.TS.mka" ("TS" means TimestampScale)):
ffmpeg -i Test.Laced.Big.TS.mka -f framehash -hash crc32 - ... #format: frame checksums #version: 2 #hash: CRC32 #software: Lavf58.13.100 #tb 0: 1/48000 #media_type 0: audio #codec_id 0: pcm_s16le #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo #stream#, dts, pts, duration, size, hash 0, 0, 0, 512, 2048, 52bbda48 0, 504, 504, 512, 2048, 2b037e9f 0, 1016, 1016, 512, 2048, f69e3985 0, 1512, 1512, 512, 2048, 04f27523 0, 2024, 2024, 512, 2048, 0c9b0963 0, 2536, 2536, 512, 2048, de6e37eb 0, 3048, 3048, 512, 2048, 2230f372 0, 3560, 3560, 512, 2048, b4275a94 0, 4072, 4072, 512, 2048, e2efc7d5 0, 4584, 4584, 512, 2048, e6ff0c6f 0, 5096, 5096, 512, 2048, 43d5c355 0, 5592, 5592, 512, 2048, f689afdb 0, 6104, 6104, 512, 2048, 7ce06f4f 0, 6616, 6616, 512, 2048, d639e9c7 0, 7128, 7128, 512, 2048, 87aee60f 0, 7640, 7640, 512, 2048, 6e32d1e1 0, 8184, 8184, 512, 2048, 99b53229 0, 8696, 8696, 512, 2048, 46803053 0, 9208, 9208, 512, 2048, 4e4143b5 0, 9720, 9720, 512, 2048, 2116fa38 0, 10232, 10232, 512, 2048, ffd0d2d3 0, 10744, 10744, 512, 2048, ab0e8d25 0, 11256, 11256, 512, 2048, d75d5dbf 0, 11768, 11768, 512, 2048, 495f20b4 0, 12280, 12280, 512, 2048, c73a83e5 0, 12792, 12792, 512, 2048, 1a8bd665 0, 13304, 13304, 512, 2048, 37baf488 0, 13800, 13800, 512, 2048, 75a43386 0, 14312, 14312, 512, 2048, bac86852 0, 14824, 14824, 512, 2048, cfa03cf6 0, 15336, 15336, 512, 2048, ec85b2cf 0, 15848, 15848, 512, 2048, 568417f0 0, 16360, 16360, 512, 2048, de55f656 0, 16872, 16872, 512, 2048, b4471f41 0, 17384, 17384, 512, 2048, a8b615d7 0, 17880, 17880, 512, 2048, 634e69bd 0, 18392, 18392, 512, 2048, 28bb1df8 0, 18904, 18904, 512, 2048, 7a2b2546 0, 19416, 19416, 512, 2048, dd67f369 0, 19928, 19928, 512, 2048, 72468c87 0, 20472, 20472, 512, 2048, 31358846 0, 20984, 20984, 512, 2048, 1b25d341 0, 21496, 21496, 512, 2048, 0f188f8e 0, 22008, 22008, 512, 2048, d4c28420 0, 22520, 22520, 512, 2048, c2a2cc15 0, 23032, 23032, 512, 2048, 97348c24 0, 23544, 23544, 512, 2048, 8266b6bd 0, 24056, 24056, 512, 2048, 9492736f 0, 24568, 24568, 512, 2048, a0eb4084 0, 25080, 25080, 512, 2048, 84f6ec09 0, 25592, 25592, 512, 2048, 050f991a 0, 26088, 26088, 512, 2048, deee9a7e 0, 26600, 26600, 512, 2048, 12b66ef5 0, 27112, 27112, 512, 2048, 38780750 0, 27624, 27624, 512, 2048, e309fbb0 0, 28136, 28136, 512, 2048, ee05c406 0, 28648, 28648, 512, 2048, fe965280 0, 29160, 29160, 512, 2048, 0e456d8f 0, 29672, 29672, 512, 2048, 8868c0a4 0, 30168, 30168, 512, 2048, b67200db 0, 30680, 30680, 512, 2048, 98452104 0, 31192, 31192, 512, 2048, 1c1d5dfa 0, 31704, 31704, 512, 2048, 3bb376e9 0, 32216, 32216, 512, 2048, 5cb27573 0, 32760, 32760, 512, 2048, ce8afcf9 0, 33272, 33272, 512, 2048, 671aafd3 0, 33784, 33784, 512, 2048, 6b0ef4ae 0, 34296, 34296, 512, 2048, 6190ea4e 0, 34808, 34808, 512, 2048, 9cc28eec 0, 35320, 35320, 512, 2048, 2fcb40a1 0, 35832, 35832, 512, 2048, c97c7941 0, 36344, 36344, 512, 2048, a8ddb89e 0, 36856, 36856, 512, 2048, 1f03cd39 0, 37368, 37368, 512, 2048, 0ae93b83 0, 37880, 37880, 512, 2048, 2f2c98d4 0, 38376, 38376, 512, 2048, 77460589 0, 38888, 38888, 512, 2048, a4d05c57 0, 39400, 39400, 512, 2048, df5b2b8d 0, 39912, 39912, 512, 2048, 19602dd2 0, 40424, 40424, 512, 2048, 53e32a7f 0, 40936, 40936, 512, 2048, 2f4acb24 0, 41448, 41448, 512, 2048, 29b3fd40 0, 41960, 41960, 512, 2048, 5cd68804 0, 42456, 42456, 512, 2048, 1a8765dd 0, 42968, 42968, 512, 2048, 5fbc5ae7 0, 43480, 43480, 512, 2048, d41ba16a 0, 43992, 43992, 512, 2048, 36614005 0, 44504, 44504, 512, 2048, b959f96a 0, 45048, 45048, 512, 2048, 764ffe4b 0, 45560, 45560, 512, 2048, 45e7b0d8 0, 46072, 46072, 512, 2048, e190ddd4 0, 46584, 46584, 512, 2048, 89c9b204 0, 47096, 47096, 512, 2048, 6d5e6559 0, 47608, 47608, 512, 2048, faa96307 0, 48120, 48120, 512, 2048, cf00e88c 0, 48632, 48632, 512, 2048, 9a7a02d3 ...
If I encode Test.Laced.Big.TS.mka with libopus to Test.Laced.Big.TS.opus, the resulting file is again defective:
Processing file "Test.Laced.Big.TS.opus"...
New logical stream (#1, serial: fb757aa2): type opus
Encoded with Lavf58.13.100
User comments section follows...
BPS-eng=1508966
DURATION-eng=00:00:30.006000000
NUMBER_OF_FRAMES-eng=2813
NUMBER_OF_BYTES-eng=5659756
_STATISTICS_WRITING_APP-eng=mkvmerge v22.0.0 ('At The End Of The World') 64-bit
_STATISTICS_WRITING_DATE_UTC-eng=2018-04-22 04:35:34
_STATISTICS_TAGS-eng=BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
encoder=Lavc58.19.100 libopus
WARNING: Samples with negative granpos in stream 1
WARNING: Sample count ahead of granule (633600>633568) in stream 1
WARNING: Sample count ahead of granule (681600>681568) in stream 1
WARNING: Sample count ahead of granule (729600>729568) in stream 1
WARNING: Sample count ahead of granule (777600>777568) in stream 1
WARNING: Sample count ahead of granule (825600>825584) in stream 1
WARNING: Sample count ahead of granule (873600>873584) in stream 1
WARNING: Sample count ahead of granule (921600>921584) in stream 1
WARNING: Sample count ahead of granule (969600>969584) in stream 1
WARNING: Sample count ahead of granule (1017600>1017584) in stream 1
Opus stream 1:
Pre-skip: 312
Playback gain: 0 dB
Channels: 2
Original sample rate: 48000Hz
Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min)
Page duration: 1020.0ms (max), 1000.7ms (avg), 780.0ms (min)
Total data length: 387229 bytes (overhead: 0.887%)
Playback length: 0m:30.005s
Average bitrate: 103.2 kb/s, w/o overhead: 102.3 kb/s
Logical stream 1 ended
The reason for the "Samples with negative granpos" warning despite the first sample not having a negative timestamp is that the ogg format doesn't explicitly signal the granule position for every packet, but only for every page (in which a packet is completed) and the ogg muxer uses a page duration of 1s by default. In order to fill this 1s, one needs 48000-312 = 47688 samples from the input file and for this one needs the first 94 packets. The first sample of the last of these 94 packets should have be sample number 47616 (zero-based), but according to the above framehash it has the timestamp 47608. Consequently sample number 47687 has the timestamp 47679 and therefore the granule position of the first page is 47679+1+312 = 47992 (the +1 comes from the fact that the granule position indicate the position after decoding the whole content of the page) and that is exactly what is in the output file. That of course means that the output file is invalid.
Because not every output packet has a granule position, not every gap/overlap in the samples that are fed to the libopus encoder end up having an influence on the output file. If the sum of the durations/number of samples just happens to conincide with the granule position delta, then everything's fine. This explains why
If one decodes the just created opus file with the reference decoder it again ignores that a) ii) has the prerequisite of the 'end of stream' flag being set and discards several samples. This leads to audible distortions at around sample 633248 (= 633568 (from above opusinfo message) - 312 (pre-skip) - 8 (the number of samples from the end of the first page that got skipped)).
f) Here is a bit more about the Matroska timestamps:
i) If one uses a timestamp-scale of 1000000 and no lacing the timestamps are fine despite the second dts packet having a timestamp of 11ms whereas the first dts packet has only a duration of 10 2/3 ms:
#format: frame checksums #version: 2 #hash: CRC32 #software: Lavf58.13.100 #tb 0: 1/48000 #media_type 0: audio #codec_id 0: pcm_s16le #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo #stream#, dts, pts, duration, size, hash 0, 0, 0, 512, 2048, 52bbda48 0, 512, 512, 512, 2048, 2b037e9f 0, 1024, 1024, 512, 2048, f69e3985 0, 1536, 1536, 512, 2048, 04f27523 0, 2048, 2048, 512, 2048, 0c9b0963 0, 2560, 2560, 512, 2048, de6e37eb 0, 3072, 3072, 512, 2048, 2230f372 0, 3584, 3584, 512, 2048, b4275a94 0, 4096, 4096, 512, 2048, e2efc7d5 0, 4608, 4608, 512, 2048, e6ff0c6f 0, 5120, 5120, 512, 2048, 43d5c355 0, 5632, 5632, 512, 2048, f689afdb 0, 6144, 6144, 512, 2048, 7ce06f4f 0, 6656, 6656, 512, 2048, d639e9c7 0, 7168, 7168, 512, 2048, 87aee60f 0, 7680, 7680, 512, 2048, 6e32d1e1 0, 8192, 8192, 512, 2048, 99b53229 0, 8704, 8704, 512, 2048, 46803053 0, 9216, 9216, 512, 2048, 4e4143b5 0, 9728, 9728, 512, 2048, 2116fa38 ...
The result is the same whether the unlaced Matroska file has a default duration or not.
ii) As has already been said, for files with audio but no video track mkvmerge uses a smaller TimestampScale (namely 20832 for 48kHz) by default. With lacing the timestamps are as follows:
#format: frame checksums #version: 2 #hash: CRC32 #software: Lavf58.13.100 #tb 0: 1/48000 #media_type 0: audio #codec_id 0: pcm_s16le #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo #stream#, dts, pts, duration, size, hash 0, 0, 0, 512, 2048, 52bbda48 0, 512, 512, 512, 2048, 2b037e9f 0, 1024, 1024, 512, 2048, f69e3985 0, 1536, 1536, 512, 2048, 04f27523 0, 2048, 2048, 512, 2048, 0c9b0963 0, 2560, 2560, 512, 2048, de6e37eb 0, 3072, 3072, 512, 2048, 2230f372 0, 3584, 3584, 512, 2048, b4275a94 0, 4096, 4096, 512, 2048, e2efc7d5 0, 4608, 4608, 512, 2048, e6ff0c6f 0, 5120, 5120, 512, 2048, 43d5c355 0, 5632, 5632, 512, 2048, f689afdb 0, 6144, 6144, 512, 2048, 7ce06f4f 0, 6656, 6656, 512, 2048, d639e9c7 0, 7168, 7168, 512, 2048, 87aee60f ... 0, 22528, 22528, 512, 2048, c2a2cc15 0, 23040, 23040, 512, 2048, 97348c24 0, 23551, 23551, 512, 2048, 8266b6bd 0, 24063, 24063, 512, 2048, 9492736f 0, 24576, 24576, 512, 2048, a0eb4084 0, 25088, 25088, 512, 2048, 84f6ec09 ...
So they are not perfect (there mustn't be any odd timestamps like 23551), but way better.
Notice that also the first eight packets get different timestamps from the timestamps they had in e) with the bigger timestamp-scale. This is despite them being in the same lace and the lace both starting precisely at the same time (namely at absolute zero which coincides for every TimestampScale). This happens even when one trims the files to contain only eight packets (which are all in the same lace). So a lower TimecodeScale in this case leads to better results despite the file with the lower TimestampScale not containing any more information about the timestamps than the file with the default 1000000 TimecodeScale.
iii) Using a small TimestampScale and no lacing leads to good timestamps (as expected).
iv) It seems that also the DefaultDuration is involved: If one uses the file from e) (laced, TimestampScale 1000000) and deletes the DefaultDuration header element (MKVToolNix has a tool named mkvpropedit for that) one gets even worse timestamps (e.g. the 3384 should actually be 3584):
#format: frame checksums #version: 2 #hash: CRC32 #software: Lavf58.13.100 #tb 0: 1/48000 #media_type 0: audio #codec_id 0: pcm_s16le #sample_rate 0: 48000 #channel_layout 0: 3 #channel_layout_name 0: stereo #stream#, dts, pts, duration, size, hash 0, 0, 0, 512, 2048, 52bbda48 0, 504, 504, 512, 2048, 2b037e9f 0, 984, 984, 512, 2048, f69e3985 0, 1464, 1464, 512, 2048, 04f27523 0, 1944, 1944, 512, 2048, 0c9b0963 0, 2424, 2424, 512, 2048, de6e37eb 0, 2904, 2904, 512, 2048, 2230f372 0, 3384, 3384, 512, 2048, b4275a94 0, 4080, 4080, 512, 2048, e2efc7d5 0, 4584, 4584, 512, 2048, e6ff0c6f 0, 5064, 5064, 512, 2048, 43d5c355 0, 5544, 5544, 512, 2048, f689afdb 0, 6024, 6024, 512, 2048, 7ce06f4f 0, 6504, 6504, 512, 2048, d639e9c7 0, 6984, 6984, 512, 2048, 87aee60f 0, 7464, 7464, 512, 2048, 6e32d1e1 0, 8208, 8208, 512, 2048, 99b53229 ...
And consequently one gets way more errors from opusinfo if one encodes the above:
Processing file "I:\Neuer Ordner (2)\test.laced.big.ts.no.defdur.opus"...
New logical stream (#1, serial: 0bcf313d): type opus
Encoded with Lavf58.13.100
User comments section follows...
BPS-eng=1508966
DURATION-eng=00:00:30.006000000
NUMBER_OF_FRAMES-eng=2813
NUMBER_OF_BYTES-eng=5659756
_STATISTICS_WRITING_APP-eng=mkvmerge v22.0.0 ('At The End Of The World') 64-bit
_STATISTICS_WRITING_DATE_UTC-eng=2018-04-22 04:35:34
_STATISTICS_TAGS-eng=BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
encoder=Lavc58.19.100 libopus
WARNING: Samples with negative granpos in stream 1
WARNING: Sample count behind granule (96960<96992) in stream 1
WARNING: Sample count behind granule (145920<145952) in stream 1
WARNING: Sample count behind granule (194880<194920) in stream 1
WARNING: Sample count ahead of granule (243840>243808) in stream 1
WARNING: Sample count behind granule (291840<291872) in stream 1
WARNING: Sample count behind granule (340800<340840) in stream 1
WARNING: Sample count behind granule (389760<389800) in stream 1
WARNING: Sample count behind granule (438720<438760) in stream 1
WARNING: Sample count behind granule (486720<486760) in stream 1
WARNING: Sample count behind granule (535680<535720) in stream 1
WARNING: Sample count behind granule (584640<584680) in stream 1
WARNING: Sample count ahead of granule (633600>633408) in stream 1
WARNING: Sample count ahead of granule (681600>681472) in stream 1
WARNING: Sample count behind granule (730560<730600) in stream 1
WARNING: Sample count ahead of granule (779520>779328) in stream 1
WARNING: Sample count ahead of granule (827520>827392) in stream 1
WARNING: Sample count ahead of granule (875520>875456) in stream 1
WARNING: Sample count behind granule (1118400<1118408) in stream 1
WARNING: Sample count behind granule (1167360<1167368) in stream 1
WARNING: Sample count ahead of granule (1216320>1216288) in stream 1
WARNING: Sample count behind granule (1264320<1264328) in stream 1
WARNING: Sample count behind granule (1313280<1313288) in stream 1
WARNING: Sample count ahead of granule (1362240>1362064) in stream 1
WARNING: Sample count ahead of granule (1410240>1410128) in stream 1
Opus stream 1:
Pre-skip: 312
Playback gain: 0 dB
Channels: 2
Original sample rate: 48000Hz
Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min)
Page duration: 1020.0ms (max), 1000.7ms (avg), 640.0ms (min)
Total data length: 387229 bytes (overhead: 0.887%)
Playback length: 0m:30.006s
Average bitrate: 103.2 kb/s, w/o overhead: 102.3 kb/s
Logical stream 1 ended
PS: Yes, mkvmerge was also buggy. In fact, I think it still is and will soon open a bug report for it. For example up until version 15.0 it used lacing in BlockGroups with DiscardPadding (the result was that the information to which audio packet the DiscardPadding actually applies is lost (upon remuxing mkvmerge treated every packet in the block as if the DiscarPadding element applied to them)). But I have never ever observed it creating bad output if the input file didn't have any issues.
PPS: And I also have some good news: The actual packets (without the container stuff) of the file created by e) and f) iv) both completely coincide with what one gets when one directly encodes test.dts. The only thing that is truly lost is the end trimming.
follow-up: 20 comment:19 by , 4 years ago
I was also bitten by this bug. The sound in a mkv video is garbage using mpv or ffplay (both from current Debian testing). The same mkv file plays fine in VLC. If I use ffmpeg to extract the audio track into an opus container, ffplay and mpv can play it fine.
From ffplay -v info (there are no warning/errors while playing):
Input #0, matroska,webm, from 'Mai_Mai_Miracle_(BD_1080p)_(Hi10)_(THORA).mkv':
Metadata:
creation_time : 2017-09-22T10:11:19.000000Z
ENCODER : Lavf57.72.101
Duration: 01:35:01.03, start: -0.007000, bitrate: 1607 kb/s
[...]
Stream #0:1(jpn): Audio: opus, 48000 Hz, 5.1, fltp (default)
Metadata:
title : OPUS Audio (6 channeled)
BPS : 254936
BPS-eng : 254936
DURATION : 01:35:01.030000000
DURATION-eng : 01:35:01.030000000
NUMBER_OF_FRAMES: 285052
NUMBER_OF_FRAMES-eng: 285052
NUMBER_OF_BYTES : 181675178
NUMBER_OF_BYTES-eng: 181675178
_STATISTICS_WRITING_APP: mkvmerge v8.5.2 ('Crosses') 64bit
_STATISTICS_WRITING_APP-eng: mkvmerge v8.5.2 ('Crosses') 64bit
_STATISTICS_WRITING_DATE_UTC: 2017-09-22 10:11:19
_STATISTICS_WRITING_DATE_UTC-eng: 2017-09-22 10:11:19
_STATISTICS_TAGS: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
_STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
Would providing samples of the mkv help to get this solved?
comment:20 by , 4 years ago
Replying to mytskine:
Would providing samples of the mkv help to get this solved?
No. Actually, there was never a problem of missing samples that exhibit this; the problem was with how these samples were created in the first place.
(mkvmerge has btw received bugfixes wrt opus; see issue 2280.)
1 minute of an MKA container with an Opus sound file