| Commit message (Collapse) | Author | Age |
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Commit e392d6610d1e35cc0190c794c151211b0aae83e6 modified the native
demuxer to use track gain as a fallback for album gain if the latter is
not present. This commit makes functionally equivalent changes in the
libavformat demuxer.
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I encountered a stream that fails with "Could not demux init fragment.".
It turns out this is a regression from the recent change to that code.
The assumption was that demux_lavf.c would treat this as concatenated
stream - which it does, but not for probing.
Doing this transparently is hard without doing it properly. Doing it
properly would mean creating some sort of stream_concat (reminiscent of
that FFmpeg security bug). I probably don't want to go there, and I
think libavformat should just support this directly, so whatever.
Hack-fix this with the knowledge that the init segment will always
contain the headers.
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There was no reason to limit this. Only some int fields had to be
changed to size_t.
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FFmpeg is retarded enough not to give us any indication whether it is
(unless we query fields not in the ABI/API). I bet FFmpeg developers
love it when library users have to litter their code with duplicated
information.
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for the rawaudio demuxer to do the expected gapless playback
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Signed-off-by: Aman Gupta <aman@tmm1.net>
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Fixes #5923
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This code shouldn't even exist in libavformat. If you still need it, you
can enable it via --demuxer-lavf-o.
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When the current packet queue was completely empty, and EOF was reached,
the queue->is_eof flag was not correctly set to true. Change this by
reading ds->eof to check whether the stream is considered EOF. We also
need to make sure update_seek_ranges() is called in this case, so change
the code to simply call it when queue->is_eof changes.
Also, read_packet() needs to call adjust_seek_range_on_packet() if
ds->eof changes. In that case, the decoder also needs to be notified
about EOF. So both of these should be called when ds->eof changes to
true. (Other code outside of this function deals with the case when
ds->eof is changed to false.)
In addition, this code was kind of shoddy about calling wakeup_ds()
correctly. It looks like there was an inverted condition, and sent a
wakeup to the decoder only when ds->eof was already true, which is
obviously bogus. The final EOF case tried to be somehow clever about
checking in->last_eof for notifying the codec, which is sort of OK, but
seems to be strictly worse than just checking whether ds->eof changed.
Fix these things.
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ffm (ffserver) was removed from ffmpeg.
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Fixes several issues playing back mpegts with video streams marked
as having "still images". For example, see this video which has
frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts
Changes include:
- start playback right away, without waiting for first video frame
- do not consider the sparse video stream in demuxer underrun detection
- do not require multiple video frames for the VO
- use audio as the master stream for demuxer metadata events
- use audio stream for playback time
Signed-off-by: Aman Gupta <aman@tmm1.net>
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Signed-off-by: Aman Gupta <aman@tmm1.net>
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With -v -v ("debug" level), which is the default for --log-file, this
would log every damn Matroska EBML element and some other uninteresting
things, which was very noisy.
Adjust the log levels to make them less noisy. Also, change some log
calls to MP_ERR for things which are actually errors.
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Going by ISO 639.2, "und" means "Undetermined". Whatever it's supposed
to mean, in practice it's user for "unset". We prefer if the language
tag remains simply unset in this case.
This removes an ugliness with mp4 in partricular, because libavformat
will export unset languages as such, which affects most mp4 files.
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This makes ICY title changes show up at approximately the correct time,
even if the demuxer buffer is huge. (It'll still be wrong if the stream
byte cache contains a meaningful amount of data.)
It should have the same effect for mid-stream metadata changes in e.g.
OGG (untested).
This is still somewhat fishy, but in parts due to ICY being fishy, and
FFmpeg's metadata change API being somewhat fishy. For example, what
happens if you seek? With FFmpeg AVFMT_EVENT_FLAG_METADATA_UPDATED and
AVSTREAM_EVENT_FLAG_METADATA_UPDATED we hope that FFmpeg will correctly
restore the correct metadata when the first packet is returned.
If you seke with ICY, we're out of luck, and some audio will be
associated with the wrong tag until we get a new title through ICY
metadata update at an essentially random point (it's mostly inherent to
ICY). Then the tags will switch back and forth, and this behavior will
stick with the data stored in the demuxer cache. Fortunately, this can
happen only if the HTTP stream is actually seekable, which it usually is
not for ICY things. Seeking doesn't even make sense with ICY, since you
can't know the exact metadata location. Basically ICY metsdata sucks.
Some complexity is due to a microoptimization: I didn't want additional
atomic accesses for each packet if no timed metadata is used. (It
probably doesn't matter at all.)
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This fixes an issue where captions stop rendering after an
in-demuxer-cache seek, because the demuxer keeps waiting to find
a keyframe (ds->skip_to_keyframe set to true in execute_cache_seek).
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ffmpeg marks audio tracks which are not meant to be played standalone
as DEPENDENT. these are typically used in DVB broadcasts for audio
descriptions, and are meant to be mixed into the main audio track during
playback.
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I changed avio_flush() and introduced avformat_flush() exactly for this
reason.
Used with DVD/BD only (on seeks and when setting the "angle" property).
Seems to work, but wasn't tested too thoroughly (I don't care about
optical discs, I only want this ugly stuff gone that might even violate
the API/ABI).
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Tries to recursively lock a non-recursive lock, which usually ends in a
deadlock. Must have been broken by some past refactor.
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Some shittily muxed files (by a certain HandBrake+libavformat combo)
contain a SeekHead pointing to a SeekHead at the end of the file, which
in turn points to track headers (also at the end of the file). This
failed because the demuxer didn't bother to actually read the elements
listed by the second SeekHead, so no track headers were read, and
playback broke.
Somehow commit 6fe75c38 broke this for no reason. It adds a "needed"
field, which seems completely pointless and replaced the "parsed" flag
in an incomplete way. In particular, the "needed" field was not set when
a _recursive_ SeekHead was read, so those elements were not read. Just
get rid of the field and use "parsed" instead.
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Quickly tested by a person who had FFmpeg linked with libaom.
Seems as simple as the VP9 mappings, where there is no extradata/
initialization data off-band, and just stuff in the packets
themselves.
Do note that the AV1 video format itself at this point is still
not frozen, so what you might produce one day might not be
decodable the following day.
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When this happens, network calls are forcibly aborted (more or less),
but demuxers might keep going, as most of them do not check for forced
exits properly. This can possibly lead to broken packets being added.
Also do not attempt to read more packets in this situation.
Also do not print a stream open failed message if opening was aborted
anyway.
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Since the demuxer cache addition, ds->queue->head can actually be set to
non-NULL, but the decoder can still be at EOF (with no packets to come).
This made it report an unknown buffered size, instead of 0. Fix this by
checking the decoder part of the packet queue instead.
Probably doesn't matter much, but fixes an annoying "???" on the CLI
status line in some situations.
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Naturally, there's more than one fourcc that indicates an mjpeg
stream.
I have a particular ancient webcam here (Logitech QuickCam Messanger)
that only supports the single 'JPEG' format, but there are other
devices out there which support both 'JPEG' and 'MJPG' with no visible
differences, and others where the streams are slightly different.
Regardless of those details, it remains correct to treat 'JPEG'
the same as 'MJPG' from a stream consumption perspective.
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Just the usual guess-what-opaque-ffmpeg-thing-supports.
See #5550. It looks like we can reduce packet drop by having the cache
enabled automatically.
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Another attempt to try to make it behave in certain situations.
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No functional changes.
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It's a mess: mp3 files have user tags as global metadata (because the
id3v2 tag is global and there is only 1 stream), while OGG files have it
per-track (because it's per-stream on the lowest level). mpv needs to
try to make something nice out of the mess.
It did so by trying to detect audio-only OGG files, and then copying the
per-stream metadata to the global metadata. Make the heuristic for
detecting this slightly more clever, so it works for files with extra,
unrelated streams, like the awful libavformat cover art hack.
Fixes #5577.
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The return value of stream_get_size() will be -1 if it fails. We
shouldn't mess up this value if a mp4 init segment is used.
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It appears some (or all) mkv files with EAC3 are muxed in a way that
breaks FFmpeg's spdifenc. I suspect it's because either dependent
substream packets are localted in their own packets, or the reverse. Or
possibly this is case where the muxer did not respect packet boundaries
at all. Enabling the EAC3 parser seems to fix this anyway, because why
waste your precious time on retarded Dolby bullshit technology? (Which
idiot came up with this shitty substream garbage?)
Observed with dolby_digital_plus_channel_check_lossless-DWEU.mkv.
Fixes #5578.
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param_names[n] is only valid for n<nparam.
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This includes codec/muxer/demuxer iteration (different iteration
function, registration functions deprecated), and the renaming of
AVFormatContext.filename to url (plus making it a malloced string).
Libav doesn't have the new API yet, so it will break. I hope they will
add the new APIs too.
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Reduce backward/forward from 400MB/400MB to 50MB/150MB. Too many
complaints about high memory usage.
Note that external tracks (like ytdl DASH with external audio tracks)
will double the amounts, because an external track uses its own demuxer
and cache.
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This reverts commit b7f90be567c3c19eb3fec30be2b76775296a6ed1.
The author agreed to the relicensing now (if that code is affected by
the original copyright at all - that was the only line possibly left of
it).
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If tags like TITLE have the whole parameter in " quotes, strip them.
Also remove the leading whitespace, since even with a single space it
was always included.
Fixes #5462.
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Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes
a source filter. vd.h mostly disappears, because mp_filter takes care of
the dataflow, but its remains are in struct mp_decoder_fns.
One goal is to simplify dataflow by letting the filter framework handle
it (or more accurately, using its conventions). One result is that the
decode calls disappear from video.c, because we simply connect the
decoder wrapper and the filter chain with mp_pin_connect().
Another goal is to eventually remove the code duplication between the
audio and video paths for this. This commit prepares for this by trying
to make f_decoder_wrapper.c extensible, so it can be used for audio as
well later.
Decoder framedropping changes a bit. It doesn't seem to be worse than
before, and it's an obscure feature, so I'm content with its new state.
Some special code that was apparently meant to avoid dropping too many
frames in a row is removed, though.
I'm not sure how the source code tree should be organized. For one,
video/decode/vd_lavc.c is the only file in its directory, which is a bit
annoying.
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This is supposed to help making data flow easier and wakeup handling
more efficient. Once that change is done, reading a packet on any
stream won't have to wakeup and poll all decoders (which helps reducing
the mess even if all decoders are on the same thread).
This also improves the accuracy of wakeups by tracking better whether
a wakeup is needed.
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AV_DISPOSITION_ATTACHED_PIC usually means the video track isn't real,
and merely reflects the presence of an embedded image in tag data (such
as ID3v2 tags), with some inconsistent hack to make libavformat return
it as video packet once.
Except it doesn't mean that. It can be randomly set on other streams
that do sort of behave like video streams, such as chapter thumbnail
tracks in mp4 files. AV_DISPOSITION_TIMED_THUMBNAILS is set in these
cases. In theory, there can supposedly be more such cases, but only the
chapter thumbnail one currently exists. So add it as exception.
This restores displaying these thumbnails as video frames, for better or
worse. (Before, only the first thumbnail was displayed.)
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Requires newest FFmpeg git, which has a change that makes the HLS
demuxer set an AVFMTCTX_UNSEEKABLE flag if seeking is not available,
which is the case for HLS live streams. This should make the player
frontend behave pretty well, instead of crapping up irrecoverably.
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And use it for 2 demuxer options. It could be used for more options
later. (Though the --cache options can not use this, because they use KB
as base unit.)
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I found that at least for mjpeg streams, FFmpeg will set packet pts/dts
anyway. The mjpeg raw video demuxer (along with some other raw formats)
has a "framerate" demuxer option which defaults to 25, so all mjpeg
streams will be played at 25 FPS by default.
mpv doesn't like this much. If AVFMT_NOTIMESTAMPS is set, it prints a
warning, that might print a bogus FPS value for the assumed framerate.
The code was originally written with the assumption that FFmpeg would
not set pts/dts for such formats, but since it does, the printed
estimated framerate will never be used. --fps will also not be used by
default in this situation.
To make this hopefully less confusing, explicitly state the situation
when the AVFMT_NOTIMESTAMPS flag is set, and give instructions how to
work it around.
Also, remove the warning in dec_video.c. We don't know what FPS it's
going to assume anyway. If there are really no timestamps in the stream,
it will trigger our normal missing pts workaround. Add the assumed FPS
there.
In theory, we could just clear packet timestamps if AVFMT_NOTIMESTAMPS
is set, and make up our own timestamps. That is non-trivial for advanced
video codecs like h264, so I'm not going there. For seeking and
buffering estimation the situation thus remains half-broken.
This is a mitigation for #5419.
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It was actually already implemented as ta_dup_ptrtype(), but that seems
like a clunky name. Also we still use the talloc_ names throughout the
source, and I'd rather use an old name instead of a mixing inconsistent
naming conventions.
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If you play a video with an external audio track, and do backwards
keyframe seeks, then audio can be missing. This is because a backwards
seek can end up way before the seek target (this is just how this seek
mode works). The audio file will be seeked at the correct seek target
(since audio usually has a much higher seek granularity), which results
in silence being played until the video reaches the originally intended
seek target.
There was a hack in audio.c to deal with this. Replace it with a
different hack. The new hack probably works about as well as the old
hack, except it doesn't add weird crap to the audio resync path (which
is some of the worst code here, so this is some nice preparation for
rewriting it). As a more practical advantage, it doesn't discard the
audio demuxer packet cache. The old code did, which probably ruined
seeking in youtube DASH streams.
A non-hacky solution would be handling external files in the demuxer
layer. Then chaining the seeks would be pretty easy. But we're pretty
far from that, because it would either require intrusive changes to the
demuxer layer, or wouldn't be flexible enough to load/unload external
files at runtime. Maybe later.
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Similar to 1eec7d2315, but for the beginning of the stream (named BOF in
this commit).
We can know this only if demuxing actually started from the beginning.
If there is a seek to the beginning (even if you use --start=-1000), we
don't know in general whether the demuxer truly returns the start of the
file. We could probably make a heuristic with assuming that this is what
happens if the seek target is before the start time or so, but this is
not included in this commit.
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libavformat's cover art hack (aka attached pictures) breaks the ability
of the demuxer cache to keep multiple seek ranges. This happens because
the cover art packet has neither position nor timestamp, and libavformat
gives us the packet even though we intended to drop it.
The cover art hack works by adding the cover art packet to the read
packet stream once when demuxing starts (or after seeks). mpv treats
cover art in a similar way internally, but we have to compensate for
libavformat's shortcomings, and add the cover art packet ourselves when
we need it. So we don't want libavformat to return the packet.
We normally prevent this in demux_lavc.c/select_tracks() and explicitly
disable cover art streams. (We add it in dequeue_packet() instead.) But
libavformat will actually add the cover art packet even if we disable
the cover art stream, because it adds it at initialization time, and
does not bother to check again in av_read_frame() (apparently). The
packet is actually read, and upsets the demuxer cache logic. In
addition, this also means we probably decoded the cover art picture
twice in some situations.
Fix this by explicitly checking/discarding this in yet another place.
(Screw this hack...)
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Benjamin Barenblat
wm4
Tom Yan
sfan5
Aman Gupta
Nicolas F
sergey.dobrodey
Jan Ekström
Philip Langdale