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.
Passing NULL to mp_get_config_group() returns the main option struct.
This is just a dumb hack to deal with inconsistencies caused by legacy
things (as I'll claim), and will probably be changed in the future. So
before littering the whole code base with hard to find NULL parameters,
require using callers an easy to find separate define.
This will enable the player core to terminate the demuxers in a "nicer"
way without having to block on network. If it just used demux_free(), it
would either have to block on network, or like currently, essentially
kill all I/O forcefully.
The API is slightly awkward, because demuxer lifetime is bound to its
allocation. On the other hand, changing that would also be awkward, and
introduce weird in-between states that would have to be handled in tons
of places.
Currently unused, to be user later.
Alway give each demuxer its own mp_cancel instance. This makes
management of the mp_cancel things much easier. Also, instead of having
add/remove functions for mp_cancel slaves, replace them with a simpler
to use set_parent function. Remove cancel_and_free_demuxer(), which had
mpctx as parameter only to check an assumption. With this commit,
demuxers have their own mp_cancel, so add demux_cancel_and_free() which
makes use of it.
Them being separate is just dumb. Replace them with a single
demux_free() function, and free its stream by default. Not freeing the
stream is only needed in 1 special case (demux_disc.c), use a special
flag to not free the stream in this case.
The properties/commands touched in this commit are all for obscure
special inputs (BD/DVD/DVB/TV), and they all block on the demuxer/stream
layer. For network streams, this blocking is very unwelcome. They will
affect playback and probably introduce pauses and frame drops. The
player can even freeze fully, and the logic that tries to make playback
abortable even if frozen complicates the player.
Since the mentioned accesses are not needed for network streams, but
they will block on network streams even though they're going to fail,
add a flag that coarsely enables/disables these accesses. Essentially it
establishes a whitelist of demuxers/streams which support them.
In theory you could to access BD/DVD images over network (or add such
support, I don't think it's a thing in mpv). In these cases these
controls still can block and could even "freeze" the player completely.
Writing to the "program" and "cache-size" properties still can block
even for network streams. Just don't use them if you don't want freezes.
Don't allow it to freeze everything when loading a playlist from network
(although you definitely shouldn't do that, but whatever).
This also affects the really obscure --ordered-chapters-files option.
The --playlist option on the other hand has no choice but to freeze the
shit, because there's no concept of aborting the player during command
line parsing.
It seems a bit inappropriate to have dumped this into stream.c, even if
it's roughly speaking its main user. At least it made its way somewhat
unfortunately to other components not related to the stream or demuxer
layer at all.
I'm too greedy to give this weird helper its own file, so dump it into
thread_tools.c.
Probably a somewhat pointless change.
If a stream starts later than the others at the start of the file, it
shouldn't restrict the seek range to the time stamp where it begins.
This is similar to the previous commit, just for the other end.
Normally, the seek range is the minimum overlap of the cached ranges of
each stream. But if one of the streams ends earlier, this leads to the
seek range getting cut off, even if you could seek there.
Change it so that EOF streams cannot restrict the end of the seek range.
They can only extend it. This is the opposite from not-EOF streams, so
they need to be handled separately. In particular, they get exluded from
normal end range calculation, but when full EOF is reached, all streams
are EOF, and the maximum end time can be used to set the seek end time.
(In theory we could also take the max with the demuxer signaled total
file duration, but let's not for now.)
Also, if a stream is completely empty, essentially skip it, instead of
considering the range unseekable. (Also, we don't need to mess with
seek_start in this case, because it will be NOPTS and is skipped
anyway.)
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.
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>
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.
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.
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.)
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).
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.
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).
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.
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.
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.
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.
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.
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.
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.
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.
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.
This reverts commit b7f90be567.
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).
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.
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.
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.
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.)
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.
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.)
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.
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.
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.
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.
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...)
The impact was that you couldn't exactly seek to the join point with a
keyframe seek, even though there was a keyframe. This commit fixes it by
preserving the necessary metadata that got lost on cached range joining.
This is so absurdly obscure that it gets a longer code comment.
This warning was printed when the demuxer cache tried to join two
adjacent seek ranges, but failed if the last keyframe in the second
range was within the (overlapping) first range. This is a weird corner
case which to support probably would not be worth it.
So this code just printed a warning and discarded the second range. As
it turns out, this can happen relatively often if you seek a lot, and
the seek ranges are very tiny (such as consisting of only 1 keyframe).
Dropping the second range in these cases is OK and probably cheaper than
trying to actually join them. Change the warning to verbose level.
(It seems this could actually be "supported", because if keyframe_latest
is not set, there will be no other keyframes, so it could just be unset,
with the exception that q1->keyframe_latest in the code below must not
be overwritten. But still, too much trouble for a special case that
likely does not matter, and it would have to be tested too.)
This means if the user tries to seek past EOF, and we know EOF was seen
already, then use a cached seek, instead of triggering a low level seek.
This requires some annoying tracking, but seems pretty simple otherwise.
One advantage of doing this is that if the user tries to do this kind of
seek, there's no unnecessary waiting for a reaction by network (and in
most cases, redundant downloading of data just to discard it again).
Another is that this avoids creating overlapping seek ranges: previously, the
low level seek would naturally create a new range. Then it would read and add
data from the end of the stream due to the low level demuxer not being able to
seek to the target and selecting the last seek point before the end of the
stream. Consequently, this new range would overlap with the previous cached
range. But since the cache joining code is written such that you join the
current range with the _next_ range (instead of the previous as it would be
needed in this case), the overlapping ranges were left alone, until seeking back
to the previous range. That was ugly, sort of harmless, and could happen in
other cases, but this avoidable case was pretty easy to trigger.
Export them as explicitly undocumented debugging fields for the
"demuxer-cache-state" property.
Should be somewhat helpful to debug "wtf is the demuxer" doing
situations better, especially when seeking. It also becomes visible how
long the demuxer is blocked on an "old" seek when you keep seeking while
the first seek hasn't finished.
update_seek_ranges() has some special code that attempts to correctly
adjust seek ranges for subtitle tracks. (Subtitle are a nightmare for
seek ranges, because they are sparse, so using the packet list is not
enough to reliably determine the valid cached range.)
This had code like this inside the modified if statement:
range->seek_start = MP_PTS_MAX(range->seek_start, <something>);
If seek_start is NOPTS, then seek_start will be set to <something>,
breaking some other code that checks seek_start for NOPTS to see if it's
empty. Fix this by explicitly checking whether seek_start is NOPTS
before adjusting it.
The crash happened in prune_old_packets() because the range was marked
as non-empty, yet there was no packet in it to prune. This was with
files with muxed subtitles, when seeking back to the start. This should
not happen anymore with the change. Also add an assert() to
check_queue_consistency() that checks for this specific case.
There's still some mess. In theory, subtitle tracks could be completely
empty, yet their seek range would span the entire file. Seek range
tracking of subtitle files is slightly broken (even before this change).
Some of this should probably be revisited later, including not just
using seek_start to determine whether a seek range should be pruned due
to being empty.
The x264 hack requires reading the first video packet, which in turn we
handle with a hack in demux_mkv.c to get the packet without having to
add special crap to demux.c. Another useless MKV feature (which they
enabled by default at one point and which caused many demuxers to break
completely, only to disable it again when it was too late) conflicts
with this, because we actually pass a block as packet contents, instead
of after "decompression".
Fix this by calling demux_mkv_decode().
This fixes when resuming certain broken h264 files encoded by x264. See
FFmpeg commit 840b41b2a643fc8f0617c0370125a19c02c6b586 about the x264
bug itself.
Normally, the unregistered user data SEI (that contains the x264 version
string) is informational only. But libavcodec uses it to workaround a
x264 bug, which was recently fixed in both libavcodec and x264. The fact
that both encoder and decoder were buggy is the reason that it was not
found earlier, and there are apparently a lot of files around created by
the broken decoder. If libavcodec sees the SEI, this bug can be worked
around by using the old behavior.
If you resume a file with mpv (i.e. seeking when the file loads),
libavcodec never sees the first video packet. Consequently it has to
assume the file is not broken, and never applies the workaround,
resulting in garbage being played.
Fix this by always feeding the first video packet to the decoder on
init, and then flushing the codec (to avoid that an unwanted image is
output). Flushing the codec does not remove info such as the x264
version. We also abuse the fact that the first avcodec_send_packet()
always pushes the frame into the decoder (so we don't have to trigger
the decoder by requsting an output frame).
This will help with things like livestreams.
As a minor detail, subtitles are excluded, because they sometimes have
"unused" events after video and audio ends. To avoid this annoying
corner case, just ignore them.
Before this change and before the seekable stream cache became a thing,
we could possibly seek using the stream cache. But we couldn't know
whether the seek would succeed. We knew the available byte range, but
could in general not tell whether a demuxer would stay within the range
when trying to seek to a specific time position. We preferred to have
safe defaults, so seeking in streams that were detected as unseekable
were not honored. We allowed overriding this via --force-seekable=yes,
in which case it depended on your luck whether the seek would work, or
the player crapped its pants.
With the demuxer packet cache, we can tell exactly whether a seek will
work (at least if there's only 1 seek range). We can just let seeks go
through. Everything to allow this is already in place, and this commit
just moves around some minor things.
Note that the demux_seek() return value was not used before, because low
level (i.e. network level) seeks are usually asynchronous, and if they
fail, the state is pretty much undefined. We simply repurpose the return
value to signal whether cache seeking worked. If it didn't, we can just
resume playback normally, because demuxing continues unaffected, and no
decoder are reset.
This should be particularly helpful to people who for some reason stream
data into stdin via streamlink and such.
Caused by the relatively recent change to packet parsing. This time it
was probably triggered by lace type 0, which reduces the byte length of
a 0 sized packet to 3 (timestamp + flag) instead of 4 (lace count for
other lace types). The thing about laces is just my guess why it worked
for other 0 sized packets, though.
Also remove the redundant and now incorrect check below.
Fixes#5271.
This log line tells us why the demuxer is trying to read more, which us
useful when debugging queue overflows. Probably barely useful, but I
think keeping that flag separately also makes the code slightly easier
to understand.
This fixes weird behavior in the following case:
- open a file
- make sure the max. demuxer forward cache is smaller than the
file's video track
- make sure the max. readahead duration is larger than the file's
duration
- disable the audio track
- seek to the beginning of the file
- once the cache has filled enable the audio track
- a queue overflow warning should appear
(- looking at the seek ranges is also interesting)
The queue overflow warning happens because the packed queue for the
video track will use up the full quota set by --demuxer-max-bytes. When
the audio track is enabled, reading an audio packet would technically
overflow the packet cache by the size of whatever packet is read next.
This means the demuxer signals EOF to the decoder, and once playback has
consumed enough video packets so that audio packets can be read again,
the decoder resumes from EOF. This interacts badly with A/V
synchronization and the whole thing can randomly crap itself until audio
has fully recovered.
We didn't care about this so far, but we want to raise the readahead
duration to something very high, so that the demuxer cache is fully
used. This means this case can be hit quite quickly by switching audio
or subtitle tracks, and is not really an obscure corner case anymore.
Fix this by always losing all cache. Since the cache can't be used
anyway until the newly selected track has been read, this is not much of
a disadvantage. The only thing that could be brought up is that
unselecting the track again could resume operation normally. (Maybe this
would be useful if network died completely without chance of recovery.
Then you could watch the already buffered video anyway by deselecting
the audio track again.) But given the headaches, this seems like the
better solution.
Unfortunately this requires adding new new strange fields and strangely
fragmenting state management functions again. I'm sure whoever works on
this in the future will hate me. Currently it seems like the lesser
evil, and much simpler and robust than the other potential solutions.
In case this needs to be revisited, here is a reminder for readers from
the future what alternative solutions were considered, without those
disadvantages:
A first attempted solution allowed the demuxer to buffer some additional
packets on track switching. This would allow it to read enough data to
feed the decoder at least. But it was still awkward, as it didn't allow
the demuxer to continue prefetching the newly selected track. It also
barely worked, because you could make the forward buffer "over full" by
seeking back with seekable cache enabled, and then it couldn't read
packets anyway.
As alternative solution, we could always demux and cache all tracks,
even if they're deselected. This would also not require a network-level
seek for the "refresh" logic (it's the thing that lets the video decoder
continue as if nothing happened, while actually seeking back in the
stream to get the missing audio packets, in the case of enabling a
previously disabled audio track). But it would also possibly waste
network and memory resources, depending on what the user actually wants.
A second solution would just account the queue sizes for each stream
separately. We could freely fill up the audio packet queue, even if the
video queue is full. Since the demuxer API returns interleaved packets
and doesn't let you predict which packet type comes next, this is not as
simple as it sounds, but it'd probably tie in nicely with the "refresh"
logic.
A third solution would be removing buffered video packets from the end
of the packet queue. Since the "refresh" logic gets these anyway, there
is no reason to keep them if they prevent the audio packet queue from
catching up with the video one. But this would require additional logic,
would interact badly with a bunch of other corner cases. And as far as
the code goes, it's rather complex, because all the logic is written
with FIFO behavior in mind (including the fact that the packet queue is
a singly linked list with no backwards links, making removal from the
end harder).
It seems like there's nothing stopping from sub-demuxers from keeping
packets in the cache, even if it's completely pointless. The top-most
demuxer (demux_timeline) already takes care of caching, so sub-demuxers
only waste space and time with this.
Add a function that can disable the packet cache even at runtime and
after packets are read. (It's not clear whether it really can happen
that packets are read before demux_timeline gets the sub-demuxers, but
there's no reason to make it too fragile.) Call it on all sub-demuxers.
For this to work, it seems we have to move the code for setting the
seekable_cache flag to before demux_timeline is potentially initialized,
because otherwise the cache would be reenabled if the demuxer triggering
timeline support is a timeline segment itself (e.g. ordered chapters).
This fixes missing audio when cycling through audio tracks with anything
that uses nested demuxers, such as demux_timeline, which us used for
EDL, --merge-files, ordered chapters, and youtube-dl pseudo DASH
support. When this bug happened, reenabling an audio track would lead to
silence for the duration of the readahead amount.
The underlying reason is the incorrectly updated buffered range on track
switch. It accidentally included the amount covered by the deselected
stream. But the cause of the observed effect was that demux_timeline
issued a refresh seek to the underlying slave demuxer, which in turn
thought it could do a cache seek, because the seek range still included
everything.
update_stream_selection_state() calls update_seek_ranges() to update the
seek ranges after a track switch. When reenabling the track, ds->eager
was set to false during update_seek_ranges(), which made it think the
stream was sparse, and thus it didn't restrict the current seek range
(making later code think everything was buffered). Fix this by moving
some code, so we first update the ds->eager flag, then the seek ranges.
Also verbose log the low level stream selection calls.
Always display the duration as "unknown" if the duration is known. Also
fix that at least demux_lavf reported unknown duration as 0 (fix by
setting the default to unknown in demux.c).
Remove the dumb _u formatter function, and use a different approach to
avoiding displaying "unknown" as playback time on playback start (set
last_seek_pts for that).
This gives the filename or URL to the libavformat probing logic, which
might use the file extension as a "help" to decide which format the file
is. This helps with mp3 files that have large id3v2 tags and prevents
the idiotic ffmpeg probing logic to think that a mp3 file is amr.
(What we really want is knowing whether we _really_ need to feed more
data to libavformat to detect the format. And without having to pre-read
excessive amounts of data for relatively normal streams.)
If the backbuffer is much larger than the forward buffer, and if you
join a small range with a large range (larger than the forward buffer),
then the seek issues to the end of the range after joining will overflow
the queue.
Normally, read_more will be false when the forward buffer is full, but
the resume seek after joining will set need_refresh to true, which
forces more reading and thus triggers the overfloe warning.
Attempt to fix this by not setting read_more to true on refresh seeks.
Set prefetch_more instead. read_more will still be set if an A/V stream
has no data.
This doesn't help with the following problems related to using refresh
seeks for track switching:
- If the forward buffer is full, then enabling another track will
obviously immediately overflow the queue, and immediately lead to
marking the new track as having no more data (i.e. EOF). We could cut
down the forward buffer or so, but there's no simple way to implement
it. Another possibility would be dropping all buffers and trying to
resume again, but this would likely be complex as well.
- Subtitle tracks will not even show a warning (because they are sparse,
and we have no way of telling whether a packet is missing, or there's
just no packet near the current position). Before this commit,
enabling an empty subtitle track would probably have overflown the
queue, because ds->refreshing was never set to true. Possibly this
could be solved by determining a demuxer read position, which would
reflect until which PTS all subtitle packets should have been demuxed.
The forward buffer limit was intended as a last safeguard to avoid
excessive memory usage against badly interleaved files or decoders going
crazy (up to reading the whole into memory and OOM'ing the user's
system). It's not good at all to limit prefetch. Possibly solutions
include having another smaller limit for prefetch, or maybe having only
a total buffer limit, and discarding back buffer if more data has to be
read. The current solution is making the forward buffer larger than the
forward duration (--cache-secs) would require, but of course this
depends on the stream's bitrate.
The option for enabling it has now an "auto" choice, which is the
default, and which will enable it if the media is thought to be via
network or if the stream cache is enabled (same logic as --cache-secs).
Also bump the --cache-secs default from 10 to 120.
Some back buffer is required to make the immediate forward range
seekable. This is because the back buffer limit is strictly enforced.
Just set a rather high back buffer by default. It's not use if
--demuxer-seekable-cache is disabled, so this is without risk.
Limit the number of cached ranges to MAX_SEEK_RANGES, which is the same
number of maximally exported seek ranges. It makes no sense to keep
them, as the user won't see them anyway. Remove the smallest ones to
enforce the limit if the number grows too high.
Helps a little bit, I guess. But in general, t(h)rashing the cache kills
us anyway.
This has a fixed number of index entries. Entries are added/removed as
packets go through the packet queue. Only keyframes after index_distance
seconds are added. If there are too many keyframe packets, the existing
index is reduced by half, and index_distance is doubled. This should
provide somewhat even spacing between the entries.
The packet queue is sorted, so we can stop the search if we have found a
packet, and the next packet in the queue has a higher PTS than the seek
PTS (for the sake of SEEK_FORWARD, we still consider the first packet
with a higher PTS).
Also, as a mostly cosmetic change, but which might be "faster", check
target for NULL, instead of target_diff for a magic float value.
Subtitle streams are sparse, and no overlap is required to correctly
join two cached ranges. This was not correctly handled iff the two
ranges had different subtitle packets close to the join point.
demux_add_packet() must completely ignore any packets that are added
while a queued seek is not initiated yet.
The main issue is that after setting in->seeking==true, the central lock
is released, and it can take "a while" until it's reacquired on the
demux thread and the seek is actually initiated. During that time,
packets could be read and added, that have nothing to do with the new
state.
If subtitles are part of the stream, determining the seekable range
becomes harder. Subtitles are sparse, and can have packets in irregular
intervals, or even completely lack packets. The usual logic of computing
the seek range by the min/max packet timestamps fails.
Solve this by making the only assumption we can make: subtitle packets
are implicitly demuxed along with other packets. We also assume perfect
interleaving for this, but you really can't do anything with sparse
packets that makes sense without this assumption.
One special case is if we prune sparse packets within the current
seekable range. Obviously this should limit the seekable range to after
the pruned packet.
Instead of weirdly deciding this on every packet read and having the
code far away from where it's actually needed, just run it where it's
actually needed.
A typical idiom for calling functions that remove something from the
array being iterated, but it's not needed here. I have no idea why this
was ever done.
Setting ds->refreshing for unselected streams could lead to a
nonsensical queue overflow warning, because read_packet() took it as a
reason to continue reading.
Also add some more information to the queue overflow warning (even if
that one doesn't have anything to do with this bug - it was for
unselected streams only).
This fixes an endless loop with threading disabled, such as for example
when playing a file with an external subtitle file, and seeking to the
beginning. Something will set in->seeking, but the seek is never
executed, which made demux_read_packet() loop endlessly. (External
subtitles will use non-threaded mode for whatever reasons.)
Fix this by by making the unthreaded code to execute the worker thread
body, which reduces the difference in logic.
Until now, the demuxer cache was limited to a single range. Extend this
to multiple range. Should be useful for slow network streams.
This commit changes a lot in the internal demuxer cache logic, so
there's a lot of room for bugs and regressions. The logic without
demuxer cache is mostly untouched, but also involved with the code
changes. Or in other words, this commit probably fucks up shit.
There are two things which makes multiple cached ranges rather hard:
1. the need to resume the demuxer at the end of a cached range when
seeking to it
2. joining two adjacent ranges when the lowe range "grows" into it (and
resuming the demuxer at the end of the new joined range)
"Resuming" the demuxer means that we perform a low level seek to the end
of a cached range, and properly append new packets to it, without adding
packets multiple times or creating holes due to missing packets.
Since audio and video never line up exactly, there is no clean "cut"
possible, at which you could resume the demuxer cleanly (for 1.) or
which you could use to detect that two ranges are perfectly adjacent
(for 2.). The way how the demuxer interleaves multiple streams is also
unpredictable. Typically you will have to expect that it randomly allows
one of the streams to be ahead by a bit, and so on.
To deal with this, we have heuristics in place to detect when one packet
equals or is "behind" a packet that was demuxed earlier. We reuse the
refresh seek logic (used to "reread" packets into the demuxer cache when
enabling a track), which checks for certain packet invariants.
Currently, it observes whether either the raw packet position, or the
packet DTS is strictly monotonically increasing. If none of them are
true, we discard old ranges when creating a new one.
This heavily depends on the file format and the demuxer behavior. For
example, not all file formats have DTS, and the packet position can be
unset due to libavformat not always setting it (e.g. when parsers are
used).
At the same time, we must deal with all the complicated state used to
track prefetching and seek ranges. In some complicated corner cases, we
just give up and discard other seek ranges, even if the previously
mentioned packet invariants are fulfilled.
To handle joining, we're being particularly dumb, and require a small
overlap to be confident that two ranges join perfectly. (This could be
done incrementally with as little overlap as 1 packet, but corner cases
would eat us: each stream needs to be joined separately, and the cache
pruning logic could remove overlapping packets for other streams again.)
Another restriction is that switching the cached range will always
trigger an asynchronous low level seek to resume demuxing at the new
range. Some users might find this annoying.
Dealing with interleaved subtitles is not fully handled yet. It will
clamp the seekable range to where subtitle packets are.
libavcodec can't deal with them, because its API doesn't distinguish
between 0 sized packets and sending EOF. As such, keeping them doesn't
do any good, ever. This actually fixes some obscure mkv sample (see
previous commit).
Fixes some obscure sample that uses fixed size laces with 0-sized lace
size. Some broken shit. (Maybe the decoder wouldn't care about these
packets, but the demuxer attempted to resync after these packet reading
errors, even though they were perfectly recoverable. But I don't care
enough about this.)
Sample link: https://samples.ffmpeg.org/Matroska/switzler084d_dl.mkv
This directly reads individual mkv sub-packets (block laces) into a
dedicated AVBufferRefs, which can be directly used for creating packets
without a additional copy of the packet data. This also means we switch
parsing of block header fields and lacing metadata to read directly from
the stream, instead of a memory buffer.
This could have been much easier if libavcodec didn't require padding
the packet data with zero bytes. We could just have each packet
reference a slice of the block data. But as it is, the only way to get
padding without a copy is to read the laces into individually allocated
(and padded) memory block, which required a larger rewrite.
This probably makes recovering from broken mkv files slightly worse if
the transport is unseekable. We just read, and then check if we've
overread. But I think that shouldn't be a real concern.
No actual measureable performance change. Potential for some
regressions, as this is quite intrusive, and touches weird obscure shit
like mkv lacing. Still keeping it because I like how it removes some
redundant EBML parsing functions.
This adds a bunch of stuff (mostly unused or redundant) as preparation
for supporting multiple seek ranges. Actual support is probably still
far away.
One change that messes deeper with the actual code is that we account
for total buffered bytes instead of just the backwards bytes now. This
way, prune_old_packets() doesn't have to iterate over all seek ranges to
determine whether something needs pruning.
The main purpose of this commit is avoiding any hidden O(n^2) algorithms
in the code for pruning the demuxer cache, and for determining the
seekable boundaries of the cache. The old code could loop over the whole
packet queue on every packet pruned in certain corner cases.
There are two ways how to reach the goal:
1) commit a cardinal sin
2) do everything incrementally
The cardinal sin is adding an extra field to demux_packet, which caches
the determined seekable range for a keyframe range. demux_packet is a
rather general data structure and thus shouldn't have any fields that
are not inherent to its use, and are only needed as an implementation
detail of code using it. But what are you gonna do, sue me?
In the future, demux.c might have its own packet struct though. Then the
other existing cardinal sin (the "next" field, from MPlayer times) could
be removed as well.
This commit also changes slightly how the seek end is determined. There
is a note on the manpage in case anyone finds the new behavior
confusing. It's somewhat cleaner and might be needed for supporting
multiple ranges (although that's unclear).
The demuxer cache seeking mechanism looks at keyframe ranges to
determine the earlierst PTS of a packet. Instead of looping over all
packets twice (once to find the next keyframe, a second time to find the
seek PTS), do it in one go.
For that basically turn recompute_keyframe_target_pts() into an
iteration functionn. Functionality should be unchanged with this commit.
The base_ts field is used to guess the decoder position, and when set to
NOPTS, it just read ahead arbitrarily. Also demux_add_packet() sets
base_ts to the new timestamp when appending a packet, which would also
make it readahead by a too large amount.
Fix this by setting base_ts after a seek. This assumes that normally, a
cached seek target will always have the timestamp set. This is actually
not quite clear (as it calls recompute_keyframe_target_pts(), which
looks at multiple packets), but maybe it works well enough.
Don't do any of the extra work related to pruning the backbuffer if
demuxer cache seeking is disabled.
As a small extra, always prune packets with no timestamps immediately,
or queue heads that are not keyframes. (Both of them would be pruned
from the backbuffer by the normal logic anyway.)
If fulfilling --demuxer-readahead-secs requires more memory than allowed
by --demuxer-max-bytes, the queue obviously overflows. But the warning
is normally only for the case when trying to find the next video or
audio packet fails, and decoding can't continue.
Use a separate variable for determining whether to prefetch, and if the
queue has overflown, skip the message. In fact, skip the EOF setting and
waking up the decoder thread as well, because the EOF flag should not be
(have been) set in this situation, and waking up the reader thread helps
only if the EOF state changed.
In a shit show of subtle corner case interactions, making the demuxer
cache buffer the entire file can display a small buffered time if
subtitles are enabled. The reason is that some subtitle decoders may
read in advance infinitely, i.e. they read the entire subtitle stream.
Then, since the other streams (audio/video) have logically reached EOF,
and the subtitle stream is set to ds->active==true. This will have to be
fixed properly later to account buffering for subtitle-only files
(another corner case) correctly, but for now this is less annoying.
We don't hope to auto-detect them at load time, as that would be too
much of a pain - even FFmpeg requires fetching and parsing of video
packets, and exposes the information only via deprecated API.
But there still needs to be a way to select them by default. This is
also needed to get the first CC packet at all (without seeking back).
This commit also attempts to clean up locking a bit, which is a PITA,
but it's better be careful & clean.
Even though only 1 seek range is supported at the time.
Other than preparation for possibly future features, the main gain is
actually that we finally separate the reporting for the buffering, and
the seek ranges. These can be subtly different, so it's good to have a
clear separation.
This commit also fixes that the ts_reader wasn't rebased to the start
time, which could make the player show "???" for buffered cache amount
in some .ts files and others (especially at the end, when ts_reader
could become higher than ts_max). It also fixes writing the cache-end
field in the demuxer-cache-state property: it checked ts_start against
NOPTS, which makes no sense.
ts_start was never used (except for the bug mentioned above), so get rid
of it completely. This also makes it convenient to move the segment
check for last_ts to the demux_add_packet() function.
It isn't all that reliable, and improving it would make startup slower
and require more complexity. There isn't even a good reason to do this
(other than semi-broken mkv files), so don't do it. Also see previous
commit.
Avoids that cache seeking is not possible with files that have audio
frames with no timestamps (such as avi, sometimes mkv sub-packets from
lacing). These would set back_pts (first seekable PTS) to NOPTS, and
thus disable cache seeking completely. Instead, prune such packets until
we find one with timestamps.
One corner case is that the new next good packet might be in the forward
cache. In this case we defer dropping until the next time this code is
run, and the reader position has possibly moved past the drop point.
In theory, start/ts_min could be set to NOPTS, in which case
"pts < start" for a valid pts would always evaluate to false.
Also remove the redundant "in-cache seek is possible.." message, as
there's always another message when cache seeks are done.
The seek range computation ignored streams with no timestamps. For
things like buffer estimation this is OK and wanted, but the seek range
needs to be conservative.
Which parts of the queue are considered forward or backward cache
depends on ds->reader_header. The packet at ds->reader_head, as well as
all packets following it (via the ->next field) are considered forward.
The fw_packs/fw_bytes/bw_bytes must be updated accordingly.
This broke in demux_add_packet(), when ds->reader_head was _not_ set on
the first packet (or before). This could happen since commit
05ae571241, which can require skipping packets (so they immediately end
up in the backbuffer).
With the timeline code, a packet at the start or end of a segment can
refer to an invisible frame. So it doesn't extend the seek range, and
the timestamp should be clipped to the actual segment range.
Also restructure recompute_keyframe_target_pts() to be hopefully less
confusing.
Restores some behavior from before the demuxer cache changes, though
affects mostly just OSD display. The unknown queue state is reserved for
streams with missing or messed up timestamps.
Fully fixes behavior of the files mentioned in the previous commit. Will
probably lead to worse behavior if someone tries to fix real video and
cover art tracks, but that's a broken fringe case anyway.
This fixes .cue files with audio files that contain attached pictures to
some degree. demux_timeline.c just discarded packets with unset index,
so the picture was never fed to the decoder.
Directory-opening never worked on Windows because MSVCRT's open()
doesn't open directories and its fstat() doesn't recognise directory
handles. These are just MSVCRT restrictions, and the Windows API itself
has no problem with opening directories as file objects, so reimplement
mpv's mp_open and mp_stat to use the Windows API directly. This should
fix directory playback.
This also populates the st_dev and st_ino fields of struct stat, so
filesystem loop checking in demux_playlist.c should now work on Windows.
Fixes#4711
The new_segment field was used to track the decoder data flow handler of
timeline boundaries, which are used for ordered chapters etc. (anything
that sets demuxer_desc.load_timeline). This broke seeking with the
demuxer cache enabled. The demuxer is expected to set the new_segment
field after every seek or segment boundary switch, so the cached packets
basically contained incorrect values for this, and the decoders were not
initialized correctly.
Fix this by getting rid of the flag completely. Let the decoders instead
compare the segment information by content, which is hopefully enough.
(In theory, two segments with same information could perhaps appear in
broken-ish corner cases, or in an attempt to simulate looping, and such.
I preferred the simple solution over others, such as generating unique
and stable segment IDs.)
We still add a "segmented" field to make it explicit whether segments
are used, instead of doing something silly like testing arbitrary other
segment fields for validity.
Cached seeking with timeline stuff is still slightly broken even with
this commit: the seek logic is not aware of the overlap that segments
can have, and the timestamp clamping that needs to be performed in
theory to account for the fact that a packet might contain a frame that
is always clipped off by segment handling. This can be fixed later.
Needed for a failed thing, leaving it anyway because it causes no harm
and might be less awkward if struct virtual_stream is possibly extended
anyway in the future.
Although seeking past the cached range will trigger a low level seek, a
seek into the region between cache end and last video key frame would
simply seek to the video key frame. This meant that you could get
"stuck" at the end of the file instead of terminating playback when
trying to seek past the end.
One change is that we fix this by _actually_ allowing SEEK_FORWARD to
seek past the last video keyframe in find_seek_target().
In that case, or otherwise seeking to cache buffer end, it could happen
that we set ds->reader_head=NULL if the seek target is after the current
packet. We allow this, because the end of the cached region is defined
by the existence of "any" packet, not necessarily a key frame. Seeking
there still makes sense, because we know that there is going to be more
packets (or EOF) that satisfy the seek target.
The problem is that just resuming demuxing with reader_head==NULL will
simply return any packets that come its way, even non-keyframe ones.
Some decoders will produce ugly soup in this case. (In practice, this
was not a problem, because seeking at the end of the cached region was
rare before this commit, and also some decoders like h264 will skip
broken frames by default anyway.)
So the other change of this commit is to enable key frame skipping.
As a nasty implementation detail, we use a separate flag, instead of
setting reader_head to the first key frame encounted (reader_head being
NULL can happen after a normal seek or on playback start, and then we
want to mirror the underlying demuxer behavior, for better or worse).
This change is relatively untested, so you get to keep the pieces for
yourself.
Seems like most code dealing with this was for setting it in redundant
cases. Now SEEK_BACKWARD is redundant, and SEEK_FORWARD is the odd one
out.
Also fix that SEEK_FORWARD was not correctly unset in try_seek_cache().
In demux_mkv_seek(), make the arbitrary decision that a video stream is
not required for the subtitle prefetch logic to be active. We might want
subtitles with long duration even with audio only playback, or if the
file is used as external subtitle.
If a packet uses segmentation, the codec field must be set. Copying the
codec field was forgotten as an oversight, which is why this just
crashes. This showed up only now, because demux_copy_packet() was not
used before in the main demux path until recently.
Fixes#5027.
This improves upon the previous commit, and partially rewrites it (and
other code). It does:
- disable the seeking within cache by default, and add an option to
control it
- mess with the buffer estimation reporting code, which will most likely
lead to funny regressions even if the new features are not enabled
- add a back buffer to the packet cache
- enhance the seek code so you can seek into the back buffer
- unnecessarily change a bunch of other stuff for no reason
- fuck up everything and vomit ponies and rainbows
This should actually be pretty usable. One thing we should add are some
properties to report the proper buffer state. Then the OSC could show a
nice buffer range. Also configuration of the buffers could be made
simpler. Once this has been tested enough, it can be enabled by default,
and might replace the stream cache's byte ringbuffer.
In addition it may or may not be possible to keep other buffer ranges
when seeking outside of the current range, but that would be much more
complex.
More the ignore_eof field to the internal demux_stream struct. This is
relatively messy, because the internal struct exists only once the
stream is created, and after that setting the ignore_eof flag is a race
condition. We could bother with adding demux_add_sh_stream() parameters
for this, but let's not. So in theory a tiny race condition is
introduced, which can never be triggered since all demux API functions
are called by the playback thread only anyway.
Fix that ts_offset is accessed without log (this was introduced much
earlier by myself).
Introduce an alternative way of avoiding the annoying EOF reached
messages by not resetting the EOF flags for CC streams when a CC packet
is added. This makes the second commit in the PR which added the
original fix unnecessary.
As another cosmetic change merge the check in cached_demux_control()
into a single if().
In the future, the CC pseudo-stream should probably be replaced with an
entire pseudo-demuxer or such, which would avoid some of the messiness
(or maybe not, we don't know yet).
iive agreed to relicense things that are still in mpv to LGPLv2.1. So
change the licenses of the affected files, and rename the configure
switch for LGPL mode to --enable-preliminary-lgpl2.
(The "preliminary" part will probably be removed from the configure
switch soon as well.)
Also player/main.c hasn't had GPL parts since a few commits ago.
See "Copyright" file for caveats.
This changes the remaining "almost LGPL" files to LGPL, because we think
that the conditions the author set for these was finally fulfilled.
This adds handling of spherical video metadata: retrieving it from
demux_lavf and demux_mkv, passing it through filters, and adjusting it
with vf_format. This does not include support for rendering this type of
video.
We don't expect we need/want to support the other projection types like
cube maps, so we don't include that for now. They can be added later as
needed.
Also raise the maximum sizes of stringified image params, since they
can get really long.
If --demuxer-mkv-probe-start-time=no is used, and a seek is triggered on
start, then cluster_start will be 0, and the packet reading code will
print an error message about not finding valid data. This fixes itself
since it invokes the resync code, but it's still pretty ugly. Avoid this
by always initializing cluster_start.
This currently is only limited to Android. Its stdlib contains the
things that mpv's POSIX check checks for, but unfortunately not
glob().
This fixes Android compilation broken in 70a70b9da .
It seems like adjusting the raw stream ID should be done only for DVD.
Otherwise, getting the subtitle language for Bluray breaks.
Untested. Regression since fb9a32977d.
Fixes#4611 (probably).
This API isn't deprecated (yet?), but it's still inferior and harder to
use than avcodec_free_context().
Leave the call only in 1 case in af_lavcac3enc.c, where we apparently
seriously close and reopen the encoder for whatever reason.
Was at least somewhat broken, and is misleading. I don't really have an
idea why FFmpeg has two AVOptions here anyway. We don't need to care,
and I'm only aware of 1 user trying this option ever.
See #4579.
The first time I saw a user try to use this option, and apparently it
didn't work. I'm not exactly sure why, but the code seems to be broken
anyway. Apart from not doing any error checking (neither mallocs nor
warning the user against invalid input), it forgets to add a 0
terminator.
Use the corresponding AVOption instead, which probably works.
See #4579.
All relevant authors have agreed.
I'm removing the NV12 FourCC, which was added in f910f3d9 by someone who
was not contacted for the relicensing. I doubt the remaining code is
copyrightable (basically all what remains the fact is that NV12 uses the
same amount of space like YV12), but in this case I feel more
comfortable removing it.
Might contain some trace amounts of "michael"'s copyright, who agrees
with LGPL only once the core is relicensed - but with the core already
mostly relicensed, I'm changing the license header to LGPL, and only
marking this in the "Copyright" file.
cehoyos, who did not agree to the relicensing, added bcb5c78ce3. If
there was copyright, we consider it gone, because the table changed. It
does not map file extension to a FourCC anymore, and codecs.conf is
gone. The new mapping is a libavcodec codec name (happens to be the same
as the file extension).
The same applies to commits 60ecafec, b749836b, 5b3e3be1. None of these
authors were contacted. These were before the code was replaced with a
table (in d0326807). The parts outside of demux_mf.c were removed a long
time ago. Like in the previous comment, we don't think any copyright
applies at least to the new code (at least after the FourCC removal).
iive authored 0aa37a0d, which is probably still left in some form, and
makes demux_mf.c "LGPL 3 or later".
stream_avdevice.c (unrelated) has been marked as LGPL before.
- Fix a signed/unsigned comparison involving info.segment_uid.len
(doesn't actually warn here, but seems fragile). Code was previously
safe though.
- Match up all printf format strings with the respective value types,
using the *int*_t printf specifiers where necessary, and fixing
multiple signed/unsigned differences. Removed some casts that
otherwise may have truncated values.
- Fix a warning when initializing ebml_info.
On some platforms, unsigned long and uint64_t aren't the same type,
after all. As this is just a MP_VERBOSE message, risking truncation in
some cases seems OK.
As usual, the history of these files is a bit murky. It starts with the
initial commit. (At which some development had already been done,
according to the AUTHORS and ChangeLog files at the time, we should be
but covered with relicensing agreements, though.) then it goes on with
complete lack of modularization, which was cleaned up later (cd68e161).
As usual, we don't consider the copyright of the stuff that has been
moved out cleanly.
There were also contributions to generic code by people who could not be
reached or who did not agree to the relicensing, but this was all
removed.
The only patches that we could not relicense and which were still in the
current code in some form are from Dénes Balatoni: 422b0d2a, 32937181.
We could not reach him, so commits f34e1a0d and 18905298 remove his
additions. It still leaves the demux_control() declaration itself, but
we don't consider it copyrightable. It's basically an idiom that existed
in MPlayer before that change, applied to the demuxer struct. (We even
went as far as making sure to remove all DEMUXER_CTRLs the original
author added.)
Commit be54f481 might be a bit of a corner case, but this was rewritten,
and we consider the old copyright removed long ago.
Similar purpose as f34e1a0dee.
Somehow this is much more natural too, and needs less code.
This breaks runtime updates to duration. This could easily be fixed, but
no important demuxer does this anyway. Only demux_raw and demux_disc
might (the latter for BD/DVD). For the latter it might actually have
some importance when changing titles at runtime (I guess?), but guess
what, I don't care.
This is more uniform, and potentially gets rid of some past copyrights.
It might be that this subtly changes caching behavior (it seems before
this, it synced to the demuxer if the length was unknown, which is not
what we want.)
MaxCLL is the more authoritative source for the metadata we are
interested in. The use of mastering metadata is sort of a hack anyway,
since there's no clearly-defined relationship between the mastering peak
brightness and the actual content. (Unlike MaxCLL, which is an explicit
relationship)
Also move the parameter fixing to `fix_image_params`
I don't know if the avutil check is strictly necessary but I've included
it anyway to be on the safe side.
List of changes:
1. Kill nom_peak, since it's a pointless non-field that stores nothing
of value and is _always_ derived from ref_white anyway.
2. Kill ref_white/--target-brightness, because the only case it really
existed for (PQ) actually doesn't need to be this general: According
to ITU-R BT.2100, PQ *always* assumes a reference monitor with a
white point of 100 cd/m².
3. Improve documentation and comments surrounding this stuff.
4. Clean up some of the code in general. Move stuff where it belongs.
This file is an leftover from when img_format.h was changed from using
the ancient FourCCs (based on Microsoft multimedia conventions) for
pixel formats to a simple enum. The remaining cases still inherently
used FourCCs for whatever reasons.
Instead of worrying about residual copyrights in this file, just move it
into code we don't want to relicense (the ancient Linux TV code). We
have to fix some other code depending on it. For the most part, we just
replace the MP_FOURCC macro with libavutil's MKTAG (although the macro
definition is exactly the same). In demux_raw, we drop some pre-defined
FourCCs, but it's not like it matters. (Instead of
--demuxer-rawvideo-format use --demuxer-rawvideo-mp-format.)
Since this demuxer is based on code by michael, this file can become
LGPL only once the mpv core becomes LGPL, and this is preparation for
it.
There were quite a lot of changes for rearranging preferred libavformat
vs. internal MPlayer demuxers, codec mappings, and filename extensions,
but all this got removed, so some of the relevant authors weren't asked.
cehoyos, who disagreed with LGPL, made a few changes in the past (mostly
codec mapping and deinterlacing related things), but all of them were
removed, mostly due to libavformat API cleanups.
adland, who could not be reached, did commit 057916ee65, but it's easy
to essentially revert the change (this is what the source changes in
this commit do), so we don't need to think about it.
Chris Welton, who could not be reached, made a simple change in commit
958c41d9b6. Fortunately, the API changed again, and his changes were
removed, so we don't need to think about this either.
There is an anonymous contribution in commit 085f35f4b4 - since this
did not introduce any original code, and the probe code was heavily
rewritten multiple times, I don't consider it relevant.
This switches back the --demuxer-lavf-probe-info default for HLS from
"no" to "yes".
Apparently the old default caused problems with the FFmpeg MediaCodec
wrapper. I'm not sure whether it's due to the extradata (which would not
make any sense as MediaCodec takes in Annex B formatted h264 data), or
something else. Reportedly, enabling probing fixes it though, so enable
it again.
Add disparaging comment about Google software/APIs here.
This affects in particular the heuristic that enables byte seeks in some
cases with .ts input. --demuxer-lavf-hacks=no should disable this
behavior now.
In the previous commit, I claimed that this GUID stuff was a libavformat
extension, but that seems to be completely wrong. (LATM might be an
extension of some kind, though.)
I don't know what Microsoft calls this GUID "suffix" though. It's
generally used to wrap wav format tags and video FourCCs as GUIDs.
I guess you could grep Microsoft headers for it to find its name,
or something.
Indeed, FFmpeg found a way to maximize the misery around VfW/AVI-style
muxing. It appears it can mux a number of random codecs by using random
format tags. To make this even more stranger, it has a probably custom
GUID for signaling them, although for unknown reasons this is done only
"sometimes" (judging from FFmpeg's riffenc.c).
Whatever, it's not too hard to support it. Also apparently fix the
incorrect interpretation of extended formats - there's absolutely no
reason to assume they're always PCM. Instead, check for the correct
GUIDs. Also while we're at it, move the channel mask handling also to
codec_tag.c, so all WAVEFORMATEXTENSIBLE handling is in one place. (With
the normal wav header handling strangely still in demux_mkv.c.)
The case I was looking at (aac_latm muxing) decodes now. While I'm not
entirely sure about its correctness (libavformat has a weird
special-case for SBR), it certainly doesn't try to play it as PCM,
which is much of an improvement.
The extradata mess in the demux_mkv.c A_MS/ACM code path is unfortunate
and ugly, but has less impact than refactoring all the code to make
this specific case nicer.
Did I mention yet that I hate VfW-style mkv muxing?
Try to read header elements stored at the end of the file in the order
of their position. (It would be nicer if mkv simply told us a range of
elements to parse, but it doesn't do that.)
This can potentially reduce seek elements, although I didn't check if
any real files trigger this. The real contribution by this change is
that it does not defer reading the CUE index if we need to seek to the
end of the file anyway. This can actually avoid 2 seeks when opening a
file and --start is used, and the file has other headers elements at the
end of the file (like tags).
All authors of the current code have agreed.
For most of its life, MPlayer used Microsoft structs like WAVEFORMATEX
to describe media content. It appears these were copied from wine in
61c5a99851. Copyright is unclear, but mpv completely removed use of
these structs anyway. (demux_mkv.c still contains code to read these
fields from a byte stream, but the struct is fully gone.)
42f97b2b820: cehoyos (who probably didn't agree with LGPL) applied a
patch by someone who agreed. It's unknown whether cehoyos modified the
patch (and thus his copyright would apply), but the changed code was
removed anyway. (The code was first moved somewhere else, then removed.)
efd53eed614: the patch author was not asked. Although the mkv_sh_sub_t
struct was later moved to stheader.h, the added field was removed
without replacement.
f6878753fbd: nick, who could not be reached, added an include guard, but
the guard was changed several times later, and it's probably not
copyrightable anyway.
afb0fd5ea17: the same nick adds a field that was later replaced and
finally removed again in 8cd6b20571.
All contributors have agreed. In 3a43f13fce, someone who potentially
disagreed reverted a commit by someone else (restoring the original
state). This shouldn't matter for Copyright, and all of the affected
code was rewritten/removed anyway.
Most contributors have agreed. This claims it's based on gstreamer code,
but this was LGPL at the time (and still is). Contributors whose code
was removed were not accounted for. There are still some potentially
problematic cases:
06eee1b67 is potentially the most problematic case. Most of these
changes are gone due to mpv not using BITMAPINFOHEADER anymore. Some
of the other changes are rather trivial. If someone contests this and
claims that copyrightable changes are left, the original change can
simply be reverted.
62bfae140 has only 2 lines left: a "char *name;" struct field, and a
line that prints a message. All other code was removed. The parsing code
in particular was made declarative, which replaced reading this element
explicitly (and other elements, see 1b22101c77). I'm putting the log
message under HAVE_GPL, but I don't think the declaration is
copyrightable, or the mere concept of reading this element. Redoing the
other 2 lines of code would result in the same program text.
d41e860ba was applied by someone who (potentially) disagreed. The patch
itself is from someone who did agree, though. It's unknown whether the
applier changed the patch. But it seems unlikely, and the change was
mostly rewritten.
50a86fcc3 all demux_mkv changes were reverted (old stdout slave mode)
3a406e94d same
2e40bfa13 the old MPlayer subtitle code was completely removed
316bb1d44 completely removed in 1cf4802c1d87f93d9d7 same
11bfc6780 relative seeks were removed in 92ba630796be54f4813 the corresponding demux_mkv code was removed in 5dabaaf093efd53eed6 all internal vobsub handling is now in FFmpeg
d7f693a20 removed in f3db4b0b93e8a1b3713 removed in 522ee6b783cfb890259 removed, see 6b1374b203 for analysis
c80808b5a same
It's all explained in the DOCS changes. Although this option was always
kind of obscure and pointless. Until it is removed, the only reason for
setting it would be to raise the static default limit, so change its
default to INT_MAX so that it does nothing by default.
Apparently fixes youtube mp4 streams if avformat_find_stream_info() is
not called.
Keeping audio/video track and other track durations separate is for
the sake of embedded subtitle streams, where we want to include the
duration of overlong subtitle streams (I think).
Includes hls, mp4, mkv by default. This also avoids stupid things like
decoding at least 1 video frame per stream in the demuxer.
This also add --demuxer-lavf-probe-info to give finer control over what
happens.
We use the metadata provided by youtube-dl to sort-of implement
fragmented DASH streaming.
This is all a bit hacky, but hopefully a makeshift solution until
libavformat has proper mechanisms. (Although in danger of being one
of those temporary hacks that become permanent.)
Instead of enabling it only when a stream-cache is enabled, also try to
enable it independently from that if the demuxer is marked as
is_network.
Also add some code to the EDL code, so EDLs containing network streams
are automatically cached this way.
Extend the OSD info line so that it shows the demuxer cache in this case
(more or less).
I didn't find where or whether options.rst describes how the demuxer
cache is enabled, so no changes there.
Because it's kind of dumb. (But not sure if it was worth the trouble.)
For stream_file.c, we add new explicit fields. The rest are rather
special uses and can be killed by comparing the stream impl. name.
The changes to DVD/BD/CD/TV are entirely untested.
"uncached_stream" is a pretty bad name. It could be mistaken for a
boolean, and then its meaning would be inverted. Rename it.
Also add a "caching" field, which signals that the stream is a cache or
reads from a cache. This is easier to understand and more flexible.
This could potentially have caused fun crashes if the --tv-channels
option was used, and something more advanced than tv:// was used to open
it. (This code is still untested.)
Fixes seeking with:
https://bugs.chromium.org/p/chromium/issues/detail?id=497889
Haali also ignores the element's contents, and interprets its presence
as the block not being a keyframe. FFmpeg is going to have an equivalent
change.
I don't know yet whether the affected sample is valid - a reference
timestamp of 0 doesn't make too much sense to me.
Some files appear to use them. In the sample I've seen, one field was
"Performer" instead of "PERFORMER".
This change is slightly risky, because it increases the chance of
misdetecting other formats as cue files.
Fixes#4057.
The FFmpeg versions we support all have the APIs we were checking for.
Only Libav missed them. Simplify this by explicitly checking for FFmpeg
in the code, instead of trying to detect the presence of the API.
This was excessively useless, and I want my time back that was needed to
explain users why they don't want to use it.
It captured the byte stream only, and even for types of streams it was
designed for (like transport streams), it was rather questionable.
As part of the removal, un-inline demux_run_on_thread() (which has only
1 call-site now), and sort of reimplement --stream-dump to write the
data directly instead of using the removed capture code.
(--stream-dump is also very useless, and I struggled coming up with an
explanation for it in the manpage.)
Disabling cache readahead by default until at least 1 track is selected
is mainly for external files and such, where you don't want them to use
up resources until they're actually used.
It doesn't make sense to disable the cache for the demuxer opened for
prefetch. Also, it's fine to let it do that for the main file too (doing
or not doing it is of little consequence). That saves us from having to
distinguish them.
Cover art handling is a disgusting hack that causes a mess in all
components. And this will stay this way. This is the Xth time I've
changed cover art handling, and that will probably also continue.
But change the code such that cover art is injected into the demux
packet stream, instead of having an explicit special case it in the
decoder glue code. (This is somewhat more similar to the cover art hack
in libavformat.)
To avoid that the over art picture is decoded again on each seek, we
need some additional "caching" in player/video.c. Decoding it after each
seek would work as well, but since cover art pictures can be pretty
huge, it's probably ok to invest some lines of code into caching it.
One weird thing is that the cover art packet will remain queued after
seeks, but that is probably not an issue.
In exchange, we can drop the dec_video.c code, which is pretty
convenient for one of the following commits. This code duplicates a
bunch of lower-level decode calls and does icky messing with this weird
state stuff, so I'm glad it goes away.
It has only 1 caller, and is too far appart within the file. I think it
used to have multiple callers, but now it just doesn't make any sense to
keep it separate anymore.
This was probably the intention all along. But I honestly have no idea
what this code even does.
Due to what ebml_read_vlen_int() is used for, this is unlikely to have
mattered anyway as it rarely/never reads huge values. Which is probably
why this has worked for over a decade.
TrueHD is a fucked up audio codec with extremely small frame sizes. Some
of these frames start with full headers, which are usually marked as
keyframes, and from which decoding can be started (or at least that's
what you'd expect).
So for such tracks we should probably trust the keyframe flags. Doesn't
really improve seek behavior, though.
Some files have audio tracks with packets that do not have a keyframe
flag set at all. I don't think there's any audio codec which actually
needs keyframe flags, so always assume an audio packet is a keyframe
(which, in Matroska terminology, means it can start decoding from that
packet).
The file in question had these set:
| + Multiplexing application: Lavf57.56.100 at 313
| + Writing application: Lavf57.56.100 at 329
Garbage produced by garbage...
There are other such files produced by mkvmerge, though. It's not
perfectly sure whether these have been produced by FFmpeg as well
(mkvmerge often trusts the information in the source file, even if it's
wrong - so other samples could have been remuxed from FFmpeg).
Fixes#3920.
This is needed to put the decoders into the correct state. In
particular, decoders will not initialize the current segment without
this flag. The intention of not setting the flag for seeks within the
segments were to avoid costly decoder reinits, but it seems this is
better handled explicitly in the decoder wrappers.
Implementation-wise, the values from the demuxer/codec header are merged
with the values from the decoder such that the former are used only
where the latter are unknown (0/auto).
Matroska actually has lots of colorimetry metadata that video tracks can
use, including mastering metadata (HDR signal peak) etc.
This commit adds the EBML definitions and parses the most basic fields.
Note that nothing uses these fields yet, this commit is just adding the
necessary parsing and infrastructure.
This deals with the estimation of buffered packets, which is used mostly
for display, but also things like pausing on low buffer levels.
If a stream is fully EOF (no more packets), we don't want to include it
in the total buffer amount. This also means we should make ds->eof less
flaky and more stable, so don't reset it in ds_get_packets() (this
function reset ds->eof just to retrigger a packet read attempt - we can
have this slightly simpler). This somewhat fixes buffering display when
e.g. issuing a refresh seek after re-enabling audio/video when playing
with subtitles only.
Commit f72a900892 (and others) added support for ordered editions that
recursively refer to other ordered editions. However, this recursion
code incorrectly activated if the source files had ordered chapters
even if the main file only wanted to use them as raw video, resulting
in broken timeline info overall.
Ordered chapters can specify a ChapterSegmentEditionUID value if they
want to use a specific edition from a source file. Otherwise the
source is supposed to be used as a raw video file. The code checked
demuxer->matroska_data.num_ordered_chapters for an opened source file
to see whether it was using a recursive ordered edition, but demux_mkv
could enable a default ordered edition for the file using the normal
playback rules even if the main file had not specified any
ChapterSegmentEditionUID. Thus this incorrectly enabled recursion if a
source file had a default edition using ordered chapters. Check
demuxer->matroska_data.uid.edition instead, and ensure it's never set
if a file is opened without ChapterSegmentEditionUID.
Also fix what seems like a memory leak in demux_mkv.c.
Signed-off-by: wm4 <wm4@nowhere>
FFmpeg recently got "support" for mov edit lists. This is a terrible
hack that will fail completely at least with some decoders (in
particular wrappers for hardware decoding might be affected). As such it
makes no point to pretend they are supported, even if we assume that the
"intended" functionality works, that there are no implementation bugs
(good luck with all that messy code added to the already huge mov
demuxer), and that it covers enough of the mov edit list feature to be
of value.
So log an error if the FFmpeg code for mov edit lists appears to be
active - AV_PKT_FLAG_DISCARD is used only for "clipping" edit list
segments on non-key frame boundaries.
In the first place, FFmpeg committed this only because Google wanted it
in, and patch review did not even pick up obvious issues. (Just look how
there was no lavc version bump when AV_PKT_FLAG_DISCARD was added.)
We still pass the new packet flag to the decoders (av_common.c change),
which means we "support" FFmpeg's edit list code now. (Until it breaks
due to FFmpeg not caring about all the details.)
When switching a subtitle track, the subtitle wasn't necessarily
updated, especially when playback was paused.
Some awfully subtle and complex interactions here.
First off (and not so subtle), the subtitle decoder will read packets
only on explicit update_subtitles() calls, which, if video is active, is
called only when a new video frame is shown. (A simply video frame
redraw doesn't trigger this.) So call it explicitly. But only if
playback is "initialized", i.e. not when it does initial track selection
and decoder init, during which no packets should be read.
The second issue is that the demuxer thread simply will not read new
packets just because a track was switched, especially if playback is
paused. That's fine, but if a refresh seek is to be done, it really
should do this. So if there's either 1. a refresh seek requested, or 2.
a refresh seek ongoing, then read more packets.
Note that it's entirely possible that we overflow the packet queue with
this in unpredicated weird corner cases, but the queue limit will still
be enforced, so this shouldn't make the situation worse.
This has all been made unnecessary recently. The change not to copy the
global option struct in particular can be made because now nothing
accesses the global options anymore in the demux and stream layers.
Some code that was accidentally added/changed in commit 5e30e7a0 is also
removed, because it was simply committed accidentally, and was never
used.
Don't access MPOpts directly, and always use the new m_config.h
functions for accessing them in a thread-safe way.
The goal is eventually removing the mpv_global.opts field, and the
demuxer/stream-layer specific hack that copies MPOpts to deal with
thread-safety issues.
This moves around a lot of options. For one, we often change the
physical storage location of options to make them more localized,
but these changes are not user-visible (or should not be). For
shared options on the other hand it's better to do messy direct
access, which is worrying as in that somehow renaming an option
or changing its type would break code reading them manually,
without causing a compilation error.
This is for text subtitles. libavformat currently always reads text
subtitles completely on init. This means the underlying stream is
useless and will consume resources for various reasons (network
connection, file handles, cache memory).
Take care of this by closing the underlying stream if we think the
demuxer has read everything. Since libavformat doesn't export whether it
did (or whether it may access the stream again in the future), we rely
on a whitelist. Also, instead of setting the stream to NULL or so, set
it to an empty dummy stream. This way we don't have to litter the code
with NULL checks.
demux_lavf.c needs extra changes, because it tries to do clever things
for the sake of subtitle charset conversion.
The main reason we keep the demuxer etc. open is because we fell for
libavformat being so generic, and we tried to remove corresponding
special-cases in the higher-level player code. Some of this is forced
due to ass/srt mkv/mp4 demuxing being very similar to external text
files. In the future it might be better to do this in a more
straight-forward way, such as reading text subtitles into libass and
then discarding the demuxer entirely, but for aforementioned reasons
this could be more of a mess than the solution introduced by this
commit.
Probably fixes#3456.
Cleaner and makes it easier to change the underlying stream.
mp_property_stream_capture() still directly accesses it directly via
demux_run_on_thread(). This is evil, but still somewhat sane and is not
getting into the way here.
Not sure if I got all field accesses.
It doesn't necessarily have to mean anything bad.
We're still too lazy to provide any more detailed information (e.g.
whether this happened to likely bad interleaving, excessive amount of
packets like with some ASS subs, or that the readahead user option is
limitted by the packet queue size).
Instead of passing through double float timestamps opaquely, pass real
timestamps. Do so by always setting a valid timebase on the
AVCodecContext for audio and video decoding.
Specifically try not to round timestamps to a too coarse timebase, which
could round off small adjustments to timestamps (such as for start time
rebasing or demux_timeline). If the timebase is considered too coarse,
make it finer.
This gets rid of the need to do this specifically for some hardware
decoding wrapper. The old method of passing through double timestamps
was also a bit questionable. While libavcodec is not supposed to
interpret timestamps at all if no timebase is provided, it was
needlessly tricky. Also, it actually does compare them with
AV_NOPTS_VALUE. This change will probably also reduce confusion in the
future.
When an ogg track upodates metadata, we have to perform a complicated
runtime update due to the demux.c architecture. A detail was broken and
an array was allocated with the previous number of streams, which
usually led to invalid memory write accesses at least on the first
update.
See github commit comment on commit b9ba9a89.
If the PEAK tag is invalid, return an error.
Make the error signalling conventions more uniform by strictly returning
a negative value on error, and treating >=0 as success.
The demuxer layer usually doesn't log per-stream information, and even
the replaygain information was logged only if it came from tags.
So log it in af_volume instead.
...and ignore it. The main purpose is for retrieving per-track
replaygain tags. Other than that per-track tags are not used or accessed
by the playback core yet.
The demuxer infrastructure is still not really good with that whole
synchronization thing (at least in part due to being inherited from
mplayer's single-threaded architecture). A convoluted mechanism is
needed to transport the tags from demuxer thread to user thread. Two
factors contribute to the complexity: tags can change during playback,
and tracks (i.e. struct sh_stream) are not duplicated per thread.
In particular, we update the way replaygain tags are retrieved. We first
try to use per-track tags (common in Matroska) and global tags
(effectively formats like mp3). This part fixes#3405.
Play a trick to make the packet pos field monotonically increasing over
segment boundaries if the source demuxers return monotonically
increasing pos values. This allows the demuxer to uniquely identify
packets with the pos field, and can do refresh seeks using that.
Normally, the packet pos field is used as a fallback for determining the
playback position if the demuxer returns no proper duration. But
demux_timeline.c always will, and the packet pos fields usually make no
sense in relation to the returned file size anyway if the timeline
source demuxers originate from separate streams.
Remove the explicit whitelisting of formats for refresh seeks. Instead,
check whether the packet position is somewhat reliable during demuxing.
If there are packets without position, or the packet position is not
monotonically increasing, then do not use them for refresh seeks.
This does not make sure of some requirements, such as deterministic
seeks. If that happens, mpv will mess up a bit on stream switching.
Also, add another method that uses DTS to identify packets, and prefer
it to the packet position method. Even if there's a demuxer which
randomizes packet positions, it hardly can do that with DTS. The DTS
method is not always available either, though. Some formats do not have
a DTS, and others are not always strictly monotonic (possibly due to
libavformat codec parsing and timestamp determination issues).
If the packet read function returns, and EOF was detected, and a seek
was issued in the meantime, then don't use the EOF result. The seek will
be processed later, and reset the EOF state anyway.
The main effect is probably that we don't return EOF to the decoders
(which the playback core resets before issuing the seek), and that we
won't log an EOF message.
Not important, but slightly more correct.
When switching tracks, we normally have the problem that data gets lost
due to readahead buffering. (Which in turn is because we're stubborn and
instruct the demuxers to discard data on unselected streams.) The
demuxer layer has a hack that re-reads discarded buffered data if a
stream is enabled mid-stream, so track switching will seem instant.
A somewhat similar problem is when all tracks of an external files were
disabled - when enabling the first track, we have to seek to the target
position.
Handle these with the same mechanism. Pass the "current time" to the
demuxer's stream switch function, and let the demuxer figure out what to
do. The demuxer will issue a refresh seek (if possible) to update the
new stream, or will issue a "normal" seek if there was no active stream
yet.
One case that changes is when a video/audio stream is enabled on an
external file with only a subtitle stream active, and the demuxer does
not support rrefresh seeks. This is a fuzzy case, because subtitles are
sparse, and the demuxer might have skipped large amounts of data. We
used to seek (and send the subtitle decoder some subtitle packets
twice). This case is sort of obscure and insane, and the fix would be
questionable, so we simply don't care.
Should mostly fix#3392.
This commit adds an --audio-channel=auto-safe mode, and makes it the
default. This mode behaves like "auto" with most AOs, except with
ao_alsa. The intention is to allow multichannel output by default on
sane APIs. ALSA is not sane as in it's so low level that it will e.g.
configure any layout over HDMI, even if the connected A/V receiver does
not support it. The HDMI fuckup is of course not ALSA's fault, but other
audio APIs normally isolate applications from dealing with this and
require the user to globally configure the correct output layout.
This will help with other AOs too. ao_lavc (encoding) is changed to the
new semantics as well, because it used to force stereo (perhaps because
encoding mode is supposed to produce safe files for crap devices?).
Exclusive mode output on Windows might need to be adjusted accordingly,
as it grants the same kind of low level access as ALSA (requires more
research).
In addition to the things mentioned above, the --audio-channels option
is extended to accept a set of channel layouts. This is supposed to be
the correct way to configure mpv ALSA multichannel output. You need to
put a list of channel layouts that your A/V receiver supports.
Aman Gupta
Nicolas F
wm4
sergey.dobrodey
Jan Ekström
Philip Langdale
sfan5
Daniel Kucera
James Ross-Gowan
Raúl Peñacoba
Rudolf Polzer
Niklas Haas
wang-bin
Uoti Urpala