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.
It used not to work - but now it apparently does. Not sure when that got
fixed in FFmpeg, but there's no longer a reason to keep this hack.
This also gets rid of the check for the read_seek2 field, which is not
part of the public API.
Since the libavformat API is crap, we have to apply tons of heuristics
to check whether seeking will work. (No, checking it at seek time isn't
going to work either, because if a seek fails, the demuxer will be in an
undefined state. Because the libavformat API is crap.)
I've got a broken webm that fails to seek correctly with "--start=0".
The problem is that every index entry points to 1 byte before cluster
start (!!!). demux_mkv tries to resync to the next cluster, but since it
already has read 2 bytes with ebml_read_id(), it doesn't get the first
cluster, but the following one. Actually, it can be any amount of bytes
from 1-4, whatever happens to look valid at this essentially random byte
position.
Improve this by resyncing from the original position, instead of the one
after the EBML element ID has been attempted to be read.
The file shows the following headers:
| + Muxing application: google at 177
| + Writing application: google at 186
Indeed, the file was downloaded with youtube-dl. I can only guess that
Google got it completely wrong.
demux_playlist.c recognizes if the source stream points to a directory,
and adds its directory entries. Until now, only 1 level was added.
During playback, further directory entries could be resolved as
directory paths were "played".
While this worked fine, it lead to frequent user confusion, because
playlist resuming and other things didn't work as expected. So just
recursively scan everything.
I'm unsure whether it's a good fix, but at least it gets rid of the
complaints. (And probably will add others.)
Now it will always be able to seek back to the start, even if the index
is sparse or misses the first entry.
This can be achieved by reusing the logic for incremental index
generation (for files with no index), and start time probing (for making
sure the first block is always indexed).
AVFormatContext.codec is deprecated now, and you're supposed to use
AVFormatContext.codecpar instead.
Handle this for all of the normal playback code.
Encoding mode isn't touched.
This was changed in 2014, so I suppose users will usually have a FFmpeg
release which includes the corresponding upstream change. If not, well
too bad for those MicroDVD-obsessed users.
Also don't try to retrieve the default framerate as exported by the
demuxer, and instead hardcode it and trust it won't ever change. this
avoids that we have to deal with a larger mess in the codecpar commit.
I don't trust it one bit, and it's a bother with the codecpar change.
If it turns out to be important for some file formats, it could be
added back (or FFmpeg fixed).
This reverts commit 503c6f7fd6.
There are situations where some decoders (MF apparently) always require
a timestamp. Also, this makes bitrate estimation more granular than
necessary. It seems it's better to try to detect fiels with broken
default durations explicitly instead. Or maybe something should be
added to smooth audio timestamps after filters.
Instead of having a separate for each, which also requires separate
additional caching in the demuxer. (The demuxer adds an indirection,
since STREAM_CTRLs are not thread-safe.)
Since this includes the cache speed, this should fix#3003.
SEEK_HR is interpreted by demux_mkv.c, and enables subtitle preroll by
prefetching additional subtitle pakcets which might overlap with the
seek destination. This should make the case work when segment boundaries
fall into the middle of subtitle events.
This still usually leaves a flicker of at least 1 frame on start,
because dec_sub.c does not ensure that enough subtitles are read before
rendering after a segment switch. (Probably a WONTFIX.)
This is simpler, because it doesn't have to wait from both threads for
synchronization.
Apart from being simpler/cleaner, this serves vague plans to stop/start
the demuxer thread itself automatically on demand (for the purpose of
reducing unneeded resource usage).
This pause stuff is bothersome and is needed only for a few corner-
cases. This commit removes it from the demuxer public API and replaces
it with a demux_run_on_thread() function and refactors the code which
needed demux_pause(). The next commit will change the implementation.
Commit 503c6f7f essentially removed timestamps from "laces" (Block sub-
divisions), which means many audio packets will have no timestamp.
There's no reason why bitrate calculation can't just delayed to a point
when the next timestamp is known.
Fixes#2903 (no audio bitrate with mkv files).
stream->info can be NULL if it's the cache wrapper. To be fair,
stream->info is considered private API anyway. So don't access it, but
check the URL instead.
wm4
Eric Toombs