This considered only index entries that were for the same track ID as
the track used for seeking. This doesn't make much sense for preroll;
it'll just possibly skip clusters, and select an earlier cluster.
One possible negative side-effect is that the preroll might be too tight
now, and miss subtitle packets more often.
The demuxer has a hack to seek to the cluster before the target cluster
in order to "catch" subtitle lines that start before the seek target,
but overlap with the video after the seek target.
Avoid this hack if the cue index indicates that there are no overlapping
subtitle packets that can be caught by seeking to the previous cluster.
Nothing is done with them yet. This is preparation for the following
commit.
CueRelativePosition isn't even saved anywhere, because I don't intend to
use it. (Too messy for no gain.)
Instead of indexing only 1 packet per cluster (which is enough for
working seeking), add every packet to the index.
Since on seek, we go through every single index entry, this probably
makes seeking slower. On the other hand, this code is used for files
without index only (e.g. incomplete files), so it probably doesn't
matter much.
Preparation for the following commits.
If no packets are queued, the readahead time is obviously 0.
If the end time is smaller than the start time, the problem is probably
that audio and video start at slightly different times - report 0 in
this case too.
Do this because seeing "???" as readahead time is a bit annoying.
This influences the demuxer readahead display. If a stream has reached
EOF, we want to ignore it for the purpose of this calculation.
Note that if a stream contains no packets, it still should cause the
value 0s to be displayed (unless it's EOF), because that's just the
actual situation.
This was relying on the fact that timestamps will always be numerically
larger than MP_NOPTS_VALUE, but the trick didn't actually work for
MP_PTS_MIN. Be a bit more sincere, and don't rely on this anymore. This
fixes the comparison, and avoids the readahead amount displaying as
"???" in some situations (since one of the values was NOPTS).
In this case, we didn't find any new packets for this stream, even
though we've read ahead as much as possible. (If reading ahead in this
case, the "Too many packets in the demuxer packet queues" error is
normally printed.)
If we do consider this an underrun, handle_pause_on_low_cache() will
pause and show the "buffering" state, which is not useful.
Could also happen on very bad interleaving.
This mechanism was introduced for Opus, and allows correct skipping of
"preroll" data, as well as discarding trailing audio if the file's
length isn't a multiple of the audio frame size.
Not sure how to handle seeking. I don't understand the purpose of the
SeekPreRoll element.
This was tested with correctness_trimming_nobeeps.opus, remuxed to mka
with mkvmerge v7.2.0. It seems to be correct, although the reported file
duration is incorrect (maybe a mkvmerge issue).
Instead of defining a separate data structure in the core.
For some odd reason, demux_chapter exported the chapter time in
nano-seconds. Change that to the usual timestamps (rename the field
to make any code relying on this to fail compilation), and also remove
the unused chapter end time.
Basically, this will mark the demuxer as seekable with rtmp* and mmsh
protocols. These protocols have network-level time seeking, and whether
you can seek on the byte level does not matter.
Until now, seeking was typically only enabled because of the cache, and
a (nonsensical) warning was shown accordingly.
It still could happen that the server doesn't actually support thse
requests (or simply rejects them), so this is somewhat imperfect.
I'm not sure if this could be done in libavformat instead. Probably not,
because libavformat doesn't seem to have any mechanism for trying one
protocol and reverting (or redirecting) to another one if needed.
This commit is sort of a hack too, because it redirects the URL by
pretending the http:// link is a playlist containing the mmsh:// link.
The list of mime types is borrowed from MPlayer (which has completely
different code to handle this).
This was originally done for DVD/BD/DVB, where the start position could
be something different from 0, and seeking back to 0 would mess it up
completely.
Since we're not quite sure that these streams are unseekable, we can
simplify this somewhat, and also make sure we also start at 0 for normal
files. Helps a little bit with the following edition reloading commit.
Although this is fine when the stream cache is active (which caches
these and returns the result immediately), it seems cleaner not to
rely on this detail.
Remove the update_cache() call from demux_thread(), because it's sort
of in the way. I forgot why it exists, and there's probably no good
reason for it to exist anyway.
It's needed for some obscure feature in combination with .rar reading.
However, it's unconditionally used by the subtitle loader code, so take
care of not blocking the main thread unnecessarily.
(Untested.)
Apparently using the stream index is the best way to refer to the same
streams across multiple FFmpeg-using programs, even if the stream index
itself is rarely meaningful in any way.
For Matroska, there are some possible problems, depending how FFmpeg
actually adds streams. Normally they seem to match though.
Especially with other components (libavcodec, OSX stuff), the thread
list can get quite populated. Setting the thread name helps when
debugging.
Since this is not portable, we check the OS variants in waf configure.
old-configure just gets a special-case for glibc, since doing a full
check here would probably be a waste of effort.
Normally, we pass libavformat demuxers a wrapped mpv stream. But in some
cases, such as HLS and RTSP, we let libavformat open the stream itself.
In these cases, set typical network properties like useragent according
to the mpv options.
(We still don't set it for the cases where libavformat opens other
streams on its own, e.g. when opening the companion .sub file for .idx
files - not sure if we maybe should always set these options.)
Fixes opening some streams.
This means the HLS playlist will be opened twice, but that's not much of
a problem, considering it's pretty small, and HLS will make many other
http accesses anyway.
The one in msg.c was mistakenly removed with commit e99a37f6.
I didn't actually test the change in ao_sndio.c (but obviously "ap"
shouldn't be static).
Commit 50e131b43e happened to make it work for DVD (because the higher
bits of the ID are masked in the DVD case), but failed for Bluray. This
probably fixes it, although I don't have a sample to multiple streams to
confirm it really does it right.
CC: @mpv-player/stable
This code meant to flush demuxer internal buffers by doing a byte seek
to the current position. In theory this shouldn't drop any stream data.
However, if the stream positions mismatch, then avio_seek() (called by
av_seek_frame()) stops being a no-op, and might for example read some
data to skip to the seek target. (This can happen if the distance is
less than SHORT_SEEK_THRESHOLD.)
The positions get out of sync because we drop data at one point (which
is what we _want_ to do). Strictly speaking, the AVIOContext flushing is
done incorrectly, becuase pb->pos points to the start of the buffer, not
the current position. So we have to increment pb->pos by the buffered
amount.
Since there are other weird reasons why the positions might go out of
sync (such as stream_dvd.c dropping buffers itself), and they don't
necessarily need to be in sync in the first place unless AVIOContext has
nothing buffered internally, just use the sledgehammer approach and
correct the position manually.
Also run av_seek_frame() after this. Currently, it shouldn't read
anything, but who knows how that might change with future libavformat
development.
This whole change didn't have any observable effect for me, but I'm
hoping it fixes a reported problem.
When flushing the AVIOContext, make sure it can't seek back to discarded
data. buf_ptr is just the current read position, while buf_end - buffer
is the actual buffer size. Since mpegts.c is littered with seek calls,
it might be that the ability to seek could read
Mark the stream (which the demuxer uses) as not seekable. The cache can
enable seeking again (this behavior is sometimes useful for other
things). I think this should have had no bad influence in theory, since
seeking BD/DVD first does the "real" seek, then flushes libavformat and
reads new packets.
Makes it behave slightly better for VP9. This is also the behavior
libavformat has.
Also while we're at it, don't set duration except for the first packet.
Normally we don't use the duration except for subtitles (which are never
parsed or "laced"), so this should make no observable difference.
This was once central, but now it's almost unused. Only vf_divtc still
uses it for extremely weird and incomprehensible reasons. The use in
stream.c is trivial. Replace these, and remove mpbswap.h.
stream_cdda's output format is linked to demux_raw's default audio
format, and at least we don't care enough to provide a separate
mechanism to let stream_cdda explicitly set the format, so they must
match.
Judging from the existing code, it looks like CDDA always outputs little
endian. stream_cdda.c changed this back to native endian (what demux_raw
expects). Just make them both little endian. This requires less code,
and also having a raw demuxer's behavior depend on the endianness of the
machine isn't very sane anyway.
See previous commits. This finally replaces directly reading the file
data into a struct with reading them manually. In theory this is more
portable (no alignment issues and other things). For the most part,
it's nice seeing this gone.
MPlayer traditionally did this because it made sense: the most important
formats (avi, asf/wmv) used Microsoft formats, and many important
decoders (win32 binary codecs) also did. But the world has changed, and
I've always wanted to get rid of this thing from the codebase.
demux_mkv.c internally still uses it, because, guess what, Matroska has
a VfW muxing mode, which uses these data structures natively.
Let codec_tags.c do the messy mapping.
In theory we could simplify further by makign demux_mkv.c directly use
codec names instead of the MPlayer-inherited "internal FourCC" business,
but I'd rather not touch this - it would just break things.
For a while, we used this to transfer PCM from demuxer to the filter
chain. We had a special "codec" that mapped what MPlayer used to do
(MPlayer passes the AF sample format over an extra field to ad_pcm,
which specially interprets it).
Do this by providing a mp_set_pcm_codec() function, which describes a
sample format in a generic way, and sets the appropriate demuxer header
fields so that libavcodec interprets it correctly. We use the fact that
libavcodec has separate PCM decoders for each format. These are
systematically named, so we can easily map them.
This has the advantage that we can change the audio filter chain as we
like, without losing features from the "rawaudio" demuxer. In fact, this
commit also gets rid of the audio filter chain formats completely.
Instead have an explicit list of PCM formats. (We could even just have
the user pass libavcodec PCM decoder names directly, but that would be
annoying in other ways.)
Before this commit, there was AF_FORMAT_AC3 (the original spdif format,
used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS
and DTS-HD), which was handled as some sort of superset for
AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used
IEC61937-framing, but still was handled as something "separate".
Technically, all of them are pretty similar, but may use different
bitrates. Since digital passthrough pretends to be PCM (just with
special headers that wrap digital packets), this is easily detectable by
the higher samplerate or higher number of channels, so I don't know why
you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs.
AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is
just a mess.
Simplify this by handling all these formats the same way.
AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3).
All AOs just accept all spdif formats now - whether that works or not is
not really clear (seems inconsistent due to earlier attempts to make
DTS-HD work). But on the other hand, enabling spdif requires manual user
interaction, so it doesn't matter much if initialization fails in
slightly less graceful ways if it can't work at all.
At a later point, we will support passthrough with ao_pulse. It seems
the PulseAudio API wants to know the codec type (or maybe not - feeding
it DTS while telling it it's AC3 works), add separate formats for each
codecs. While this reminds of the earlier chaos, it's stricter, and most
code just uses AF_FORMAT_IS_IEC61937().
Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to
include special formats, so that it always describes the fundamental
sample format type. This also ensures valid AF formats are never 0 (this
was probably broken in one of the earlier commits from today).
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.
From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.
This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
The last demuxed file position (demuxer->filepos) is used to estimate
the total playback percentage in files with possible timestamp resets
(like MPEG-PS). Until know, reading from any stream set this position
freely. This makes the position jump around.
Fix this by allowing icnreasing file position only. Reset it on seeking.
With crazy formats, this still could go wrong, but there's only so much
you can do.
HLS streams as demuxed by libavformat have no track title metadata. So
show the HLS bitrate if no title is set. Could be useless or annoying,
so it's a bit controversial, I guess.
--hls-bitrate=min/max lets you select the min or max bitrate. That's it.
Something more sophisticated might be possible, but is probably not even
worth the effort.
This inserts an automatic conversion filter if a Matroska file is marked
as 3D (StereoMode element). The basic idea is similar to video rotation
and colorspace handling: the 3D mode is added as a property to the video
params. Depending on this property, a video filter can be inserted.
As of this commit, extending mp_image_params is actually completely
unnecessary - but the idea is that it will make it easier to integrate
with VOs supporting stereo 3D mogrification. Although vo_opengl does
support some stereo rendering, it didn't support the mode my sample file
used, so I'll leave that part for later.
Not that most mappings from Matroska mode to vf_stereo3d mode are
probably wrong, and some are missing.
Assuming that Matroska modes, and vf_stereo3d in modes, and out modes
are all the same might be an oversimplification - we'll see.
See issue #1045.
In the else branch pict_type is always 3, so pict_type != 3 is always
false. (Note that I have no idea of what it was supposed to do and it is
just an equivalent of the old behaviour.)
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.
The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
Use the "native" underrun detection, instead of guessing by a low cache
duration. The new underrun detection (which was added with the original
commit) might have the problem that it's easy for the playloop to miss
the underrun event. The underrun is actually not stored as state, so if
the demuxer thread adds a new packet before the playloop happens to see
the state, it's as if it never happened. On the other hand, this means
that network was fast enough, so it should be just fine.
Also, should it happen that we don't know the cached range (the
ts_duration < 0 case), just wait until the demuxer goes idle (i.e.
read_packet() decides to stop). This pretty much should affect broken or
unusual files only, and there might be various things that could go
wrong. But it's more robust in the normal case: this situation also
happens when no packets have been read yet, and we don't want to
consider this as reason to resume playback.
The cache percentage was useless. It showed how much of the total stream
cache was in use, but since the cache size is something huge and
unrelated to the bitrate or network speed, the information content of
the percentage was rather low.
Replace this with printing the duration of the demuxer-cached data, and
the size of the stream cache in KB.
I'm not completely sure about the formatting; suggestions are welcome.
Note that it's not easy to know how much playback time the stream cache
covers, so it's always in bytes.
The "buffering" logic was active even if the stream cache was disabled.
This is contrary to what the manpage says. It also breaks playback
because of another bug: the demuxer cache is smaller than 2 seconds,
and thus the resume condition never becomes true.
Explicitly run this code only if the stream cache is enabled. Also, fix
the underlying problem of the breakage, and resume when the demuxer
thread stops reading in any case, not just on EOF.
Broken by previous commit. Unbreaks playback of local files.
Add the --cache-secs option, which literally overrides the value of
--demuxer-readahead-secs if the stream cache is active. The default
value is very high (10 seconds), which means it can act as network
cache.
Remove the old behavior of trying to pause once the byte cache runs
low. Instead, do something similar wit the demuxer cache. The nice
thing is that we can guess how many seconds of video it has cached,
and we can make better decisions. But for now, apply a relatively
naive heuristic: if the cache is below 0.5 secs, pause, and wait
until at least 2 secs are available.
Note that due to timestamp reordering, the estimated cached duration
of video might be inaccurate, depending on the file format. If the
file format has DTS, it's easy, otherwise the duration will seemingly
jump back and forth.
The purpose of the unconditional pthread_cond_signal() when reading
cached DEMUXER_CTRLs and STREAM_CTRLs was apparently to update the
stream cache state. Otherwise, the cached fields would never be updated
when the stream is e.g. paused.
The same could be said about other CTRLs, but these aren't as important,
since they are normally updated while reading packet data.
In order to reduce wakeups, make this logic explicit.
This is a simplification, because it lets us use the AVPacket
functions, instead of handling the details manually.
It also allows the libavcodec rawvideo decoder to use reference
counting, so it doesn't have to memcpy() the full image data. The change
in av_common.c enables this.
This change is somewhat risky, because we rely on the following AVPacket
implementation details and assumptions:
- av_packet_ref() doesn't access the input padding, and just copies the
data. By the API, AVPacket is always padded, and we violate this. The
lavc implementation would have to go out of its way to make this a
real problem, though.
- We hope that the way we make the AVPacket refcountable in av_common.c
is actually supported API-usage. It's hard to tell whether it is.
Of course we still use our own "old" demux_packet struct, just so that
libav* API usage is somewhat isolated.
If a packet is appended to a stream, and there were already packets
queued, nothing about the state changed, as far as the user (i.e. the
player) is concerned. Thus no wakeup is needed.
The pthread_cond_signal() call following this is not interesting - it
will simply be a NOP if there are actually no waiters.
--demuxer-readahead-secs now controls how much the demuxer should
readahead by an amount of seconds. This is based on the raw packet
timestamps. It's not always very exact. For example, h264 in Matroska
does not store any linear timestamps (only PTS values which are going
to be reordered by the decoder), so this heuristic is usually off by
several hundred milliseconds.
The decision whether to readahead is basically OR-ed with the other
--demuxer-readahead-packets options. Change the manpage descriptions
to subtly convey these semantics.
Switching tracks caused cached_demux_control() to catch the command to
switch tracks, even if no thread was running. Thus, the tracks were
never really switched, and EOF happened immediately on playback start.
Fix it by not using the cache at all if the demuxer thread is disabled.
The cache code still has to be called somewhere, though, because it
handles stream metadata update.
Regression from today.
Because why not.
This can lead to reordering of operations between seeking and track
switching (happens when the demuxer wakes up after seek and track
switching operations were queued). Do the track switching strictly
before seeks if there is a chance of reordering, which guarantees that
the seek position will always start with key frames. The reverse
(seeking, then switching) does not really have any advantages.
(Not sure if the player relies on this behavior.)
Use OPT_KEYVALUELIST() for all places where AVOptions are directly set
from mpv command line options. This allows escaping values, better
diagnostics (also no more "pal"), and somehow reduces code size.
Remove the old crappy option parser (av_opts.c).
This happens apparently randomly with rtmp:// and after seeks. This
eventually leads to audio decoding returning an EOF status, which
basically disables audio sync. This will lead to audio desync, even if
audio decoding later "recovers" when the demuxer actually returns audio
packets.
Hack-fix this by special-casing EAGAIN.
This didn't work, because the timebase was wrong. According to the
ffmpeg doxygen, if the stream index is -1 (which is what we used), the
timebase is AV_TIME_BASE. But this didn't work, and it really expected
the stream's timebase. Quite "surprising", since this feature
(avio_seek_time) is used by rtmp only.
Fixing this properly is too hard, so hack-fix our way around it.
STREAM_CTRL_SEEK_TO_TIME is also used by DVD/BD, so a new
STREAM_CTRL_AVSEEK is added. We simply pass-through the request
verbatim.
This tells the demuxer thread that it should seek, instead of waiting
until the demuxer thread is ready.
Care has to be taken about the state between seek request and actual
seeking: newly demuxed packets have to be discarded. We can't just
flush when doing the actual seek, because the user thread could read
these packets.
I'm wondering if this could lead to issues due to relaxed ordering of
operations. But it should be fine, since seeking influences packet
reading only, and seeking is always strictly done before that.
Currently, this will have no advantages; unless audio is disabled. Then
seeking as well as normal playback can be non-blocking.
Instead of starting to fill the packet queue if at least 1 stream is
selected, wait until there is at least 1 stream had new packets
requested.
In theory this is cleaner, because it allows you to e.g. do a seek and
then reselect streams without losing packets. Seeking marks all streams
as inactive, and without this new logic, the thread would read new
packets anyway right after seek.
This fixes the same symptom as the previous commit, but when the demuxer
thread is enabled. In this case, if nothing was read from the demuxer,
the STREAM_CTRLs weren't updated either. To the player, this looked like
the stream cache was never making progress, so playback was kept paused.
It can happen that read_packet() doesn't read a packet, even if it
succeeds. Typically this is because a packet was read, but then thrown
away, because it's not part of a selected stream. The result would be a
bogus EOF condition.
Fix by explicitly checking for EOF.
In corner cases, it might be possible that a demux_read_packet_async()
call fails to make the demuxer thread to read more packets.
If a packet is queued, the function will simply return a packet, without
marking the stream as active. As a consequence, read_packet() might
decide not to read any further packets, and the demuxer will never read
a packet and wake up the playback thread.
This was originally done to align it with demux_read_packet() semantics;
just drop this.
demux_read_any_packet() attempts to call read_packet(), but if no stream
is active, it can decide not to read anything. The function will return
NULL, which implies EOF. Fix this by explicitly
setting demux_stream->active if needed.
Also use dequeue_packet() instead of demux_read_packet(), because it's
cleaner. (Shouldn't change behavior.)
Possibly fixes#938.
We told the demuxer that a pipe (if stream cache is enabled) is
seekable. This is because the stream cache is technically seekable, it's
just that seeking may fail at runtime if a non-cached byte range is
requested.
This caused libavformat to issue seeks on initialization (at least when
piping mp4 youtube videos). Initialization failed completely after
spamming tons of error messages.
So, if an unseekable stream is cached, tell the demuxer that the file is
not seekable. This gets reversed later (when printing a message about
caching an unseekable stream), so the user can still try his luck by
issuing a seek command. The important part is that libavformat
initialization will not take code paths that will unnecessarily seek for
whatever reasons.
CC: @mpv-player/stable: regression from 0.3.x
It was easy to get into a wakeup feedback loop on EOF. The reason that
EOF is complicated is that we try to retry reading when EOF is reached,
in case the EOF state actually disappears (e.g. when watching a
currently downloaded file).
This feature is probably worthless, since in practice you have to do a
seek to "unstuck" it anyway, but since the old code also did this, we
want to keep this behavior for now.
Avoid the feedback loop by introducing another EOF flag (last_eof), that
contains the actual previous EOF state, and is not overwritten when
retrying reading. Wakeup is skipped if the EOF state didn't change.
Also, actually call the wakeup callback when EOF is detected.
The line that adds "ds->active = false;" actually does nothing, but in
theory it's cleaner.
The old FFmpeg API and the new Libav API disagree about mp4 display
rotation direction. Well, whatever, fix it trial-and-error-style.
CC: @mpv-player/stable: add
This adds a thread to the demuxer which reads packets asynchronously.
It will do so until a configurable minimum packet queue size is
reached. (See options.rst additions.)
For now, the thread is disabled by default. There are some corner cases
that have to be fixed, such as fixing cache behavior with webradios.
Note that most interaction with the demuxer is still blocking, so if
e.g. network dies, the player will still freeze. But this change will
make it possible to remove most causes for freezing.
Most of the new code in demux.c actually consists of weird caches to
compensate for thread-safety issues (with the previously single-threaded
design), or to avoid blocking by having to wait on the demuxer thread.
Most of the changes in the player are due to the fact that we must not
access the source stream directly. the demuxer thread already accesses
it, and the stream stuff is not thread-safe.
For timeline stuff (like ordered chapters), we enable the thread for the
current segment only. We also clear its packet queue on seek, so that
the remaining (unconsumed) readahead buffer doesn't waste memory.
Keep in mind that insane subtitles (such as ASS typesetting muxed into
mkv files) will practically disable the readahead, because the total
queue size is considered when checking whether the minimum queue size
was reached.
For OGG audio files, we usually merge the per-stream metadata back to
the file-global metadata. Don't do that for OGM, because with OGM most
metadata is actually per-stream.
libdvdnav can actually jump into the middle of the DVD (e.g. scene
selection menus do that). Then time display is incorrect: we start from
0, even though playback time is somewhere else. This really matters when
seeking. If the display time mismatches, a small relative seek will
apparently jump to the beginning of the movie.
Fix this by initializing the PTS stuff on opening. We have to do this
after some small amount of data has been read from the stream (because
libdvdnav is crap and doesn't always update the time between seeks and
the first read; also see STREAM_CTRL_GET_CURRENT_TIME remarks in
cache.c; although this was not observed when testing with scene
selection menus). On the other hand, we want to do it before opening the
demuxer, because that will read large amounts of data and likely will
change the stream position.
Also see commit 49813670.
Technically needed, but not strictly. It seems it works without in
practice, because demux_lavf.c reads exactly one packet for fill_buffer
call, so there are never packets queued.
We used a complicated and approximate method to cache the stream
timestamp, which is basically per-byte. (To reduce overhead, it was only
cached per 8KB-block, so it was approximate.)
Simplify this, and read/keep the timestamp only on discontinuities. This
is when demux_disc.c actually needs the timestamp.
Note that caching is currently disabled for dvdnav, but we still read
the timestamp only after some data is read. libdvdread behaves well, but
I don't know about libbluray, and the previous code also read the
timestamp only after reading data, so try to keep it safe.
Also drop the start_time offset. It wouldn't be correct anymore if used
with the cache, and the idea behind it wasn't very sane either (making
the player to offset the initial playback time to 0).
This should work now, at least kind of. Note that actual success depends
on the behavior of the underlying lib{dvd{nav,read},bluray}
implementation, which could go very wrong.
In the worst case, it could happen that the underlying implementation
seeks a long time before the seek target time. In this case, the player
will just decode video until the target time is reached, even if that
requires e.g. decoding 30 mintues of video before refreshing.
In the not-so-bad but still bad case, it would just miss the seek
target, and seek past it.
In my tests, it works mostly ok, though. Seeking backwards usually
fails, unless something like --hr-seek-demuxer-offset=1 is used (this
makes it seek to 1 second before the target, which may or may not be
enough to compensate for the DVD/BD imprecision).
This is a pretty big change. Instead of doing a half-hearted passthrough
of the playback timestamp, we attempt to rewrite the raw MPEG timestamps
such that they match with the playback time.
We add the offset between raw start timestamp and playback time to the
packet timestamps. This is the easy part; but the problem is with
timestamp resets. We simply detect timestamp discontinuities by checking
whether they are more than 500ms apart (large enough for all video
faster than 2 FPS and audio with reasonable framesizes/samplerates), and
adjust the timestamp offset accordingly.
This should work pretty well. There may be some problems with subtitles.
If the first packet after a timestamp reset is a subtitle instead of
video, it will fail. Also, selecting multiple audio or video streams
won't work (but mpv doesn't allow selecting several anyway). Trying to
demux subtitles with no video stream enabled will probably fail.
Untested with Bluray, because I have no Bluray sample.
Background:
libdvdnav/libdvdread/libbluray make this relatively hard. They return a
raw MPEG (PS/TS) byte stream, and additionally to that provide a
function to retrieve the current "playback" time. The playback time is
what should be displayed to the user, while the MPEG timestamps can be
completely different. Even worse, the MPEG timestamps can reset. Since
we use the libavformat demuxer (instead of parsing the MPEG packets in
the DVD/BD code), it's hard to associate between these timestamps. As a
result, the time display is special cased in the playloop, and of low
quality (updates only all 1 or 2 seconds, sometimes is incorrect). The
fact that the stream cache can be between demuxer and the stream source
makes things worse.
All the libs seem to provide an event that tells whether timestamps are
resetting. But since this signalling is byte based, it's hard to connect
it to the demuxed MPEG packets. It might be possible to create some sort
of table mapping file positions to discontinuities and new timestamps.
(For simplicity, this table could be 2 entries large, sufficient to
catch all discontinuities if the distance between them is larger than
the total buffering.)
It can happen that demux_fill_buffer() adds more than 1 packet, and then
the packets would add up. Affects demux_disc.c only (nothing else uses
this function).
Suggested by tholin on github issue #882.
This is not entirely clean, but the fields we're accessing might be
considered internal to libavformat. On the other hand, existence of the
fields is guaranteed by the ABI, and nothing in the libavformat doxygen
suggestes they're not allowed to be accessed.
CC: @mpv-player/stable
DVD and Bluray (and to some extent cdda) require awful hacks all over
the codebase to make them work. The main reason is that they act like
container, but are entirely implemented on the stream layer. The raw
mpeg data resulting from these streams must be "extended" with the
container-like metadata transported via STREAM_CTRLs. The result were
hacks all over demux.c and some higher-level parts.
Add a "disc" pseudo-demuxer, and move all these hacks and special-cases
to it.
(Again.)
This time, we simply make it event-based, as it should be. This is done
for both demuxer metadata and stream metadata.
For some ogg-over-icy streams, 2 updates are reported on stream start.
This is because libavformat reports an update right on start, while
including the same info in the "static" metadata. I don't know if that's
a bug or a feature.
It's unlikely that files with multiple audio tracks and with replaygain
actually happen, but this change might help avoid minor corner cases
with later changes.
Recently, libavformat added demuxers to open image files like normal
demuxers. This is a good thing, but for now they interfere with the
operation of demux_mf. Add them to the blacklist until there is a proper
solution.
(The list doesn't contain _all_ recognized image formats, just those
that might interfere with demux_mf.)
CC: @mpv-player/stable
This returned a stream error value directly to libavformat, which can't
make sense. For example STREAM_ERROR (0) means success in libavformat
error codes. (The meaning of the libavformat read_seek return value is
underdocumented too.)
Something like "char *s = ...; isdigit(s[0]);" triggers undefined
behavior, because char can be signed, and thus s[0] can be a negative
value. The is*() functions require unsigned char _or_ EOF. EOF is a
special value outside of unsigned char range, thus the argument to the
is*() functions can't be a char.
This undefined behavior can actually trigger crashes if the
implementation of these functions e.g. uses lookup tables, which are
then indexed with out-of-range values.
Replace all <ctype.h> uses with our own custom mp_is*() functions added
with misc/ctype.h. As a bonus, these functions are locale-independent.
(Although currently, we _require_ C locale for other reasons.)
Some of these might be security relevant.
The RealAudio code was especially bad. I'm not sure if all RealAudio
stuff still plays correctly; I didn't have that many samples for
testing. Some checks might be unnecessary or overcomplicated compared
to the (obfuscated) nature of the code.
CC: @mpv-player/stable
Also clarify the semantics.
It seems --idx didn't do anything. Possibly it used to change how the
now removed legacy demuxers like demux_avi used to behave. Or maybe
it was accidental.
--forceidx basically becomes --index=force. It's possible that new
index modes will be added in the future, so I'm keeping it
extensible, instead of e.g. creating --force-index.
Probably "needed" to get the correct alignment, although I'm not aware
of actual breakages or performance issues.
In fact we should probably always just allocate AVPackets, but for now
use the simple fix.
FFmpeg requires a bullshit padding after each input buffer, and they
just increased that padding without warning and without ABI or API bump.
We need this only in one file (although mp_image hardcodes something
similar, for which no FFmpeg API define is available), so drop our own
define.
While I'm not very fond of "const", it's important for declarations
(it decides whether a symbol is emitted in a read-only or read/write
section). Fix all these cases, so we have writeable global data only
when we really need.
Convert all these commands to properties. (Except tv_last_channel, not
sure what to do with this.) Also, internally, don't access stream
details directly, but dispatch commands with stream ctrls.
Many of the new properties are a bit strange, because they're write-
only. Also remove some OSD output these commands produced, because I
couldn't be bothered to port these.
In general, this makes everything much cleaner, and will also make it
easier to e.g. move the demuxer to its own thread.
Don't bother updating input.conf, but changes.rst documents how old
commands map to the new ones.
Mostly untested, due to lack of hardware.
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
wm4
shdown
Andrey Morozov
Marcoen Hirschberg