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.
The i_bps members of the sh_audio and dev_video structs are mostly used
for displaying the average audio and video bitrates. Keeping them in
bits-per-second avoids truncating them to bytes-per-second and changing
them back lateron.
Stop using it in most places, and prefer STREAM_CTRL_GET_SIZE. The
advantage is that always the correct size will be used. There can be no
doubt anymore whether the end_pos value is outdated (as it happens often
with files that are being downloaded).
Some streams still use end_pos. They don't change size, and it's easier
to emulate STREAM_CTRL_GET_SIZE using end_pos, instead of adding a
STREAM_CTRL_GET_SIZE implementation to these streams.
Make sure int64_t is always used for STREAM_CTRL_GET_SIZE (it was
uint64_t before).
Remove the seek flags mess, and replace them with a seekable flag. Every
stream must set it consistently now, and an assertion in stream.c checks
this. Don't distinguish between streams that can only be forward or
backwards seeked, since we have no such stream types.
stream.start_pos was needed for optical media only, and (apparently) not
for very good reasons. Just get rid of it.
For stream_dvd, we don't need to do anything. Byte seeking was already
removed from it earlier.
For stream_cdda and stream_vcd, emulate the start_pos by offsetting the
stream pos as seen by the rest of mpv.
The bits in discnav.c and loadfile.c were for dealing with the code
seeking back to the start in demux.c. Handle this differently by
assuming the demuxer is always initialized with the stream at start
position, and instead seek back if initializing the demuxer fails.
Remove the --sb option, which worked by modifying stream.start_pos. If
someone really wants this option, it could be added back by creating a
"slice" stream (actually ffmpeg already has such a thing).
Drop: sami, vplayer, rt, pjs, mpsub, aqt, jacosub. None of these seem
to be actually in use, except sami. Sami is very complex, and the
results subreader produces are not very useful.
For all these formats, there are still parsers in FFmpeg. We remove the
subreader implementation, because it might contain security relevant
bugs and such. (This is old, unmaintained C string parsing code, written
in times where absolutely nobody cared about security. The kind of
awesome code.)
We keep the other formats, because they're (mostly) commonly used and
relatively simple, for UTF16 support (still missing in FFmpeg), and for
the sake of Libav.
Caused failure to detect .pls files, because they were misdetected as
m3u. The problem is that "forcing playlist files" and "forcing a
specific playlist format" are not really treated separate, and in both
cases p->force is set to true. This made m3u detect all files as m3u
if --playlist was used. So correctly check whether the file format is
actually being probed or not.
Because the http playlist URL I had for testing claimed to be m3u by
file extension and mime type, but didn't have the header.
Note that this actually changes behavior only in the case the format is
detected by mime type. Then p->force will be set before calling the
parser, and the header becomes optional.
mp3 has a hack lowering the probescore for format detection. This is
because detecting mp3s is hard due to their nature, and the fact that
ID3v2 tags are sometimes several megabytes big.
When playing mp3 from network, the mime-type is usually set, and that
matches the format hack entry meant for webradios, overriding the normal
mp3 entry. This can lead to network mp3s not being detected. Lower the
network case to the same probescore as on-disk mp3s. The difference is
that for network mp3s, we don't load the full probe-buffer, and we lower
the amount of audio the demuxer will read to collect data on opening
(0.5 seconds instead of typically 5 seconds).
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
VP9 packets can contain 2 frames in some video packets (from which 1
frame is invisible). Due to a design mismatch between libvpx and the
libavcodec vp9 decoder, libvpx can take the "full" packets, but lavc vp9
can not. The consequence is that we have to split the packets if we want
to feed them to the lavc codec.
This is not entirely correct yet: timestamp handling is missing.
--demuxer=lavf and ffmpeg native utilities have the same problem. We can
fix this only once the ffmpeg VP9 parser is fixed.
For some reason, some files appear to have broken mp3 packets, or at
least in a form that libavcodec can't deal with. The audio in the sample
file in question could not be decoded using libavcodec.
The problematic file had variable packet sizes, and the libavcodec
decoder kept printing "mp3: Header missing" for each packet it was fed.
Remuxing with mkvmerge fixes the problem. The mp3 data is probably not
VBR, and remuxing resulted in fixed-size mp3 frames. So I don't know why
the sample file was muxed this way - it might just be incorrect.
The sample file had "libmkv 0.6.4" as MuxingApp (although I could not
get mkvinfo to print this element, maybe the file uses an incorrect
element ID), and "HandBrake 0.9.4" as WritingApp.
Note that the libmpg123 decoder does not have any issues with it. It's
probably more robust, because libmpg123 was made to decode whole mp3
files, not just single frames.
Fixes issue #742.
glob() is mandated by POSIX. For the only non-POSIX platform we support,
Windows, we have our own replacement. So the ifdeffery is not needed.
Still leave the checks in the configure scripts, because they have to
decide whether to compile the replacement or not. (Although this could
be special cased to mingw-only, the wscript seems to make this hard.)
glob-win.c wasn't big, so it was easier to rewrite it. The new version
supports Unicode, handles directories properly, sorts the output and
puts all its allocations in the same talloc context to simplify the
implementation of globfree.
Notably, the old glob had error checking code, but didn't do anything
with the errors since the error reporting code was commented out. The
new glob doesn't copy this behaviour. It just treats errors as if there
were no more matching files, which shouldn't matter for mpv, since it
ignores glob errors too.
To match the other Windows I/O helper functions, the definition is moved
to osdep/io.h.
I hate tabs.
This replaces all tabs in all source files with spaces. The only
exception is old-makefile. The replacement was made by running the
GNU coreutils "expand" command on every file. Since the replacement was
automatic, it's possible that some formatting was destroyed (but perhaps
only if it was assuming that the end of a tab does not correspond to
aligning the end to multiples of 8 spaces).
This was broken at some unknown point (even before the recent cache
changes). There are several problems:
- stream_dvd returning a random stream position, confusing the cache
layer (cached data and stream data lost their 1:1 corrospondence by
position)
- this also confused the mechanism added with commit a9671524, which
basically triggered random seeking (although this was not the only
problem)
- demux_lavf requesting seeks in the stream layer, which resulted in
seeks in the cache or the real stream
Fix this by completely removing byte-based seeking from stream_dvd. This
already works fine for stream_dvdnav and stream_bluray. Now all these
streams do time-based seeks, and pretend to be infinite streams of data,
and the rest of the player simply doesn't care about the stream byte
positions.
Instead, always use the mpctx->chapters array. Before this commit, this
array was used only for ordered chapters and such, but now it's always
populated if there are chapters.
Stream-level chapters (like DVD etc.) did potentially not have
timestamps for each chapter, so STREAM_CTRL_SEEK_TO_CHAPTER and
STREAM_CTRL_GET_CURRENT_CHAPTER were needed to navigate chapters. We've
switched everything to use timestamps and that seems to work, so we can
simplify the code and remove this old mechanism.
Instead of parsing the ASS file in demux_libass.c and trying to pass the
ASS_Track to the subtitle renderer, just read all file data in
demux_libass.c, and let the subtitle renderer pass the file contents to
ass_process_codec_private(). (This happens to parse full files too.)
Makes the code simpler, though it also relies harder on the (messy)
probe logic in demux_libass.c.
The mplayer decoder (spudec.c) actually handled this. There was explicit
code for binary palettes (16 32 bit values), and the subtitle resolution
was handled by video resolution coincidentally matching the subtitle
resolution.
Whoever puts vobsub into mp4 should be punished.
Fixes the sample gundam_sample.mp4, closes github issue #547.
This skipped all audio packets before the first video key frame was
found. I'm not really sure why this would be needed; most likely it
isn't. So get rid of it. Even if audio packets are returned to the
player too soon, the player will sync the audio start to the video
start by decoding and discarding audio data.
Note that although the removed code was just added in the previous
commit, it merely kept the old keeping semantics which demux_mkv
always followed. This commit removes these special semantics.
v_skip_to_keyframe is set to true while non-keyframe video packets are
skipped. Until now, audio packets were also skipped when doing this. I
can't see any good reason why this would be done, but for now I want to
keep the old logic when audio+video seeks are done.
However, for audio-only mode, do proper seeking, which also fixes
behavior when trying to seek past the end of the file: playback is
terminated properly, instead of starting playback on the start of the
last cluster.
Note that a_no_timecode_check is used only for audio+video seek. I'm
not sure what this is needed for, but it might influence A/V sync after
seeking.
Trying to set a non-existent flag (like +keepside on Libav) causes
libavutil print an incomprehensible warning (something about eval;
probably the overengineered libavutil option parser tripping over the
'+' normally used for flags, and trying to interpret it as formula).
There's apparently no easy way to check for the existence of a flag,
so add some more ifdeffery to shut it up.
There is some logic to discard packets from streams that are not
selected. Run the metadata update code before this, just to make 100%
sure that no metadata updates can be lost when streams are deselected.
(I'm not sure why this logic would be needed, since both libavformat and
the generic demuxer code do this already. But a quick test shows that
av_read_frame() can return a packet from a stream even if the stream has
AVStream.discard set to AVDISCARD_ALL. This happened after stream
switching. Maybe libavformat doesn't discard already queued packets.)
Instead of printing lines like:
Demuxer info GENRE changed to Alternative Rock
Just output all tags once they change. The assumption is that individual
tags rarely change, while all tags change in the common case.
This changes tag updates to use polling. This could be fixed later,
although the ICY stuff makes it a bit painful, so maybe it will remain
this way.
Also remove DEMUXER_CTRL_UPDATE_INFO. This was intended to check for tag
updates, but now we use a different approach.
Use an arbitrary constant instead, which is as good as PATH_MAX.
This helps us to avoid having to think about pull request #523.
Also fix a case where a potentially signed char was passed to isspace().
This used to work; I'm not sure when or why it regressed. When setting
AVProbeData.filename to NULL, libavformat will crash in rtp_probe() by
unconditionally accessing the string.
We used to set the filename to NULL to prevent probing by file extension
when we don't deem it as necessary. Using an empty string also works for
this purpose.
This generally affects mp3 files that don't have any (or many) mp3
frames in the first 2 MB. 2 MB is the maximum probe size, and
libavformat returns a low probescore even if we give it the full 2 MB.
Trying to probe a larger buffer (or even the full file) doesn't work for
mysterious reasons.
The workaround consists in accepting a very weak probescore if the
format is detected as mp3 and we probed already 2 MB.
Restructure it a bit, so we can use the format hack list even if no mime
type applies. Shouldn't change anything functionally yet. Preparation
for the next commit.
If there's more than one edition, print the list of editions, including
the edition name, whether the edition is selected, whether the edition
is default, and the command line option to select the edition. (Similar
to stream list.)
Move reading the tags to a separate function process_tags(), which is
called when all other state is parsed. Otherwise, that tags will be lost
if chapters are read after the tags.
Pretty worthless. This is called from the seek code, which will
reinitialize these anyway. Even if seeking somehow decides to fail, the
new values are still valid.
One could say a failed seek (if that happens) should jump back to the
original position, and thus it would be better to make sure the state
is restored. But then demux_mkv_seek needs to do this correctly,
including not setting up skipping to the target timestamp. But not
bothering with this.
Extremely obscure corner case with concatenated segments, in which EOF
wasn't recognized correctly, and it tried to demux clusters from the
next segment.
See [MKV]_Editions,_Linked_Segments,_&_Tracksets.mkv from the CCCP test
file collection.
This basically used to be part of the user interface, before mpv moved
printing the track list to the frontend, and this code was raised to
verbose output level.
For some reason, if an error happened when reading headers, it merely
stopped reading the headers, and then continued normally. (It looks like
the case to exit hard (-2) was mainly used for skipping unwanted ordered
chapter segments.)
I can't comprehend this. Always exit on error when reading headers.
(Maybe some more error tolerance would be good, but I have no test case,
and there's some danger of entering endless loops.)
This makes everything more robust, and also somewhat simpler (even if
the diffstat isn't very impressive).
Instead of recursively following SeekHeads while reading headers, just
read the headers until the first cluster, and then possibly use
SeekHeads to read the remaining missing headers.
As of this commit, stream_read_line() can't actually error (except in
the case the passed in buffer is 0, which never happens here). This
commit is preparation for the following commit, which checks harder
whether the read data is actually text. Before this commit, an error
was treated as end-of-file, but the data read so far was considered
valid.
Many ebml_read_* functions have a length int pointer parameter, which
returns the number of bytes skipped. Nothing actually needed this
(anymore), and code using it was rather hard to understand, so get rid
of them.
