The end of the current segment will be the end of the file if there is
no next segment. Normally, this didn't matter much, since UNIX files
allow seeking past the end of the file. But when opening files from
HTTP, this would print confusing error messages. So explicitly check for
EOF before trying to read a segment.
This readds a more or less completely new dvdnav implementation, though
it's based on the code from before commit 41fbcee. Note that this is
rather basic, and might be broken or not quite usable in many cases.
Most importantly, navigation highlights are not correctly implemented.
This would require changes in the FFmpeg dvdsub decoder (to apply a
different internal CLUT), so supporting it is not really possible right
now. And in fact, I don't think I ever want to support it, because it's
a very small gain for a lot of work. Instead, mpv will display fake
highlights, which are an approximate bounding box around the real
highlights.
Some things like mouse input or switching audio/subtitles stream using
the dvdnav VM are not supported.
Might be quite fragile on transitions: if dvdnav initiates a transition,
and doesn't give us enough mpeg data to initialize video playback, the
player will just quit.
This is added only because some users seem to want it. I don't intend to
make mpv a good DVD player, so the very basic minimum will have to do.
How about you just convert your DVD to proper video files?
There are 3 users of the image format option type: demux_raw,
vf_format, vf_noformat. Allow the hwaccel formats (like vdpau etc.)
in general, so that the filters can use it. This won't work for
demux_raw, so explicitly reject these formats there.
So, FFmpeg/Libav requires us to figure out video timestamps ourselves
(see last 10 commits or so), but the methods it provides for this aren't
even sufficient. In particular, everything that uses AVI-style DTS (avi,
vfw-muxed mkv, possibly mpeg4-in-ogm) with a codec that has an internal
frame delay is broken. In this case, libavcodec will shift the packet-
to-image correspondence by the codec delay, meaning that with a delay=1,
the first AVFrame.pkt_dts is not 0, but that of the second packet. All
timestamps will appear shifted. The start time (e.g. the time displayed
when doing "mpv file.avi --pause") will not be exactly 0.
(According to Libav developers, this is how it's supposed to work; just
that the first DTS values are normally negative with formats that use
DTS "properly". Who cares if it doesn't work at all with very common
video formats? There's no indication that they'll fix this soon,
either. An elegant workaround is missing too.)
Add a hack to re-enable the old PTS code for AVI and vfw-muxed MKV.
Since these timestamps are not reorderd, we wouldn't need to sort them,
but it's less code this way (and possibly more robust, should a demuxer
unexpectedly output PTS).
The original intention of all the timestamp changes recently was
actually to get rid of demuxer-specific hacks and the old timestamp
sorting code, but it looks like this didn't work out. Yet another case
where trying to replace native MPlayer functionality with FFmpeg/Libav
led to disadvantages and bugs. (Note that the old PTS sorting code
doesn't and can't handle frame dropping correctly, though.)
Bug reports:
https://trac.ffmpeg.org/ticket/3178https://bugzilla.libav.org/show_bug.cgi?id=600
This was broken by the recent commits. Apparently realvideo timestamps
are severely mangled, and Matroska _of course_ doesn't have the sane,
umangled timestamps, but something unusable. The existing unmangling
code in demux_mkv.c didn't output proper timestamps either. Instead,
it was something weird that triggered sorting. Without sorting (it was
disabled by default recently), you'd get decreasing PTS warnings
In order to fix this, steal some code from libavcodec. Basically copy
the contents of rv34_parser.c (with some changes), which makes
everything magically work. (Maybe it would be better to use the
libavcodec parser API, but I don't want to do that just for this. An
alternative idea would be refusing to read files that have realvideo
tracks, and delegate this to demux_lavf.c, but maybe that's too redical
too.)
I wish I hadn't notice this...
These packets have to be explicitly dropped, because usually libavcodec
uses 0-sized packets to flush delayed frames, meaning just passing
through these packets would have bad consequences.
Normally, libavformat doesn't output 0-sized packets anyway. But I don't
want to take any chances, so don't delete it, and just move it out of
the way to demux.c.
It appears PTS sorting was useful only for avi files (and VfW-muxed
mkv). Maybe it was historically also important for decoders with broken
or non-existent PTS reordering (win32 codecs?). But now that we handle
demuxers which outputs DTS only correctly, it just seems dead weight.
Disable it by default. The --pts-association-mode option is now forced
to always use the decoder's PTS value. You can still enable the old
default (auto) or force sorting. But we will probably remove this option
entirely at some point.
Make demux_mkv export timestamps at DTS when it's in VfW mode. This is
needed to get correct timestamps with the new default mode. demux_lavf
already does that.
This was needed to determine PTS from DTS, but the previous commits
make it unnecessary.
The builtin genpts hack was used for DVD, because libavformat's genpts
essentially went amok on DVD timestamp resets. See commit 65d87091 for
details.
Having the DTS directly can be useful for restoring PTS values.
The avi file format doesn't actually store PTS values, just DTS. An
older hack explicitly exported the DTS as PTS (ignoring the [I assume]
genpts generated non-sense PTS), which is not necessary anymore due to
this change.
This used to be needed to access the generic stream header from the
specific headers, which in turn was needed because the decoders had
access only to the specific headers. This is not the case anymore, so
this can finally be removed again.
Also move the "format" field from the specific headers to sh_stream.
This is similar to the sh_audio commit.
This is mostly cosmetic in nature, except that it also adds automatical
freeing of the decoder driver's state struct (which was in
sh_video->context, now in dec_video->priv).
Also remove all the stheader.h fields that are not needed anymore.
sh_audio is supposed to contain file headers, not whatever was decoded.
Fix this, and write the decoded format to separate fields in the decoder
context, the dec_audio.decoded field. (Note that this field is really
only needed to communicate the audio format from decoder driver to the
generic code, so no other code accesses it.)
Move all state that basically changes during decoding or is needed in
order to manage decoding itself into a new struct (dec_audio).
sh_audio (defined in stheader.h) is supposed to be the audio stream
header. This should reflect the file headers for the stream. Putting the
decoder context there is strange design, to say the least.
This was forgotten when the parser for mplayer2 EDL files was removed.
Change the header of the mpv EDL format to include a '#', so a naive
parser could skip the header as comment. (Maybe this is questionable;
on the other hand, if it can be simpler, why not.)
Also, strip the header in demux_edl.c before passing on the data, so the
header check doesn't need to be duplicated in tl_mpv_edl.c.
Edit Decision Lists (EDL) allow combining parts from multiple source
files into one virtual file. MPlayer had an EDL format (which sucked),
which mplayer2 tried to improve with its own format (which sucked). As
logic demands, mpv introduces its very own format (which sucks).
The new format should actually be much simpler and easier to use, and
its implementation is simpler and smaller too.
demuxer->filepos contains the byte offset of the last read packet. This
is so that the player can estimate the current playback position, if no
proper timestamps are available. Simplify it to use demux_packet->pos in
the generic demuxer code, instead of bothering every demuxer
implementation about it.
(Note that this is still a bit incorrect: it relfects the position of
the last packet read by the demuxer, not that returned to the user. But
that was already broken, and is not that trivial to fix.)
This was originally added for better seeking where libavformat's seek
function won't work well: files with timestamp resets. In these cases,
the code tried to calculate an average bitrate, and then do byte based
seeks by multiplying the seek target time with the bitrate.
Apparently this was unreliable enough that the code was just commented
(and other parts became inactive). Get rid of it.
Note that the player still does byte based seeks in these cases when
doing percent-seeks.
Slightly simplifies memory management. This might make adding a demuxer
cache wrapper easier at a later point, because you can just copy the
complete stream header, without worrying that the wrapper will free the
individual stream header fields.
Most libavcodec decoders output non-interleaved audio. Add direct
support for this, and remove the hack that repacked non-interleaved
audio back to packed audio.
Remove the minlen argument from the decoder callback. Instead of
forcing every decoder to have its own decode loop to fill the buffer
until minlen is reached, leave this to the caller. So if a decoder
doesn't return enough data, it's simply called again. (In future, I
even want to change it so that decoders don't read packets directly,
but instead the caller has to pass packets to the decoders. This fits
well with this change, because now the decoder callback typically
decodes at most one packet.)
ad_mpg123.c receives some heavy refactoring. The main problem is that
it wanted to handle format changes when there was no data in the decode
output buffer yet. This sounds reasonable, but actually it would write
data into a buffer prepared for old data, since the caller doesn't know
about the format change yet. (I.e. the best place for a format change
would be _after_ writing the last sample to the output buffer.) It's
possible that this code was not perfectly sane before this commit,
and perhaps lost one frame of data after a format change, but I didn't
confirm this. Trying to fix this, I ended up rewriting the decoding
and also the probing.
This affects 64 bit floats and big endian integer PCM variants
(basically crap nobody uses). Possibly not all MS-muxed files work, but
I couldn't get or produce any samples.
Remove a bunch of format tags that are not needed anymore. Most of these
were used by demux_mov, which is long gone. Repurpose/abuse 'twos' as
mpv-internal tag for dealing with the PCM variants mentioned above.
This member was redundant. sh_audio->sample_format indicates the sample
size already.
The TV code is a bit strange: the redundant sample size was part of the
internal TV interface. Assume it's really redundant and not something
else. The PCM decoder ignores the sample size anyway.
Somehow the new parser ends up much smaller. Much of it is because we
don't parse some additional information. We just skip it, instead of
parsing it and then throwing it away.
More importantly, we use the physical order of entries, instead of
trying to sort them by entry number. Each "File" entry is followed by a
number that is supposed to be the entry number, and "File1" is first.
(Should it turn out that this is really needed, an additional field
should be added to playlist_entry, and then qsort().)
Make TOOLS/matroska.pl output structs with fields sorted by name in
ebml_types.h to make the order of fields deterministic. Fix warnings in
demux_mkv.c caused by the first struct fields switching between scalar
and struct types due to non-deterministic ebml_types.h field order.
Since it's deterministic now, this shouldn't change anymore.
The warnings produced by the compilers are bogus, but we want to silence
them anyway, since this could make developers overlook legitimate
warnings.
What commits 7b52ba8, 6dd97cc, 4aae1ff were supposed to fix. An earlier
attempt sorted fields in the generated C source file, not the header
file. Hopefully this is the last commit concerning this issue...
The configure followed 5 different convetions of defines because the next guy
always wanted to introduce a new better way to uniform it[1]. For an
hypothetic feature 'hurr' you could have had:
* #define HAVE_HURR 1 / #undef HAVE_DURR
* #define HAVE_HURR / #undef HAVE_DURR
* #define CONFIG_HURR 1 / #undef CONFIG_DURR
* #define HAVE_HURR 1 / #define HAVE_DURR 0
* #define CONFIG_HURR 1 / #define CONFIG_DURR 0
All is now uniform and uses:
* #define HAVE_HURR 1
* #define HAVE_DURR 0
We like definining to 0 as opposed to `undef` bcause it can help spot typos
and is very helpful when doing big reorganizations in the code.
[1]: http://xkcd.com/927/ related
Instead of having each demuxer do it (only demux_mkv actually did...),
let generic code determine whether the file is seekable. This requires
adding exceptions to demuxers where the stream is not seekable, but the
demuxer is.
Sort-of try to improve handling of unseekable files in the player. Exit
early if the file is determined to be unseekable, instead of resetting
all decoders and then performing a pointless seek.
Add an exception to allow seeking if the file is not seekable, but the
stream cache is enabled. Print a warning in this case, because seeking
outside the cache (which we can't prevent since the demuxer is not aware
of this problem) still messes everything up.
Pointless, using stream->start_pos/end_pos instead.
demux_mf was the only place where this was used specially, but we can
rely on timestamps instead for this case.
There are some Microsoft Windows symbols which are traditionally used by
the mplayer core, because it used to be convenient (avi was the big
format, using binary windows decoders made sense...). So these symbols
have the exact same definition as the Windows one, and if mplayer is
compiled on Windows, the symbols from windows.h are used.
This broke recently just because some files were shuffled around, and
the symbols defined in ms_hdr.h collided with windows.h ones. Since we
don't have windows binary decoders anymore, there's not the slightest
reason our symbols should have the same names. Rename them to reduce the
risk for collision, and to fix the recent regression.
Drop WAVEFORMATEXTENSIBLE, because it's mostly unused. ao_dsound defines
its own version if the windows headers don't define it, and ao_wasapi is
not available on systems where this symbol is missing.
Also reindent ms_hdr.h.
Now that matroska.pl generates struct fields in deterministic order,
this should be the last time I change this.
(gcc and clang shouldn't warn about this line of code, but since they
do, we want to workaround and silence the warning anyway.)
Unfortunately, we can't avoid this warning 100%, because ebml_info is
written by a Perl script. I think the script writes the struct fields in
random order (thanks Perl), so there's no way to know whether the first
struct field is a scalar or a struct.
At least {0} is always valid here, even if it shows a warning. (The
compilers are wrong, see e.g. [1].)
[1] http://gcc.gnu.org/bugzilla/show_bug.cgi?id=53119
This one really did bite me hard (see previous commit), so enable it by
default.
Fix some cases of shadowing throughout the codebase. None of these
change behavior, and all of these were correct code, and just tripped up
the warning.
gcc and clang happen to allow {} to default-initialize a struct, but
strictly speaking, C99 requires at least {0}. In one case, we use {{0}},
but that's only because gcc as well as clang are too damn stupid not
to warn about {0}, which is a perfectly valid construct in this case.
(Sure is funny, don't warn about the non-standard case, but warn about
another standard conform case.)
Leaving these braces away just because the syntax allows them is really
obnoxious. It removes the visual cues which help understanding the code
at the first look.
For the record,
if (cond)
something();
is ok, as long as there's no else branch, and the if body is one
physical line. But everything else should have braces.
This was probably not a real problem. But it's not entirely clear
whether this could actually happen or not, so it's better to be
defensive. The code is now also somewhat easier to understand.
This adds support for ChapterSegmentEditionUID (pull request #258),
and also fixes issue #278 (pull request #292).
In fact, this is a straight merge of pr/292, which also contains pr/258.
Note that you still need --vd-lavc-o='strict=-2' to enable the decoder.
Also, there's no guarantee that all required features for HEVC demuxing
are actually implemented, nor that the current muxing schema is the
final one.
Change talloc destructor so that they can never signal failure, and
don't return a status code. This makes our talloc copy even more
incompatible to upstream talloc, but on the other hand this is
preparation for getting rid of talloc entirely.
(The talloc replacement in the next commit won't allow the talloc_free
equivalent to fail, and the destructor return value would be useless.
But I don't want to change any mpv code either; the idea is that the
talloc replacement commit can be reverted for some time in order to
test whether the talloc replacement introduced a regression.)
To support edition references in matroska chapters, editions need to be
remembered for each chapter and source. To facilitate easier management
of these now-paired uids, a single structure is used.
There is uninitialized memory access if the actual size isn't passed
along. In the worst case, this can cause a source to be loaded against
the uninitialized memory, causing a false count of found versus required
sources, preventing the "Failed to find ordered chapter part" message.
By default, libavformat uses UDP for rtsp playback. This doesn't work
very well. Apparently the reason is that the buffer sizes libavformat
chooses for UDP are way too small, and switching to TCP gets rid of this
issue entirely (thanks go to Reimar Döffinger for figuring this out).
In theory, you can set buffer sizes as libavformat options, but that
doesn't seem to help.
Add an option to select the rtsp transport, and make TCP the default.
Also remove an outdated comment from stream.c.
In insane files with a very huge number of subtitle events, and if the
--demuxer-mkv-subtitle-preroll option is given, seeking can still
overflow the packet queue. Normally, the subtitle_preroll variable
specifies the maximum number of packets that can be added. But once this
number is reached, the normal seeking behavior is enabled, which will
add all subtitle packets with the right timestamps to the packet queue.
At this point the next video keyframe can still be quite far away, with
enough subtitle packets on the way to overflow the packet queue.
Fix this by always setting an upper limit of subtitle packets read
during seeking. This should provide additional robustness even if the
preroll option is not used.
This means that even with normal seeking, at most 500 subtitle packets
are demuxed. Packets after that are discarded.
One slightly questionable aspect of this commit is that subtitle_preroll
is never reset in audio-only mode, but that is probably ok.
The quicktime html scripting guide suggests to wrap urls not
necesarly associated with quicktime in a .mov file.
(so that when <embed>ing videos quicktime would be forced.)
These mov files may contain several "Text Hacks".
One of these is RTSPtext.
The suggested/allowed format (as regex) is like:
RTSPtext[ \r]RTSP://url
See also p.51 of:
https://developer.apple.com/library/mac/documentation/QuickTime/Conceptual/QTScripting_HTML/QTScripting_HTML.pdf
In reality there are also files like (e.g. zdfmediathek.de):
RTSPtext\nrtsp://url\n\n
Lets handle these files as a playlist with one element.
The --deinterlace option does on playback start what the "deinterlace"
property normally does at runtime. You could do this before by using the
--vf option or by messing with the vo_vdpau default options, but this
new option is supposed to be a "foolproof" way.
The main motivation for adding this is so that the deinterlace property
can be restored when using the video resume functionality
(quit_watch_later command).
Implementation-wise, this is a bit messy. The video chain is rebuilt in
mpcodecs_reconfig_vo(), where we don't have access to MPContext, so the
usual mechanism for enabling deinterlacing can't be used. Further,
mpcodecs_reconfig_vo() is called by the video decoder, which doesn't
have access to MPContext either. Moving this call to mplayer.c isn't
currently possible either (see below). So we just do this before frames
are filtered, which potentially means setting the deinterlacing every
frame. Fortunately, setting deinterlacing is stable and idempotent, so
this is hopefully not a problem. We also add a counter that is
incremented on each reconfig to reduce the amount of additional work per
frame to nearly zero.
The reason we can't move mpcodecs_reconfig_vo() to mplayer.c is because
of hardware decoding: we need to check whether the video chain works
before we decide that we can use hardware decoding. Changing it so that
this can be decided in advance without building a filter chain sounds
like a good idea and should be done, but we aren't there yet.
Retrieve per-chapter metadata, but don't do much with it. We just make
the metadata of the _current_ chapter available as chapter-metadata
property. Returning the full chapter list with metadata would be no
problem, except that the property interface isn't really good with
structured data, so it's not available for now.
Not sure if it's worth it, but it was requested via github issue #201.
Consider the cluster used for prerolling contains an insane amount of
subtitle packets. Then the demuxer packet queue would be full of
subtitle packets, and demux.c would refuse to read any further packets -
including video and audio packets, resulting in EOF. Since everything
involving Matroska and subtitles is 100% insane, this can actually
happen.
Fix this by putting a limit on the number of subtitle packets read by
preroll, and throw away any further packets if the limit is exceeded. If
this happens, the preroll mechanism will stop working, but the player's
operation is unaffected otherwise.
The really funny thing about this commit is that this code is added on
top of another work around. Basically, subtitle seeking in libavformat
is completely broken. To make it useful, we have to add yet another
workaround.
The basic problem is that libavformat's subtitle seeking code always
uses the stream time base, instead of AV_TIME_BASE if stream index -1 is
passed to the avformat_seek_file() function.
Fixes github issue #216. Hopefully this will be fixed in ffmpeg too at
some point.
Port it from playlist_parser.c to demux_playlist.c. Also, change the m3u
parser to drop whitespace from the trailing part of the line (will make
it work properly with windows line endings).
(I hoped that this would make MMS URIs with http instead of mmsh
prefixes work, but it doesn't. Instead, it leads to a playlist loop. So
solving this issue would require a change in ffmpeg, probably.)
Apparently, it is popular to store large files in uncompressed rar
archives. Extracting files is not practical, and some media players
suport playing directly from uncompressed rar (at least VLC and some
DirectShow components).
Storing or accessing files this way is completely idiotic, but it is
a common practice, and the ones subjected to this practice can't do
much to change this (at least that's what I assume/hope). Also, it's
a feature request, so we say yes.
This code is mostly taken from VLC (commit f6e7240 from their git tree).
We also copy the way this is done: opening a rar file by itself yields
a playlist, which contains URLs to the actual entries in the rar file.
Compressed entries are simply skipped.
Modeled after the old playlist_parser.c, but actually new code, and it
works a bit differently.
Demuxers (and sometimes streams) are the component that should be used
to open files and to determine the file format. This was already done
for subtitles, but playlists still use a separate code path.
The way this was added to FFmpeg is less than ideal, because it requires
text parsing in the Matroska demuxer. But in order to use the FFmpeg
webvtt-to-ass converter, we still have to mimic this in some way. We do
this by putting the parsing into sd_lavc_conv.c, before the subtitle
packet is passed to libavcodec. At least this keeps the ugliness out of
unrelated code.
There is some change that FFmpeg will fix their design eventually.
Instead of rewriting the parsing code, we simply borrow it from FFmpeg's
Matroska demuxer.
Otherwise, this would just try to demux a good chunk of the file, even
though the operation can't succeed anyway.
This caused some pretty strange issues, where perfectly valid use cases
would print a "Too many packets in the demuxer packet queue..." message.
The rawaudio demuxer read one frame per packet, basically a few bytes,
which caused insane overhead. (I found this when I couldn't play raw
audio without dropouts when using -v, which printed a line per packet
read.)
Fix this and read 1 second of audio per packet. This is a regression
since cfa5712 (merging of demux_rawaudio and demux_rawvideo).
Originally, the objective of this commit was changing --edition to be
1-based, but this was cancelled. I'm still leaving the change to
demux_mkv.c though, which is now only of cosmetic nature.
This is completely useless, and in this particular case, it broke the
fallback for MLP2 subtitles (stored as .txt files) to demux_subreader.
(Yes, libavformat should be fixed to handle this, but for now this will
_always_ break playback of subtitle files stored in .txt.)
You can still force this demuxer, but by default we will just pretend
that the "tty" demuxer does not exist.
Perhaps not very useful, but reserved for situations when a user reports
awful latency and experimentation/debugging might be required to find
out why or to fix it (happens often).
avio_alloc_context() is documented to require an av_malloc'ed buffer. It
appears libavformat can even reallocate the buffer while it is probing,
so passing a static buffer can in theory lead to crashes.
I couldn't reproduce such a crash, but apparently it happened to
mplayer-svn. This commit follows the mplayer fix in svn commit r36397.
Move the decoder parts from vo_vdpau.c to a new file vdpau_old.c. This
file is named so because because it's written against the "old"
libavcodec vdpau pseudo-decoder (e.g. "h264_vdpau").
Add support for the "new" libavcodec vdpau support. This was recently
added and replaces the "old" vdpau parts. (In fact, Libav is about to
deprecate and remove the "old" API without deprecation grace period,
so we have to support it now. Moreover, there will probably be no Libav
release which supports both, so the transition is even less smooth than
we could hope, and we have to support both the old and new API.)
Whether the old or new API is used is checked by a configure test: if
the new API is found, it is used, otherwise the old API is assumed.
Some details might be handled differently. Especially display preemption
is a bit problematic with the "new" libavcodec vdpau support: it wants
to keep a pointer to a specific vdpau API function (which can be driver
specific, because preemption might switch drivers). Also, surface IDs
are now directly stored in AVFrames (and mp_images), so they can't be
forced to VDP_INVALID_HANDLE on preemption. (This changes even with
older libavcodec versions, because mp_image always uses the newer
representation to make vo_vdpau.c simpler.)
Decoder initialization in the new code tries to deal with codec
profiles, while the old code always uses the highest profile per codec.
Surface allocation changes. Since the decoder won't call config() in
vo_vdpau.c on video size change anymore, we allow allocating surfaces
of arbitrary size instead of locking it to what the VO was configured.
The non-hwdec code also has slightly different allocation behavior now.
Enabling the old vdpau special decoders via e.g. --vd=lavc:h264_vdpau
doesn't work anymore (a warning suggesting the --hwdec option is
printed instead).
Remove the (now unused) code for determining correct-pts mode based on
the demuxer in use. Change its description in the manpage to reflect
what this option does now.
Gives really funky results with PNG attachments otherwise. The main
problem is that avcodec_flush_buffers() does not fully reset the
decoder, so passing multiple PNG packets without keyframe flags will
attempt to combine the new picture with the previously decoded
contents. (Makes no sense with proper PNG - maybe this codepath is
intended for MNG or APNG.)
In general, this warning can hint to actual bugs. We don't enable it
yet, because it would conflict with some unmerged code, and we should
check with clang too (this commit was done by testing with gcc).
This also affects --audiofile. The previous behavior wasn't really
useful. There are even separate switches for that: --audio-demuxer and
--sub-demuxer.
This fixes the sample RA_missing_timestamps.mkv. Pretty funny how this
code got it almost right, but not quite, so it was broken all these
years. And then, after everyone stopped caring, someone comes and fixes
it. (By the way, I know absolutely nothing about realaudio.)
This fixes playback of the sample linked by FFmpeg ticket 2508. The fix
follows ffmpeg commit 6158a3b (although it's not exactly the same).
The problem here is that the file contains an apparently non-sense
DefaultDuration value. DefaultDuration for audio tracks is used to
derive PTS values for packets with no timestamps, like they can happen
with frames inside a laced block. So the first packet of a SimpleBlock
will have a correct PTS, while the PTS values of the following packets
are calculated using DefaultDuration, and thus are broken.
This leads to seemingly ok playback, but broken A/V sync. Not using the
DefaultDuration value will leave the PTS values of these packets unset,
and the audio decoder can derive them from the output instead.
The fix more or less uses a heuristic to detect the broken case: if the
sample rate is 8 KHz (Matroska default, can assume unset), and the codec
is AC3 (as the broken file did), don't use it. I'm not sure why this
should be done only for AC3, maybe the muxing application (mkvmerge
v4.9.1) has known issues with AC3. AC3 also doesn't support 8 KHz as
sample rate natively.
(By the way, I'm not sure why we should honor the DefaultDuration at all
for audio. It doesn't seem to be needed. You can't seek to these frames,
and decoders should always be able to produce perfect PTS values by
adding the duration of the decoded audio to the first PTS.)
Matroska has an output sample rate (OutputSamplingFrequency), which in
theory should be forced instead of whatever the decoder outputs. But it
appears no software (other than mplayer2 and mpv until now) actually
respects this. Even worse, there were broken files around, which played
correctly with (in theory) broken software, but not mplayer2/mpv. Hacks
were added to our code to play these files correctly, but they didn't
catch all cases.
Simplify this by doing what everyone else does, and always use the
decoder's sample rate instead. In particular, we try to handle all
sample rate issues like libavformat's Matroska demuxer does.
Guess the colorspace directly in mpcodecs_reconfig_vo(), instead of in
set_video_colorspace(). The difference is that the latter function just
makes the video filter chain (and VOs) force the detected colorspace,
and then throws it away, while the former is a bit more general and
central. Not really a big difference and it doesn't matter much in
practice, but it guarantees that there is no internal disagreement about
the colorspace.
DVD playback had some trouble with PTS resets: libavformat's genpts
feature would try reading until EOF (worst case) to find a new usable
PTS in case a packet's PTS is not set correctly. Especially with slow
DVD access, this would make the player to appear frozen.
Reimplement it partially in demux_lavf.c, and use that code in the DVD
case. This is heavily "inspired" by the code in av_read_frame from
libavformat/utils.c. The difference is that we stop reading if no PTS
has been found after 50 packets (consider this a heuristic). Also, we
don't bother with the PTS wrapping and last-frame-before-EOF handling.
Even with normal PTS wraps, the player frontend will go to hell for the
duration of a frame anyway, and should recover quickly after that.
The terribleness of this commit is mostly that we duplicate libavformat
functionality, and that we suddenly need a packet queue.
