Since the libavformat API is crap, we have to apply tons of heuristics
to check whether seeking will work. (No, checking it at seek time isn't
going to work either, because if a seek fails, the demuxer will be in an
undefined state. Because the libavformat API is crap.)
I've got a broken webm that fails to seek correctly with "--start=0".
The problem is that every index entry points to 1 byte before cluster
start (!!!). demux_mkv tries to resync to the next cluster, but since it
already has read 2 bytes with ebml_read_id(), it doesn't get the first
cluster, but the following one. Actually, it can be any amount of bytes
from 1-4, whatever happens to look valid at this essentially random byte
position.
Improve this by resyncing from the original position, instead of the one
after the EBML element ID has been attempted to be read.
The file shows the following headers:
| + Muxing application: google at 177
| + Writing application: google at 186
Indeed, the file was downloaded with youtube-dl. I can only guess that
Google got it completely wrong.
demux_playlist.c recognizes if the source stream points to a directory,
and adds its directory entries. Until now, only 1 level was added.
During playback, further directory entries could be resolved as
directory paths were "played".
While this worked fine, it lead to frequent user confusion, because
playlist resuming and other things didn't work as expected. So just
recursively scan everything.
I'm unsure whether it's a good fix, but at least it gets rid of the
complaints. (And probably will add others.)
Now it will always be able to seek back to the start, even if the index
is sparse or misses the first entry.
This can be achieved by reusing the logic for incremental index
generation (for files with no index), and start time probing (for making
sure the first block is always indexed).
AVFormatContext.codec is deprecated now, and you're supposed to use
AVFormatContext.codecpar instead.
Handle this for all of the normal playback code.
Encoding mode isn't touched.
This was changed in 2014, so I suppose users will usually have a FFmpeg
release which includes the corresponding upstream change. If not, well
too bad for those MicroDVD-obsessed users.
Also don't try to retrieve the default framerate as exported by the
demuxer, and instead hardcode it and trust it won't ever change. this
avoids that we have to deal with a larger mess in the codecpar commit.
I don't trust it one bit, and it's a bother with the codecpar change.
If it turns out to be important for some file formats, it could be
added back (or FFmpeg fixed).
This reverts commit 503c6f7fd6.
There are situations where some decoders (MF apparently) always require
a timestamp. Also, this makes bitrate estimation more granular than
necessary. It seems it's better to try to detect fiels with broken
default durations explicitly instead. Or maybe something should be
added to smooth audio timestamps after filters.
Instead of having a separate for each, which also requires separate
additional caching in the demuxer. (The demuxer adds an indirection,
since STREAM_CTRLs are not thread-safe.)
Since this includes the cache speed, this should fix#3003.
SEEK_HR is interpreted by demux_mkv.c, and enables subtitle preroll by
prefetching additional subtitle pakcets which might overlap with the
seek destination. This should make the case work when segment boundaries
fall into the middle of subtitle events.
This still usually leaves a flicker of at least 1 frame on start,
because dec_sub.c does not ensure that enough subtitles are read before
rendering after a segment switch. (Probably a WONTFIX.)
This is simpler, because it doesn't have to wait from both threads for
synchronization.
Apart from being simpler/cleaner, this serves vague plans to stop/start
the demuxer thread itself automatically on demand (for the purpose of
reducing unneeded resource usage).
This pause stuff is bothersome and is needed only for a few corner-
cases. This commit removes it from the demuxer public API and replaces
it with a demux_run_on_thread() function and refactors the code which
needed demux_pause(). The next commit will change the implementation.
Commit 503c6f7f essentially removed timestamps from "laces" (Block sub-
divisions), which means many audio packets will have no timestamp.
There's no reason why bitrate calculation can't just delayed to a point
when the next timestamp is known.
Fixes#2903 (no audio bitrate with mkv files).
stream->info can be NULL if it's the cache wrapper. To be fair,
stream->info is considered private API anyway. So don't access it, but
check the URL instead.
This reverts commit af66fa8fa5.
The reverted commit caused AVCodecContext.channel_layout to be set,
while requesting stereo downmix will make libavcodec output a stupid
message:
ac3: Channel layout '5.1' with 6 channels does not match specified number of channels 2: ignoring specified channel layout
The same happens with --demuxer=lavf (without this change too).
I'm not quite sure what acrobatics are required to shut up libavcodec,
but for now revert the commit. It was a rather minor, almost cosmetic
issue, which I consider less important than clean CLI terminal output.
Ever since a change in mplayer2 or so, relative seeks were translated to
absolute seeks before sending them to the demuxer in most cases. The
only exception in current mpv is DVD seeking.
Remove the SEEK_ABSOLUTE flag; it's not the implied default. SEEK_FACTOR
is kept, because it's sometimes slightly useful for seeking in things
like transport streams. (And maybe mkv files without duration set?)
DVD seeking is terrible because DVD and libdvdnav are terrible, but
mostly because libdvdnav is terrible. libdvdnav does not expose seeking
with seek tables. (Although I know xbmc/kodi use an undocumented API
that is not declared in the headers by dladdr()ing it - I think the
function is dvdnav_jump_to_sector_by_time().) With the current mpv
policy if not giving a shit about DVD, just revert our half-working seek
hacks and always use dvdnav_time_search(). Relative seeking might get
stuck sometimes; in this case --hr-seek=always is recommended.
If a stream is marked as EOF (due to no packets found in reach), then we
need to wakeup the decoder. This is important especially if no packets
are found at the start of the file, so the A/V sync logic actually
starts playback, instead of waiting for packets that will never come.
(It would randomly start playback when running the playback loop due to
arbitrary external events like user input.)
Commit 943f76e6, which already tried this, was very stupid: it didn't
actually override the samplerate for Opus, but overrode it for all
codecs other than Opus. And even then, it failed to use the overridden
samplerate. (Sigh...)
Fixes relative seeks. Without this, a seek back could skip so much data
that the seek would effectively jump forward. (Or insert silence for
files with video.)
There's the question whether the frontend should do this instead (by
using information from the decoders), but for now this seems more
proper.
demux_mkv.c does this already, sort of.
libavformat doesn't for seeks in .ogg (aka .opus), but might be doing it
for mkv. Seems to be a mess as well.
I think the conclusion is that AV_PKT_DATA_SKIP_SAMPLES is misdesigned
(at least for some formats), and an alternative mechanism using
durations would be better. (Combining it with a proper timebase would
keep sample-accuracy.)
This is achieved indirectly by deslecting all streams for the non-
current segment (and if the segment doesn't share the demuxer with the
currently active one).
Restores functionality added with commit 46bcdb70.
This uses a different method to piece segments together. The old
approach basically changes to a new file (with a new start offset) any
time a segment ends. This meant waiting for audio/video end on segment
end, and then changing to the new segment all at once. It had a very
weird impact on the playback core, and some things (like truly gapless
segment transitions, or frame backstepping) just didn't work.
The new approach adds the demux_timeline pseudo-demuxer, which presents
an uniform packet stream from the many segments. This is pretty similar
to how ordered chapters are implemented everywhere else. It also reminds
of the FFmpeg concat pseudo-demuxer.
The "pure" version of this approach doesn't work though. Segments can
actually have different codec configurations (different extradata), and
subtitles are most likely broken too. (Subtitles have multiple corner
cases which break the pure stream-concatenation approach completely.)
To counter this, we do two things:
- Reinit the decoder with each segment. We go as far as allowing
concatenating files with completely different codecs for the sake
of EDL (which also uses the timeline infrastructure). A "lighter"
approach would try to make use of decoder mechanism to update e.g.
the extradata, but that seems fragile.
- Clip decoded data to segment boundaries. This is equivalent to
normal playback core mechanisms like hr-seek, but now the playback
core doesn't need to care about these things.
These two mechanisms are equivalent to what happened in the old
implementation, except they don't happen in the playback core anymore.
In other words, the playback core is completely relieved from timeline
implementation details. (Which honestly is exactly what I'm trying to
do here. I don't think ordered chapter behavior deserves improvement,
even if it's bad - but I want to get it out from the playback core.)
There is code duplication between audio and video decoder common code.
This is awful and could be shareable - but this will happen later.
Note that the audio path has some code to clip audio frames for the
purpose of codec preroll/gapless handling, but it's not shared as
sharing it would cause more pain than it would help.
FFmpeg can generate such files. It's unclear whether they're allowed by
Matroska. mkvinfo shows packet timestamps in both forms (one of them
must be a bug), and at last libavformat's demuxer treats timestamps
as signed.
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.
There are probably more files to which this applies, but I'm being
conservative here.
A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).
common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.
codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.
From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).
misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.
screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
Slightly helps with timeline stuff, like EDL. There is no need to keep
network (or even just disk I/O) busy for all segments at the same time,
because 1. the data won't be needed any time soon, and 2. will probably
be discarded anyway if the stream is seeked when segment is resumed.
Partially fixes#2692.
