| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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Before this commit, the af_instance->mul/delay values were in bytes.
Using bytes is confusing for non-interleaved audio, so switch mul to
samples, and delay to seconds. For delay, seconds are more intuitive
than bytes or samples, because it's used for the latency calculation.
We also might want to replace the delay mechanism with real PTS
tracking inside the filter chain some time in the future, and PTS
will also require time-adjustments to be done in seconds.
For most filters, we just remove the redundant mul=1 initialization.
(Setting this used to be required, but not anymore.)
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Replace the code that used a single buffer with mp_audio_buffer. This
also enables non-interleaved output operation, although it's still
disabled, and no AO supports it yet.
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Set the PulseAudio stream title, just like the VO window title is set.
Refactor update_vo_window_title() so that we can use it for AOs too.
The ao_pulse.c bit is stolen from MPlayer.
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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.
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Note that the change in seek_reset is not entirely equivalent: we even
drop the remainder of buffered audio when seeking. This should be more
correct, because the whole point of the reset_ao parameter is to control
whether audio queued for output should be dropped or not.
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ao_lavc.c accesses ao->buffer, which I consider internal. The access was
done in ao_lavc.c/uninit(), which tried to get the left-over audio in
order to write the last (possibly partial) audio frame. The play()
function didn't accept partial frames, because the AOPLAY_FINAL_CHUNK
flag was not correctly set, and handling it otherwise would require an
internal FIFO.
Fix this by making sure that with gapless audio (used with encoding),
the AOPLAY_FINAL_CHUNK is set only once, instead when each file ends.
Basically, move the hack in ao_lavc's uninit to uninit_player.
One thing can not be entirely correctly handled: if gapless audio is
active, we don't know really whether the AO is closed because the file
ended playing (i.e. we want to send the buffered remainder of the audio
to the AO), or whether the user is quitting the player. (The stop_play
flag is overwritten, fixing that is perhaps not worth it.) Handle this
by adding additional code to drain the AO and the buffers when playback
is quit (see play_current_file() change).
Test case: mpv avdevice://lavfi:sine=441 avdevice://lavfi:sine=441 -length 0.2267 -gapless-audio
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There's no real use-case for this, and is wasn't documented (didn't even
appear on the "undocumented commands" list).
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Conflicts:
configure
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mpvcore/player/playloop.c: In function 'seek':
mpvcore/player/playloop.c:209:54: warning: declaration of 'seek' shadows a global declaration [-Wshadow]
mpvcore/player/playloop.c:209:12: warning: shadowed declaration is here [-Wshadow]
mpvcore/player/playloop.c: In function 'queue_seek':
mpvcore/player/playloop.c:360:25: warning: declaration of 'seek' shadows a global declaration [-Wshadow]
mpvcore/player/playloop.c:209:12: warning: shadowed declaration is here [-Wshadow]
Signed-off-by: wm4 <wm4@nowhere>
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The code made no sense at all.
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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.
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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.
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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
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mpv crashed when linked files were not found. The reason was that the
chapters array contained some uninitialized data.
I have no idea how this code works (after the merge). The old code
actually seems to remove missing chapters, while the new code just
leaves them unintiialized. Work around the crash by initializing the
chapters array (and a bunch of other things) with 0, which means the
missing chapter will be located at 00:00:00 and have no name.
There is a regression since commit af0306d.
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The compatibility layer still takes care of the old speed_mult command.
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Essentially works like "add".
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Better prefer real English...
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Just doing this because mp_osd.h and osd.c is not consistent.
There are some other header files (command.h and screenshot.h), but
since I don't feel too good about inflating mp_core.h, I'm not merging
them, at least not yet.
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I considered making a header file for each .c file, but decided against
it. Asking around, not making separate headers was deemed acceptable. In
the end, all of these depend on MPContext and store state inside of it,
so separate headers aren't all that useful anyway.
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mplayer.c was a bit too big. Split it into multiple files. I hope the
way it's split makes sense. Maybe some things don't make too much sense,
or go against intuition. These will fixed as soon as I notice them.
Some files are a bit questionable (misc.c, osd.c, configfiles.c), and
suggestions how to organize this better are welcome.
Regressions are possible due to reorganized include statements.
Obviously I didn't just copy mplayer.c's orgy of include statements, but
recreated them for each file. It's easily possible that there are
oversights and mistakes, which will show up on other platforms.
There is one actual change: the public avutil.h include is removed from
encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I
consider using libavutil too much as dangerous, because the set of
include files they recursively pull in is rather arbitrary and is
different between FFmpeg and Libav.
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All these files access mp_core.h and MPContext, and form the actual
player application. They should be all in one place, and separate
from the other sources that are mere utility helpers.
Preparation for splitting mplayer.c into multiple smaller parts.
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