| Commit message (Collapse) | Author | Age | Files | Lines |
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Allocate af_instance->data in generic code before filter initialization.
Every filter needs af->data (since it contains the output
configuration), so there's no reason why every filter should allocate
and free it.
Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min().
Interestingly, most code becomes simpler, because the new function takes
the size in samples, and not in bytes. There are larger change in
af_scaletempo.c and af_lavcac3enc.c, because these had copied and
modified versions of the RESIZE_LOCAL_BUFFER macro/function.
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No AO can handle these, so it would be a problem if they get added
later, and non-interleaved formats get accepted erroneously. Let them
gracefully fall back to other formats.
Most AOs actually would fall back, but to an unrelated formats. This is
covered by this commit too, and if possible they should pick the
interleaved variant if a non-interleaved format is requested.
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Based on earlier work by Stefano Pigozzi.
There are 2 changes:
1. Instead of mp_audio.audio, mp_audio.planes[0] must be used.
2. mp_audio.len used to contain the size of the audio in bytes. Now
mp_audio.samples must be used. (Where 1 sample is the smallest unit
of audio that covers all channels.)
Also, some filters need changes to reject non-interleaved formats
properly.
Nothing uses the non-interleaved features yet, but this is needed so
that things don't just break when doing so.
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Also add a corresponding function to audio/format.c, which fills an
audio block with silence.
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My main problem with this is that the output format will be incorrect.
(This doesn't matter right, because there are no samples output.)
This assumes all audio filters can deal with len==0 passed in for
filtering (though I wouldn't see why not).
A filter can still signal an error by returning NULL.
af_lavrresample has to be fixed, since resampling 0 samples makes
libavresample fail and return a negative error code. (Even though it's
not documented to return an error code!)
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When blending OSD and subtitles onto the video, we write bogus alpha
values. This doesn't normally matter, because these values are normally
unused and discarded. But at least on Wayland, the alpha values are used
by the compositor and leads to transparent windows even with opaque
video on places where the OSD happens to use transparency.
(Also see github issue #338.)
Until now, the alpha basically contained garbage. The source factor
GL_SRC_ALPHA meant that alpha was multiplied with itself. Use GL_ONE
instead (which is why we have to use glBlendFuncSeparate()). This should
give correct results, even with video that has alpha. (Or at least it's
something close to correct, I haven't thought too hard how the
compositor will blend it, and in fact I couldn't manage to test it.)
If glBlendFuncSeparate() is not available, fall back to glBlendFunc(),
which does the same as the code did before this commit. Technically, we
support GL 1.1, but glBlendFuncSeparate is 1.4, and I guess we should
try not to crash if vo_opengl_old runs on a system with GL 1.1 drivers
only.
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Apparently we were using FFmpeg-specific APIs. I have no idea whether
this code is correct on both FFmpeg and Libav (no examples, bad
doxygen... why do they even complaint aht people are using their APIs
incorrectly?), but it appears to work on FFmpeg. That was also the case
before commit ebc4ccb though, where it used internal libavformat
symbols.
Untested on Libav, Travis will tell us.
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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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VDPAU handles are integers, but for some reasons the VDPAU GL extension
expects them as void*.
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In theory, af_volume could use separate volume levels for each channel.
But this was never used anywhere.
MPlayer implemented something similar before (svn r36498), but kept the
old path for some reason.
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I accidentally copied the AES4/ORIGFS constants from the ALSA headers,
instead of the AES3/FS ones. The difference is probably important.
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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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Also do some cosmetic changes, like merging definition and
initialization of local variables.
Remove an annoying debug mp_msg() from af_open(). It just printed the
command line parameters; if this is really needed, it could be added
to af.c instead (similar as to what vf.c does).
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If filter initialization fails anyway, we don't need to reconfigure
libavresample.
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Helps with readability. Also remove the ctx_opt_set_* helper macros and
use av_opt_set_* directly (I think these macros were used because the
lines ended up too long, but this commit removes two indentation levels,
giving more space).
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The conversion works for native endian only. The correct check lists
supported format combination explicitly, but is also much simpler.
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This did strange things; perhaps caused by the channel layout changes.
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This should allow to make format negotiation much simpler, since it
takes the responsibility to compare actual input and accepted input
formats from the filters. It's also backwards compatible. Filters which
have expensive initialization still can use the old method.
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Also remove the rather bad/incomplete log calls from ao_alsa and ao_oss.
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I have no idea what these do, but apparently they are needed to inform
ALSA about spdif configuration. First, replace the literal constant "6"
for the AES0 parameter with the symbolic constants from the ALSA
headers (the final value is the same). Second, copy paste some funky
looking parameter setup from VLC's alsa output for setting the AES1,
AES2, AES3 parameters. (The code is actually not literally copy-pasted,
but does exactly the same.)
My small but non-zero hope is that this could make DTS-HD work, or at
least work into that direction. I can't test spdif stuff though, and
for DTS-HD not even opening the ALSA device succeeds on my system.
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Using spdif with alsa requires adding magic parameters to the device
name, and the existing code tried to deal with the situation when the
user wanted to add parameters too.
Rewrite this code, in particular remove the duplicated parameter string
as preparation for the next commit. The new code is a bit stricter, e.g.
it doesn't skip spaces before and after '{' and '}'. (Just don't add
spaces.)
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This accessed tons of private libavformat symbols all over the place.
Don't do this and convert all code to proper public APIs. As a
consequence, the code becomes shorter and cleaner (many things the code
tried are done by libavformat APIs).
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It's probably better if all auto-inserted filters are removed when doing
an af_add operation. If they're really needed, they will be
automatically re-added.
Fix the error message. It used to be for an actual internal error, but
now it happens when format negotiation fails, e.g. when trying to use
spdif and real audio filters.
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Obviously I didn't test commit 1b8cd01, and it just crashed. Oops.
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Lately Travis sends out many notifications that are false positives caused by
timeout. We are annoyed.
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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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It spams these in verbose mode. It's caused by format negotiation code
in af.c. It's for the mpv format to ffmpeg-format case, and that one is
very uninteresting. (The ffmpeg supported audio formats are practically
never extended.)
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This was supposed to handle preemption better. I still think the current
state isn't very nice, since the decoder can "accidentally" call the
previous render function after preemption (instead of calling the
reloaded function), so there might be issues. But all in all, this
dummy_render function is a bit confusing, and still not entirely
correct, so it's not worth it.
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The code should be equivalent; a compatibility macro definition is left.
(It should be mass-replaced later.)
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Turn the sample format definitions into an enum. (The format bits are
still macros.) The native endian versions of the new definitions don't
have a NE suffix anymore, although there are still compatibility defines
since too much code uses the NE variants.
Rename the format bits for special formats to help to distinguish them
from the actual definitions, e.g. AF_FORMAT_AC3 to AF_FORMAT_S_AC3.
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Also, remove all af_fmt2str usages.
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We found that the stretching - although it usually improves the looks of
the fonts - is incorrect.
On DVD, subtitles can cover the full area of the picture, and they have
the same pixel aspect as the movie itself.
Too bad many commercially released DVDs use bitmap fonts made with the
wrong pixel aspect (i.e. assuming 1:1) - --stretch-dvd-subs will make
these more pretty then.
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The previous code used the output video's pixel aspect for stretching
purposes, breaking rendering with e.g. -vf scale in the chain. Now
subtitles are stretched using the input video's pixel aspect only,
matching the intentions of the original subtitle author.
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Basically I introduced an inverted condition, and the line removed was
inactive before commit ce72aaa.
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I overlooked the fact that the ffmpeg/libav build system only supports `--cc`
and completly ignores $CC. Hopefully this makes the build times a little
faster.
Fixes #332
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There's no reason why we should e.g. handle SIGQUIT, but not SIGTERM.
Note that sending SIGTERM twice still kills the player.
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This is a confusing mechanism, so the explanation should bit more clear.
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Besides cosmetic changes, also change memcpy_pic return type and remove
config.h include.
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This removes "--hwdec=crystalhd".
I doubt anyone even tried to use this. But even if someone wants to
use it, the decoders can still be explicitly invoked with e.g.:
--vd=lavc:h264_crystalhd
The only advantage our special code provided was fallback to
software decoding. (But I'm not sure how the ffmpeg crystalhd
pseudo-decoder actually behaves.)
Removing this will allow some simplifications as soon as we don't need
vdpau_old.c anymore.
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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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This drops autorepeated key events for a number of commands. This should
help with slow situations accidentally triggering too many repeats. (I'm
not sure if this actually happened to users - maybe.)
It's not clear whether MP_CMD_COMMAND_LIST (multiple commands on one
binding separated by ';') should be repeated, or whether it should try
to do something clever. For now, disallow autorepeat with it.
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Could possibly leading to failing compilation on systems with headers
that miss the vdpau extension.
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This uses vdpau OpenGL interop to convert a vdpau surface to a texture.
Note that this is a bit weak and primitive. Deinterlacing (or any other
form of vdpau postprocessing) is not supported. vo_opengl chroma scaling
and chroma sample position are not supported. Internally, the vdpau
video surfaces are converted to a RGBA surface first, because using the
video surfaces directly is too complicated. (These surfaces are always
split into separate fields, and the vo_opengl core expects progressive
frames or frames with weaved fields.)
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The goal is being able to use vdpau decoding independently from
vo_vdpau.c.
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Instead of checking for resolution and image format changes, always
fully reinit on any parameter change. Let init_video do all required
initializations, which simplifies things a little bit.
Change the gl_video/hardware decoding interop API slightly, so that
hwdec initialization gets the full image parameters.
Also make some cosmetic changes.
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More correct, might make things slightly faster (probably
insignificant).
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These formats are helpful for distinguishing surfaces with and without
alpha. Unfortunately, Libav and older version of FFmpeg don't support
them, so code will break. Fix this by treating these formats specially
on the mpv side, mapping them to RGBA on Libav, and unseting the alpha
bit in the mp_imgfmt_desc struct.
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This is the last release in the 1.2.4 line.
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Signed-off-by: Timothy Gu <timothygu99@gmail.com>
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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().)
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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...
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