| Commit message (Collapse) | Author | Age | Files | Lines |
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This possibly enables code that has never been tested before
(accidentally), so let's hope this works out ok.
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It was a bit too complicated and inconvenient, and I doubt anyone
actively used it. The mpv EDL format should cover all use cases.
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Questionable change from user perspective, but internally needed to
implement the next commit. Also useful for testing timeline stuff.
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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.
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The intention of the existing code was trying to match demuxer-reported
stream IDs, instead of using possibly arbitrary ordering of the frontend
track list. But EDL files can consist of quite different files, for
which trying to match the stream IDs doesn't always make sense.
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This didn't have any consequences, other than suddenly reinitializing
video when it works again (such as with EDL timeline mixing video and
audio-only files).
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OpenGL interop was essentially disabled, because the decoder didn't
request vdpau device creation from vo_opengl.
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The size accidentally defaulted to 200 seconds instead of 200
milliseconds, which had fatal consequences when trying to use it.
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This always bothered me.
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This always bothered me.
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This could lead to (barely) audible artifacts with --af=scaletempo and
modified playback speed.
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The AF control commands used an elaborate and unnecessary organization
for the command constants. Get rid of all that and convert the
definitions to a simple enum. Also remove the control commands that
were not really needed, because they were not used outside of the
filters that implemented them.
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And by "cleanup", I mean "remove". Actually, only remove the parts that
are redundant and doxygen noise. Move useful parts to the comment above
the function's implementation in the C source file.
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Some decoders used to read packets and decode data when calling
resync_audio_stream(). This required a special case in mp_seek() for
audio. (A comment mentions liba52, which is long gone; but until
recently ad_mpg123.c actually exposed this behavior.)
No decoder does this anymore, and resync_audio_stream() works similar
as resync_video_stream(). Remove the special case.
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The priv struct is now allocated by talloc in stream.c. It doesn't need
to be manually freed, and using free() instead of talloc_free() probably
crashes.
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Same as MPlayer svn commit r36515 "Chose cheaper alpha blend equation."
No idea if this is actually faster, but can't hurt.
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When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
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Commit 22b3f522 not only redid major aspects of audio decoding, but also
attempted to fix audio format change handling. Before that commit, data
that was already decoded but not yet filtered was thrown away on a
format change. After that commit, data was supposed to finish playing
before rebuilding filters and so on.
It was still buggy, though: the decoder buffer was initialized to the
new format too early, triggering an assertion failure. Move the reinit
call below filtering to fix this.
ad_mpg123.c needs to be adjusted so that it doesn't decode new data
before the format change is actually executed.
Add some more assertions to af_play() (audio filtering) to make sure
input data and configured format don't mismatch. This will also catch
filters which don't set the format on their output data correctly.
Regression due to planar_audio branch.
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Also split the function itself into 3.
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Regression introduced with commit a89549e8.
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Don't call ao_play() if there's nothing left. Of course this still asks
the AO to play internally buffered audio by setting drain=true.
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Simulate proper handling of AOPLAY_FINAL_CHUNK. Print when underruns
occur (i.e. running out of data). Add some options that control
simulated buffer and outburst sizes.
All this is useful for debugging and self-documentation. (Note that
ao_null always was supposed to simulate an ideal AO, which is the reason
why it fools people who try to use it for benchmarking video.)
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Also break that line a bit.
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Might be helpful later.
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These were replaced with DEMUXER_CTRL_SWITCHED_TRACKS a while ago.
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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.)
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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.
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This should be almost equivalent, but is slightly better because the
EOF flag is reset earlier.
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This should allow it to select better fallback formats, instead of
picking the first encoder sample format if ao->format is not equal to
any of the encoder sample formats.
Not sure what is supposed to happen if the encoder provides no
compatible sample format (or no sample format list at all), but in this
case ao_lavc.c still fails gracefully.
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The added function af_format_conversion_score() can be used to select
the best sample format to convert to in order to reduce loss and extra
conversion work.
It calculates a "loss" score when going from one format to another, and
for each conversion that needs to be done a certain score is subtracted.
Thus, if you have to convert from one format to a set of other formats,
you can calculate the score for each conversion, and pick the one with
the highest score.
Conversion between int and float is considered the worst case. One odd
consequence is that when converting from s32 to u8 or float, u8 will be
picked.
Test program used to develop this follows:
#define MAX_FMT 200
struct entry {
const char *name;
int score;
};
static int compentry(const void *px1, const void *px2)
{
const struct entry *x1 = px1;
const struct entry *x2 = px2;
if (x1->score > x2->score)
return 1;
if (x1->score < x2->score)
return -1;
return 0;
}
int main(int argc, char *argv[])
{
for (int n = 0; af_fmtstr_table[n].name; n++) {
struct entry entry[MAX_FMT];
int entries = 0;
for (int i = 0; af_fmtstr_table[i].name; i++) {
assert(i < MAX_FMT);
entry[entries].name = af_fmtstr_table[i].name;
entry[entries].score =
af_format_conversion_score(af_fmtstr_table[i].format,
af_fmtstr_table[n].format);
entries++;
}
qsort(&entry[0], entries, sizeof(entry[0]), compentry);
for (int i = 0; i < entries; i++) {
printf("%s -> %s: %d \n", af_fmtstr_table[n].name,
entry[i].name, entry[i].score);
}
}
}
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This was accidentally equivalent to floatp.
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These must be written even if there was no "final frame", e.g. due to
the player being exited with "q".
Although the issue is mostly of theoretical nature, as most audio codecs
don't need the final encoding calls with NULL data. Maybe will be more
relevant in the future.
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You get the native format by not appending any suffix to the format.
This change includes user-facing names, e.g. for the --format option.
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This used to be in bytes, now it's in samples. Divide the value by 8
(assuming a typical audio format, float samples with 2 channels).
Fix some editing mistake or non-sense about the extra buffering added
(1<<x instead of x<<5).
Also sneak in a s/MPlayer/mpv/.
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This was needed for ao_jack.c., but not anymore.
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This changes option parsing as well as filter defaults slightly. The
default is now to encode to spdif (this is way more useful than writing
raw AC3 - what was this even useful for, other than writing broken ac3
-in-wav files?). The bitrate parameter is now always in kbps.
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Apparently this was completely broken after commit 22b3f522. Basically,
this locked up immediately completely while decoding the first packet.
The reason was that the buffer calculations confused bytes and number of
samples. Also, EOF reporting was broken (wrong return code).
The special-casing of ad_mpg123 and ad_spdif (with DECODE_MAX_UNIT) is a
bit annoying, but will eventually be solved in a better way.
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This could cause the bundle to recache stuff because of differences with
configuration of other software using fonconfig. The defaults OS X directories
should be added to fontconfig at build time (through configure).
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I thought this would always work... how disappointing.
Revert to interleaved format if requesting non-interleaved fails.
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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.
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Practically all audio decoding and filtering code leaked sample data
memory after uninitialization due to a simple logic bug (or typo).
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rejected it
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The existing code tried to remove the "extra" profile flags for h264.
FF_PROFILE_H264_INTRA doesn't matter for us at all, because it's set
only for profiles the vdpau/vaapi APIs don't support.
The FF_PROFILE_H264_CONSTRAINED flag on the other hand is added to
H264_BASELINE, except that it makes the file a real subset of H264_MAIN
and H264_HIGH. Removing that flag would select the BASELINE profile,
which appears to be rarely supported by hardware decoders. This means we
accidentally rejected perfectly hardware decodable files. Use MAIN for
it instead.
(vaapi has explicit support for CONSTRAINED_BASELINE, but it seems to be
a new thing, and is not reported as supported where I tried. So don't
bother to check it, and do the same as on vdpau.)
See github issue #204.
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We got rid of this some time ago, but apparently not completely.
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This used to be needed for teletext support. Teletext commit has been
removed (see commit ebaaa41f), and it appears this code is inactive.
It was just forgotten with the removal. Get rid of it completely.
Untested. (Like all changes to the TV code.)
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Signed-off-by: wm4 <wm4@nowhere>
Significant modifications over the original patch by not overriding
syscalls with macros ("#define open v4l2open") for fallback, but the
other way around ("#define v4l2open open"). As consequence, the calls
have to be replaced throughout the file.
Untested, although the original patch probably was tested.
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Apparently this is not portable to FreeBSD. It turns out that we
(probably) don't use any symbols defined by this header directly, so
the includes are not needed.
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These use the _oldargs_ hack, which failed in combination with playback
resume. Make it work.
It would be better to port all filters to new option parsing, but that's
obviously too much work, and most filters will probably be deleted and
replaced by libavfilter in the long run.
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ao_null should simulate a "perfect" AO, but framestepping behaved quite
badly with it. Framstepping usually exposes problems with AOs dropping
their buffers on pause, and that's what happened here.
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Fixes a problem where the passed size doesn't match the actuall string.
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Conflicts:
audio/out/ao_lavc.c
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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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Drop support for anything before 1.14.0.
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Decoder overwrites parameters in sh_audio, but we still have old audio
in the old format to filter.
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libav* is generally freaking horrible, and might do bad things if the
data pointer passed to it are not aligned. One way to be sure that the
alignment is correct is allocating all pointers using av_malloc().
It's possible that this is not needed at all, though. For now it might
be better to keep this, since the mp_audio code is intended to replace
another buffer in dec_audio.c, which is currently av_malloc() allocated.
The original reason why this uses av_malloc() is apparently because
libavcodec used to directly encode into mplayer buffers, which is not
the case anymore, and thus (probably) doesn't make sense anymore.
(The commit subject uses the word "cargo cult", after all.)
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Remove the awkward planarization. It had to be done because the AC3
encoder requires planar formats, but now we support them natively.
Try to simplify buffer management with mp_audio_buffer.
Improve checking for buffer overflows and out of bound writes. In
theory, these shouldn't happen due to AC3 fixed frame sizes, but being
paranoid is better.
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The format negotiation is the same, except don't confusingly copy the
input format into af->data, just to overwrite it later. af->data should
alwass contain the output format, and the existing code was just a very
misguided use of the af_test_output() helper function.
Just set af->data to the output format immediately, and modify the input
format properly.
Also, if format negotiation fails (and needs another iteration), don't
initialize the libavcodec encoder.
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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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Since ao_openal simulates multi-channel audio by placing a bunch of
mono-sources in 3D space, non-interleaved audio is a perfect match for
it. We just have to remove the interleaving code.
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ALSA supports non-interleaved audio natively using a separate API
function for writing audio. (Though you have to tell it about this on
initialization.) ALSA doesn't have separate sample formats for this,
so just pretend to negotiate the interleaved format, and assume that
all non-interleaved formats have an interleaved companion format.
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Simply change internals from using byte counts to sample counts.
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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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Implementation wise, this could be much improved, such as using a
ringbuffer that doesn't require copying data all the time. This is
why we don't use mp_audio directly instead of mp_audio_buffer.
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This comes with two internal AO API changes:
1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.
2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)
Change all AOs to use the new API.
The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
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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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It's a dead format that was never used anywhere.
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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.
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Now to shift audio pts when outputting to e.g. avi, you need an explicit
facility to insert/remove initial samples, to avoid initial regions of
the video to be sped up/slowed down.
One such facility is the delay filter in libavfilter.
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Previously, using it led to no single frame being output, ever.
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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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Before the bitstream_buffers field was deprecated, you had to free it,
otherwise you would leak memory.
(Although vdpau.c uses a new API, they managed to introduce a new
deprecation this quickly. This is a complaint.)
This introduces a memory leak of 12 bytes per file on every file on some
_older_ libavcodec versions. This is minor enough that I don't care.
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Assume a rtmp prefix means rtmp instead.
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Video has up to 4 textures, if you include obscure formats with alpha.
This means alpha formats could always overwrite the first scaler
texture, leading to corrupted video display. This problem was recently
brought to light, when commit 571e697 started to explicitly unbind all 4
video textures, which broke rendering for non-alpha formats as well.
Fix this by reserving the correct number of texture units.
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Holy inconsistency, let's just kill it with waf.
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Conflicts:
configure
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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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Pretty bad, although it should actually not cause any misbehavior. Comes
from the hardware decoding interop commit.
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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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VA-API's OpenGL/GLX interop is pretty bad and perhaps slow (renders a
X11 pixmap into a FBO, and has to go over X11, probably involves one or
more copies), and this code serves more as an example, rather than for
serious use. On the other hand, this might be work much better than
vo_vaapi, even if slightly slower.
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Most hardware decoding APIs provide some OpenGL interop. This allows
using vo_opengl, without having to read the video data back from GPU.
This requires adding a backend for each hardware decoding API. (Each
backend is an entry in gl_hwdec_vaglx[].) The backends expose video data
as a set of OpenGL textures.
Add infrastructure to support this. The next commit will add support for
VA-API.
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This enw function is similar to mp_image_set_params(), but doesn't force
setting "hard" parameters like image size and format.
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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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Never check s->seek (except in init), because it'd have to check
s->flags anyway. Also, for fast skippable streams (like pipes), don't
set the bit that indicates support for seek forward.
Make sure s->end_pos is always 0 for unseekable streams. Lots of code
outside of stream.c uses this to check seeking support.
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This stops mpv from being stuck in reconnecting at the end of the file
with some unseekable streams.
Test URL: http://playerservices.streamtheworld.com/pls/CBC_R1_VCR_H.pls
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Same semantics with respect to fullscreen as x11.
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This is a bit more intuitive, since before, the window size was just set
to something random when setting the window-scale property during
fullscreen.
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Not really cargo cult, but an unexplainable, needless difference that
just exists to annoy us.
Fixes that gcc on MinGW treats format specifiers in MSVC mode. Just why?
Why?
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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.
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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.)
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Newer versions of perl randomize the hash used for hashes every time
it's run; this makes the order of the fields be non-deterministic. Tack
a sort there to make it deterministic. Needed to fix (or allow fixing) a
buggy gcc warning.
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These aren't printed with newer gcc or clang versions for some reason.
All of them seem to be about local variables shadowing global functions.
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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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We had some code for checking profiles earlier, which was removed in
commits 2508f38 and adfb71b. These commits mentioned that (working) hw
decoding was sometimes prevented due to profile checking, but I can't
find the samples anymore that showed this behavior. Also, I changed my
opinion, and I think checking the profiles is something that should be
done for better fallback to software decoding behavior.
The checks roughly follow VLC's vdpau profile checks, although we do
not check codec levels. (VLC's profile checks aren't necessarily
completely correct, but they're a welcome help anyway.)
Add a --vd-lavc-check-hw-profile option, which skips the profile check.
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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
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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.
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E.g. "-vf scale=848:480" set the w argument twice, instead of setting w
and then h.
This was caused by accidental shadowing of a local variable.
Regression since probably 4cd143e.
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Let's see if I get annoyed reactions...
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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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As preparation for resizing the window with input commands in the
following commit.
Since there are already so many functions which somehow resize the
window, add the word "highlevel" to the name of this new function.
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DVD subs (rarely) have subtitle events without end timestamp. The
duration is unknown, and they should be displayed until they're
replaced by the next event.
FFmpeg fails hard to make us aware whether duration is unknown or
actually 0, so we can't distinguish between these two cases. It fails
at this twice: AVPacket.duration is set to 0 if duration is unknown,
and AVSubtitle.end_display_time has the same issue.
Add a hack that considers all bitmap subtitles with duration==0 as
events with uknown length. I'd rather accidentally display a hidden
subtitle (if they exist at all), instead of not displaying random
subtitles at all.
See github issue #325.
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The code was selecting PA_CHANNEL_POSITION_MONO for MP_SPEAKER_ID_FC,
which is correct only with the "mono" channel layout, but not anything
else. Remove the mono entry, and handle mono separately.
See github issue #326.
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We mixed the "old" AVFrame management functions (avcodec_alloc_frame,
avcodec_free_frame) with reference counting. This doesn't work
correctly; you must use av_frame_alloc and av_frame_free. Of course
ffmpeg doesn't warn us about the bad usage, but will just mess up
things silently. (Thanks a lot...)
While the alloc function seems to be 100% compatible, the free function
will do bad things, such as freeing memory that might still be
referenced by another frame. I didn't experience any actual bugs, but
maybe that was pure luck.
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Looks the gl.h header in XQuartz is incompatible with the one in OS X 10.9.
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mplayer.c split, and some other things.
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Was overlooked.
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This stopped working when the code was changed to create a window even
if --wid is used.
It appears we can't create our own window in this case, because in X11
there is no difference between a window with the root window as parent,
and a window that is managed by the WM. So make this (kind of worthless)
special case use the root window itself.
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Tested with recent metacity; this code is not triggered anymore. The
code was added in 2003 and probably has been unused for years.
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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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No functional change, because the only user of ao_play() ignores return
values below 1.
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Also, replace broken noup= by lavfi expressions.
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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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On systems that provide legacy OpenGL (up to 2.1), but not GL3 and
later, creating a GL3 context will fail. We then revert to legacy GL.
Apparently the error message printed when the GL3 context creation
fails is confusing. We could just silence it, but there's still a X
error ("X11 error: GLXBadFBConfig"), which would be quite hard to
filter out. For one, it would require messing with the X11 error
handler, which doesn't even carry a context pointer (for application
private data), so we don't even want to touch it. Instead, change
the error message to inform the user what's actually happening: a
fallback to an older version of OpenGL.
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When starting mpv with nohup, file descriptor 0 seems to be invalid for
some reason. (I'm not quite sure why it should be... /proc/pid/fd/0
seems to indicate it's just /dev/null, and using /dev/null explicitly
shows that it works just fine.)
select() will always immediately return, and this causes mpv to burn CPU
without reason. Fix this by treating it as EOF when read() returns
EBADF.
Also add EINVAL to this condition, because it seems like a good idea.
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Using static variables for mutable state inside functions is a bad idea,
because it's not immediately obvious that it is what it is.
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This might actually fix an issue with DVB channel switching, because
that uses some sort of hack to reinitialize the demuxer, which causes
the subtitle renderer to initialize twice. As consequence, the assert in
add_subtitle_fonts_from_sources() would trigger.
I didn't actually test the DVB case, because I don't have the hardware.
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This is completely pointless, but it still somehow bugged me.
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Regression from bc49957
Fixes #321
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Disable autorepeat for the "add"/"cycle" commands when changing
properties that are choices (such as fullscreen, pause, and more).
In these cases, autorepeat is not very useful, and can rather harmful
effects (like triggering too many repeated commands if command
execution is slow).
(Actually, the commands are just dropped _after_ being repeated, since
input.c itself does not know enough about the commands to decide whether
or not they should be repeated.)
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Trying to toggle the border during fullscreen (with "cycle border")
would leave the window stuck without border, and it couldn't be
restored. This was because vo_x11_decoration() always excepted to be
called when toggling the state, and thus confusing the contents of the
olddecor variable. Add got_motif_hints to hopefully prevent this.
Also, when changing the border, don't take fs in account. May break on
older/broken WMs, but all in all is in fact more robust and simpler,
because you do not need to update the border state manually when
returning from fullscreen.
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Defining names like min, max etc. in an often used header is not really
a good idea.
Somewhat similar to MPlayer svn commit 36491, but don't use libavutil,
because that typically causes us sorrow.
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If the caller requests at least OpenGL 2.1 (which --vo=opengl does),
but we get OpenGL 1.x, suggest using opengl-old.
Based on a patch by pfor on IRC.
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Also, use more C99 and remove "register" keywords.
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On the float path. Note that this skips clipping, but we expect that
everything on the audio-path is pre-clipped anyway.
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Roughly follows MPlayer svn commits 36492 and 36493. We also remove
the volume peak reporting. (There are much better libavfilter filters
for this, I think.)
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Turnes out, when playing stuff over WiFi, the cache may go below 48%
quite frequently.
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Not up to my quality standards (need ASS boundingboxes), so disabled by default and undocumented for now.
Can be enabled with seektooltip=yes in plugin_osc.conf
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There are certain cases where mpv will automatically set options, such
as per-file configs, per protocol/VO/AO/extension profiles, and
watch_later resume configs. All these were overwriting the user's
options, even when they were specified on command line.
Add something that explicitly preserves command line options. This
means you can now actually use the command line to override any
options that the playback resume functionality backups and restores.
It still happily overrides options set at runtime (e.g. changed via
properties while playing a file; then playing the next file might
override them again), but maybe that's not a problem with typical use.
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It's true that ALSA uses alloca() in some of its API functions, but
since this is hidden behind macros in the ALSA headers, we have no
reason to include alloca.h ourselves.
Might help with portability (FreeBSD).
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Apparently this fixes issues with Gnome, which has a smaller timeout.
See github issue #315.
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The intention of this is to not make the window go outside the screen when
changing dimensions from 2x to .5x.
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We always use the file's timestamps, so the start time can be easily
something different from 0. Make the --length option respect this.
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We don't need that anymore, because the old way to get default values
by recreating the m_config is gone.
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In cases we're just listing options or checking their values (as it
happens with -vo/-vf etc. suboption parsing), we don't need to allocate
abd initialize the actual option struct. All we're interested in is
the list of options.
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I had to look up what the surf_x and surf_y members do.
That means the name is not very good. Also a comment.
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The default behavior of weston changed some time ago to not fill the surface
black for fullscreen windows.
Now let mpv draw the whole screen in fullscreen mode.
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Fixup commit for bbc146927
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Keep track of the default values directly, instead of creating a new
instance of the option struct just to get the defaults.
Also get rid of the special handling of m_obj_desc.init_options.
Instead, handle it purely by the option parser. Originally, I wanted to
handle --vo=opengl-hq and --vo=direct3d_shaders with this (by making
them aliases to the real VOs with a different preset), but since --vo
=opengl-hq=help prints the wrong values (as consequence of the
simplification), I'm not doing that, and instead use something
different.
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This didn't make sense and caused issues with the following commit: if
it's an option with a data pointer, it should be writable.
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Instead, track the opstruct explicitly. It turns out we don't need the
parent at all. Also, add something rudimentary to track the default
values.
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I think in this case, having a separate function is confusing.
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Might reduce memory overhead, and is also less annoying.
Since pointers to m_config_options are kept around, this assumes
they're added only during initialization. Otherwise you'd get
dangling pointers.
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Allocate strings only if needed (when we have to prefix sub-options).
Prepare for storing m_config_options in an array instead of a list.
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We can assume memcpy is enough, because the source should be from
static data. (It wouldn't work if the data could contain pointers
back into itself.)
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Minor simplification. String options are now not allowed to use the
same variable/field anymore. (Also affects other "dynamic" options
which require memory allocation.)
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It looks like on Mavericks the Finder is passing different arguments to mpv
depending on whether the opened file is quarantined or not.
Fixes #285
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This seems to be a problem only in OS X 10.9. I guess they improved the
general speed of the Cocoa startup and suddenly mpv core takes more time
than the Cocoa thread to initialize.
Fixes #285
(hopefully!)
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This time I didn't bother to move the contents of the author field to
the file headers. "git log" is your friend.
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The author and comment fields were printed only in -v mode.
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Drop the author and comment fields. They were completely unused - not
even printed in verbose mode, just dead weight.
Also use designated initializers and drop redundant flags.
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Set the input/output format in filter init. This doesn't change anything
functionally, but it makes the forced format show up in the filter chain
init verbose output (which sometimes prints the filter chain before all
filters have been configured).
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The filter itself was already renamed earlier - but rename the file too.
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The speed advantages are marginal (at least with the way it's currently
used), and it might actually be slower on some drivers, like Mesa.
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Maybe this can be the default in the future, but I'm not sure yet.
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af_format is the old audio conversion filter. It could do all possible
conversions supported by the audio chain. However, ever since the
addition of af_lavrresample, most conversions are done by
libav/swresample, and af_format is used as fallback.
Separate out the fallback cases and remove af_format. af_convert24 does
24 bit <-> 32 bit conversions, while af_convertsignendian does sign and
endian conversions. Maybe the way the conversions are split sounds a bit
odd. But the former changes the size of the audio data, while the latter
is fully in-place, so there's at least different buffer management.
This requires a quite complicated algorithm to make sure all these
"partial" conversion filters can actually get from one format to
another. E.g. s24le->s32be always requires convertsignendian and
convert24, but af.c has no idea what the intermediate format should
be. So I added a graph search (trying every possible format and
filter) to determine required format and filter. When I wrote this,
it seemed this was still better than messing everything into
af_lavrresample, but maybe this is overkill and I'll change my
opinion. For now, it seems nice to get rid of af_format though.
The AC3->IEC61937 conversion isn't supported anymore, but I don't think
this is needed anywhere. Most AOs test all formats explicitly, or use
the AF_FORMAT_IS_IEC61937() macro (which includes AC3).
One positive consequence of this change is that conversions always
include dithering (done by libav/swresample), instead of possibly going
through af_format, which doesn't do anything fancy.
Rename af_force to af_format. It's essentially compatible with command
line uses of af_format. We retain a compatibility alias for af_force.
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At least not with ffmpeg.
Honestly, I have no idea how little endian AC3 works at all, since
ao_pcm doesn't do anything special about it, and treats it like s16le.
Maybe it's broken and ffmpeg has special logic to detect it.
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This is obviously not needed, and just creates potential for bad
breakages (e.g. what happens if libquvi tries to open a normal filename
as http URL?).
Note that for simplicity, we still pass file:// URIs to quvi, and we
don't exclude other protocol prefixes either. In general, we don't know
what protocols quvi might support, so we don't try to second-guess it.
(Even though in practice, it's probably only "http" and "https".)
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The code did not set and unset the current context inside sync sections. I am
not sure if this was an actual problem but this is better since the context is
linked to a single thread. In my brief tests this seems to avoid garbage to
show up in fullscreen.
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Crashed, instead of disabling auto-repeat.
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Instead of removing dragging we now test if it we should drag the window or
not. Because if the OSC shows up we can not drag the window because that would
cause mouse events that makes the OSC disappear.
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This reverts commit 3308bc2bc919cb8fadddad04e48b94d4e9324a34.
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This reverts commit d75cfef49c271e43317666451f17c55e6747e564.
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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.)
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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.
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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.
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If cache was enabled, using ordered chapters could easily crash. The
reason is that enable_cache() reopens the demuxer and closes the old
one. The code after that (reading m->ordered_chapters etc.) then
accessed freed data.
This commit also avoids enabling cache for files which are not used
(which would make opening much slower).
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Implement MP_GROW_ARRAY using MP_TARRAY_GROW. MP_GROW_ARRAY is basically
the earlier version of MP_TARRAY_GROW, and had different semantics. When
I added MP_TARRAY_GROW, I didn't dare to change it, but I think all code
that relied on the exact semantics of MP_GROW_ARRAY is gone now, or the
difference doesn't matter anyway. The semantic change is that now
(n+1)*2 elements are preallocated instead of n*2.
Also, implement MP_TARRAY_GROW using MP_RESIZE_ARRAY, which saves 1 line
of code.
In future, these macros should be part of TA.
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--hwdec=vdpau did nothing with older ffmpeg/libav versions. Oops.
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Apparently, Samba has .pc files now.
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Fixes #304.
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Was disabled by default, was never used, internal support was
inconsistent and poor, and there has been virtually no interest in
creating translations.
And I don't even think that a terminal program should be translated.
This is something for (hypothetical) GUIs.
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Not 100% complete; not touching the more complicated cases.
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see boxalpha option
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Apparently this was for MorphOS. If you really want to use that, ask
the devs to provide dummy headers and declarations for ioctl() instead.
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Too ugly to leave it be.
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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.
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If a source is shorter than the requested chapter length, account for
the time missing.
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When adding or removing frames to avoid 1-frame seeks on chapter
boundaries, the end of the chapter needs to be pushed or pulled by the
same amount to keep the intended end frame the same.
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Since ordered editions can reference external files, when an ordered
chapter references another ordered edition, we have to go and figure out
what is is referencing as well.
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When playing a Matroska file with ordered chapters, it may reference
another file by edition uid. When this edition is also ordered, it may
reference other files. When this occurs, the new segment/edition pair is
added to the list of sources to search for.
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This will need to be recursive to support chapters referencing ordered
editions in other files.
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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.
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This caused the equalizer controls to appear stuck.
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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.
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Also moved OPTIONS headline into options.rst
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camelCase from some of the option names removed. If you were using those, you'll have to update them in your config.
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This could cause the OSC to be displayed without mouse interaction: for
example, starting mpv with --fs, and putting the mouse to where the OSC
area is beforehand, would cause the OSC to appear and stay visible. We
don't want that. The simplest solution is not generating artificial
mouse move events from mouse enter events, because they make the OSC
think the mouse was actually moved.
Also see commit 0c7978c, where handling of mouse enter events was added.
This was supposed to fix certain corner cases, but they're not relevant
anymore due to changes in OSC behavior.
Commit 9777047 fixed this as well (by resetting the mouse state on
MOUSE_LEAVE), but all the behavior reverted with this commit as perhaps
a bad idea. It wasn't very robust, made it hard to distinguish real
events from artificial ones, and finally made the mouse cursor more
often visible than needed. (Now switching between workspaces doesn't
make the cursor visible again when switching to a fullscreened mpv.)
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and store it freshly on first mouse_move event when entering again
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A documentation explaing where and how to create one instead will follow.
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When the main mpv config file, "info" is used, so this makes things
consistent.
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Now you can pretend the config file is quite literally command line
values dumped into a file, e.g.
--option1=value
--option2=value
...
although the underlying mechanisms are quite different.
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Until now it used both char[] and bstr variants in the same code, which
was nasty. For example, the next commit would have additionally required
using memmove() to remove the prefix from the char[] string.
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Can can disable it with --osc=no.
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This fixes handling of e.g. "--vf=lavfi=[ noise ]" when used with
playback resume functionality. The spaces made it bug out, and there are
more cases where it could potentially break.
We could always escape for simplicity, but for now make old and new mpv
builds somewhat interoperable and use this new escaping only if needed.
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This parses "%len%string" escapes, where string can contain any
characters. This method of escaping has also been used in other parts
of mplayer and mpv, so it's not a new idea.
(Also, don't confuse with URL encoding.)
Needed by the following commit.
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With the way I've been doing releases in the release/0.1 branch, this
proved completely useless. You need to write the VERSION file anyway,
and since we use github's automatically generated tarballs (via its
release system), the VERSION file needs to be in the git revision of
a release anyway. git master on the other hand displayed the v0.1.0
tag, because the tag is within the master branch.
This essentially reverts commit b27f65a. Now just print the contents
of the VERSION file if it exists, and likewise, we append the git
revision if it exists. (Do that even if VERSION exists, because this
way we can tell if someone is using the release branch at an in-between
point where no new release has been made yet.)
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Also change what the FFmpeg version info looks like, and additionally
dump lavfi/lavr/lswr versions. (Don't bother with libavdevice and
libpostproc, they're not important enough.)
Unfortunately, there's no "single" FFmpeg/Libav version due to fatal
braindeath on the FFmpeg/Libav side. We can't map the versions to
releases either (it simply isn't accessible anywhere).
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when OSC not visible
The mouse area that covers the OSC is now only activated when the OSC is actually visible, to make sure the mouse still hides if it happens to be parked in the OSC area without making the OSC show up.
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vo_image didn't handle OSD redrawing correctly anymore after OSD
redrawing behavior was changed in commit ed9295c (or maybe it has been a
problem for a longer time, and only showed up now). Basically, flip_page
was called unexpectedly and when no image was stored, which made it
crash trying to access the image. This could happen when for example
provoking OSD redrawing by pausing while using --vo=image, or by using
this command line: mpv --vo=image '-vf=lavfi="select=not(mod(n\,3))"'
Fix by removing the code that pretends vo_image can redraw OSD, and by
removing the framestepping fallback, which could make bad things happen
if the VO didn't support OSD redrawing. By now, there aren't any real
VOs that can't redraw the OSD properly, so this code is not needed and
just complicates things like vo_image.
This change likely will also be useful for vo_lavc (encoding).
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Libass is technically an optional dependency, but in practice users
tend to disable libass accidentally or for the hell of it to get
something "minimal", without being aware of the consequences.
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The deadzone-size is now by default zero, so movement on the entire window will make the OSC show up. To avoid it showing up by randomly moving mice, the option 'minmousemove' controls how many pixels movement (default: 1) between ticks (frames) are necessary to make the OSC show up.
The deadzone can be reenabeled by setting the option 'deadzonesize' (default: 0 = no deadzone, 1 = entire area between OSC and opposite window border), to restore the old behavior, set it to ~0.92.
The OSC will hide immediately when leaving the window or entering the deadzone (if existing) or after the time specified with 'hidetimeout' (default: 500ms) passed without any new movement. Set to negative value to disabling auto-hide (thus restoring old behavior). The OSC will never hide if hovered by the mouse.
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Now that talloc has been removed, the license can be switched back to
GPLv2+. Actually, there never was a GPLv2+ licensed MPlayer (fork or
not) until now, but removal of some GPLv2-only code makes this possible
now. Rewrite the Copyright file to explain the reasons for the licenses
MPlayer and forks use. The old Copyright file didn't contain anything
interesting anymore, and all information it contained is available at
other places in the source tree.
The reason for the license change itself is that it should improve
interoperability with differently licensed code in general.
This essentially reverts commit 1752808.
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There are multiple reasons to do this. One big reason is the license:
talloc is LGPLv3+, which forces mpv to be licensed as GPLv3+.
Another one is that our talloc copy contains modifications, which makes
it essentially incompatible with upstream talloc (in particular, our
version aborts on out of memory conditions - well, it wasn't my idea).
Updating from upstream is also a bit involved - the talloc source is
not really organized in a way to allow copying it into projects (and
this isn't an intended use-case).
Finally, talloc is kind of big and bloated. The replacement halves the
amount of code - mainly because we didn't use all talloc features. It's
even more extreme if you compare upstream talloc (~4700 lines) and the
new allocator without talloc compat (~900 lines).
The replacement provides all features we need. It also doesn't clash
with talloc. (The talloc compatibility wrapper uses macros to avoid
introducing linker-level symbols which could clash with libtalloc.)
It also tries to lower the overhead (only 4 words opposed to 10 words
in talloc for leaf nodes in release mode). Debugging features like leak
reporting can be enabled at compile time and add somewhat more overhead.
Though I'm not sure whether the overhead reduction was actually
successful: allocations with children need an "extra" header, which adds
plenty of overhead, and it turns out that almost half of all allocations
have children. Maybe the implementation could be simplified and the
extra header removed - even then, overhead would be lower than talloc's.
Currently, debugging features can be entirely deactivated by defining
NDEBUG - I'm not sure if anything defines this directly yet, though.
Unlike in talloc, the leak reporting stuff is thread-safe. (That's also
why it's far less elegant, and requires extra list pointers.)
Comes with a compatibility layer, so no changes to mpv source code
are needed. The idea is that we will pretend to be using talloc for
a while, so that we can revert to our old talloc implementation at
any time for debugging purposes.
Some inspiration was taken from Mesa's ralloc:
http://cgit.freedesktop.org/mesa/mesa/tree/src/glsl/ralloc.h
This is another talloc replacement, but lacks some features we need
(getting size of an allocation, debugging features, being able to
access children in the dtor).
There's some information in ta/README what will happen next and how the
transition is expected to progress.
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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.)
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This only shows any differences when mpv isn't frontmost and is in fullscreen.
Cmd+Tab overlay is still at a higher level as to avoid complete usability fail.
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Before this, they were shown on terminal only. Now they use the OSD
mechanism, which shows them on the video window or the terminal,
depending on settings and what's available.
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Changing volume when audio is disabled was a feature request (github
issue #215), and was introduced with commit 327a779.
But trying to fix github issue #280 (volume is not correct in no-audio
mode, and if audio is re-enabled, the volume set in no-audio mode isn't
set), I concluded that it's not worth the trouble and the current
implementation is questionable all around. (For example, you can't
change the real volume in no-audio mode, even if the AO is open - this
could happen with gapless audio.) It's hard to get right, and the
current mixer code is already hilariously overcomplicated. (Virtually
all of mixer.c is an amalgamation of various obscure corner cases.)
So just remove this feature again.
Note that "options/volume" and "options/mute" still can be used in
idle mode to adjust the volume used next time, though these properties
can't be used during playback and thus not in audio-only mode.
Querying the volume still "works" in audio-only mode, though it can
return bogus values.
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Letting some playlist functions randomly accept NULL, while others do
not isn't such a good idea.
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The argument or this change is that --loop should set how often the
file is played, not the number of additional repeats.
Based on pull request 277, with additions to the manpage and removal
of "--loop=0".
Signed-off-by: wm4 <wm4@nowhere>
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The primary effect of this commit is that it fixes playback resume if
libquvi 0.9 and German locale are used, and libkdecore is on the system.
(See github issue #279.)
libquvi uses libproxy to determine system-wide proxy settings. libproxy
in turn loads libkdecore (if present) to determine KDE proxy settings.
libkdecore has a global constructor, which calls setlocale(LC_ALL, ""),
switching the current locale from "C" to what the user's settings. This
breaks some basic C string processing functions. Note that the locale
won't be set on program start, but only when libproxy calls dlopen() on
its config_kde module, which actually causes libkdecore to be loaded and
initialized.
In particular, with German locale, snprintf() would use "," instead of
"." when formatting float values, which in combination with playback
resume, would lead to parse errors the next time mpv was started, which
is how this issue was found.
I'd consider this a bug with libkdecore or at least libproxy. No library
should ever even touch locale: it might break basic expectations on C
string handling functions, might override program settings, and it's not
thread-safe. But this is so nasty and severe, that a quick hack to fix
it hurts less.
See github issue #279 and KDE bug #325902.
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Before this patch, those will cause a crash:
mpv -playlist /dev/null
mpv -playlist <(bla) # if the result of bla is empty
Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com>
Signed-off-by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
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Fixes #274
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previously, mp_ring_read_cb() would allow reading past the end of the
ringbuffer when buffered > available. mp_ring_read() had logic for
splitting the read into two, which I duplicated into mp_ring_read_cb().
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First, don't try to seek if the result is 0 (i.e. nothing found, or
subtitle event happens to be exactly on spot).
Second, since we never can make sure that we actually seek to the exact
subtitle PTS (seeking "snaps" to video PTS), offset the seek by 10ms.
Since most subtitle events are longer than 10ms, this should work fine.
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As expected, this is not liked.
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The OSC will now display cache fill status between the timecodes, but only if it's below 48% to not clutter the interface with erratically changing values.
By default, the displayed value is multiplied by 2 to not confuse users who are unfamillar with the inner workings of the caching system. This can be disabled using the iAmAProgrammer=true setting.
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If the OSC is rendered on a forced window (by --force-window), the scale can be configured with scaleForcedWindow.
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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@36461 b3059339-0415-0410-9bf9-f77b7e298cf2
Fixes playback of http://mpg123.org/test/44and22.mp3
Cherry-picked from MPlayer SVN rev. #36461, a patch by
Thomas Orgis, committed by by Reimar Döffinger.
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The window wasn't cleared in this mode before doing the redrawing.
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Requested by github issue #255.
Does not work where mmap is not available (i.e. Windows).
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You couldn't jump to the first entry in the playlist, if that entry was
marked as "too short". But this is usually fine, because it doesn't get
into the way of the user. (This feature is meant to allow being able to
jump backwards through the playlist even if a file immediately stops
playback right after initialization, not to prevent playing these files
at all.)
Also, apply the skip logic when wrapping around the playlist when going
backwards. This happens with --loop, because looping pretends the
playlist is infinitely repeated forwards and backwards (as long as the
playlist_prev/next commands are concerned).
Also add some comments.
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Just specifying a VO that doesn't exist would make it crash.
Regression caused by commit 6a0408b.
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I forgot these when I split the code. They will hopefully fix compilation on
10.7.
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When for example switching off the video stream, and --force-window is
used, forcefully reconfigure the VO. This will reset the size, and will
clear the window with black.
Needed some effort, because you don't always want to clear the window
on "discontinuity" points like going to a next file: this would
introduce some flicker.
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glXGetVisualFromFBConfig() specifies specifies that it can return NULL
if there is no associated X visual. Instead of crashing, let
initialization fail. I'm not sure if this is actually supposed to work
with a fallback visual (passing a NULL visual to vo_x11_config_vo_window
would just do this), but let's play safe for now.
Apparently this can happen when trying to use vo_opengl over a remote
X display.
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Output silence to the output buffer during underruns. This removes small
occasional glitches that happen before the AUHAL is actually paused from the
`audio_pause` call.
Fixes #269
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Based on an earlier patch for mplayer by Alexandre Ratchov <alex@caoua.org>
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This was the only thing in video/out/ left with major mp_msg() use.
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Because why not.
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Essentially reuses the sub_step command, and is subject to the same
restrictions.
Seems to behave a bit strange sometimes, but generally works.
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A leftover from 10 years of refactoring.
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Can happen when seeking in audio files with attached picture.
MP_NOPTS_VALUE doesn't really make sense outside of mpv internals, so
don't return it.
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Reverts a small change made in commit ed9295c. This is needed, because
otherwise mplayer.c/update_video_attached_pic() thinks it never has to
update the picture after initialization. (Maybe there would be more
elegant ways to handle this, but not without adding extra state.)
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The OSD bar wasn't displayed, and instead a fallback message was shown.
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Didn't handle VO events, didn't handle OSD message management.
There is probably still some strangeness left. --idle mode was never
made for direct interaction.
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Closer to the ubiquitous 16:9 screen aspect ratio. Requested by ChrisK3.
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Seems appropriate.
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We can render subtitles if a VO is open. Whether we're decoding video
(i.e. if mpctx->sh_video is set) doesn't really matter.
Subtitle display with --force-window still doesn't quite work yet,
because there's nothing to actually force redrawing of subtitles in this
mode.
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This commit adds the --force-window option, which will cause mpv always
to create a window when started. This can be useful when pretending that
mpv is a GUI application (which it isn't, but users pretend anyway), and
playing audio files would run mpv in the background without giving a
window to control it.
This doesn't actually create the window immediately: it only does so
only after initializing playback and when it is clear that there won't
be any actual video. This could be a problem when starting slow or
completely stuck network streams (mpv would remain frozen in the
background), or if video initialization somehow is stuck forever in
an in-between state (like when the decoder doesn't output a video
frame, but doesn't return an error either). Well, we can pretend only
so much that mpv is a GUI application.
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Seeking normally resets the VO and throws away the currently displayed
frame, so if you seek outside the video with --keep-open enabled, the
window would normally be "stuck" and not redraw properly, because there
is no source video frame that could be redrawn. To deal with this, a
precise seek to the position of the last displayed video frame was
issued.
This usually worked, but it can fail for formats where seeking is broken
or not possible (consider reading from a pipe).
Fix this by changing the semantics for vo_seek_reset(): now the video
frame is remembered even after seeking.
Note that this changes behavior a little when trying to seek outside of
a file with --keep-open enabled. Since no actual seek is done anymore,
the video will remain "frozen" on the previous position, and you can't
unpause or framestep to see the video between current position and
actual end of the video. If users complain, I might revert this commit.
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vo_vdpau is the only VO which implements VOCTRL_RESET. Redrawing the
last output frame is hard, because the output could consist of several
source video frames with certain types of post-processing
(deinterlacing). Implement redrawing as special case by keeping the
previous video frames aside until at least one new frame is decoded.
This improves the previous commit, but is separate, because it's rather
complicated.
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Before, a VO could easily refuse to respond to VOCTRL_REDRAW_FRAME,
which means the VO wouldn't redraw OSD and window contents, and the
player would appear frozen to the user. This was a bit stupid, and makes
dealing with some corner cases much harder (think of --keep-open, which
was hard to implement, because the VO gets into this state if there are
no new video frames after a seek reset).
Change this, and require VOs to always react to VOCTRL_REDRAW_FRAME.
There are two aspects of this: First, behavior after a (successful)
vo_reconfig() call, but before any video frame has been displayed.
Second, behavior after a vo_seek_reset().
For the first issue, we define that sending VOCTRL_REDRAW_FRAME after
vo_reconfig() should clear the window with black. This requires minor
changes to some VOs. In particular vaapi makes this horribly
complicated, because OSD rendering is bound to a video surface. We
create a black dummy surface for this purpose.
The second issue is much simpler and works already with most VOs: they
simply redraw whatever has been uploaded previously. The exception is
vdpau, which has a complicated mechanism to track and filter video
frames. The state associated with this mechanism is completely cleared
with vo_seek_reset(), so implementing this to work as expected is not
trivial. For now, we just clear the window with black.
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The previous code fetched the device name regardless of log level and then
only printed it if log level was verbose.
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Trying to connect multiple mpv clients to JACK with the
JackUseExactName option would fail unless the user manually
specifies a unique client name. This changes the behavior
to automatically generate a unique name if the requested
one is already in use.
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vo_x11 had a clever trick to implement a video equalizer: it requested a
DirectColor visual. This is a X11 mechanism which allows you to specify
a lookup table for each color channel. Effectively, this is a safe
override for the graphic card's gamma ramp. If X thinks the window
deserves priority over other windows in the system, X would temporarily
switch the gamma ramp so that DirectColor visuals can be displayed as
the application intends. (I'm not sure what the exact policy is, but in
practice, this meant the equalizer worked when the mouse button was
inside the window.)
But all in all, this is just lots of useless code for a feature that is
rarely ever useful. Remove it and use the libswscale equalizer instead.
(This comes without a cost, since vo_x11 already uses libswscale.)
One worry was that using DirectColor could have made it work better in
8-bit paletted mode. But this is not the case: there's no difference,
and in both cases, the video looks equally bad.
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Will be used by vo_x11.
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Considering that we use version control, maintaining an AUTHORS file
doesn't make much sense, and indeed it was barely touched since the time
mplayer2 forked from MPlayer. None of the contributors to mplayer2 or
mpv are listed in this file (even uau's work is not appropriately
reflected, despite having an entry in this file).
I'm removing this file because it's obviously pointless to keep around,
and moreover it's misleading.
In theory removing this file might be questionable: the git log reaches
back into CVS and SVN times, which don't have separate committer and
author fields for each comment. The commit messages could be incomplete
and not mention the original author, while the AUTHORS file somehow
could. I doubt this is a real concern, since you can always reconstruct
the original AUTHORS file from git.
Another concern is that this could be interpreted as disrespectful to
the people mentioned in this file; but it isn't meant to be
disrespectful.
Also point out a slight irregularity in the Copyright file.
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Commit 884c179 attempted to make it possible to skip backwards through
the playlist, even for files which fail to intitialize, or play for a
very short time. This was also used to prevent mpv from looping forever
and doing nothing if --loop=inf is used, and no file in the playlist is
playable.
This broke looping of very short files, because mpv was assuming that
this case happened. But there are legitimate use cases.
Fix this by making the looping case special. Instead of checking whether
playback was "very short", check whether something could be decoded and
displayed/played. If yes, allow looping.
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These probably don't work. libavcodec doesn't seem to support them,
and neither did the original mplayer-vaapi patch.
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Attempting signed comparison on unsigned value.
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How embarrassing. This could make --hwdec=vaapi-copy as well as
screenshots with vo_vaapi randomly fail. Regression since
commit b8382aa.
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glClear needs GL headers and we don't want those in `cocoa_common`. Create
a callback in `gl_cocoa` and register it `cocoa_common`.
Fixes #264
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After rebasing my dev branch it turned out that the code deadlocked on
recursive calls of `vo_control`. Make the locking code a little bit smarter
by making always skip locking/unlocking if we are executing a chunck of
code that is already synchronized with `dispatch_sync`.
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This fixes the position reporting on retina displays. Doesn't make any
difference on normal displays where 1px = 1pt.
Fixes: #260
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Split the code to several files. The GUI elements now each have they own files
and private state. The original code was a mess to respect the retarded mplayer
convention of having everything in a single file.
This commit also seems to fix the long running bug of artifacts showing
randomly when going fullscreen using nVidia GPUs.
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The boolean-options showWindowed and showFullscreen can now be set in plugin_osc.conf to control the behaviour.
enhancement for #262
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Or at least I hope it fixes it, since I can't test.
Broken by commit 0d90dd0.
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This can just be not supported, so making it look like a real error
doesn't make much sense.
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Just for robustness. Also print a warning in vo_vaapi if this happens.
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Don't allocate a VAImage and a mp_image every time. VAImage are cached
in the surfaces themselves, and for mp_image an explicit pool is
created. The retry loop runs only once for each surface now.
This also makes use of vaDeriveImage() if possible.
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Until now, mouse positions were just passed to the core as-is, even if
the mouse coordinates didn't map to any useful coordinate space, like
OSD coordinates. Lua scripting (used by the OSC, the only current user
of mouse input) had to translate mouse coordinates manually to OSD space
using mp_get_osd_mouse_pos(). This actually didn't work correctly in
cases mouse coordinates didn't map to OSD (like vo_xv): the mouse
coordinates the OSC got were correct, but input.c was still expecting
"real" mosue coordinates for mouse areas.
Fix this by converting to OSD coordinates before passing the mouse
position to the core.
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Do this so that MOUSE_LEAVE can't be combined with other keys. (E.g.
keep 'w' pressed, then move the mouse outside of the mpv window; it will
print a warning what w-MOUSE_LEAVE is not mapped.)
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It appears the last run missed all mp_tmsg().
Since the command parser also prints messages, it needs a new parameter
to pass a log context. We go the easy way and just require the input_ctx
to be passed, and use its mp_log.
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The old code prevented it since uids being NULL makes a 0 talloc length.
Now that we're iterating over a specific length, NULL can be
dereferenced.
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Should fix #259
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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.
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This is pretty much a hack for the OSC. It will allow it to rely on a
somewhat predictable style, instead of having to overwrite all user
OSD settings manually with override tags.
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Nothing really accesses it. Subtitle initialization actually does in a
somewhat meaningful way, but there container size is probably fine, as
subtitles were always initialized before the first video frame was
decoded.
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Now writing -1 to the 'aspect' property resets the video to the auto
aspect ratio. Returning the aspect from the property becomes a bit more
complicated, because we still try to return the container aspect ratio
if no frame has been decoded yet.
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This function would probably be useful in other places too.
I'm not sure why vd.c doesn't apply the aspect if it changes size by
less than 4 pixels. Maybe it's supposed to avoid ugly results with bad
scalers if the difference is too small to be noticed normally.
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As of now, this function is called only after decoding a full, valid
video frame, so the image parameters should be reliable.
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Signed-off-by: wm4 <wm4@nowhere>
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This is preliminary. There are still tons of issues, and any aspect
of scripting may change in the future. I decided to merge this
(preliminary) work now because it makes it easier to develop it, not
because it's done. lua.rst is clear enough about it (plus some
sarcasm).
This requires linking to Lua. Lua has no official pkg-config file, but
there are distribution specific .pc files, all with different names.
Adding a non-pkg-config based configure test was considered, but we'd
rather not.
One major complication is that libquvi links against Lua too, and if
the Lua version is different from mpv's, you will get a crash as soon
as libquvi uses Lua. (libquvi by design always runs when a file is
opened.) I would consider this the problem of distros and whoever
builds mpv, but to make things easier for users, we add a terrible
runtime test to the configure script, which probes whether libquvi
will crash. This is disabled when cross-compiling, but in that case
we hope the user knows what he is doing.
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This time it broke because I didn't actually test compiling vo_vaapi.c,
and it was using a macro from mp_image.h, which implicitly assumed
FFALIGN was available. Screw that too, and copy the definition of
ffmpeg's FFALIGN to MP_ALIGN_UP, and move these macros to mp_comnon.h.
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The code using FFSWAP was moved from vo_vaapi.c to vaapi.c, which didn't
include libavutil/common.h anymore, just libavutil/avutil.h. The header
avutil.h doesn't include common.h recursively in Libav, so it broke
there.
Add FFSWAP as MPSWAP in mp_common.h (copy pasted from ffmpeg) to make
sure this doesn't happen again. (This kind of stuff happens all too
often, so screw libavutil.)
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I'm using the word "languages" instead of "scripts" in the manpage, but
I think that's easier to understand with a smaller amount of
descriptions.
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options.c still doesn't use the constants defined by the libass headers,
but it's still better than exposing those to the user directly.
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This code is actually quite inefficient: it reuses the (slow, simple)
screenshot code. It uses an inefficient method to read the image
(vaGetImage() instead of vaDeriveImage()), allocates new memory for
each frame that is read, and it tries all image formats again each
time.
Also, in my tests it always picked NV12 as image format, which is not
ideal if you actually want to filter the video, and vo_xv can't handle
this format without conversion either.
However, a user confirmed that it worked for him, so everything is fine.
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This will allow GPU read-back with process_image.
We have to restructure how init_vo() works. Instead of initializing the
VO before process_image is called, rename init_vo() to
update_image_params(), and let it update the params only. Then we really
initialize the VO after process_image.
As a consequence of these changes, already decoded hw frames are
correctly unreferenced if creation of the filter chain fails. This
could trigger assertions on VO uninitialization, because it's not
allowed to reference hw frames past VO lifetime.
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Merged from pull request #246 by xylosper. Minor cosmetic changes, some
adjustments (compatibility with older libva versions), and manpage
additions by wm4.
Signed-off-by: wm4 <wm4@nowhere>
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There shouldn't be any functional changes. Just reduce the amount of
pointless temporary variables.
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This code was made inactive some months ago. At this time it wasn't
entirely clear whether this code was still needed, but now I'm pretty
sure it isn't. Even if it is, it didn't work anymore.
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This wasn't enough and could lead to a cut off message shown on OSD.
Just make it dynamic, since we already use dynamic memory allocation
at this point anyway.
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Moving the window was convenient but generates a MOUSE_LEAVE event
which it shouldn't. Now we remove it, because it is still possible
to move the window in weston with MOD+BTN0.
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Setting CompanyName (probably) helps with issue #256.
Update some other fields as well, because why not. (They're full of
MPlayer remnants.)
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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: wm4 <wm4@nowhere>
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The problem with DVD/BD and playback resume is that most often, the
filename is just "dvd://", while the actual path to the DVD disk image
is given with --dvd-device. But playback resume works on the filename
only.
Add a pretty bad hack that includes the path to the disk image if the
filename starts with dvd://, and the same for BD respectively. (It's a
bad hack, but I want to go to bed, so here we go. I might revert or
improve it later, depending on user feedback.)
We have to cleanup the global variable mess around the dvd_device.
Ideally, this should go into MPOpts, but it isn't yet. Make the code
paths in mplayer.c take MPOpts anyway.
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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.
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Mainly for debugging. Usually, we just set options for all possible
protocols, and we can't really know whether a certain protocol is used
beforehand. That's also the reason why avio_open2() takes a dictionary,
instead of letting the user set options directly with av_opt_set(). Or
in other words, we don't know whether an option that could be set is an
error or not, thus we print the messages only at verbose level.
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So nobody has to wonder why everything is silent, even if they raise the
volume.
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This is for properties that normally show a bar, and thus do not show an
OSD message (as per classic mplayer behavior). Setting an extra_msg
allows showing an OSD message anyway, except if OSD messages are
explicitly suppressed.
This refactors the whole show_property_osd() function a bit, and
replaces the weird sep field with a more general method.
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Allows for example: --status-msg='${?pause==yes:(Paused) } ...' to
emulate the normal terminal status line. It's useful in other situations
too.
I'm a bit worried about extending this mini-DSL, and sure hope nobody
will implement a generic formula evaluator at some point in the future.
But for now we're probably safe.
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In most cases, it's better if deinterlacing happens before any other
filtering, so prepend the filter to the user's filter list, instead
of appending it.
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Also restores consistent log message prefixes with ao_lavc.
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Instead of hardcoding a single filter. This might be helpful for
modeling the vaapi deinterlacer as a video filter. The idea is that a
software deinterlacer would be tried first, and if that fails (because
vaapi hardware decoding uses HW surfaces, which a software deinterlacer
does not accept), the vaapi filter would be tried.
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Normally, we need this for Xutf8LookupString(). But we can just fall
back to XLookupString(). In fact, the code for this was already there,
the code was just never tested and was actually crashing when active
(see commit 2115c4a).
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vf_open returns 0 on error, 1 on success. Oops. Accidentally broken
with 6629a95.
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Calling them separately doesn't really make sense, and all existing
calls to them usually combined them. One subtitle difference was that
af_init() didn't wipe the filter chain if initialization of the chain
itself failed, but that didn't really make sense anyway.
Also remove af_init() from the code for setting balance in mixer.c. The
mixer should be in the initialized state only if audio is fully
initialized, so the af_init() call made no sense.
Note that the filter "editing" code in command.c doesn't really do a
nice job of handling errors in case recreating an _old_ (known to work)
filter chain unexpectedly fails, and this obscure/rare case might be
differently handled after this change.
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Also remove stray include statements from ao_alsa and ao_oss.
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The volume is set as soon as the audio chain is created again. This
works only in softvol mode. For system wide volume or otherwise
externally user controllable volume, this code is intentionally
disabled. It would be extremely weird if changing volume (while audio is
not initialized) would do nothing, and then suddenly change it when the
audio chain is created.
There's another odd corner case: the user-set volume will be thrown away
if it's set before the _first_ audio chain initialization. This is
because the volume restore logic recognizes a change from nothing to
softvol or an AO, and circumventing that would require additional code.
Also, we don't even know the start volume before that point.
Forcing the volume with --volume will can override the volume set during
no-audio mode, depending on the situation.
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Note that this is intentionally never done if the AO or softvolume is
different, or if the current volume control method is thought to control
system wide volume (such as ALSA) or otherwise user controllable (such
as PulseAudio). The intention is to keep things robust and to avoid
messing with the user's audio settings as far as possible, while still
providing the ability to resume volume if it makes sense.
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This is rather strange behavior, away with it.
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Refactor how mixer.c does volume/mute restoration and initialization.
Move to handling of --volume and --mute to mixer.c. Simplify the
implementation of these and hopefully fix bugs/strange behavior related
to using them as file-local options (this uses a somewhat dirty trick:
the option values are reverted to "auto" after initialization). Put most
code related to initialization and volume restoring in probe_softvol()
and restore_volume(). Having this code all in one place is less
confusing.
Instead of trying to detect whether to use softvol at runtime, detect it
at initialization time using AOCONTROL_GET_VOLUME (same with mute,
AOCONTROL_GET_MUTE). This implies we expect SET_VOLUME/SET_MUTE to work
if the GET variants work. Hopefully this is always the case.
This is also preparation for being able to change volume/mute settings
if audio is disabled, and for allowing restoring value with playback
resume.
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Softvol always used a linear multiplier for volume control. This was
converted to dB, and then back to linear in af_volume. Remove this non-
sense. We still try to keep the command line argument to af_volume in
dB, though.
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Let struct mixer access access MPOpts to simplify some things. Rename
some variables and functions. There should be no functional changes.
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95a2151d1 introduced a crash on systems lower than 10.9 when opening files
with a single argument.
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Commit broke text subtitles without embedded fonts. Will look for a better
solution later. Revert it for now, since I'm starting to get bug reports.
This reverts commit 4a9f618d9f3b2d964bcc819426cc3042b8e447f7.
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XOpenIM can fail to find a valid input method, in which case it
returns NULL. Passing a NULL pointer to XCreateIC would cause a
crash, so fail VO init before that happens.
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Before this commit there was just an error message, but the file descriptor was
still open. Now we close the file descriptor and prevent it from calling
endlessly. Also a CLOSE_WIN event is sent which closes the window eventually if
the action of CLOSE_WIN is set to quit or quit_watch_later.
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Improves display of images and video with alpha channel, especially if
the transparent regions contain (supposed to be invisible) garbage
color values.
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Blargh.
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This is to avoid the 30s hang while mpv caches fonts. In practice all the
fonts an average user is going to use are embedded in mkv files so there is
no reason to build fontconfig's cache on all of OS X system directories.
I might add something similar for terminal usage, but I am highly undecided.
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It's quite unlikely, but functions like mp_find_user_config_file() can
return NULL, e.g. if $HOME is unset.
Fix all the code that didn't check for this correctly yet.
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They're not really needed, so kill them.
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This makes the code uniform to how stuff was handled for Windows in 1cb55ceb.
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Remove the ifdef hell from mp_find_user_config_file(). Move the win32
specific code (for MinGW and Cygwin) to path-win.c. The behavior should
be about the same, but I can't be sure due to lack of testing and
because the old path.c code was hard to follow. (I expect those who care
about windows will fix things, should issues pop up - sorry.)
One difference is that the new code will always force MPV_HOME. It looks
like the old code preferred the mpv config dir in the exe dir if it
exists.
Also, make sure MP_PATH_MAX has enough space, even if the equivalent
wchar_t string is not 0-terminated with PATH_MAX (because apparently the
winapi doesn't require this). (Actually, maybe we should just kill all
uses of PATH_MAX/MP_PATH_MAX.)
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This is a bit "hard", because getenv() returns a static string, and we
can't just return an allocated string. We also want getenv() to be
thread-safe if possible. (If the mpv core is going to be more threaded,
we sure do want the lower layers to be thread-safe as well.)
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The homepath variable was static, and its value was set to a stack
buffer. This means a second invocation of the function would trigger
undefined behavior. Moreover the stack buffer always went out of scope
before homepath was used.
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This is mainly to avoid spurious cursor states due to the mouse moving inside
or outside the window as a result of the window resize (with cmd-0/1/2).
This avoids complex logic and triggers a mouse move so that the player
recomputes the correct cursor state based on the autohide configuration of
the user.
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This keeps the state in sync with the current state in cocoa_common. Infact the
cocoa code in mpv can decide wether it really wants to hide the cursor based on
the result of the `canHideCursor` method (this is so that the cursor is only
hidden when hovering on the video window).
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Turns out that these checks were for versions of OS X that mpv doesn't even
support anymore. So just remove the checks since they cause a deprecation
warning.
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NULL pointer deref when a .sub file with the same filename as the video
file was present.
I was probably half asleep when writing this code.
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GetCurrentProcess() is deprecated on 10.9. Make a universal solution by
checking OS version number.
get_system_version() function is the recommended Apple way of getting the
OS version, since Gestalt is also deprecated (and does pretty much the same
thing anyway)
Updating HIDRemote.m to use a similar function would allow to get rid of the
2 other warnings.
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This is supposed to reduce the amount of useless error messages shown
during initialization of vo_opengl. If multiple backends are compiled,
usually only one of them will work. For example, on Linux both X and
Wayland backends can be compiled, but usually either Wayland or X is
running. Then, if Wayland is not running, but X is, trying to initialize
the Wayland backend should not spam the terminal with error messages.
Signed-off-by: Andreas Sinz <andreas.sinz@aon.at>
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Call update_subtitles() on every iteration of the playloop, so that
subtitle packets are read as soon as possible, instead of every time a
video frame is displayed. This helps in case the packet queue is swamped
with subtitle packets, which can happen with certain insane mkv files.
The change will simply cause the subtitle queue to be emptied on each
playloop iteration.
The timestamps update_subtitles() uses for display are the same before
and after this commit. (Important for files which have subtitle packets
with timestamps or duration not set.)
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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.
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Pausing the player and changing the aspect would leave the VO without a
frame to display.
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The mpv.desktop file is taken from the Arch package [1]. It appears to
be based on the mplayer2 git mplayer.desktop file (e.g. very similar
MimeType field), with minor modifications applied by Arch package
maintainers.
Note that for now, this doesn't show a terminal (Terminal=false), which
might not always be ideal. For example, if the file is audio only, or
if VO initialization fails for some reason, mpv will run in the
background and play audio without showing a window. But users prefer
running it without terminal, and don't want to play audio files with
it.
Maybe a --force-window option will be added in the future, which would
always create a VO window, and compensate for these issues.
[1] https://aur.archlinux.org/packages/mpv-git/
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These would have to be updated manually all the time. Replacing them
automatically would be possible, but additional work, and would force
regeneration of the manpage way too often.
We decided that we don't need these fields.
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Instead of containing a format string within %w{...}, simply allow %w
to specify one item of a time format string. This is simpler, more like
other format specifiers (%t), and probably easier to use too.
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This is for situations when repeated attempts at playing a playlist
entry failed, and playlist navigation becomes impossible due to that.
For example, it wasn't possible to skip backwards past an unplayable
playlist entry:
mpv file1.mkv doesntexist.mkv file3.mkv
You couldn't skip back to file1.mkv from file3.mkv. When running a
single "playlist_prev" command, doesntexist.mkv would be played, which
would fail to load. As reaction to the failure to load it, the next file
would be played, which is file3.mkv.
To make this even worse, the file could successfully load, but run only
for a split second. So just loading successfully isn't good enough.
Attempt to solve this by marking problematic playlist entries as failed,
and by having playlist_prev skip past such playlist entries. We define
failure as not being able to play more than 3 seconds (or failing to
initialize to begin with). (The 3 seconds are in real time, not file
duration.)
"playlist_prev force" still exhibits the old behavior.
Additionally, use the same mechanism to prevent pointless infinite
reloading if none of the files on the playlist exist. (See github issue
All in all, this is a heuristic, and later adjustments might be
necessary.
Note: forward skips (playlist_next) are not affected at all. (Except for
the interaction with --loop.)
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This is commonly used to disable the screensaver with broken/non-
standard X screensavers. During pause, the screensaver should not be
disabled, so not calling this command while paused seems sensible.
See github issue #236.
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They are rarely useful in my opinion.
This commit was mainly motivated by this message:
Video uses a non-standard and wasteful way to store B-frames ('packed B-frames'). Consider using a tool like VirtualDub or avidemux to fix it.
It's what's left over from the "Invalid and inefficient vfw-avi..."
warning that used to be printed when playing avi/divx files. Although
the new message is much better, it's still rather useless and poses
more questions than it answers. Besides, nobody wants to remux a file
when playing it, especially not if playback appears to be completely
fine. (There are some claims that these files raise CPU usage, but even
my old crappy CPU can decode low res avi/divx files at real time at
about x35 playback speed.)
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Code was using %d format instead %zd to print size_t data.
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I have a sample where some final chapters are missing. This was causing a
segmentation fault when trying to fetch chapter times for them.
This makes the code ignore those chapters.
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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.
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Change it from "Playing file." to "Playing: file". The idea is that it
looks nicer without that trailing dot. (See github issue #229.)
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In init_vo(), if sh->aspect is 0 or last_sample_aspect_ratio is set,
sh->aspect is overwritten. With software decoding fallback behaviour,
this makes the aspect ratio from container ignored since
last_sample_aspect_ratio is already set in first try with hardware
decoding.
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This requires adding a function that converts the filter list back to a
string.
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