When the audio format is not known yet and the audio chain is still
initializing, filter reinit will fail. Normally, attempts to
reinitialize filters at this stage should be rare (e.g. user commands
editing the filter chain). But it sometimes happened with track
switching in combination with the video code calling
update_playback_speed() at arbitrary times.
Get rid of the message by not trying to change the filters for the sake
of playback speed update while decoding is still being initialized.
Until now, we've relied on the following things:
- you can send flush packets to the decoder even if it's fully flushed,
- you can send new packets to a flushed decoder,
- you can send new packers to a partially flushed decoder.
("flushing" refers to sending flush packets to the decoder until the
decoder does not return new pictures, not avcodec_flush_buffers().)
All of these are questionable. The libavcodec API probably doesn't
guarantee that these work well or at all, even though most decoders have
no issue with these. But especially with hardware decoding wrappers
(like MMAL), real problems can be expected. Isolate us from these corner
cases by handling them explicitly.
This applies to unexpected freezes or deadlocks, not e.g. normal
framedrops. The verbose messages also might remind an API user if the
API usage is incorrect, such as not calling mpv_opengl_cb_draw() when a
redraw request was issued.
This commit introduces logic to read other volumes from the same source
as the primary archive. Both .rar formats as well as 7z are supported for now.
It also changes the libarchive callback structure to be per-volume
consistent with the libarchive intenal client data array constructed
with archive_read_append_callback_data.
Added open, close and switch callbacks. Only the latter is strictly
required to make sure that the streams always start at position 0, but
leaving all volumes open can eat a lot of memory for archives with many
parts.
The nnedi3 prescaler requires a normalized range to work properly,
but the original implementation did the range normalization after
the first step of the first pass. This could lead to severe quality
degradation when debanding is not enabled for NNEDI3.
Fix this issue by passing `tex_mul` into the shader code.
Fixes#2464
vo_opengl_cb is a special case, because we somehow have to render video
asynchronously, all while "trusting" the API user to do it correctly.
This didn't quite work, and a while ago a compromise using a timeout to
prevent theoretically possible deadlocks was added.
Make it even more synchronous. Basically, go all the way, and
synchronize rendering between VO and user renderer thread to the
full extent possible.
This means the silly frame queue is dropped, and we event attempt to
synchronize the GL SwapBuffer call (via mpv_opengl_cb_report_flip()).
The changes introduced with commit dc33eb56 are effectively dropped. I
don't even remember if they mattered.
In the future, we might make all VOs fetch asynchronously from a frame
queue, which would mostly remove the differences between vo_opengl and
vo_opengl_cb, but this will take a while (if it will even be done).
Pick the correct GLSL version from the GL_SHADING_LANGUAGE_VERSION
string. Might be somewhat questionable, as we expect the minor version
number not to have leading 0s.
Should help with cases when the reported GLSL version is much higher
than the equivalent of the reported GL version. This problem was
observed in combination with GL_ARB_uniform_buffer_object, which
can't be used if the declared GLSL version is too low.
Has the same function as setting the option.
This commit changes the property in a bunch of other ways. For example
if the VO is not created, it will return the option value.
Simplifies some auto detection matters.
I _still_ don't want to remove the lazy loading mechanism, because it's
still slightly useful for filters using the hwdec APIs. My main
motivation for not always preloading them is actually that libva prints
random useless crap to the terminal with no way to prevent this.
The examples demonstrates use with optical media, which is far from
mpv's main purpose.
The authors section is a leftover from MPlayer times. There are enough
other places which reiterate how mpv is based on mplayer2/MPlayer,
copyright statements, and so on.
Hint that the linked section contains information for Windows. (Well,
that's a lie, but it has a link to the Windows section.)
Avoid implying that lines in the config file end with ';'. Also, the <>
are probably just confusing.
Deal with jittering Matroska crap timestamps. This reuses the mechanism
that is needed for frames without PTS, and adds a heuristic to it. If
the interpolated timestamp is less than 1ms away from the real one, it
might be due to Matroska timestamp rounding (or other file formats with
such rounding, or files remuxed from Matroska).
While there actually isn't much of a need to do this (audio PTS
jittering by such a low amount doesn't negatively influence much), it
helps with identifying jitter from other sources.
Instead of requiring the decoder to set the PTS directly on the
dec_audio context (including handling absence of PTS etc.), transfer the
packet PTS to the decoded audio frame. Marginally simpler, and gives
more control to the generic code.
Notes:
- Unfortunately the only way to talk to EGL from within DRM I could find
involves linking with GBM (generic buffer management for Mesa.)
Because of this, I'm pretty sure it won't work with proprietary NVidia
drivers, but then again, last time I checked NVidia didn't offer
proper screen resolution for VT.
- VT switching doesn't seem to work at all. It's worth mentioning that
using vo_drm before introduction of VT switcher had an anomaly where
user could switch to another VT and input text to it, while video
played on top of that VT. However, that isn't the case with drm_egl:
I can't switch to other VT during playback like this. This makes me
think that it's either a limitation coming from my firmware or from
EGL/KMS itself rather than a bug with my code. Nonetheless, I still
left (untestable) VT switching code in place, in case it's useful to
someone else.
- The mode_id, connector_id and device_path should be configurable for
power users and people who wish to watch videos on nonprimary screen.
Unfortunately I didn't see anything that would allow OpenGL backends
to register their own set of options. At the same time, adding them to
global namespace is pointless.
- A few dozens of lines could be shared with vo_drm (setting up VT
switching, most of code behind page flipping). I don't have any strong
opinion on this.
- Sometimes I get minor visual glitches. I'm not sure if there's a race
condition of some sort, unitialized variable (doubtful), or if it's
buggy driver. (I'm using integrated Intel HD Graphics 4400 with Mesa)
- .config and .control are very minimal.
Signed-off-by: wm4 <wm4@nowhere>
While it seemed like a pretty good idea at first, it's just a dead end
and works only in the simplest cases. While it may or may not help
slightly with audio sync mode, the display-sync mode already compensates
this in a better way. The main issue is that timestamps at this layer
are not in order, so it can look at single timestamps only.
Essentially we'd use something random, just because it's part of the srt
of traditionally used ALSA channel mappings. But each driver can do its
own things.
This doesn't let me sleep at night, so remove it.
This could accidentally change some spdif formats to AAC (because AAC is
the first on the list and will match first). spdif formats are
inherently uninterchangeable, so treat them as their own class of
formats (like int vs. float).
Might fix some issues with ao_wasapi.c.
glXCreateContextAttribsARB() by design can throw some X11 errors. We
ignore these, but we generally still print error messages to the
terminal. This was confusing/annoying users, so silence it. The stupid
part is that the Xlib error handler is global, so we have to be slightly
careful here.
They are evil and should be eradicated. Some of these were pretty dumb
anyway.
There are probably some more around in platform specific code or other
code not enabled by default on Linux.
This is based on an older patch by James Ross-Gowan. It was rebased and
cleaned up. Also, the DWM API usage present in the older patch was
removed, because DWM reports nonsense rates at least on Windows 8.1
(they are rounded to integers, just like with the old GDI API - except
the GDI API had a good excuse, as it could report only integers).
Signed-off-by: wm4 <wm4@nowhere>
This simplifies update_screen_rect a bit. Unless --fs-screen=all is
used, it will always get an HMONITOR and call GetMonitorInfo to
determine its dimensions. This will make it easier for the next few
commits to determine the colour profile and the refresh rate from the
HMONITOR.
There is a slight change in behaviour. When selecting a screen that is
out of range, such as --screen=9 on a machine with only two monitors,
the old code would silently select the last existing monitor. The new
code prints an error message and falls back to the default screen (same
as the Cocoa code.)
Signed-off-by: wm4 <wm4@nowhere>
The call to EnumDisplaySettings seems to be a relic from when MPlayer
ran on systems that didn't have GetMonitorInfo or SM_CX/CYVIRTUALSCREEN.
GetMonitorInfo was loaded dynamically, so it was possible for MPlayer to
run without it and use the values returned by EnumDisplaySettings.
These are always present in modern versions of Windows, so the values
returned from EnumDisplaySettings are always overwritten. Remove the
call to EnumDisplaySettings and assume SM_CX/CYVIRTUALSCREEN is always
present.
Signed-off-by: wm4 <wm4@nowhere>
Commit 27dc834f added it as such.
Also remove the check for glUniformBlockBinding() - it's part of an
extension, and the check glGetUniformBlockIndex() already checks whether
the extension is fully available.
Implement NNEDI3, a neural network based deinterlacer.
The shader is reimplemented in GLSL and supports both 8x4 and 8x6
sampling window now. This allows the shader to be licensed
under LGPL2.1 so that it can be used in mpv.
The current implementation supports uploading the NN weights (up to
51kb with placebo setting) in two different way, via uniform buffer
object or hard coding into shader source. UBO requires OpenGL 3.1,
which only guarantee 16kb per block. But I find that 64kb seems to be
a default setting for recent card/driver (which nnedi3 is targeting),
so I think we're fine here (with default nnedi3 setting the size of
weights is 9kb). Hard-coding into shader requires OpenGL 3.3, for the
"intBitsToFloat()" built-in function. This is necessary to precisely
represent these weights in GLSL. I tried several human readable
floating point number format (with really high precision as for
single precision float), but for some reason they are not working
nicely, bad pixels (with NaN value) could be produced with some
weights set.
We could also add support to upload these weights with texture, just
for compatibility reason (etc. upscaling a still image with a low end
graphics card). But as I tested, it's rather slow even with 1D
texture (we probably had to use 2D texture due to dimension size
limitation). Since there is always better choice to do NNEDI3
upscaling for still image (vapoursynth plugin), it's not implemented
in this commit. If this turns out to be a popular demand from the
user, it should be easy to add it later.
For those who wants to optimize the performance a bit further, the
bottleneck seems to be:
1. overhead to upload and access these weights, (in particular,
the shader code will be regenerated for each frame, it's on CPU
though).
2. "dot()" performance in the main loop.
3. "exp()" performance in the main loop, there are various fast
implementation with some bit tricks (probably with the help of the
intBitsToFloat function).
The code is tested with nvidia card and driver (355.11), on Linux.
Closes#2230
Add the Super-xBR filter for image doubling, and the prescaling framework
to support it.
The shader code was ported from MPDN extensions project, with
modification to process luma only.
This commit is largely inspired by code from #2266, with
`gl_transform_trans()` authored by @haasn taken directly.
This check disables the display-sync resample method. If the filters
convert PCM to AC3, we can still insert a filter to change speed. This
is because filters are inserted at the beginning of the filter chain.
Actually, it didn't really require that before (most work was avoided),
but some bits had to be run anyway. Separate the speed change into a
light-weight function, which merely updates already created filters, and
a heavy-weight one which messes with filter insertion.
This also happens to fix the case where the filters would "forget" the
current speed (force resampling, change speed, hit a volume control to
force af_volume insertion - it will reset speed and desync).
Since we now always run the light-weight function, remove the
af_scaletempo verbose message that is printed on speed setting. Other
than that, all setters are cheap.
Move it (in a cosmetic sense), and also move its invocation to below all
the video handling.
All other changes remain cosmetic, including moving the framedrop
calculation code, and getting rid of the video_speed_correction
variable.
For some reason, the encoder didn't like that the AVPacket already had
fields set. I'm not quite sure, but this might just be invalid API
usage. Do it as it's recommended.
We still have a sample-based buffer between filters and audio outputs.
In order to avoid cutting frames into half (which can upset receivers),
we strictly need to align the boundaries on which we cut the audio.
Update msg.c state immediately if a terminal or logging setting is set.
Until now, this was delayed until mp[v]_initialize() was called. When
using the client API, you could easily miss logged error messages, even
when logging was initialized early on by calling
mpv_request_log_messages().
(Properties can't be used for this either, because properties do not
work before mpv_initialize().)
The noframe event is logged whenever there is no new frame. This can
happen due to normal redraws, but also due to video frame queue
underflow.
The mpv_opengl_cb_report_flip() API function is currently pretty
useless, because blocking on the video frame queue is more reliable and
simpler. But at least we can log the actual vsync.
next_vsync/prev_vsync was only used to retrieve the vsync duration. We
can get this in a simpler way.
This also removes the vsync duration estimation from vo_opengl_cb.c,
which is probably worthless anyway. (And once interpolation is made
display-sync only, this won't matter at all.)
We need to effectively swap the last channel pair. See commit 4e358a96
and 5a18c5ea for details.
Doing this seems rather strange, as 7.1 just extends 5.1 with 2 new
speakers, and 5.1 doesn't need this change. Going by the HDMI standard
and the Intel HDA sources (cited in the referenced commits), it also
looks like 7.1 should simply append two channels to 5.1 as well. But
swapping them is apparently correct. This is also what XBMC does. (I
didn't find any other applications doing 7.1 PCM using the ALSA channel
map API. VLC seems to ignore the 7.1 case.) Testing reveals that at
least the end result is correct.
"Normal" ALSA 7.1 is unaffected by this, as it reports a different
(and saner) channel layout.
Instead of constructing an ALSA channel map from mpv ones from scratch,
try to find the original ALSA channel map again. Th result is that we
need to convert channel maps only in one direction. If we need to map
a mp_chmap to ALSA, we fetch the device's channel map list, convert
each entry to mp_chmap, and find the first one which fits.
This seems helpful for the following commit. For now, this only gets rid
of mapping back the trivial MONO mapping, which alone would still be
acceptable, but with other channel layout mogrifications it gets messy
fast. While we need to do something awkward to keep our channel map
reordering for VAR chmaps (which basically gives nicer output and
possibly slightly better performance), this is still the better
solution.
This reverts commit 4e358a9636.
Testing shows the channel pairs must indeed be swapped (details see
commit message of the reverted commit). Making the downmix code move
sl/sr to sdl/sdr is not an appropriate solution anymore, and it's
better to fix the unusual channel layout in ao_alsa.c directly.
(Not reverting the change in chmap.c; this is still correct.)
This affects only the display-sync code path, as for normal timing the
wakeup_pts stuff handles proper wakeup. It's probably mostly a
theoretical issue.
Discontinuities (like toggling fullscreen) can cause multiple frames to
be dropped in succession, which sounds very weird. It's better to drop
some video frames instead to compensate for larger desyncs.
We roughly base it on the maximum allowed speed changes (audio change is
"additional" to the video change to account for deviations when playing
at max. video speed change).
A hw decoder might fail to decode a frame for multiple reasons, and not
always just because decoding is impossible. We can't generally
distinguish these reasons well. Make it more tolerant by accepting
failures of 3 frames, but not more. The threshold can be adjusted by the
repurposed --vd-lavc-software-fallback option.
(This behavior was suggested much earlier in some PR, but at the time
the "proper" hwdec fallback was indistinguishable from decoding error.
With the current situation, "proper" fallback is still instantious.)
ao_alsa: attempt to fix 7.1 over HDMI
The last 2 channels of 7.1 (RLC/RRC in ALSA) were exported as sdl/sdr
instead of sl/sr (I don't even know why I chose sdl/sdr, but SL/SR
and RLC/RRC are different in the ALSA API). libsw/avresample do not
move the sl/sr channels to sdl/sdr when rematrixing, so silence was
sent for 2 channels. If my selection of sdl/sdr is essentially API
abuse, there's no reason why they should do this differently.
The mess here is really that ALSa doesn't map the HDMI layouts cleanly.
Most ALSA drivers export 7.1 in a way compatible to our expectations,
but Intel HDA/HDMI does not:
mpv/ffmpeg: fl-fr-fc-lfe-bl-br-sl-sr
ALSA/generic: FL FR FC LFE RL RR SL SR [1]
ALSA/HDMI: FL FR LFE FC RL RR RLC RRC [2]
The HDMI layout is layout 0x13 (going by CEA-861-B). The comment in
the kernel code has to be correct too. The early standard defines only
1 other layout, which replaces RLC/RRC with FRC/FLC - this probably
corresponds to what we call "7.1(wide)".
So it appears when ALSA requests RLC/RRC, we should feed it sl/sr.
To make it more complicated, Kodi/xbmc apparently also have to deal with
ALSA being special, but instead of sending sl/sr to RLC/RRC, they swap
the last two pairs of the layout, and send sl/sr to RL/RR and bl/br to
RLC/RRC. Or I might have misunderstood their code. I don't have a
7.1-capable A/V receiver, so I can't test this.
For now, go with the simpler solution, and wait until someone tests it.
If the speakers end up swapped, a completely different solution will be
needed.
[1] https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/core/pcm_lib.c?id=refs/tags/v4.3#n2434
[2] https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/pci/hda/patch_hdmi.c?id=refs/tags/v4.3#n307
These calls actually can leave the ALSA configuration space empty (how
very useful), which is why snd_pcm_hw_params() can fail. An earlier
change intended to make this non-fatal, but it didn't work for this
reason.
Backup the old parameters, so we can retry with the non-empty
configuration space. (It has to be non-empty, because the previous
setters didn't fail.)
Note that the buffer settings are not very important to us. They're
a leftover from MPlayer, which needed to write enough data to the
audio device to not underrun while decoding and displaying a video
frame. In mpv, most of these things happen asynchronously, _and_
there is a dedicated thread just for feeding the audio device, so
we should be pretty imune even against extreme buffer settings. But
I suppose it's still useful to prevent PulseAudio from making the
buffer too large, so still keep this code.
Again, this could have bad access, is unlikely, and has no bad
consequences. It's noteworthy that vlc and the ALSA PCM example both do
this first, even if they set the sample rate later.
I'm worried that not restricting the access type before restricting the
format will cause problems. While it's unlikely, it might prevent
failures in some corner cases. Also, since we by default always use
interleaved access (buggy ALSA plugins), this will have no effects at
all.
If the API doesn't list padded channel maps, but the final device
channel map is padded, and if unpadded output is not possible (unlike in
the somewhat similar dmix case), then we shouldn't apply the channel
count mismatch fallback in the beginning. Do it after channel map
negotiation instead.
Doesn't matter much; effectively this prevents just log spam in some
cases where the map is legitimately padded. Normally this is really
only needed for the dmix ALSA case. (See git blame for details.)
The uninit() function was called twice if the uninit() function failed
(once by init(), once by vd_lavc.c code), which caused crashes due to
double-free. (This failure is a corner case, and all other hwdec
backends appear to handle this case gracefully.)
I do not think this code should be able to deal with uninit() being
called other than once. Guarantee that it's called exactly once.
Quoting MSDN: "Notifies the Desktop Window Manager (DWM) to opt in to or
out of Multimedia Class Schedule Service (MMCSS) scheduling while the
calling process is alive.". Whatever this means. (An application can
change the scheduling priority of the window manager?)
Does this improve anything? I have no idea. Certainly this is a program
that does multimedia and graphics, so we seem to be a good match for
this.
Is it bad if we enable this even while playback is inactive or paused? I
have no idea either.
Is there a magic cargo cult function that will mark our renderer thread
as multimedia thing? I have no idea. (We use a function to enable MMCSS
for our audio thread in ao_wasapi.)
Enable it by default, but not unconditionally. Add an "auto" mode, which
disable DwmFlush if the compositor is (probably) inactive. Let's see how
this goes.
Since I accidentally enabled DwmFlush always by default (more or less)
in a previous commit touching this code, this is probably mostly just
cargo-culting, and it's uncertain whether it does anything.
Note that I still got bad vsync behavior when fullscreening mpv, and
making another window visible on the same screen. This happens even if
forcing DWM.
update_av_diff() works on the timestamps, while time_left is in real
time. When playing at not-1 speed, these are very different, and cause
the A/V difference to jitter. Fix this by scaling the expected A/V
desync to the correct range.
Commit acd5816a broke this. It was stopping playback occasionally.
Another case where the non-display-sync interpolation mode
(in->vsync_timed==true) is causing a lot of subtle issues and will be
removed soon.
PAL8 is the only format that is RGB, has only 1 component, is byte-
aligned. It was accidentally detected by the GBRP case as planar RGB.
(It would have been ok if it were gray; what ruins it is that it's
actually paletted, and the color values do not correspond to colors (but
palette entries).
Pseudo-pal formats are ok; in fact AV_PIX_FMT_GRAY is rightfully marked
as MP_IMGFLAG_YUV_P.
This didn't show up with cases where the frame pattern has a cycle of 1
or 2 like it is the case with 24-on-24 fps, or 24-on-60 fps. It did show
up with 25-on-60 fps. (We don't slow down 25 fps video to 24 on default
settings.)
In this case, we must not add the timing error of the next frame to the
A/V difference estimation of the current frame. Use the previous timing
error instead.
This is another bug resulting from the confusion about whether we
calculate parameters for the currently playing frame, or the one we're
about to queue.
Regression since commit 93db4233. I think the bit that was forgotten
here was to remove the vo_w32_config() return value completely. The VO
failed to init because that function always returned 0. This commit
removes these bits and fixes the VO.
Fixes#2434.
Yet another relatively useless option that tries to make OpenGL's sync
behavior somewhat sane. The results are not too encouraging. With a
value of 1, vsync jitter is gone on nVidia, but there are frame drops
(less than with glfinish). With 2, I get the usual vsync jitter _and_
frame drops.
There's still some hope that it might prevent too deep queuing with some
GPUs, I guess.
The timeout for the wait call is 1 second. The value is pretty
arbitrary; it should just not be too high to freeze the process (if
the GPU is un-nice), and not too low to trigger the timeout in normal
cases, even if the GPU load is very high. So I guess 1 second is ok
as a timeout.
The idea to use fences this way to control the queue depth was stolen
from RetroArch:
https://github.com/libretro/RetroArch/blob/df01279cf318e7ec90ace039d60515296e3de908/gfx/drivers/gl.c#L1856
Commit a1315c76 broke this slightly. Frame drops got counted multiple
times, and also vo.c was actually trying to "render" the dropped frame
over and over again (normally not a problem, since frames are always
queued "tightly" in display-sync mode, but could have caused 100% CPU
usage in some rare corner cases).
Do not repeat already dropped frames, but still treat new frames with
num_vsyncs==0 as dropped frames. Also, strictly count dropped frames in
the VO. This means we don't count "soft" dropped frames anymore (frames
that are shown, but for fewer vsyncs than intended). This will be
adjusted in the next commit.
vo_frame.num_vsyncs can be != 1 in some cases in normal sync mode too.
This is not a very exact fix, but in exchange it's robust. (These
vo_frame flags are way too tricky in combination with redrawing and
such.)
There were occasional shader compilation and rendering failures if FBOs
were unavailable. This is caused by the FBO caching code getting active,
even though FBOs are unavailable (i.e. dumb-mode).
Boken by commit 97fc4f.
Fixes#2432.
Fixes linker failure. How did this ever work? Apparently it did most of
the time, but apparently we just got the first case where it didn't.
Fixes#2433.
Bump it to 80, and 2 vsyncs. This is another measure against vsync
jitter. Admittedly this is a bit simplistic (and we should probably
estimate a stable estimated vsync phase instead), but for now this will
do.
It's not needed, because the additional data is not appended, but is the
total size of the audio buffer. The maximum size is the static audio
drop size (or twice, if the audio is duplicated).
Calculate the A/V difference directly in the display sync code, instead
of the awkward current way, which reuses the fields for audio sync.
We still set time_frame, because it makes falling back to audio sync
somewhat smoother.
When dropping or repeating frames, we essentially influence when the
frame after the next frame will be shown, not the next frame. This led
to dropping/repeating frames 2 times, because the A/V difference had a
delay of one frame. Compensate it with the expected value.
av_free_packet() got finally deprecated. Use av_packet_unref() instead,
which has almost the same semantics, has existed for a while, and is
available in all FFmpeg and Libav versions we support.
This was not very reliable.
In the normal vo_opengl case, this didn't deal well enough with vsync
jitter. Vsync timings can jitter quite extremely, up to a whole vsync
duration, in which case the "missed" frame counter keeps growing, even
though nothing is wrong. This behavior also messes up the A/V difference
calculation, but as long as it's within tolerance, it won't provoke
extra frame dropping/repeating. Real misses are harder to detect, and I
might add such detection later.
In the vo_opengl_cb case, this was additionally broken due to the
asynchronity between renderer and VO threads.
This is all kinds of stupid - update_avsync_after_frame() will multiply
this value with the speed at a later point, and we only update this
field for this function. (This should be refactored.)
Until recently, the channel layout code happened to catch this, but now
an explicit check is needed. Otherwise, it'd try to pad the missing
channels with NA in the channel map fallback code.
This is intended for the case when CoreAudio returns only unknown
channel layouts, or no channel layout matches the number of channels the
CoreAudio device forces. Assume that outputting stereo or mono to the
first channels is safe, and that it's better than outputting nothing.
It's notable that XBMC/kodi falls back to a static channel layout in
this case. For some messed up reason, the layout it uses happens to
match with the channel order in ALSA's/mpv's "7.1(alsa)" layout.
Share some code between ca_init_chmap() and ca_get_active_chmap(), which
also makes it look slightly nicer. No functional changes, other than the
additional log message.
If no channel layouts were determined (which can actually happen with
some "strange" devices), the selection code was falling back to mono,
because mono is always added as a fallback. This doesn't seem quite
right.
Allow a fallback to stereo too, if no channel layout could be retrieved
at all. So we always assume that mono and stereo work, if no other
layouts are available.
(I still don't know what the CoreAudio stereo layout is supposed to do.
It could be used to swap left and right channels. It could also be used
to pad/move the channels, but I have never seen that. And it can be set
to non-stereo channels, which breaks mpv. Whatever.)
The main reason is that ao_coreaudio_exclusive needs this for some OSX
devices. They want packed audio, and special-casing this in the
coreaudio code would be too much of a pain.
The maximum of channels we can support is 64 (because FFmpeg uses 64 bit
masks for channel layouts), but since struct mp_audio can get pretty
big (has static allocations of 2 pointers for each channel for planar
mode), it's less wasteful to stay lower for now.
av_get_default_channel_layout() fails with channel counts larger than 8.
The channel layout doesn't need to make sense, so pick an arbitrary
fallback.
libswresample also has options for setting the channel counts directly,
but better not introduce new concepts in the code. Also, libavresample
doesn't have these options.
Change it so that it will always return a bitmask with the correct
number of channels set if an unknown channel map is passed. This didn't
work for channel counts larger than 8, as there are not any standard
channel layouts defined with more than 8 channels (both in mpv and
FFmpeg). Instead, it returned 0.
This will help when raising the maximum allowed channel count in mpv.
Some code in af_lavrresample relies on it, more or less.
One change is that unknown channel maps won't result in lavc standard
channel layouts anymore, just a set of random speakers. This should be
fine, as the caller of mp_chmap_to_lavc_unchecked() should handle these
cases. For mp_chmap_reorder_to_lavc() this is not so clear anymore, but
should also be ok.
For normal channel maps, simply dropping NA channels is still the
correct and wanted behavior.
Currently, the mpv maximum channel count is 8. This commit is
preparation for raising this limit.
mNumberChannelDescriptions being 0 is pretty much an error, but if it
can happen, then the code checking the chmap below will trigger UB, as
chmap is not initialized at all.
Also, simplify the code a little: we never change the number of
channels, so this is just fine.
This code removes filters which can not take spdif inout. This was made
so that PCM filters are transparently dropped in spdif mode.
This entered an endless loop with:
--af=lavcac3enc:::2 --audio-channels=5.1
The forced number of output channels is incompatible with spdif. It's
trying to insert af_lavrresample as conversion filter to compensate for
it. Of course this doesn't work, which triggers the PCM filter removal.
Then it goes on normally - since the new state is exactly as before, it
will try the same thing again, forever.
Fix by reusing the retry counter, which is a very dumb but very
effective measure against these cases of filter negotiation failure. We
could try to be more clever (for example, if the removed filter is a
conversion filter, we can be sure this won't work, and error out
immediately). But better keep it simple and robust.
This speeds up redraws considerably (improving eg. <60 Hz material on a 60 Hz
monitor with display-sync active, or redraws while paused), but slightly
slows down the worst case (eg. video FPS = display FPS).
The IME is not useful for key-bindings. Handle the base ASCII chars
instead and don't show the IME window. For the sake of libmpv users, the
IME should only be disabled on mpv's GUI thread and not application-
wide.
No IME on the GUI thread should also mean that VK_PROCESSKEY will never
have to be handled, so the logic for that can be removed as well.
Older systems have certain EGL extension definitions missing. We
redefine them to make the build system easier, and because it's trivial.
But we forgot to define the EGL_LINUX_DMA_BUF_EXT identifier. (I hope
it's the only missing one.)
This makes the bitrate properties unavailable, instead of
returning 0 when:
1. No track is selected, or
2. Not enough packets have been read to have a bitrate estimate yet
Some mkv files can have this. The chapter times are still timestamps
(and thus not affected by the start time), but it misplaces the OSD
chapter ticks.
The equalizer code as it exists in vo_opengl works perfectly fine. The
situation in vo_opengl_cb is pretty different. The playback thread can't
communicate with the renderer thread synchronously (essentially to give
the API user more flexibility). So the equalizer communication has to be
done in an asynchronous way too.
There were two problems. First, the eq capabilities can change with the
pixel format, and the renderer initializes them on config only. This
means equalizers were disabled on the first config run, and options like
--video-output-levels or --brightness would not work. So we just
initialize the caps with a known superset. The player will not correctly
indicate when setting an eq doesn't work, but we're fine with it, as it
is a relatively cosmetic issue.
Second, it copied back the eq settings in the "wrong" moment (what
for?), which overwrote the settings in some cases.
Third, the eq was not reset correctly on vo init. This is needed to make
it behave the same as vo_opengl.
If the aspect ratio of the video resolution and the subtitle resolution
(the implied subtitle coordinate system) mismatch, the subtitles
obviously can't be overlayed over the video perfectly. Either you get
video that can't be covered by subtitles, or the subtitles could go
beyond the video. We don't want to stretch the subtitle to compensate
for the aspect ratio, because it would look terrible.
Until now, mpv used to fit the subtitle rectangle into the video
rectangle (letterboxing/pillarboxing). This looks odd with some sample
files with subtitle canvas being wider than the video. Also, mpc-hc
displays them in a better way. vlc stretches them, which looks bad.
While you probably can't win this game with all those broken files
around, pick the mpc-hc method to handle this.
There was a complaint that the naming is inaccurate. That's probably
right. Just use the official name instead, which is a bit clunky, but
surely correct.
Helps with broken vobsubs, which have an incorrect resolution header
set.
So we just extend the subtitle resolution to the video size, if the
video size is larger. This helps somewhat with readability, or makes
them visible at all. It should be a pretty safe change, because normally
no sub pictures are supposed to go outside of the area. It should make a
difference with broken files only.
The sample in question had a video resolution of 1888x1072, and a
subtitle resolution of 720x480. Note that always using video resolution
as subtitle resolution would break other files.
Coreaudio gives us a channel map with all entries set to
kAudioChannelLabel_Unknown. This is translated to a mpv channel map with
all channels set to NA, which has special meaning: it's an "unknown"
channel map, which acts as wildcard and can be converted from/to any
channel layout. Not really what we want.
I've got this with USB audio, playing stereo. The multichannel layout
consisted of 2 unknown channels, while the stereo channel map was
stereo (as expected).
Note that channel maps with _some_ NA entries are not affected by this,
and must still work.
If the device returns an unexpected number of channels instead of the
requested count on init, don't immediately error out. Instead, look if
there's a channel map with the given number of channels.
If there isn't, still error out, because we don't want to guess the
channel layout.
It's great that the new algorithm supports multiple placebo iterations
and all, but it's really not necessary and hurts performance in the
general case for the sake of the 0.1% that actually pause the screen
and look for minute differences.
Signed-off-by: wm4 <wm4@nowhere>
Reportedly fixes operation with "USB connected Parasound ZDAC v.2". (OSX
and USB audio sure is not nice at all.)
This might be perceived as hang by some users, so it's quite possible
that this will have to be adjusted again somehow.
Fixes#2409.
This reverts commit c10fb4ce9f.
This is already done in vo_wayland.c:resize,324 doing it here makes the window bigger before the video resizes showing a black area while dragging the border.
Apparently this function caused weird problems to me. I have no idea
why. The usage of the function looks perfectly fine to me, and even
rounding issues can be excluded. In any case, getting rid of this solved
my problem, and makes the code actually more readable.
The previous commit moved the av_frame_unref() after the got_picture
check. This accidentally also deferred the software fallback
reinitialization to until a software picture was decoded (instead of the
exact time of the fallback), which is not ideal.
Just rely on the fact that calling av_frame_unref() on a frame is ok
even if nothing was decoded.
Commit 12cd48a8 started setting the hwdec_failed field even if hwdec was
not active, and because it also checked this field even if hwdec was not
active, broke decoding forever.
Fix this, and also avoid a memory leak or API misuse by releasing the
decoded picture. Passing an unreleased frame to the decoder has as far
as I know no defined effects.
This fixes initial decoding of some samples. See #1341.
According to Libav devs, this should be considered a libavcodec bug, but
as it's hard to fix, here we go.
Adds support for AV_PIX_FMT_GBRP9, AV_PIX_FMT_GBRP10, AV_PIX_FMT_GBRP12,
AV_PIX_FMT_GBRP14, AV_PIX_FMT_GBRP16, AV_PIX_FMT_GBRAP, and
AV_PIX_FMT_GBRAP16.
(Not that it matters, because nobody uses these anyway.)
This VA_FOURCC isn't even defined by latest drivers, so I'm just
assuming it doesn't exist and never existed. For planar 4:2:0,
VA_FOURCC_YV12 is normally preferred, and there's even a VA_FOURCC_IYUV
for 4:2:0 with unswapped planes.
0.34 and 0.35 don't have the buffer API, such as vaAcquireBufferHandle.
This is only needed for the EGL interop, but why bother staying
compatible for such old things (0.36 was released over a year ago).
We also can drop some minor compatibility ifdeffery.
MKV files can very well start with timestamps other than 0. While mpv
has support for such files in general, and demux_lavf enables this
feature, demux_mkv didn't export a start time.
Implement this by simply reading the first cluster timestamp. This in
turn is done by reading 1 block. While we don't need the block for this
prupose at all, it's the easiest way to get the cluster timestamp read
correctly without code duplication. In theory this could be wrong, and
a packet could start at a much later time, but in practice this won't
happen.
This commit also adds an option to disable this feature. It's not
documented because nobody should use it. (But I happen to have a need
for this.)
Let's hope this doesn't confuse client API users too much. It's still
the best solution to get rid of corner cases where it actually return
the wrong timestamp on start, and then suddenly jump.
matplotlib is pathetically slow, which makes using stats-conv.py to view
dumps longer than a few seconds a huge pain.
pyqtgraph is slightly faster than matplotlib. Other than that, it seems
to be worse in every aspect (at least for plotting), but such is life.
When seeking, the current frame will have the wrong timestamp, until the
seek is done and a new frame from the seek target is shown. We still use
the "old" frame for redrawing during seeks. This frame has to be
explicitly marked with still=true in order not to confuse the
interpolation queue. If this is not done, it will ignore frames until a
frame with approximately the same timestamp as the "old" frame is
reached. (Does this mean interpolation handles timestamp resets
incorrectly? I have no idea.)
Of course we also have to clear possibly queued frames on seeks.
Also, in pausing, explicitly let the frame redraw.
This adjustment is supposed to improve the audio speed calculation in
case of unexpected desync. The flipped sign made it actually worse,
although the total impact of this bug was very minor.
Small adjustments to the playback speed use swr_set_compensation()
to stretch the audio as it is required. But since large adjustments
are now handled by actually reinitializing libswresample, the small
adjustments get rounded off completely with typical frame sizes.
Compensate for this by accounting for the rounding error and keeping
track of fractional samples that should have been output to achieve
the correct ratio.
This fixes display sync mode behavior, which requires these adjustments
to be relatively accurate.
Newer nVidia drivers support EGL, but they seem to work badly,
apparently don't support some needed features or not in a form we want
(such as swap control), and vdpau interop is not available. Disable it
by default, because I'm tired of explaining this issue.
Can be reverted as soon as nVidia release working drivers.
The libavcodec h264 decoder contains some idiotic code with unknown
purpose (no sample or explanation known that necessitates its
existence), that causes the AVCodecContext.get_format callback to be
invoked at a time when hwaccels can't be initialized. By definition, the
get_format callback is supposed to initialize hwaccels (another idiotic
thing now part of the API, but different story). This causes hwdec
initialization sometimes to fail (WolfensteinTwitch.mp4): the first
get_format callback will mark it as failed, so the second get_format
(the "proper" normal one) will not bother restoring the state, and hwdec
init fails.
While this should be fixed in libavcodec (good luck with that), it's
quite easy to workaround.
This affects the subtitle preroll mode during seeking. It could matter
somewhat with insane files with ten-thousands of subtitle events, which
now seem to pop up, and will avoid packet queue overflow.
swr/avresample_set_compensation() was made for small speed adjustments.
Non-documentation says it should be used for changes not larger than 1%,
so reinitialize the sampler if the change is larger than that.
This fixes a regression since commit f4d62da8. The original code run
vo_cocoa_config_window() once without creating the window, which had the
effect that the last part of the function was run at least once before
the actual window was created. Fix the regression by moving it to before
the window is created.
The regression itself is hard to describe. One test case: start mpv from
a fullscreened terminal window. It should switch to another desktop,
with the mpv window visible. This didn't happen anymore.
mpv_opengl_cb_uninit_gl() still needs the OpenGL context. It makes calls
to free OpenGL objects. Strictly speaking, this is probably unnecessary,
because the OpenGL context is destroyed afterwards (implicitly freeing
all related objects). But mpv_opengl_cb_uninit_gl() does not require the
destruction of the OpenGL context, and thus has to free resources
manually.
It's also true that OpenGL normally simply ignores API calls (or returns
errors) if no context is set, but doing it properly is cleaner.
That makeCurrent() can be called in the destructor is explicitly allowed
and recommended for freeing GL resources in the Qt docs.
This fixes a mpv error message on exit.
Thanks to rcombs, ffmpeg now properly supports DASH and we can
remove our hacks for it and use it by default whenever
available. If you don't like this for whatever reason, you
can get the "normal" streams back with --ytdl-format=best .
Closes#579Closes#1321Closes#2359
If video EOF happens during playback restart, and audio is syncing, and
the demuxer packet queue overflows (i.e. no new packets will be read),
then it could happen that the player accidentally enters sleeping, and
continues playing anything only after e.g. user input wakes it up.
libass 0.13.0 breaks this due to removal of fontconfig from its core
(instead, fontconfig is one possible backend, and pattern lookup is
apparently not possible anymore).
FFmpeg supports all formats the old subreader code does, and is better
at it. On the other hand, subreader.c's probing is bad and can lead to
false positives easily.
swr_set_compensation() changes the apparent sample rate on the fly (who
would have guessed). It is thus very well-suited for adjusting audio
speed on the fly during playback (like needed by the display-sync mode).
It skips the relatively slow resampler reinitialization.
If this doesn't work (libswresample soxr backend), then fall back to the
old method.
The previous commit handled not falling back to normal decoding if the
AO was reloaded (I think...), and this tries to re-engage spdif pass-
through if it was previously falling back to normal decoding (e.g.
because it temporarily switched to an audio device incapable of
passthrough).
The stop command didn't always stop. In this case, opening a HLS URL and
then sending "stop" during loading would actually make it fallback to
parsing it as a playlist, and then continued to play the playlist items.
(This corner case makes several unfortunate factors come together to
produce this really odd behavior.)
Another issue is that the "stop" was not always explicitly set. This
could be a problem when sending several commands at once. Only the
"quit" command should have priority over the "stop" command, so this is
still checked.
The Copyright file explains the whole license mess. The earlier change
was apparently confusing, because the link reading "details" merely
linked to the GPLv2 license instead of explaining anything. In fact, I
meant to link to the Copyright file in the first place.
Use the first encountered packet PTS/DTS as base, instead of the last
one. This does not add the amount of frames buffered in the codec to the
PTS offset, and thus is better.
Also, don't add the frame time if there was no decoded frame yet. The
first frame should obviously have the timestamp of the first packet
(going by this heuristic).
While b-frame reordering limits the maximum required number to around
16, the number of additionally buffered frames can be much higher.
Guess when this actually matters? (For the libavcodec MMAL wrapper.)
Useless. Sometimes it might be useful to make some extremely broken
files work, but on the other hand --no-correct-pts is sufficient for
these cases.
While we still need some of the code for AVI, the "auto" mode in
particular inflated the size of the code.
This can't be handled correctly at all. Other cases when the decoder
might drop a frame (such as completely failing to decode a frame) will
shift timestamps by a frame, and it can't be avoided.
While we could maybe find a better way to handle this with libavcodec's
main decoders, this seems to be much harder if it should work with
certain HW decoders, which don't passthrough the DTS field (such as
MMAL). Another problem are .avi files with b-frames. So just leave it
as it is.
The manpage entry explains this.
(Maybe this option could be always enabled and removed. I don't quite
remember what valid use-cases there are for just disabling audio
entirely, other than that this is also needed for audio decoder init
failure.)
Make the code a bit more uniform. Always build a "dummy" audio output
list before probing, which means that opening preferred devices and
pure auto-probing is done with the same code. We can drop the second
ao_init() call.
This also makes the next commit easier, which wants to selectively
fallback to ao_null. This could have been implemented by passing a
different requested audio output list (instead of reading it from
MPOptions), but I think it's better if this rather special feature
is handled internally in the AO code. This also makes sure the AO
code can handle its own options (such as the audio output list) in
a self-contained way.
This was used only by the timestamp sorting code, which is a fallback
for avi files (as well as avi-muxed mkv files). This was supposed to
prevent accumulating timestamps in case the decoder consumes more
packets than it outputs frames (i.e. frames are dropped). This didn't
work very well (timestamps could be off by a large amount), the
estimation of the delay was fragile, and the interdependencies with the
decoder were annoying, so kill it.
This parameter has been unused for years (the last flag was removed in
commit d658b115). Get rid of it.
This affects the general VO API, as well as the vo_opengl backend API,
so it touches a lot of files.
The VOFLAGs are still used to control OpenGL context creation, so move
them to the OpenGL backend code.
Make handling of metadata slightly more generic, and add reading of the
"PERFORMER" fields. There are some more fields, but for now let's leave
it at this.
TRACK-specific PERFORMER fields have to be read from the per-chapter
metadata (somewhat obscure).
Fixes#2328.
This gets rid of an old hack, VOFLAG_HIDDEN. Although handling of it has
been sane for a while, it used to cause much pain, and is still
unintuitive and weird even today.
The main reason for this hack is that OpenGL selects a X11 Visual for
you, and you're supposed to use this Visual when creating the X window
for the OpenGL context. Which means the X window can't be created early
in the common X11 init code, but the OpenGL code needs to do something
before that. API-wise you need separate functions for X11 init and X11
window creation. The VOFLAG_HIDDEN hack conflated window creation and
the entrypoint for resizing on video resolution change into one
function, vo_x11_config_vo_window(). This required all platform backends
to handle this flag, even if they didn't need this mechanism.
Wayland still uses this for minor reasons (alpha support?), so the
wayland backend must be changed before the flag can be entirely removed.
If interpolation is enabled, then this causes heavy artifacts if done
while unpaused. It's preferable to allow a latency of a few frames for
the change to take full effect instead. If this is done paused, the
frame is fully redrawn anyway.
This reverts commit d11184a256.
Unfortunately, there was a lot of unexpected resistance.
Do note that this is still extremely slow, crappy, etc.
Note that vo_x11.c was further edited. Compared to the removed vo_x11.c,
an additional ~200 lines of code was removed in order to simplify it. I
tried to strip it down as much as possible. In particular, support for
odd non-32 bit formats (24, 16, 15, 8 bit) is dropped.
Closes#2300.
Don't print the URL that is opened twice. stream.c and stream_lavf.c
each printed it once. Remove the logging from stream_lavf.c, and move
the log call to a more interesting point.
The vf_format suboption is replaced with --video-output-levels (a global
option and property). In particular, the parameter is removed from
mp_image_params. The mechanism is moved to the "video equalizer", which
also handles common video output customization like brightness and
contrast controls.
The new code is slightly cleaner, and the top-level option is slightly
more user-friendly than as vf_format sub-option.
This AVPacket field was a hack against the fact that the duration field
was merely an int (too small for things like subtitle durations). Newer
libavcodec drops this field and makes duration 64 bit.
This essentially reverts commit 009dfbe3. FFmpeg VideoToolbox support
is being wacky, and can cause major issues, such as not being able
to decode a single frame. (E.g. by playing a .ts file. This should be
fixed in FFmpeg eventually.)
This is not a straight revert of the commit; just a functional one. We
keep the slightly simpler code structure.
It doesn't deal with VDA at all anymore. Rename it to hwdec_osx.c. Not
using hwdec_videotoolbox.c, because that would give it the longest
source path in this project yet. (Also, this code isn't even
VideoToolox-specific, other than the name of the pixel format used.)
VideoToolbox is preferred. Now that FFmpeg released 2.8, there's no
reason to support VDA anymore. In fact, we had a bug that made VDA not
useable with older FFmpeg versions in some newer mpv releases.
VideoToolbox is supported even on slightly older OSX versions, and if
not, you still can run mpv without hw decoding.
Definitely not needed anymore, and fixes a crash in some weird corner-
cases.
The extradata freeing is apparently still needed, though. (Because a
codec context can be opened again, which makes no sense, but ok.)
This can happen with USB audio. There was already code for this, but
something in mpv and ALSA changed - and now the old code is not
necessarily triggered anymore. It probably depends on the exact
situation.
This could sometimes cause crashes in hotplug events. (Apparently in
cases when CoreAudio changes its state asynchronously, or such.)
CA_GET_STR() does not set the string if there was an error, so errors
have to be strictly checked before using it.
There are at least 2 ways of using VAAPI without X11 (Wayland, DRM).
Remove the X11 requirement from the decoder part and the EGL interop.
This will be used by a following commit, which adds Wayland support.
The worst about this is the decoder part, which includes a bad hack for
using the decoder without any VO interop (also known as "vaapi-copy"
mode). Separate the X11 parts so that they're self-contained. For the
EGL interop code we do something similar (it's kept slightly simpler,
because it essentially only has to translate between our silly
MPGetNativeDisplay abstraction and the vaGetDisplay...() call).
It looks like my hope that we can unconditionally include EGL headers in
the OpenGL code is not coming true, because OSX does not support EGL at
all. So I prefer loading the VAAPI EGL/GL specific extensions manually,
because it's less of a mess. Partially reverts commit d47dff3f.
While EGL 1.4 seemed a bit ambiguous about this to me, it actually says
quite clearly that core functions are not supported with
eglGetProcAddress() in the following paragraph.
Normally, we prefer GLX on X11. But for the VAAPI EGL interop, we
obviously want EGL. Since nvidia does not provide EGL with desktop GL
yet, we can leave it to the autoprobing. Just make sure some failure
messages don't unnecessarily show up in the nvidia case.
This breaks VAAPI GLX interop by default, but I don't care much. If
you use --hwdec=auto (which you should if you want hw decoding), this
should fallback to vaapi-copy instead.
Probe the surface format, and check whether it's really something we
support. This also does a complete check whether the EGL interop works
at all (the only way to find this out is actually running this code).
Also, support YV12. Under some circumstances, vaapi (with Intel
drivers) can be made to use this format.
Unfortunately, the Intel drivers show some very weird behavior, which
is hopefully a bug. insane_hack() provides a very evil workaround (see
comments). A proper solution might be passing the hw format as part of
mp_image_params, but as long as hw surfaces appear to be able to change
the format on the fly, attempting this is probably not worth the extra
complexity and likely fragility. The hack allows us to pretend that
there is sane behavior for now.
This makes it much faster if the surface is really mapped from GPU
memory. It's slightly slower than system memcpy if used on system
memory. We don't really know definitely in which type of memory
it's located, so we use the GPU memcpy in all cases.
Fixes#2317.
Make the GPU memcpy from the dxva2 code generally useful to other parts
of the player.
We need to check at configure time whether SSE intrinsics work at all.
(At least in this form, they won't work on clang, for example. It also
won't work on non-x86.)
Introduce a mp_image_copy_gpu(), and make the dxva2 code use it. Do some
awkward stuff to share the existing code used by mp_image_copy(). I'm
hoping that FFmpeg will sooner or later provide a function like this, so
we can remove most of this again. (There is a patch, bit it's stuck in
limbo since forever.)
All this is used by the following commit.
Broken by commit d47dff3f. If something is going to include EGL.h,
header_fixes.h has to know. This definitely affected vo_rpi, and
probably affects wayland builds (with x11egl didabled) as well.
Checking and resetting the VAImage.buf field is non-sense, even if it
happened to work out in the normal case. buf is actually freed when
vaDestroyImage() is called (not quite intuitive), and we need an extra
field to know whether vaReleaseBufferHandle() has to be called.
Printing "Using vaDeriveImage()" every frame is too verbose, so raise
the log level.
mp_image strides are in int and not unsigned int; fix this. It's not
like it actually matters, though.
Finish a comment.
Add the upstream symbolic names as comments. Normally, these should be
defined in libdrm's drm_fourcc.h header. But DRM_FORMAT_R8 and
DRM_FORMAT_GR88 are not defined anywhere, except in the kernel userland
headers of Linux 4.3 (!). We don't want mpv to depend on bleeding-edge
Linux kernel headers, so this will have to do.
Also, just for completeness, add fourccs for the 3 and 4 channel
formats. I didn't manage to test them, though.
Reduces the amount of hardcoded assumptions about the layout
drastically. (Now adding yuv420 support would be just adjusting an if,
if you ignore the other problems, such as determining the hw format at
all early enough.)
Don't call eglDestroyImageKHR() on the same ID possibly more than once.
Clear the image reference on termination, or we would leak up to 1 image
per VO recreation.
Should work much better than the old GLX interop code. Requires Mesa 11,
and explicitly selecting the X11 EGL backend with:
--vo=opengl:backend=x11egl
Should it turn out that the new interop works well, we will try to
autodetect EGL by default.
This code still uses some bad assumptions, like expecting surfaces to be
in NV12. (This is probably ok, because virtually all HW will use this
format. But we should at least check this on init or so, instead of
failing to render an image if our assumption doesn't hold up.)
This repo was a lot of help: https://github.com/gbeauchesne/ffvademo
The kodi code was also helpful (the magic FourCC it uses for
EGL_LINUX_DRM_FOURCC_EXT are nowhere documented, and
EGL_IMAGE_INTERNAL_FORMAT_EXT as used in ffvademo does
not actually exist).
(This is the 3rd VAAPI GL interop that was implemented in this player.)
The VAAPI EGL interop code will need access to the X11 Display. While
GLX could return it from the current GLX context, EGL has no such
mechanism. (At least no standard one supported by all implementations.)
So mpv makes up such a mechanism.
For internal purposes, this is very rather awkward solution, but it's
needed for libmpv anyway.
These extensions use a bunch of EGL types, so we need to include the EGL
headers in common.h to use our GL function loader with this.
In the future, we should probably require presence of the EGL headers to
reduce the hacks. This might be not so simple at least with OSX, so for
now this has to do.
Surfaces used by hardware decoding formats can be mapped exactly like a
specific software pixel format, e.g. RGBA or NV12. p->image_params is
supposed to be set to this format, but it wasn't.
(How did this ever work?)
Also, setting params->imgfmt in the hwdec interop drivers is pointless
and redundant. (Change them to asserts, because why not.)
This is a pseudo-OpenGL extension for letting libmpv query native
windowing system handles from the API user. (It uses the OpenGL
extension mechanism because I'm lazy. In theory it would be nicer to let
the user pass them with mpv_opengl_cb_init_gl(), but this would require
a more intrusive API change to extend its argument list.)
The naming of the extension and associated function was unnecessarily
Windows specific (using "D3D"), even though it would work just fine for
other platforms. So deprecate the old names and introduce new ones. The
old ones still work.
This turns the old scalers (inherited from MPlayer) into a pre-
processing step (after color conversion and before scaling). The code
for the "sharpen5" scaler is reused for this.
The main reason MPlayer implemented this as scalers was perhaps because
FBOs were too expensive, and making it a scaler allowed to implement
this in 1 pass. But unsharp masking is not really a scaler, and I would
guess the result is more like combining bilinear scaling and unsharp
masking.
Usually, libavcodec ignores errors reported by the hardware decoding
API, so it's not like we can actually escape if the hardware is somehow
acting up.
For normal fallback purposes, or if parts of the hw decoding API which
we actually check fails, we do this by setting and checking the
hwdec_failed flag anyway.
This can happen if the hw decoder allocates padded surfaces (e.g.
mod16), but the VPP output surface was allocated with the exact size.
Apparently VPP requires matching input and output sizes, or it will add
artifacts. In this case, it added mirrored pixels to the bottom few
pixels.
Note that the previous commit should have fixed this. But it didn't
work, while this commit does.
Fixes#2320.
If not set, VPP will use the whole surface. This is a problem if the
surfaces are padded, and especially if the surfaces are padded by
different amounts.
This is an attempt to fix#2320, but it appears to do nothing at all.
The comment was largely outdated, and described the old situation when
we used a "violent" fallback by making get_buffer2 fail completely.
Also, for the case when the hw decoder initialization succeeded (in
get_format), but get_buffer2 for some reason requests something
unexpected, we also can fallback more gracefully and in the same way.
Window classes are process-wide (or at least DLL-wide), so you can't
have 2 classes with the same name. Our code attempted to do this when
for example 2 libmpv instances were created within the same process.
This failed, because RegisterWindowEx() fails if the class already
exists.
Fix this by ignoring RegisterWindowEx() errors. If the class can really
not be registered, we will fail on CreateWindowEx() instead. Of course
we also can't unregister the class, as another thread might be using it.
Windows will free the class automatically if the DLL is unloaded or the
process terminates.
Fixes#2319 (hopefully).
It's possible to build vf_vapoursynth with either the Python or Lua
backend (or both or none). The check for the vapoursynth core itself was
hidden away and couldn't be disabled, which would link mpv with
vapoursynth even if all backends were disabled.
Rearrange the checks so that the core will be disabled if no backend is
found (or both are disabled). This duplicates the check for
vapoursynth.pc, but since it's trivial, this is not that bad.
Caused by one of the --force-window commits. The unconditional
uninit_video_out() call (which normally should be idempotent) raised
sporadic MPV_EVENT_VIDEO_RECONFIG events. This is ok, except for the
fact that clients (like a Lua script or libmpv users) would cause the
event loop to run again after receiving it, triggering a feedback loop.
Fix it by sending the events really only on a change.
Sigh... After the recent changes, another regression appeared. This
time, the VO window wasn't cleared when changing from video to a non-
video file (such as audio-only with no cover art). Fix this by properly
taking the handle_force_window() bool parameter into account.
Also, the info message could be printed twice, which is harmless but
ugly. So just remove the message.
Also, do some more minor cleanups (like fixing the comment, which was
completely outdated).
If --force-window wasn't used, this would destroy the VO while a file
is still being loaded, resulting in flicker and other interruptions
when switching from one playlist entry to another. Recent regression.
The condition used here is pretty tricky, but it boils down to that it
should trigger either in idle mode, or when loading has been fully done
(at these points we definitely know whether the VO will be needed).
This was in sub/, because the code used to be specific to subtitles. It
was extended to automatically load external audio files too, and moving
the file and renaming it was long overdue.
The previous commit was incomplete (and I didn't notice due to a broken
test procedure).
The annoying part is that actually creating the VO was separate; redo
this and merge the code for this into handle_force_window() as well.
This will also make implementing proper reaction to runtime option
changes easier. (Only the part for actually listening to option changes
is missing.)
This is a bad hack; the correct way to handle this would be implementing
profiles differently, and then listen to option changes and act on them
dynamically.
This flag was used by some filters and made sure none of these filters
were inserted twice. This triggers only if the user explicitly tries to
add multiple filters (and not e.g. due to auto-insertion), so at best
this warned the user from doing something potentially pointless. At
worst, it blocked some (mildly) legitimate use-cases. Get rid of it.
Also see #2322.
I took this out because I thought the filter chain would auto-negotiate
using nv12 without the explicit hint, and it does in the basic case
with no intermediate filter, but once you start adding filters, it
can end up negotiating a different format and then failing.
Today, vdpaurb will fail if it's used with non-hardware-decoded
content. This created work for the user as they have to explicitly
add or not add it, depending on the content.
As an improvement, we can make vdpaurb pass through any frames
that aren't hardware decoded, so that it can always be present in the
filter chain, if desired.
I see no point in keeping these around. Keeping wrappers for some select
libavfilter filters just because MPlayer had these filters is not a good
reason.
Ultimately, all real filtering work should go to libavfilter, and users
should get used to using vf_lavfi directly. We might even not require
the awful double-nested syntax for using libavfilter one day.
vf_rotate, vf_yadif, vf_stereo3d are kept because mpv uses them
internally. (They all extend the lavfi filters or change their
defaults.) vf_mirror is kept for symmetry with vf_flip. vf_gradfun and
vf_pullup are probably semi-popular, so I'll remove them not yet - only
after some more discussion.
The reason MPlayer traditionally duplicated them all over the place is
that it wanted every component to be a self-contained library (e.g.
audio filters were in "libaf"). But this is not necessarily helpful, and
this change makes the following commit a bit simpler.
The "PWD" enviornment variable is described by POSIX. We don't go to
length to verify its contents, but just trust it.
This affects the logic for resuming playback.
* (de)planarize -1
* pad 1 byte -8
* truncate 1 byte -1024
* float -> int 1048576 * (8 - dst_bytes)
* int -> float -512
Now the score is negative if and only if the conversion is lossy
(e.g. previously s24 -> float was given a negative (lossy) score),
However, int->float is still considered bad
(s16->float is worse than than s16->s32).
This penalizes any loss of precision more than performance / bandwidth hits.
For example, previously s24->s16p was considered equal to s24->u8.
Finally, we penalize padding more than (de)planarizing as this will
increase the output size for example with ao_lavc.
This is just a refactor, which makes it use the previously introduced
function, and allows us to make af_format_conversion_score() private.
(We drop 2 unlikely warning messages too... who cares.)
This mixed up the returned score for some interleaved/non-interleaved
comparisons. Changing interleaving subtracted 1 point, while extending
sample size by 1 byte also subtracted 1 point.
(This scoring system is not ideal - it'd be much cleaner to do a 3-way
sample format comparison instead, and sort the formats according to the
comparison instead of the score.)
These were normalized and are saner now. We want to use the new fields,
and also get rid of the deprecation warnings, so use them. There's no
release yet which uses these, so some ifdeffery is unfortunately needed.
2 things are being stupid here: Apple for requiring rectangle textures
with their IOSurface interop for no reason, and OpenGL having a
different sampler type for rectangle textures.
The value 0 was treated specially, and effectively forced the increment
to 1. Interestingly, passing 0 or no value also does not include the
scale (from touchpads etc.), but this is probably an accidental behavior
that was never intentionally added.
Simplify it and make the default increment 1. 0 now means what it
should: the value will not be changed. This is not particularly useful,
but on the other hand there is no need for surprising and unintuitive
semantics.
OARG_CYCLEDIR() failed to apply the default value, because
m_option_type_cycle_dir was missing a copy handler - add this too.
This causes weirdness with the "cache-size" property and option. Only
the read handler of the property included the backbuffer, while all
others did not. Make it consistent, and subtract the backbuffer size
from the cache size.
Fixes#2305.
Make it so that they don't need to be edited on release (no change in
tense). Also, move the "stray" changes after client API version 1.19 to
1.19 itself.
The removal of source-shader is a side effect, since this effectively
replaces it - and the video-reading code has been significantly
restructured to make more sense and be more readable.
This means users no longer have to constantly download and maintain a
separate deband.glsl installation alongside mpv, which was the only real
use case for source-shader that we found either way.
This is mostly to cut down somewhat on the amount of code bloat in
video.c by moving out helper functions (including scaler kernels and
color management routines) to a separate file.
It would certainly be possible to move out more functions (eg. dithering
or CMS code) with some extra effort/refactoring, but this is a start.
Signed-off-by: wm4 <wm4@nowhere>
Instead of the other way around of disabling disallowed options. This is
more robust and also slightly simpler, at least conceptually. If new
vo_opengl features are added, they don't need to be explicitly disabled
for dumb-mode just to avoid that it accidentally breaks.
Sigh...
Hopefully this code will be completely unnecessary one day, as it's
only needed due to the sub-option parser craziness.
Move dumb_mode to the top of the struct, so the C universal initializer
doesn't cause warnings with all those broken compilers.
Not sure why struct af_resample_opts even exists. It seems useful to
group the fields set by user options. But storing the current format
conversion parameters doesn't seem very elegant, and having a separate
instance in the "ctx" field isn't helpful either.
The single path optimization, rendering the video in one shader pass and
without FBO indirections, was removed soem commits ago. It didn't have a
place in this code, and caused considerable complexity and maintenance
issues.
On the other hand, it still has some worth, such as for use with
extremely crappy hardware (GLES only or OpenGL 2.1 without FBO
extension). Ideally, these use cases would be handled by a separate VO
(say, vo_gles). While cleaner, this would still cause code duplication
and other complexity.
The third option is making the single-pass optimization a completely
separate code path, with most vo_opengl features disabled. While this
does duplicate some functionality (such as "unpacking" the video data
from textures), it's also relatively unintrusive, and the high quality
code path doesn't need to take it into account at all. On another
positive node, this "dumb-mode" could be forced in other cases where
OpenGL 2.1 is not enough, and where we don't want to care about versions
this old.
This change makes vo_opengl slightly less compatible (ancient devices
without FBOs will no longer work) and decreases performance in the
simplest case (vo=opengl), in exchange for significantly reducing code
complexity and making everything easier to reason about.
Until now, most OSD objects created the associated ASS_Renderer instance
as soon as possible, even if nothing was going to be rendered. Maybe
this was even intentional.
Change this for the sake of lowering resource usage, and strictly
initialize ASS_Renderer only when it's really needed.
For the OSC, initialization has to be forced, because of the insane
mechanism for translating mouse coordinates to OSD coordinates.
This was completely broken. It was checked manually in some config
loading paths, so it appeared to work. But the intention was always to
completely disable reading from the normal config dir. This logic was
broken in commit 2263f37d.
The manual checks are actually redundant, and are not needed if
--no-config is implemented properly - remove them.
Additionally, the change to load the libmpv defaults from an embedded
profile also failed to set "config=no". The option is marked as not
being settable by a config file, and the libmpv default profile is
parsed as a config file, so this option was rejected. Fix it by removing
the CONF_NOCFG flag. (Alternatively, m_config_set_profile() could be
changed not to set the "config file" flag by default, but I'm not
bothering with this.)
This didn't seem entirely sane. It probably worked by accident, because
"position" is always the first attribute, and thus the default value 0
for the location was always correct.
Some users still use this filter, so the filter was going to be kept.
But I overlooked that libavfilter provides this filter. Remove the
redundant wrapper from mpv. Something like --af=lavfi=bs2b should work
and give exactly the same results.
All of these filters are considered not useful anymore by us. Some have
replacements in libavfilter (useable through af_lavfi).
af_center, af_extrastereo, af_karaoke, af_sinesuppress, af_sub,
af_surround, af_sweep: pretty simple and useless filters which probably
nobody ever wants.
af_ladspa: has a replacement in libavfilter.
af_hrtf: the algorithm doesn't work properly on most sources, and the
implementation was buggy and complicated. (The filter was inherited from
MPlayer; but even in mpv times we had to apply fixes that fixed major
issues with added noise.) There is a ladspa filter if you still want to
use it.
af_export: I'm not even sure what this is supposed to do. Possibly it
was meant for GUIs rendering audio visualizations, but it couldn't
really work well. For example, the size of the audio depended on the
samplerate (fixed number of samples only), and it couldn't retrieve the
complete audio, only fragments. If this is really needed for GUIs, mpv
should add native visualization, or a proper API for it.
While unknown lengths are supported in some important cases like
segments and clusters, they are not for small and complex metadata
elements like the track list. Such elements are simply rejected.
This case was caught by the size sanity check below, but the message is
misleading and wrong.
(There are likely no files in the wild which require support for this.
The sample file I've seen was muxed by libavformat, but in a case where
it aborted when writing the header. Clearly a broken file.)
Often, we don't know whether hardware decoding will work until we've
tried. (This used to be different, but API changes and improvements in
libavcodec led to this situation.) We will often output that we're going
to use hardware decoding, and then print a fallback warning.
Instead, print the status once we have decoded a frame.
Some of the old messages are turned into verbose messages, which should
be helpful for debugging. Also add some new ones.
The fallback at initialization time was basically duplicated, maybe for
the sake of showing a different error message. This doesn't matter
anymore; not much can fail at initialization anymore. Most meaningful
and common errors happen either at probing or in get_format (when the
actual hw decoder is initialized).
If PBO upload fails, disable PBOs and revert to the normal codepath. In
theory we should retry PBO upload on failure (because OpenGL specifies
that it can sporadically fail), but since it normally doesn't happen,
and the fallback will work, I'm not bothering.
Some restructuring is needed, since glUnmapBuffer needs to be called
earlier. In fact, the old code structure didn't make too much sense, and
is a leftover from MPlayer's direct rendering support, which let the
decoder decode to a PBO-mapped region. This means the buffer_ptr field
can be dropped. Drop buffer_size as well, since it only had 2 possible
values (0 or the size required for the current config).
Can significantly help with very large video resolutions on nvidia
drivers. It doesn't seem to have negative effects on Intel drivers
either. (Although it could have on Intel drivers for older hardware.)
For now, this is only for --vo=opengl-hq. Maybe --vo=opengl should use
it too, but it's still meant to be the crappy, fail-safe default.
Setup a dummy image for the given image params, and get the plane sizes
from that. Admittedly not much of a simplification, but conceptually
it's simpler and less error-prone, as the image layout is guaranteed to
be the same, rather than essentially duplicating the way it is
determined.
This is from times when we supported padded/non-NPOT textures. The
difference is not useful anymore, and theoretical support for different
sizes is most likely buggy and unmaintained. So remove it.
Also remove the tex_ prefix wherever it appears.
Use mp_image_copy() instead of copying manually. (This function checks
whether the destination is regarded writeable, which it is not, because
the destination is the source image with changed pointers, so
refcounting has to be removed from the destination image by resetting
mpi->bufs.)
This shouldn't be needed anymore. Textures are now always allocated with
the exact size. Any padding (including non-NPOT support) is gone. The
texture sizes will always match the memory plane sizes.
Drop the unused and forgotten "npot" field from the option struct too.
Previously, with mpv --force-window=yes --idle=yes --screenshot-template="%f",
mpv would display an error saying that the template was incorrect, which it
isn't, there's just no file to put in the format. In this case, just use the
string "NO_FILE".
Drop the whitelist. It's annoying to maintain. Instead, accept any
subtitle decoder. Since this code path will now also be taken by bitmap
subtitle decoders not whitelisted by sd_lavc.c, add a warning when
bitmap subtitles are decoded. (To reduce or increase potential user
confusion.)
To some degree, a whitelist is needed to distinguish text and bitmap
subtitles. FFmpeg has an API to distinguish them in a generic way to
some degree, but Libav doesn't. So we just stick with this for now.
Undo 292266f2. Reapply 3e12e79b.
An additional copy is not really justified, as it could reduce
performance. On the other hand, we can force API users to create
a GL 3.x context.
eglTerminate() affects the EGLDisplay in all threads. Since the RPI
firmware apparently only ever uses EGL_DEFAULT_DISPLAY, this means it
will trash all other contexts on other threads in the same process.
Thus we don't call eglTerminate() at all, at least on RPI. Call
eglReleaseThread() instead (which may or may not be a NOP).
yuva444p worked, yuva420p didn't. This happened because the chroma pass
discards the alpha plane, which is referenced by the alpha blend code
later.
Add a terrible hack to work this around, actually using the same hack as
was used for the Y plane. (A terrible hack for terrible code.)
If the drag and drop action is anything other than
XdndActionCopy, append the dropped files rather than
replacing the existing playlist. With most file managers,
this will mean at least pressing shift while dropping.
This puts in place the machinery to merely append dropped file to the playlist
instead of replacing the existing playlist. In this commit, all front-ends
set this to false preserving the existing behaviour.
This takes care of the corner case where the player is started with a
single playlist entry so that the next/prev arrows are greyed out, but
remain that way even after new elements are added to the playlist.
Even after it has been disabled with the `disable-osc` message, the OSC
continues to run the tick function. Completely preventing tick from
being called is impractical since there are several different places
that it's called in the code, so just make it immediately return if the
OSC has been disabled.
This prevents the OSC from continuing the clear the OSD on every tick,
allowing other scripts to disable it so that they may draw to the OSD.
The FFmpeg can officially not distinguish between unknown subtitle
durations, and subtitle durations being 0. (It documents the value 0
meaning unknown duration.)
In practice, at least the LRC demuxer signals unknown subtitle durations
with a negative value.
Assume negative durations mean unknown duration. Show subtitles with
unknown duration forever. Unless there's a subtitle event following it;
then reset the duration so that it ends on the new subtitle event.
Fixes#2244.
Add a simplistic heuristic for detecting broken indexes. This includes
indexes with very few elements (apparently libavformat sometimes writes
such indexes, or used to), and indexes with broken timestamps.
The latter was apparently produced by very old HandBrake versions:
| + Muxing application: libmkv 0.6.1.2
| + Writing application: HandBrake 0.9.1
These broken files seem to be common enough that libavformat added a
workaround for them in 2008 (and maybe again in 2015). Apparently all
timestamps are multiplied with the file's tc_scale twice, and FFmpeg
attempts to fix them. We should throw away the whole thing.
Actually, this never happened, because there's logic for ignoring
duplicate header elements (which includes the seek index). This is
mostly for robustness and readability.
This might fix some problems when framestepping with interpolation
enabled. The problem here is that we want to show the non-interpolated
frame while paused. Framestepping is like unpausing the video for a
frame, and then pausing again. This draws an interpolated frame, and
redrawing on pausing is supposed to take care of this.
This possibly didn't always work, because vo->want_redraw is not checked
by the vo_control() code path. So wake up the VO thread (which takes
care of servicing redraw requests, kind of) explicitly.
The correct solution is getting rid of the public-writable want_redraw
field and replacing it with a new vo_request_redraw() function, but this
can come later.
So snd_device_name_get_hint() return values do in fact have to be freed.
Also, change listing semantics slightly: if io==NULL, skip the entry,
instead of assuming it's an output device.
libavcodec does not support HEVC via VAAPI yet, so this won't work.
However, there is ongoing work to add HEVC support to VAAPI, and this
change might help with testing. (Or maybe not - but there is no harm in
this change.)
Instead, allow reading 2KB only. This seems to be sufficient for
libarchive to recognize zip, 7z, rar, tar. Good enough.
This is implemented by creating an in-memory stream with a copy of
the file header. If libarchive succeeds opening this, the actual
stream is opened.
Allowing unlimited reading could break unseekable streams, such as
playing from http servers with no range request support or pipes.
Also, we try not to read too much data in the first probe pass. Some
slow network streams like shoutcast services could make probing much
slower if we allow it to read too much. In the second probing pass,
actually allow 200KB.
Should not happen, but since we don't control decoder video surface
allocation, anything could happen, and the code should be able to deal
with it. Untested.
This should avoid unnecessary sleeping when audio playback start resync
has finished and goes into the normal playback state.
This is tricky; see e.g. commit 402fe381.
As expected, probing with libarchive is a disaster. Both libavformat and
libarchive are too eager to misdetect file formats just because files
"might" be of a specific type. In this case, it's mp3 vs. tar. To be
fair, neither file format has an actual header. I'm not sure why we'd
need tar support, but since libarchive provides it, and idiots on the
internet apparently pack media files in tar sometimes (really, idiots),
keep it for now, and probe tar last.
Provides a simplistic way to seek without having to care about weird
situations like timestamp vs. playback time. This is good, because the
seek command is currently timestamp based, so when using the seek
command the user _does_ have to care.
Always compute the estimated absolute time of the seek target, and
display this as playback time during seeks.
Improves behavior with e.g. .ts files, for which we try to avoid seeks
by timestamp.
Fixes regression in dfd8a5f that made autoload not add more files
at the end of the current playlist if playlist was made by the
script.
This still prevents loading the script if more than one (media or
playlist) file was manually added.
This significantly reduces the amount of noticeable flashing when using
tscale kernels with negative lobes, by cutting them off completely.
I'm not sure if this has any negative effects. It needs a bit of
subjective testing over a period of time, so I just made it an option.
Fixes#2155.
A client API user might count on the fact that audio and video outputs
have already been uninitialized. (They remain uninitialized before
entering idle mode in order to allow smooth transition to the next
playlist entry.) Since event delivery is asynchronous, this has to
happen after actually doing the uninitialization, or the client will
essentially run into a race condition.
libarchive uses a quite confusing ifdeffery mess for some of the types
used in callbacks. Currently, archive_read_set_seek_callback() causes a
warning at least on Windows due to mismatching return type. The header
file uses __LA_INT64_T as return type, so I think the user is intended
to use int64_t.
(The ssize_t return type for the read_cb seems correct, on the other
hand.)
Some particular checks can define the HAVE_ key on their own. To make sure
they work correctly when composed (with compose_checks) we force the HAVE_
key to be undefined if a check fails.
Since vo_rpi uses MMAL for video output, which is completely
independent from the GLES overlay, we can just not redraw the
GLES screen if subtitles do not change.
(As a furhter optimization, the dispmanx overlay could be removed
if nothing is visible. But I'm not sure if adding and removing the
overlay frequently is a good idea for performance, so this could
just as well go the other way.)
Slightly faster than using the dispmanx mess (perhaps to a large amount
due to the rather stupid C-only unoptimized ASS->RGBA blending code).
Do this by reusing vo_opengl's subtitle renderer, and vo_opengl's RPI
backend.
Things like .gz etc., which have no real file header. A mixed bag,
because it e.g. tends to misdetect mp3 files as compressed files or
something (of course it has no mp3 support - I don't know as what it
detects them). But requested by someone (or maybe not, I'm not sure
how to interpret that).
This affects vo_opengl_cb in particular: it'll most likely auto-load
VDA, and then the VideoToolbox decoder won't work. And everything fails.
This is mainly caused by FFmpeg using separate pixfmts for the _same_
thing (CVPixelBuffers), simply because libavcodec's architecture demands
that hwaccel backends are selected by pixfmts. (Which makes no sense,
but now we have the mess.)
So instead of duplicating FFmpeg's misdesign, just change the format to
our own canonical one on the image output by the decoder. Now the GL
interop code is exactly the same for VDA and VT, and we use the VT name
only.
This works similar to the existing .rar support, but uses libarchive.
libarchive supports a number of formats, including zip and (most of)
rar.
Unfortunately, seeking does not work too well. Most libarchive readers
do not support seeking, so it's emulated by skipping data until the
target position. On backwards seek, the file is reopened. This works
fine on a local machine (and if the file is not too large), but will
perform not so well over network connection.
This is disabled by default for now. One reason is that we try
libarchive on every file we open, before trying libavformat, and I'm not
sure if I trust libarchive that much yet. Another reason is that this
breaks multivolume rar support. While libarchive supports seeking in
rar, and (probably) supports multivolume archive, our support of
libarchive (probably) does not. I don't care about multivolume rar, but
vocal users do.
The waf build system generates this already. No point in redoing it in
the header file.
The legacy build system (which we really should drop) didn't; fix it.
Make a difference between different mpv_handles pointing to the same mpv
core, or different mpv cores. While we're not explicit about it, at
least avoid that someone could misunderstand when really looking for the
case of mpv_handles pointing to the same core.
Instead, force everyone to use the metadata struct and set a "title"
field. This is only a problem for the timeline producers, which set up
chapters manually. (They do this because a timeline is a separate
struct.)
This fixes the behavior of the chapter-metadata property, which never
returned a "title" property for e.g. ordered chapters.
We must not use the frame PTS in any case. In this case, it fails
because nothing sets it up to wake up. This typically caused the player
to apparently "pause", until something else waked it up, like moving the
mouse and other events.
enable_key_bindings()/disable_key_bindings() now prints a log message on
each call, thus we should avoid makign redundant calls.
This could probably be solved more elegantly, but since this is all
legacy/private API, don't bother.
This VO is special because it normally doesn't block on vsync, but can
be made to do so. Supposedly the MMAL video output API merely sets a
"current frame" field when sending an output frame, and the firmware
will pick up whatever frame that field is set to at the time of a
vsync.
If this mode is enabled, the player tries to strictly synchronize video
to display refresh. It will adjust playback speed to match the display,
so if you play 23.976 fps video on a 24 Hz screen, playback speed is
increased by approximately 1/1000. Audio wll be resampled to keep up
with playback.
This is different from the default sync mode, which will sync video to
audio, with the consequence that video might skip or repeat a frame once
in a while to make video keep up with audio.
This is still unpolished. There are some major problems as well; in
particular, mkv VFR files won't work well. The reason is that Matroska
is terrible and rounds timestamps to milliseconds. This makes it rather
hard to guess the framerate of a section of video that is playing. We
could probably fix this by just accepting jittery timestamps (instead
of explicitly disabling the sync code in this case), but I'm not ready
to accept such a solution yet.
Another issue is that we are extremely reliant on OS video and audio
APIs working in an expected manner, which of course is not too often
the case. Consequently, the new sync mode is a bit fragile.
For video sync, we want separate playback speed controls for user-
requested speed and the "correction" speed for video timing. Further, we
use this separation to make sure only a resampler is inserted if
playback speed is only changed for video sync correction.
As of this commit, this is basically inactive code. It's just
preparation for the video sync code (the following commit).
Additionally to taking the average, this tries to use the demuxer FPS to
eliminate jitter, and applies some other heuristics to check if the
result is sane.
This code will also be used for the display sync code (it will actually
make use of the require_exact parameter).
(The value of doing this over keeping the simpler demux_mkv hack is
somewhat questionable. But at least it allows us to deal with other
container formats that use jittery timestamps, such as mp4 remuxed
from mkv.)
This doesn't work too well if sections of the file change to a different
framerate. It lowers our chances to guess the correct FPS in the display
sync code.
For normal playback, this (probably) doesn't help that much anyway,
except that the "estimated-vf-fps" property will regress in the simplest
mkv case. This will be fixed with the next commit.
The now disabled code will probably be removed; it's not useful anymore.
While the "old" libavcodec vdpau API is not deprecated (only the very-
old API is), it's still relatively complicated code that badly
duplicates the much simpler newer vdpau code. It exists only for the
sake of older FFmpeg releases; get rid of it.
Lua script that enables handling of certain commands depending on where
the mouse pointer is. Mostly useful for mouse-wheel handling via
input.conf.
Example:
MOUSE_BTN3 script-message-to zones commands "*-left: add volume +5"
"default: seek +10"
MOUSE_BTN4 script-message-to zones commands "*-left: add volume -5"
"default: seek -10"
Will changes behavior of Mouse Wheel Up/Down to alter volume if mouse is
in the left zone of the window, but re-adds the default to seek if wheel
is used on other parts.
Since we're on the topic of consistency, I've seen multiple users
complain about the presence of this period, which does not really match
other programs' behavior.
As the removed comment says, not copying this field may cause problems
on older libav* releases. See also commit 5f7de399.
Remove this, as newer FFmpeg releases are available. As of this commit,
use of mpv with FFmpeg 2.5.x and below, or Libav 11 and below is not
recommended, and may lead to random video decoding issues. (Although the
failure cases are apparently somewhat obscure.)
Removes some more internal API calls from the Lua scripting backend.
Which is good, because ideally the scripting backend would use libmpv
functions only.
One awkwardness is that mouse sections are still not supported by the
public commands (and probably will never), so flags like allow-hide-
cursor make no sense to an outside user.
Also, the way flags are passed to the Lua function changes. But that's
ok, because they're only undocumented internal functions, and not
supposed to be used by script users. osc.lua only does due to historical
reasons.
This was requested. It was more or less present internally already and
used for Lua scripting. Lua will switch to the "public" functions in
the following commits.
Add --demuxer-max-packets and --demuxer-max-bytes, which control the
maximum size of the packet queue. These can be helpful to avoid
excessive memory usage.
Memory usage is the reason why there's a limit in the first place. If a
file is more or less broken, and audio and video don't line up, the
decoders will fill up the packet queue trying to read more audio or
video, and the maximum sizes are required to avoid unbounded memory
allocation. Being able to override the maximum sizes is useful; either
for restricting memory usage further, or enlarging the sizes when
attempting to play various broken files.
Remove --demuxer-readahead-packets and --demuxer-readahead-bytes. These
were a bit useless. They could force a minimum packet queue size, but
controlling the queue size with --demuxer-readahead-secs is much nicer.
It's fairly certain nobody ever used these options.
Also add a note that you should not use the old build system. (It's only
kept because of the "asshole maintainer" rule: no matter how broken or
pointless someting is, as long as you're the maintainer and want to keep
it, it stays in the repo.)
VDA is being deprecated in OS X 10.11 so this is needed to keep hwdec working.
The code needs libavcodec support which was added recently (to FFmpeg git,
libav doesn't support it).
Signed-off-by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
Pretty stupid: vo_get_vsync_interval() returns a negative value if the
display FPS is unknown (e.g. xrandr not compiled), and the comparison
whether the value is below 0 fails later because it's assigned to an
unsigned int.
Regression since commit e3d85ad4.
Also, fix some comments in vo.c.
Some charsets can look like valid UTF-8, but aren't UTF-8. One example
is ISO-2022-JP. While ENCA apparently likes to get misdetect real UTF-8,
this is not the case with uchardet. uchardet can detect ISO-2022-JP
correctly, but didn't even get to try, because our own UTF-8 check
succeeded. So run the UTF-8 check when using ENCA only.
Fixes#2195.
Now it can always be read. Normally returns the value of the video-
aspect option. Writing it sets the option. If the aspect is not forced,
it will attempt to return whatever is the current video aspect.
That just makes no sense, but seems to be a somewhat common user error.
The detection is not perfect. It's conceivable that EXT-X-... headers
are used in normal m3u playlists. After all, HLS playlists are by
definition a compatible extension to m3u playlists, as stupid as it
sounds.
Drop d for toggling framedrop. Toggling this is way too special to be at
such a prominent place, and in fact I believe toggling it is pointless.
Remap deinterlacing from D to d. It's relatively useful and non-
destructive.
As suggested in #973 (almost).
Instead of opening a stream and then a demuxer, do both at once with
demux_open_url().
This requires some awkward additions to demuxer_params, because there
are some weird features associated with opening the main file. E.g. the
relatively useless --stream-capture features requires enabling capturing
on the stream before the demuxer is opened, but on the other hand
shouldn't be done on secondary files like external subtitles.
Also relatively bad: since demux_open_url() returns just a demuxer
pointer or NULL, additional error reporting is done via demuxer_params.
Still, at least conceptually, it's ok, and simpler than before.
Nobody wanted to restore this, so it gets the boot.
If anyone still wants to volunteer to restore menu support, this would
be welcome. (I might even try it myself if I feel masochistic and like
wasting a lot of time for nothing.) But if it does get restored, it
should be done differently. There were many stupid things about how it
was done. For example, it somehow tried to pull mp_nav_events through
all the layers (including needing to "buffer" them in the demuxer),
which was needlessly complicated. It could be done simpler.
This code was already inactive, so this commit actually changes nothing.
Also keep in mind that normal DVD/BD playback still works.
Normally when there's a timestamp reset, we make audio resync to make
sure audio and video line up (again). But in video-only mode, just
setting audio to resyncing breaks EOF detection, because there's no code
which would get audio_status out of this bogus state.
When full_redraw is set, we always need to take the draw_image path. If
it's not set, we can try VOCTRL_REDRAW_FRAME (and fallback to draw_image
if that fails).
Fixes#2184.
Commit c5818046 fixed one case of audio EOF handling, and caused a new
one. This time, the ao_buffer doesn't actually contain everyting that
should be played - because if --end is used, only a part of it is
played. Of course this is stupid, and it will be changed later. For now,
this smaller change fixes the bug.
Fixes#2189.
The jpeg-optimize and jpeg-baseline options were undocumented, and
they're also pretty useless. There's no reason to ever change them.
Also, don't write jpeg baseline images. This just makes compression
worse for the sake of rather questionable compatibility with ancient
decoders.
For now, it needs to be explicitly selected. ENCA is still the default.
This assumes uchardet returns iconv names. This doesn't seem to be
always the case, and the result are lots of iconv errors. So
explicitly check for this situation, and print a warning if it
occurs. It's entirely possible that uchardet support is actually
useless, because names are not necessarily iconv-compatible (but
uchardet doesn't seem to document whether it attempts to return
iconv-compatible names if possible).
Fixes#908.
uchardet is written in C++, and thus doesn't appreciate the value of
using static strings, and internally stores the guessed charset as
allocated std::string. Add a minimal hack to deal with this. (I don't
appreciate that the code is potentially harder to understand by
returning either a static or allocated string, but I do appreciate for
not having to litter the existing code with strdups.)
time_frame is when the next video frame should be shown. It's normally
overwritten by the video timing code. This also says something about
"nosound mode" (--no-audio today), but at least these days we don't use
it at all if video is disabled.
Remove it; it likely has no function at all.
If the framedrop count happens to be incremented with
vo_increment_drop_count() during rendering, these increments were
counted twice, because these events also set in->dropped_frame.
This was originally done for zsh; but zsh can manage the terminal state
correctly when foregrounding/backgrounding applications if you enable it
with "ttyctl -f". So I see no reason to wake up the mpv process once
every second anymore.
Revert "win32: more wchar_t -> WCHAR replacements"
Revert "win32: replace wchar_t with WCHAR"
Doing a "partial" port of this makes no sense anymore from my
perspective. Revert the changes, as they're confusing without
context, maintenance, and progress. These changes were a bit
premature anyway, and might actually cause other issues
(locale neutrality etc. as it was pointed out).
This was essentially missing from commit 0b52ac8a.
Since L"..." string literals have the type wchar_t[], we can't use them
for UTF-16 strings. Use C11 u"..." string literals instead. These have
the type char16_t[], but we simply assume char16_t is the same
underlying type as WCHAR. In practice, they're both unsigned short.
For this reason use -std=c11 on Windows. Since Windows is a "special"
environment (we require either MinGW or Cygwin), we don't need to worry
too much about compiler compatibility.
A user complains that it leads to the dxva driver failing, leading to
messages like this:
[ffmpeg/video] h264: Failed to execute: 0x8007000e
[ffmpeg/video] h264: hardware accelerator failed to decode picture
Reportedly, this happens only with vo_direct3d, not with vo_opengl. The
only difference is that vo_direct3d attempts to share the D3D device
with the decoder. Possibly the error is that the device in the VO is not
created with D3DCREATE_MULTITHREADED. Change this.
Probably fixes#2178.
The user probably doesn't want these. Conveniently, this also skips the
unwanted "." and ".." entries.
(This code is triggered if the input stream is a directory - and it's in
demux_playlist.c because it's convenient.)
WCHAR is more portable. While at least MinGW, Cygwin, and MSVC actually
use 16 bit wchar_t, Midipix will have 32 bit wchar_t. In that context,
using WCHAR instead is more portable.
This affects only non-MinGW parts, so not all uses of wchar_t need to
be changed. For example, terminal-win.c won't be used on Midipix at
all. (Most of io.c won't either, so the search & replace here is more
than necessary, but also not harmful.)
(Midipix is not useable yet, so this is just preparation.)
We used double-checked locking on pthread_mutex_t.requires_init in order
to lazily initialize static mutexes (since CRITICAL_SECTION has no
native way to do this). This was kind of unclean: we relied on MSVC
semantics for volatile (which apparently means all accesses are weakly
atomic), which is not such a good idea since mpv can't even be compiled
with MSVC.
Since it's too much of a pain to get weak atomics, just use INIT_ONCE
for initializing the CRITICAL_SECTION. Microsoft most likely implemented
this in an extremely efficient way. Essentially, it provides a mechanism
for correct double-checked locking without having to deal with the
tricky details. We still use an extra flag to avoid calling it at all
for normal locks.
(To get weak atomics, we could have used stdatomic.h, which modern MinGW
provides just fine. But I don't want this wrapper depend on MinGW
specifics if possible.)
See manpage additions.
The main reason for adding this is that we can't guess whether the user
wants his config in his Windows profile or not. The user basically has
to tell mpv what should be done, and the "portable_config" directory
does this implicitly.
Fixes#2042 (approximately).
Remove the exception for decoding only 1 frame if VO framedrop is
disabled. This was originally done to be able to test potential
regressions when we enabled VO framedrop and decoding 2 frames by
default. It's not needed anymore.
Instead of special-casing hwdec in the place where the video textures
are used, just set the textures in the image upload function. The
renderer code doesn't need to know whether hwdec interop is used at all.
This is an unfortunate fact of life. Maybe making this the default
wasn't such a good idea after all.
Also update etc/example.conf. It used an obsolete alias for "auto".
As suggested in #2033.
Additionally, change "LGPLv2+" to "LGPLv2.1+". Since this was always
used with the "and later" phrase, this doesn't require any further
discussion.
Also clarify that the "+" means "or later". This should be clear to
anyone with a brain, but you never know with copyright.
This detected whether an OpenGL context still provided legacy OpenGL if
the OpenGL version is modern (>= 3.0). This was actually only needed for
vo_opengl_old, because it relied on legacy functions. Since it's gone,
this code isn't needed either.
(Also, the removed comment about OpenGL 3.0 was wrong: you could just
query GL_CONTEXT_FLAGS and see if the forward compatible bit was set.)
In paused mode, we never entered the audio EOF state. This shows e.g. in
--keep-open mode, which will not set the eof-reached property correctly.
Regression since commit c06cd1b9. This commit was the wrong fix. We need
to respect the buffer state, and pausing has nothing to do with this.
Fixes#2167.
This reverts commit fb8d158366.
Reallocating the FBOs on every resize is very slow. It affects resizing
the window, as well as changing the video size itself with e.g.
panscan. Since the original change was done based on a single user
complaint, but the change itself caused a lot of complaints, we decided
to just revert it.
Allow setting an arbitrary amount, instead of the fixed 50%.
This is nto striclty backwards compatible. The defaults don't change,
but the --cache/--cache-default options now set the readahead portion.
So in practice, users who configured this until now will see the
double amount of cache being used, _plus_ the 75MB default backbuffer
will be in use.
Currently, this is perfectly equivalent, because back_size is hardcoded
to buffer_size/2. But this fixes the logic for the case the back_size
can be configured freely.
If filters are disabled or reconfigured, attempt to remove and probe the
deinterlace filter again. This fixes behavior if e.g. a software deint
filter was automatically inserted, and then hardware decoding is enabled
during playback. Without this commit, initializing hw decoding would
fail because of the software filter; with this commit, it'll replace it
with the hw deinterlacer instead.
Instead of calling it "future frames" and adding or subtracting 1 from
it, always call it "requested frames". This simplifies it a bit.
MPContext.next_frames had 2 added to it; this was mainly to ensure a
minimum size of 2. Drop it and assume VO_MAX_REQ_FRAMES is at least 2;
together with the other changes, this can be the exact size of the
array.
Handle a relatively recently introduced hack, that allows FLAC audio to
have arbitrary channel layouts, instead of just the predefined fixed
ones. This is actually supported by FFmpeg, but since the demuxer
(instead of the decoder) handles this in FFmpeg, we need to add special-
code to our mkv demuxer.
(The way FFmpeg does this seems a bit backwards, since now every demuxer
for a format that can handle FLAC needs to contain this logic as well.)
The FLAC hack is relatively terrible: we need to parse the FLAC headers,
look for a VorbisComment, parse the VorbisComment, and then retrieve
the magic WAVEFORMATEXTENSIBLE_CHANNEL_MASK entry. But the hack is
officially endorsed, as the official FLAC tools use it. (Although I
couldn't find a trace of it in the format specification. Should I be
surprised?)
This was a minor optimization to potentially avoid resampler
reconfiguration when the filter is reinitialized. But filter
reinitialization is a rare event, and the case when no reconfiguration
is needed is even rarer. As such, this is an unnecessary micro-
optimization and only adds potential for bugs.
This message bloats verbose log output if e.g. audio speed is frequently
readjusted, such as when syncing audio to video. So don't print the
message if only speed is changed. (This case requires reconfiguration,
but can't change the input/output channel maps.)
Also do not print the message if no remixing is done at all.
This was requested by someone.
All code was written by myself; some minor changes by 2 contributors who
agreed to general LGPL relicensing. 1 line of code is by someone unknown
who possibly wasn't asked (setting the "display_fps" variable), and
which can be reasonably ignored as it makes up only 0.1% of the file.
I still have no idea why this is needed, maybe some weird off-by-one
in some shitty driver? Either way, the difference for a working setup
shouldn't be too major, the most noticeable effect would be somewhat worse
performance when resizing the video during playback with interpolation
enabled using the mouse.
That's a specific enough side effect for me to not care as much about it.
Fixes#1814.
Probably makes users happy who want bitmap subtitles to show up in the
screen margins, and stops them from doing idiotic crap with vf_expand.
Fixes#2098.
There are some situations when redrawing is requested, but the current
frame was deleted. This could happen when switching e.g. hw decoding
mid-stream.
Separate uploading/drawing and fix the condition.
Just avoid some code duplication. Also, gl_video_set_options() having a
queue size output parameter is weird at best. While I don't appreciate
that this commit suddenly requires gl_video.c to deal with vo.c directly
in a special case, it's simply the best place to put this function.
The VO will be provided with future frames even if the format changes
mid-stream. This caused a crash if these frames were actually used (i.e.
interpolation mode was enabled).
Fixes a crash when deinterlacing is toggled during playback, and the
deinterlacer changes the stream format (as it can happen e.g. if the
decoder outputs nv12, which in turn happens with hw decoding).
(On a side note, future frames are always non-NULL. Also, the current
frame is of course always in the correct format.)
vaQueryImageFormats() returns a randomly ordered list - so we shouldn't
assume the first format on the list which works is the best. This
effectively switches to nv12 instead of yuv420p on some drivers.
We handle this by reusing va_to_imgfmt[], and ordering it by preference.
We hardcode that GPUs prefer nv12 pver yuv420p. In theory we could do
complicated probing (allocate dummy surface + use vaDeriveImage on it,
then retrieve the FourCC) - but all things which could break assumption
in the future are not supported yet (like 10 bit or 4:4:4), so this is
fine.
Fixes problems with --vo=opengl:interpolation. The issue here is that
vo_opengl retains more surfaces than what was preallocated for the
decoder. Until now, we just explicitly failed to decode frames for which
no additional surfaces are available. Since modern drivers usually are
fine with not "registering" surfaces before the decoder is created, just
allow allocating additional surfaces if needed.
(We also could probably recreate the HW decoder, since the HW decoder
should be stateless. But let's try to avoid raising the overall
complexity of the code.)
The interlaced frame test needs to be aware that the input mpi might be
NULL - this happens at the end of a stream when the input frames have
all been submitted but frames still need to be drained from the
decoder.
ass_set_fonts() is called by mp_ass_configure_fonts(), which was called
every time a subtitle renderer was initialized. I'm not sure why this
was done - I can't find a good reason, and most likely there's none.
However, it did cause problems with an experimental libass branch. It
crashed some time after switching to a second subtitle track. The branch
will hopefully be merged soon, and it seems unlikely that libass wants
to fix its problems with its ridiculous API (rather it should normalize
its API so that the issue doesn't happen in the first place), so just
apply this change. It makes our code simpler too.
Outputting the detected OpenGL features was useless and redundant with
the extension loading output.
Also, remove MPGL_CAP_3D_TEX from OpenGL(ES) 3.0. This block didn't
include the glTexImage3D function, so that was pointless and couldn't
have worked. The OpenGL 2.1 block does it correctly.
VDPAU has explicit support for rotating surfaces, and it is far less
expensive than using the normal rotation filter (which would require
reading video frames back into system memory), it is desirable to
implement the VO rotation capability.
To do this, we need to render the video frames to an output surface,
without rotation, and then render from that surface to the final
output surface to apply the rotation. It is important that the
intermediate surface is the same size as the final one (only not
rotated) so that hqscaling can be applied if requested by the user.
(hqscaling is a mixer capability and so takes effect when the video
surface is rendered to an output surface)
Finally, we must remember to explicitly clear the final output
surface as VDPAU only auto-clears output surfaces when rendering video
surfaces.
Normally, vdpau decoded frames are passed directly to a suitable
vo (vo_vdpau or vo_opengl) without ever touching system memory. This
is efficient for output purposes, but prevents any of the regular
filters from being used with such frames.
This new filter implements a read-back step to pull the frames back
into system memory where they can be acted on by other filters.
Eventually the frames will be sent to the vo as if they were normal
software-decoded frames.
Note that a vdpau compatible vo must still be used to ensure that
the decoder is properly initialised.
Signed-off-by: wm4 <wm4@nowhere>
mp_seek_chapter() had only 1 caller. Also the code was rather
roundabout; the entire function can be compressed to 5 lines of code.
(The new code is functionally the same - "mpctx->last_chapter_seek =
-2;" was effectively a dead assingment.)
Extend the --demuxer-mkv-probe-video-duration behavior to work with
files that are partial and are missing an index. Do this by finding a
cluster 10MB before the end of the file, and if that fails, just read
the entire file. This is actually pretty trivial to do and requires only
5 lines of code.
Also add a mode that always reads the entire file to estimate the video
duration.
Until now, if a stream wasn't seekable, but the stream cache was enabled
(--cache), we've enabled seeking anyway. The idea was that at least
short seeks would typically fall within the cache. And if not, the user
was out of luck and terrible things happened. In other words, it was
unreliable.
Be stricter about it and remove this behavior. Effectively, this will
for example disable seeking in piped data.
Instead of trying to be clever, add an --force-seekable option, which
will always enable seeking if the user really wants it.
This is a real pain: if a quit command is received, it's set to PT_QUIT.
And then other code could overwrite it, making it not quit. The annoying
bit is that stop_play is written and read in many places. Just not
overwriting it unconditionally seems to be the best course of action.
Some code called by vf_vdpaupp.c calls mp_image_new_custom_ref(), but
out of convenience doesn't reset the buffers. Make this behavior ok.
(The assert() was there to catch usage errors, but the same error could
already happen before the refcount changes were made, so the check is
not overly helpful.)
Fixes#2115.
Drop libva versions below 0.34.0. These are ancient, so I don't care.
Drop the vo_vaapi deinterlacer as well. With 0.34.0, VPP is always
available, and deinterlacing is done with vf_vavpp.
The vaCreateSurfaces() function changes its signature - actually it did
in 0.34.0 or so, and the <va/va_compat.h> defined a macro to make it use
the old signature.
Sometime recently, hardware decoding started to fail if h264 with full
reference frames was decoded, and --vo=vaapi was used. VAAPI requires
registering all surfaces that the decoder will ever use in advance, so
if the playback chain uses more surfaces than originally allocated, we
fail and drop back to software decoding.
I'm not really sure why or when this started happening. Commit 7b9d7265
for one is not the cause - it can be reproduced with earlier commits. It
also seems to be timing dependent. Possibly it has to do with the way
vo.c retains previous surfaces, and the way they can be queued/unqueued
asynchronously.
Increasing the number of reserved additional surfaces by 1 fixes it.
(Though I have no idea where exactly all these surfaces are being used.
Or rather, _when_.)
See manpage additions. This is mainly useful for vo_opengl_cb, but can
also be applied to vo_opengl.
On a side note, gl_hwdec_load_api() should stop using a name string, and
instead always use the IDs. This should be cleaned up another time.
Now there's a "canonical" table for mapping the names, that other code
can use, without having to rely too much on option code magic.
Also, use the central HWDEC constants, instead of magic values. (There
used to be semi-ok reasons to do this, but now it makes no sense
anymore.)
Some filter chains require a huge number of auto-inserted conversion
filters. There is an overly stupid safeguard against infinite filter
insertions, which counts the number of conversion filters inserted. This
triggered accidentally in this case. Fix by resetting this counter after
a non-conversion filter was successfully configured.
Each subtitle track gets its own decoder instance (sd_ass). But they use
a shared ASS_Renderer. This is done mainly because of fontconfig.
Initializing fontconfig is very slow when using it with memory fonts, so
there's a practical need to cache this memory font state, which is done
by not creating separate ASS_Renderers. This is very dirty and very
evil, but we probably can't get rid of it any time soon.
The shared ASS_Renderer was not properly synchronized. While the program
logic guarantees that only one sd_ass instance is visible at a time,
there are other interactions that require synchronization. In
particular, I suspect concurrent execution of mp_ass_configure_fonts()
and sd_ass.get_bitmaps cause issues in a newer libass development
branch.
So here's a shitty hack that hopefully fixes things, hopefully only
until libass becomes less dependent on fontconfig.
Basically, we need to make sure to allocate enough data for the pretty
dumb copy_nv12 function. (It could be avoided by making the function
less dumb, but this fix is simpler.)
ao_coreaudio (using AudioUnit) accounted only for part of the latency -
move the code in ao_coreaudio_exclusive to utils, and use that for the
AudioUnit code.
(There's still the question why CoreAudio and AudioUnit require you to
jump through hoops this much, but apparently that's how it is.)
mpv had refcounted frames before libav*, so we were not using
libavutil's facilities. Change this and drop our own code.
Since AVFrames are not actually refcounted, and only the image data
they reference, the semantics change a bit. This affects mainly
mp_image_pool, which was operating on whole images instead of buffers.
While we could work on AVBufferRefs instead (and use AVBufferPool),
this doesn't work for use with hardware decoding, which doesn't
map cleanly to FFmpeg's reference counting. But it worked out. One
weird consequence is that we still need our custom image data
allocation function (for normal image data), because AVFrame's uses
multiple buffers.
There also seems to be a timing-dependent problem with vaapi (the
pool appears to be "leaking" surfaces). I don't know if this is a new
problem, or whether the code changes just happened to cause it more
often. Raising the number of reserved surfaces seemed to fix it, but
since it appears to be timing dependent, and I couldn't find anything
wrong with the code, I'm just going to assume it's not a new bug.
This caused issues with hardware decoding. The VOs by definition dictate
the lifetime of the hardware context, so no surface allocations must
survive the VO. Fixes assertions on exit with vdpau.
It's conceivable that the OS time source is subject to clock changes.
The time could jump back to before when mpv was started, which would
cause mp_time_us() to return values smaller than 1. This is unexpected
by the code and could trigger assertions. If there's no monotonic time
source there's not much we can do anyway, so just sanitize the return
value. It will cause strange behavior until the "lost" time offset has
passed, but if you make such huge changes to the system clock while
everything is running, you're asking for trouble anyway.
(Normally we try to get a monotonic time source, though. This problem
sometimes happened on Windows when compiled without winpthreads, when
the code was falling back to gettimeofday(). This was already fixed by
always using another method.)
clock_gettime is implemented in winpthreads, so it's unavailable when
mpv is compiled with its internal pthreads implementation. This makes
mp_raw_time_us fall back to gettimeofday(), which can cause an assert
failure in mp_add_timeout() when the system clock is changed. Use
QueryPerformanceCounter instead.
The clock_gettime(CLOCK_MONOTONIC) implementation in winpthreads uses
QueryPerformanceCounter anyway, so there shouldn't be any change in
behaviour.
If the request contains a "request_id", copy it back into the
response. There is no interpretation of the request_id value by mpv; the
only purpose is to make it easier on the requester by providing an
ability to match up responses with requests.
Because the IPC mechanism sends events continously, it's possible for
the response to a request to arrive several events after the request was
made. This can make it very difficult on the requester to determine
which response goes to which request.
Until now, this was for AC3 only. For PCM, we used AudioUnit in
ao_coreaudio, and the only reason ao_coreaudio_exclusive exists
is that there is no other way to passthrough AC3.
PCM support is actually rather simple. The most complicated
issue is that modern OS X versions actually do not support
copying through the data; instead everything must go through
float. So we have to deal with virtual and physical format
being different, which causes some complications.
This possibly also doesn't support some other things correctly.
For one, if the device allows non-interleaved output only, we
will probably fail. (I couldn't test it, so I don't even know
what is required. Supporting it would probably be rather
simple, and we already do it with AudioUnit.)
Mapping of spdif formats was imperfect. Since the first format on the
list is somehow AAC, it was returned first, which is confusing, because
CoreAudio calls all spdif formats AC3. Since the spdif formats have some
rather arbitrary, reverse mapping the formats didn"t actually work
either. Fix by explicitly ignoring these when spdif is used.
Also, don't forget to set the samplerate in ca_asbd_to_mpformat(), or it
will work only in some cases.
This is basically a hack for drivers which prevent the mpv DXVA2 decoder
glue from working if OpenGL is in fullscreen mode.
Since it doesn't add any "hard" new API to the client API, some of the
code would be required for a true zero-copy hw decoding pipeline, and
sine it isn't too much code after all, this is probably acceptable.
For the sake of removing the separate stream/demuxer loading code.
This could probably be reimplemented in some other way, but I have no
DVB hardware for testing. The most preferred way would be making DVB to
not quit, and just rerun the stream selection.
The final goal is making opening the demuxer and opening the stream the
same operation.
Stream dumping is a rather uninteresting feature, but has a small
number of vocal users, and it's easy to keep.
When seeking to a different position, and seeking takes long, the OSD
might get redrawn. This means that the VO will receive a request to
redraw an old frame using whatever the previous PTS was. This breaks the
interpolation logic: the old frame will be added to the queue, and then
the next frames (with lower PTS if you seeked backwards) are not drawn
as the logic assumes they're past frames.
Fix this by using the non-interpolation code path when redrawing after a
seek reset, and no "real" frame has been drawn yet.
It's a recent regression caused by the redrawing code simplification.
The old code simply sent a VOCTRL for redrawing the frame, and the VO
had to deal with retaining the old frame on its own.
This is a hack as in there's probably a better solution.
Fixes#2097.
Less code, and avoids a black flash on start.
In theory it could happen that we map the window, and then don't have a
frame to draw - but mapping the window is done in the exact moment we
have a new frame to display.
This flags stuff tried to be too clever - if there are overlapping flags
(e.g. exclusive or combined flags), the one matching with most bits has
to be chosen.
This fixes logging of the seek command. E.g. "relative" and "absolute"
overlap to make them exclusive, but "relative" was always printed as it
happened to match first.
This is not the most theoretically perfect solution, ideally we could
check to see if the frame in question has already been rendered
somewhere in the queue and then avoid re-rendering it, at the cost of a
few extra lines of code. But I don't think the performance trade-off is
dramatic enough here.
draw_image_timed is renamed to draw_frame. struct frame_timing is
renamed to vo_frame. flip_page_timed is merged into draw_frame (the
additional parameters are part of struct vo_frame). draw_frame also
deprecates VOCTRL_REDRAW_FRAME, and replaces it with a method that
works for both VOs which can cache the current frame, and VOs which
need to redraw it anyway.
This is preparation to making the interpolation and (work in progress)
display sync code saner.
Lots of other refactoring, and also some simplifications.
This should make interpolation work much better in general, although
there still might be some side effects for unusual framerates (eg. 35 Hz
or 48 Hz). Most of the common framerates are tested and working fine.
(24 Hz, 30 Hz, 60 Hz)
The new code doesn't have support for oversample yet, so it's been
removed (and will most likely be reimplemented in a cleaner way if
there's enough demand). I would recommend using something like robidoux
or mitchell instead of oversample, though - they're much
smoother for the common cases.
For now, this is trivial (and actually redundant). The future display
sync code will make better use of it. The main point is that the new
internal API pretty much makes this transparent to the vo_opengl
interpolation code.
Now the VO can request a number of future frames with the last parameter
of vo_set_queue_params(). This will be helpful to fix the interpolation
code.
Note that the first frame (after playback start or seeking) will usually
not have any future frames (to make seeking fast). Near the end of the
file, the number of future frames will become lower as well.
Until now, it only used the hash from the previous configure run,
instead of trying to get the latest hash. The "old" build system did
this correctly - we just have to use the existing logic in version.sh.
Since waf supports separate build dirs, extend version.sh with an
argument for setting the path of version.h.
May help with (supposedly) bad drivers, which can put the device into
some sort of broken state when trying to set a different physical
format. When the previous format is restored, it apparently recovers.
This might make the change-physical-format suboption more robust.
We can be pretty sure that AudioUnit will remix for us.
Before this commit, we usually upmixed to stereo, because the
stereo and multichannel layouts were the only whitelisted ones.
Again. With the old OpenGL interop dropped, this probably works better
than vaapi-copy now. Last time we defaulted to vaapi-copy, because the
OpenGL interop could swap U/V planes and other stupid crap. We'll see.
Work around that FFmpeg doesn't distinguish between surface and cropped
size. The decoder always aligns the surface size to something
"convenient" (e.g. 16 for h264), and to get to the correct cropped size,
the output image's width/height is reduced. Using the cropped size
instead of the real surface size breaks the libva API in certain cases,
so we simply store and use the original size in our per-surface struct.
(If data is cropped on the left/top borders, hw decoding will simply
display these - FFmpeg doesn't let us do better.)
In theory, this code path avoids a copy. In practice, it never seems
to get enabled at all. But it does have potential for weird bugs or
performance issues (like being mapped from non-cacheable memory),
so kill it.
Some window managers let you change the fullscreen state of any window
using a key combination. For example, on XFWM you can use Alt+F11 and
on Compiz you can configure a key combination with the
"Extra WM actions" plugin.
With this change mpv will handle these fullscreen state changes. So, if
you enter into fullscreen mode using the WM's shortcut and then you use
mpv's fullscreen toggle, you will get back into window mode.
Merges PR #2081.
Signed-off-by: wm4 <wm4@nowhere>
If the EditionFlagOrdered is set, chapters without ChapterTimeEnd make
no sense. Ordered chapters will play the chapters in the order they
appear, but will play the ranges the chapters cover. So if the end time
is missing, the range is incomplete and it's not clear what should be
played. If you assume the start of the next chapter as end time, the
ordered flag will have no observable effect, so that's not a useful
assumption.
This fixes playback of a file which (apparently) had the
EditionFlagOrdered set accidentally, with normal chapters.
At least Matroska files have a "forced" flag (in addition to the
"default" flag). Export this flag. Treat it almost like the default
flag, but with slightly higher priority.
The "FrameRate" element is probably deprecated (it's greyed out in the
"spec", and described as "Informational only" in bold). Normally files
use DefaultDuration. In fact, the FrameRate field was preferred over
DefaultDuration for determining framerate if present. Do not do this and
rely on DefaultDuration only.
Also, if no framerate is set, do not assume PAL (25 FPS). Such a
fallback makes little sense and will cause more problems than it solves.
We want to distinguish actual errors, and just aborting the program
intentionally.
Also be a bit more careful with handling the wait() exit status: do not
called WEXITSTATUS() without checking WIFEXITED() first.
mpv usually sets the terminal to non-canonical mode (which in particular
disables line buffering). But the old mode is restored if the process is
not foregrounded. This is supposed to make mpv behave nicer when it is
backgrounded.
getch2_poll() enables canonical mode. Unfortunately, this was only
called after the poll timeout elapsed, so non-canonical mode is first
enabled after about a second after program start. Fix this by moving the
poll call before the timeout.
(As far as we're aware, there's no event-based way to determine when the
FD's process group changes, thus we're polling.)
DVD/BD menu support never worked right, and are a pain to maintain. In
particular, DVD menus never actually worked correctly, because
highlights were not rendered correctly. Fixing this requires major
effort, which I'm not interested to spend.
Most importantly, the requirement to switch streams without losing the
DVD/BD state caused major weirdness in the playback core. It was
implemented by somehow syncing the playback state to the DVD/BD
implementation (in stream_dvdnav.c etc.), and then reloading the demuxer
without destroying and recreating the stream. This caused a bunch of
special-cases which I'm looking forward to remove.
For now, don't just remove everything related to menu support and just
disable it. If someone volunteers, it can be restored (i.e. rewritten)
in a reasonable way. If nobody volunteers soon, it goes.
Replace all the check macros with function calls. Give them all the
same case and naming schema.
Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes().
Introduce af_fmt_is_pcm(), and use it in situations that used
!AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format
was. It simply meant "not PCM".
Audio formats used a semi-clever schema to encode the properties of the
PCM encoding as bitfields into the format integer value.
The af_fmt_change_bits() implementation becomes a bit weird, but it's
an improvement to the rest of the code.
(I've always disliked it, so why not get rid of it.)
This may or may not fix some issues with the format switching
code. Actually, it seems somewhat unlikely, but then checking
the stream type isn't incorrect either, and is probably
something the API user should always be doing.
Originally, this was written for comparing the sample format only, but
ca_change_physical_format_sync() actually expects that the full format
is compared. (For all other uses it doesn't matter.)
The speaker replacement nonsense sometimes made blatantly incorrect
decisions. In this case, it prefered a 7.1(rear) upmix over outputting
5.1(side) as 5.1, which makes no sense at all. This happened because 5.1
and 7.1(rear) appeared equivalent to the final selection, as both of
them lose the sl-sr channels. The old code was too stupid to select the
one with the lower number of channels as well.
Redo this. There's really no reason why there should be a separate final
decision, so move the speaker replacement logic into the
mp_chmap_is_better() function.
Improve some other details. For example, we never should compare the
plain number of channels for deciding upmix/downmix, because due to NA
channels this is essentially meaningless. Remove the NA channels when
doing this comparison. Also, explicitly handle exact matches.
Conceptually this is not necessary, but it avoids that we have to
needlessly shuffle audio data around.
This reverts commit fc9695e63b.
Users were complaining that both mpv and something else (what? I don't
know) respond to some multimedia keys, such as volume change.
Integer and float elements are encoded as a sequence of bytes prefixed
by a variable-length encoded length specifier. If the length is 0, then
there is no data. Whether this is valid or not is not really clear, but
some sample files which do this have surfaced. It's not particularly
hard to handle this, so just do it.
Use char* for strings instead of bstr (data ptr + length pair). Matroska
actually (probably) allows "padding" strings with \0 bytes, so using
normal C strings instead of byte strings is more appropriate.
500ms is a bit too high. Change it to 50ms. This improves client API
(and Lua) playback state update frequency.
Updating absolutely every time the audio PTS changes would be possible,
but is not helpful. Audio samplerates are high to trigger a wakeup
feedback loop, so the process would waste CPU time on updating the
playback position all the time.
(If a client application wants to ensure smooth update of the playback
position, it should update the position manually using a timer and by
reading the property - the application can make a much better decision
at how often the playback has to happen.)
This avoids keeping "bad" state from previous reconfig calls, such as
the internal_sample_format option (which is set only on the first
reconfig call).
There's no advantage to keeping the resample contexts around anyway.
Basically, af_fix_format_conversion() behaves stupid you insert a
conversion filter that won't work, and adding back the conversion test
function is the simplest fix to it.
So apparently, this essentially happens when the kernel driver doesn't
implement write accesses in the channel map control. Which doesn't
necessarily mean that the channel map is unsupported, or that there is a
bug - it's just lazyness and a consequence of the terrible ALSA kernel
API for the channel mapping stuff.
In these cases, the channel count implicitly selects the channel map,
and snd_pcm_set_chmap() always fails with ENXIO.
I'm actually not sure what happens if dmix is on top of e.g. HDMI, which
actually lets you change the channel mapping.
I'm also not sure why commit d20e24e5d1614354e9c8195ed0b11fe089c489e4
(alsa-lib git repository) does not take care of this.
MPlayer traditionally had completely separate sh_ structs for
audio/video/subs, without a good way to share fields. This meant that
fields shared across all these headers had to be duplicated. This commit
deduplicates essentially the last remaining duplicated fields.
Always use the already existing extradata[_len] variable, instead of the
awkward switch between manually changed extradata and falling back to
passing through extradata at the end.
When using --hwdec=auto, about half of all systems will print:
"[vdpau] Error when calling vdp_device_create_x11: 1"
this happens because usually mpv will be linked against both vdpau and
vaapi libs, but the drivers are not necessarily available. Then trying
to load a driver will fail. This is a normal part of probing, but the
error messages were printed anyway. Silence them by explicitly
distinguishing probing.
This pretty much goes through all the layers. We actually consider
loading hw backends for vo_opengl always "auto probed", even if a hw
backend is explicitly requested. In this case vd_lavc will print a
warning message anyway (adjust this message a bit).
Client API users can enable log output with mpv_request_log_messages().
But you can enable only a single log level. This is normally enough, but
the --msg-level option (which controls the terminal log level) provides
more flexibility. Due to internal complexity, it would be hard to
provide the same flexibility for each client API handle. But there's a
simple way to achieve basically the same thing: add an option that sends
log messages to the API handle, which would also be printed to the
terminal as by --msg-level.
The only change is that we don't disable this logic if the terminal is
disabled. Instead we check for this before the message is output, which
in theory can lower performance if messages are being spammed. It could
be handled with some more effort, but the gain would be negligible.
This happens with av_log(NULL, ...) calls. Drop the "?: " fallback
prefix, because it was confusing.
(Of course FFmpeg should not do this at all, but it's a very long way to
making the FFmpeg log callback sane.)
That's how mime types are.
(This makes redirection with a specific HLS URL work, because some idiot
thought it'd be a great idea to spell the mime type as
"application/x-mpegURL".)
No particular reason, but it's still possible that it causes additional
corner cases, and it's not really needed to test this on wine (other
than testing fullscreen stuff, which should be done on a real Windows
anyway).
The only decoders I could find and which (possibly) require this field
are codecs which can be used via VfW only, and realaudio sipr. For VfW
we still passthrough this field.
Native Matroska codec support has to map the Matroska codec IDs to
libavcodec ones, and also has to undo codec-specific Matroska
strangeness, such as restoring AAC extradata and realaudio handling. The
VfW codec support doesn't need it, because AVI maps well enough to
libavcodec conventions (possibly because AVI was a dominant codec when
libavcodec was created). But there's still some need for generic codec
handling, such as enabling parsers and messing with various codec
parameters.
Separate these two, and move the parts which are guaranteed not to be
needed by VfW to the if-else tree that handles the VfW case
("A_MS/ACM"), making the cases exclusive.
(This should probably be done more radically, since it's very unlikely
that we should or have to mess with the VfW parameters at all - they
should just be passed through to the decoder.)
This removes the last traces of the old MPlayer FourCC-based codec
mapping code. Forcing all codec IDs through a FourCC table and then
back to codec names was confusing at best, so this is a nice cleanup.
Handling of PCM (non-VfW case) is redone to some degree.
Handling of AC3 is moved below realaudio handling, since "A_REAL/DNET"
is apparently AC3, and we must not skip realaudio-specific handling.
(It seems unlikely that anything would actually break, but on the other
hand I don't have any A_REAL/DNET samples for testing.)
Instead of explicitly matching all the specific AAC codec names, just
match them all as prefix.
Some codecs don't need special handling other than their mapping
entries, so they fall away (like Vorbis and Opus).
The prores check in mkv_parse_and_add_packet() is not strictly related
to this, but is done for consistency with the wavpack check above.
When showing cover art, the decoding logic pretends that the source has
an infinite number of frames. This slightly simplifies dealing with
filter data flow. It was done by feeding the same packet repeatedly to
the decoder (each decode run produces new output).
Change this by decoding once at the video initialization. This is easier
to follow, and increases robustness in case of broken images. Usually,
we try to tolerate decoding errors, so decoding normally continues, but
in this case it would just burn the CPU for no reason.
Fixes#2056.
This is slightly "dangerous", because it could overwrite a log callback
another library has set, after we've set our own callback. But it's
probably still slightly better than leaving our own callback, which will
run the fallback code if no mpv instance is set. (Multiple mpv instances
sharing the same global state will safely avoid overwriting each other's
log callback.)
Note that we can't do much better, because the global state in FFmpeg is
obviously insane.
The previous behavior is confusing if the B point is near EOF (consider
B being the duration of the file, which is strictly speaking past the
last video timestamp). The new behavior is fine as well for B being far
past EOF.
Achieve this by checking the EOF state in addition to whether playback
has reached the B point. Also, move the A-B loop code out of
command_event(). It just isn't useful anymore, and obfuscates the code
more than it makes it loop simple.
Fixes#2046.
Seems logical.
Note that if playback otherwise ends while playback is active and a seek
is still queued, we still exit. Otherwise you couldn't end playback by
seeking past the end of the file (which is classic MPlayer and mpv
behavior).
This attempted to find a minimal filter graph for a format conversion
involving multiple conversion filters. With the last 2 commits it
becomes dead code - remove it.
Now af_lavrresample can output 24 bit samples directly, by doing the
conversion "inline". Luckily, S32->S24 can be done in-place, so this
isn't too much work. But the output conversion logic (which seems to be
adding up) gets slightly more complicated again.
Normally this is done by af_convert24. But having multiple conversion
filters complicates some aspects of the filter chain. S24 output is the
only thing the code for multiple conversion filters is still needed for,
and getting rid of that is preferable.
If the code path for additional output conversion is active,
reorder_planes() is always called, even if the reorder_out array wasn't
filled. This is obviously wrong - always fill this array.
They are useless. Not only are they actually rarely in use; but
libavcodec doesn't even output them, as libavcodec has no such sample
formats for decoded audio.
Even if it should happen that we actually still need them (e.g. if doing
direct hardware output), there are better solutions. Swapping the sign
is a fast and lossless operation and can be done inplace, so AO actually
needing it could do this directly.
If you wonder why we keep U8 instead of S8: because libavcodec does it.
The existing code avoided doing this for some codecs. I see no point in
this, and it seems the original reason this exists was due to some
cleanup in 2007. libavformat doesn't do this. So just drop it.
It's well possible that we've always ended up invoking the
AV_CODEC_ID_MPEG1VIDEO codec, but it's hard to tell. Mangling everything
through FourCCs (and then back) makes it hard to analyze. Also,
libavformat's Matroska demuxer uses AV_CODEC_ID_MPEG2VIDEO here, so it
should be quite safe to do anyway.
Inherited from MPlayer times, we used FourCCs to identify video codecs.
This was later changed to libavcodec codec names (which made life a
whole lot simpler). But demux_mkv still uses FourCCs a lot.
Change this for video. It's pretty simple, because some preparation was
done in the past. We just have to replace some "internal" FourCCs with
different handling.
One potentially complicated issue is that there is no natural way to
set the sh->format (AVCodecContext.codec_tag) field anymore. Most
decoders do not need it, though mjpeg is an exception.
Note that the AVI compatibility code still requires codec mappings, but
these are provided by FFmpeg. Also, the audio code is not changed.
For the MKV_V_MPEG2 -> mpeg1video thing see next commit.
Channel maps reported by the device as SND_CHMAP_TYPE_VAR can be freely
reordered. We don't use this much (out of laziness), but in this case
it's a simple way to reduce necessary reordering (which would be an
extra libavresample invocation), and to make debug output more readable.
Until now, we didn't do this, because it required some effort, and
didn't seem to be necessary. It probably still isn't, but it sounds
like a good idea not to output arbitrary data on these channels.
The situation is complicated by the fact that just adding new channels
to a planar frame would require messing with buffers. So we would have
to allocate new buffers and add them to the frame. We could have to
maintain an extra buffer pool for this. Avoid this by being "clever",
and just allocate a frame with enough channels in the first place.
libav/swresample won't know about these channels and won't write to
them, but we can grab them in reorder_planes() and use them for the
NA channels.
This is just a conceptual issue, since for now every channel count has
an associated standard layout.
But should the max. channel count ever be bumped, some things would stop
function if mp_chmap_from_channels() refused to work for any channel
count within the allowed range.
In the AVFrame-style system (which we inreasingly map our internal data
stuctures on), buffers and plane pointers don't necessarily have a 1:1
correspondence. For example, a single buffer could cover 2 or more
planes, all while other planes are covered by a second buffer, and so
on. They don't need to be ordered in the same way.
Change mp_audio_get_allocated_size() to retrieve the maximum size all
planes provide. This also considers the case of planes not pointing to
buffer start.
Change mp_audio_realloc() to reset all planes, even if corresponding
buffers are not reallocated. (The caller has to be careful anyway if it
wants to be sure the contents are preserved on realloc calls.)
If you try to play surround with dmix, it will advertise surround and
lets you set more than 2 channels, but will report a stereo channel map,
with the extra channels identified as NA. We could handle this now, but
we don't want to (because it's excessively stupid).
Do it only if the channel map is not what we requested, instead of just
acting if it contains NA entries at all. This avoids that we hurt
ourselves in the unlikely but possible case we actually have to use
channel maps with NA entries.
So successful playback and user quit can be distinguished, for whatever
reason you may want to do this.
Normally, the "quit" command can be customized, but this does not work
for quit commands sent by the terminal signal handler. One solution
would be introducing something like "ON_SIGNAL" (equivalent to
"CLOSE_WIN"), but considering there are a bunch of possible signals, I'd
rather not get into this. So go with the dumb solution.
Probably fixes#2029.
gl_video_set_options() does not acquire ownership of the opts parameter
or its contents. In case of vo_cmdline, opts will point to temporary
memory. This memory will be free'd at a later point, and p->opts will
point to free'd memory on the next reinitialization.
The fix is pretty ugly, but it's a quick bug fix. This can probably be
removed once VO sub-options are exposed as properties.
Fixes#2035.
If the audio API takes a while for starting the audio callback, the
current heuristic can be off. In particular, with very short files, it
can happen that the audio callback is not called before playback is
stopped, so no audio is output at all.
Change draining so that it essentially waits for the ringbuffer to
empty. The assumption is that once the audio API has read the data
via the callback, it will always output it, even if the audio API
is stopped right after the callback has returned.
If a frame could only be partially filled with real audio data, the
silence wasn't written at the correct offset. It could have happened
that the remainder of the frame contained garbage.
(This didn't happen in the more common case of playing dummy silence.)
Used a wrong condition, and I suppose it could crash in some situations.
Change it to lazily initialize the hotplug stuff, like the
audio-device-list property does.
This was matching e.g. both "foo/bar" and "foobar" against "foo", when
only the former should match. This could cause more property
notifications than necessary.
Absence of license header implies LGPL, as mentioned in the "Copyright"
file. But vaapi.h contains some code taken from the mplayer-vaapi
patch, which was under the typical MPlayer license.
All vo_gl.c related code has been GPL+LGPL dual-licensed. The OSD code
is no exception and is also derived from vo_gl.c. Thus it should have
the same license (although I think technically speaking sub-licensing
it by removing one of the licenses is ok).
Commits 92b27be and f4ce99d removed high-fps logic to to a bug. That bug was
a missing parenthesis around everything after duration >= 0 && ... at the
removed code.
This patch restores the removed code, fixes the bug and then refactors the
code a bit.
This reverts commit f1746741de.
Together with the other revert, this fixes#2023 (the reason being
broken framedrop handling - it was dropping frames when it shouldn't).
read_output_surface() could fail and return NULL.
Also, make sure we don't set the image to a size larger than the
allocated size. Normally this shouldn't happen, but in theory it could
in corner cases; this is for robustness.
This provides a new method for enabling spdif passthrough. The old
method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated
method will probably stop working at some point.
This also supports PCM fallback. One caveat is that it will lose at
least 1 audio packet in doing so. (I don't care enough to prevent this.)
(This is named after the old S/PDIF connector, because it uses the same
underlying technology as far as the higher level protoco is concerned.
Also, the user should be renamed that passthrough is backwards.)
This deprecates the --ad-spdif-dtshd option, and replaces it with a
pseudo decoder. This means ad_spdif will report two decoders, "dts" and
"dts-hd", of which the second simply enables what the option did.
The --ad-spdif-dtshd option will actually be deprecated in the next
commit.
This makes no sense, because the format can't be converted anyway. It
just sets up the filter chain init code, which will vomit a bunch of
useless and confusing messages. So uninit and fail explicitly when this
happens.
We can't do much in this case, but at least we can not call the vdpau
API functions with too large sizes. Apparently the API considers this
undefined behavior, and random stuff might happen.
The previous code was not wrong, but I'd claim this makes the code more
robust. If a situation could happen in which the passed surface size is
incorrect, we could have passed a too small image, and
VdpOutputSurfaceGetBitsNative could have randomly overwritten memory.
Check the maximum size of video surfaces, and refuse initialization if
the video is too large for them.
Maybe we could do something more sophisticated, like inserting a
software scaler. On the other hand, this would have a very questionable
benefit, as it would be guaranteed to be too slow.
When starting in paused mode, no audio is written to the device at all,
because writing audio implicitly unpauses the AO. If the file is very
small, and all audio fits within the AO buffer, this accidentally
triggered the EOF condition. (In unpaused mode, it would write all
audio, end playback, and then wait until the AO has everything played.)
This is better, because now we call swr_get_delay() with the output
samplerate, instead of with the input samplerate and then multiplying it
with the ratio and rounding it up.
Apparently, just running ./waf and hoping that it will be run with a
Python interpreter doesn't necessarily work. The workaround is pretty
simple and reliable.
The "osd" command cycles between 4 states (OSD level 0-3), which is
probably confusing and inconvenient. OSD levels 0 and 2 are rarely
needed. I would claim there is normally not much of a need to completely
disable OSD by setting level 0 during playback. Level 2 is just slightly
less information than level 3, and I'm not sure why it exists at all.
Change it so that it toggles between level 3 and 1. Note that this
ignores the default OSD level. If the default is 3, the first use of
this key will set it to 3 again. Just assume 1 is the default. If
someone complains, this could be improved.
There's a short time during loading where external commands can add
external streams even before the main file is loaded (like during ytdl
hook execution). The track list is printed every time an external track
is added via commands. This was quite awkward when ytdl was adding
multiple streams, so don't print it in this stage. They are printed
anyway at the end of the loading process.
Listening to kAudioDevicePropertyDeviceHasChanged does not send any
property change notifications when the device dies. Makes no sense,
but I suppose in CoreAudio logic a dead/removed device can't send
any notifications.
This caused the player to essentially pause playback if the audio
device was removed during playback.
Fix by listening to the kAudioHardwarePropertyDevices property too,
which will actually be sent in this specific case. Then, if
querying the already dead device fails, we know we have to reload.
In short, instead of letting the coreaudio property listener set atomic
flags (which are then polled), make the property listeners actually
active.
The format change listener used during audio output now simply calls
ao_request_reload() on its own. All code involved is thread-safe, so
there's no need to do it during this audio callback (we assumed the
callback was never run concurrently with itself).
The listener installed temporarily during ca_change_format() is changed
to post a semaphore. Get rid of the weird retry logic and replace it
with a flat loop + timeout. It appears the maximum wait time could be
2500ms; reduce the total timeout to 500ms instead.
This manually retrieved the remaining audio from the resampler. It
subtly missed a conversion which could leave to an unsubtle crash.
This could happen if reorder_planes() was supposed to insert NA
channels, and the resampler/actual output format were different.
Simplify it by reusing the normal drain path. One oddness is that
the filter will add an output frame outside of normal filtering,
but that should be fine.
This was missing for extended deinterlacer.
Unfortunately, these deinterlacer still do not work. The provided future
frame (which is all the deinterlacers want) seems to be correct, though.
One minor behavioral change is that this always keeps the previous frame
for PTS computations. This could be avoided (in order to keep exactly
the same behavior as before), but it seems more elegant and should not
do any harm. (Also, if we really cared about reducing hw frame refs,
a more worthy goal is producing the field output incrementally.)
The mapped data (pointed to by the param variable) is not needed before,
so the call can be moved down. Also, this prevents that the buffer
remains mapped forever if the other vaMapBuffer() call above fails (the
cleanup code forgets to unmap the buffer - this commit makes it
unnecessary).
This used a do-while loop, which runs only once, as replacement for a
cleanup goto. While this is ok, doing a goto directly is easier to
follow and is closer to idiomatic C. But mainly remove it so that the
indentation can be reduced.
Reconfiguring with the same video size should never cause the window to
resize back to the video size (if the user changed its size). This was
broken and it resized anyway.
We do not fill them, so we would pass random IDs to the driver. The code
was originally written to handle bob deinterlacing only, so I guess it
originally passed always 0 anyway, despite having code for reference
surface list allocation.
Also, move down the vaUnmapBuffer() call. This call actually "unmaps"
the param pointer, so accessing it after the unmap call would be
undefined behavior. The "example" in <va/vavpp.h> does this too, but
it's most likely an error.
(Additionally, not even bob deinterlacing worked correctly in my test,
sigh.)
Normally, the cache keeps 50% of the buffer for seeking backwards. Until
now, the cache just used the full buffer size at the beginning of a
file, because the 50% normally reserved for the backbuffer are unused.
This caused a problem: when streaming from http, the player would first
read about 150MB (default cache size), then stop until 75MB of the cache
has been played. (Until the 75MB position, the cache is fully used, so
nothing new can be read. After that, part of the backbuffer starts
getting unreserved, and can be used for readahead.) This long read pause
can cause the server to terminate the connection. Reconnecting may be
possible, but if youtube-dl is used, the media URL may have become
invalid.
Fix this by limiting readahead to 50% even if unnecessary. The only
exception is when the whole file would fit in the cache. In this case,
it won't matter if we can't reconnect, because the cache covers
everything anyway, and hopefully the cache will stay valid.
Likely fixes#2000.
This must have been some non-sense in the original vaapi mplayer patch.
While I still have no good idea what this "direct mapping" business is
about, it appears to be pretty much pointless. Nothing can hold
additional "real" surface references (due to how the API and mpv/lavc
refcounting work), so removing the additional surfaces won't break
anything. It still could be that this was for achieving additional
buffering (not reusing surfaces as soon), but we buffer some additional
data anyway. Plus, the original intention of the vaapi mplayer code was
probably increasing surface count just by 1 or 2, not actually doubling
it, and/or it was a "trick" to get to the maximum count of 21 when h264
is in use.
gstreamer-vaapi uses "ref_frames + SCRATCH_SURFACES_COUNT" here, with
SCRATCH_SURFACES_COUNT defined to 4. It doesn't appear to check the
overlay attributes at all in the decoder.
In any case, remove this non-sense.
It polluted the global namespace, instead of exporting the function
properly.
For now, keep it compatible by explicitly keeping the bogus export.
Also fix a mistake in the manpage example.
On hw decoder reinit failure we did not actually always return a sw
format, because the first format (fmt[0]) is not always a sw format.
This broke some cases of fallback. We must go through the trouble to
determine the first actual sw format.
This reduces spam while preempted a bit.
The remaining message, "hardware accelerator failed to decode picture"
on every frame, can not be prevented because it's hardcoded in
libavcodec.
If gl_hwdec_driver.map_image fails, all textures will be set to 0. This
in turn makes pass_prepare_src_tex() skip generation of the texture
uniforms, which leads to a shader compilation error, as e.g. texture0 is
not defined but expected to exist and accessed.
Set the textures to an invalid non-0 ID instead. OpenGL can deal with
it.
Yet another of these dozens of hwaccel changes. This time, libavcodec
provides utility functions, which initialize the vdpau decoder and map
codec profiles. So a lot of work the API user had to do falls away.
This also will give us support for high bit depth profiles, and possibly
HEVC once libavcodec supports it.
...instead of relying on the hw decoding API to align it for us. The old
method could in theory have gone wrong if the video is cropped by an
amount large enough to step over several blocks.
Always configure the vdpau mixer based on the current surface sent to
it. Before this, we just hardcoded the chroma type, and the surface size
was essentially a guess.
Calling VdpVideoSurfaceGetParameters() on every surface is a bit
suspicious, but it appears it's a cheap function (just requiring some
locks and a table lookup). This way we avoid creating another
complicated mechanism to carry around the actual surface parameters
with a mp_image/AVFrame.
There's not much of a reason to keep get_surface_hwdec() and
get_buffer2_hwdec() separate. Actually, the way the mpi->AVFrame
referencing is done makes this confusing. The separation is probably
an artifact of the pre-libavcodec-refcounting compatibility glue.
Most of hardware decoding is initialized lazily. When the first packet
is parsed, libavcodec will call get_format() to check whether hw or sw
decoding is wanted. Until now, we've returned AV_PIX_FMT_NONE from
get_format() if hw decoder initialization failed. This caused the
avcodec_decode_video2() call to fail, which in turn let us trigger the
fallback. We didn't return a sw format from get_format(), because we
didn't want to continue decoding at all. (The reason being that full
reinitialization is more robust when continuing sw decoding.)
This has some disadvantages. libavcodec vomited some unwanted error
messages. Sometimes the failures are more severe, like it happened with
HEVC. In this case, the error code path simply acted up in a way that
was extremely inconvenient (and had to be fixed by myself). In general,
libavcodec is not designed to fallback this way.
Make it a bit less violent from the API usage point of view. Return a sw
format if hw decoder initialization fails. In this case, we let
get_buffer2() call avcodec_default_get_buffer2() as well. libavcodec is
allowed to perform its own sw fallback. But once the decode function
returns, we do the full reinitialization we wanted to do.
The result is that the fallback is more robust, and doesn't trigger any
decoder error codepaths or messages either. Change our own fallback
message to a warning, since there are no other messages with error
severity anymore.
Vobsubs come as .idx/.sub pair of files. The .idx file is the one that
should be opened, but the name of the .sub file is unknown. We can now
make our own guess what the name of that file is. In particular, improve
support with URLs (as these can have the file extension in the middle of
the filename string if there are HTTP parameters).
Note that this works only with newer ffmpeg versions, because the
recently added sub_name demuxer option is used for this.
This command has been deprecated in the 0.8.x and 0.9.x releases - get
rid of it. Its only point ever was MPlayer compatibility, which broke
years ago anyway.
They're completely orthogonal concepts, merged in the past due to
convenience and ease of implementing it in the old #ifdef hell renderer.
Especially after the CMS stuff was generalized by 634b4a, this was a
trivial change to implement and also means color management will be much
higher quality when enabled with vo=opengl (which had quantization
issues in the past due to the 8 bit FBO format and upscaling), since it
can be done in a single pass now.
While all functions of input_ctx are inherently thread-safe, access to
the _inputContext field itself is not. It could be unset any time by
cocoa_set_input_context(). So even trivial input_ctx calls must be under
a lock, so that the input_ctx can not be destroyed while the function
call is "starting". (Even a function call in progress wouldn't be fine,
because mp_input_uninit() requires the caller to "own" the object, i.e.
no other threads can access it at this point.)
For certain reasons, we allow adding external tracks even before the
main file is loaded. This somewhat breaks in old assumption, which uses
mpctx->num_sources to determine whether a command can be applied in the
current state. Use the newer playback_initialized instead, which is a
much better choice for this purpose.
The previous commit removed this. Although mp_switch_track() can now be
called in all situations, we still don't want it to be called here.
Setting a track property while no file is loaded would simply deselect
the track instead of setting the underlying option to the requested
value.
Likewise, if the "cycle" command (M_PROPERTY_SWITCH) is used, don't just
deselect the track.
Adding an external audio track before loading the main file didn't work
right. For one, mp_switch_track() assumes it is called after the main
file is loaded. (The difference is that decoders are only initialized
once the main file is loaded, and we avoid doing this before that for
whatever reason.)
To avoid further messiness, just allow mp_switch_track() to be called at
any time. Also make it do what mp_mark_user_track_selection() did, since
the latter requires current_track to be set. (One could probably simply
allow current_track to be set at this point, but it'd interfere with
default track selection anyway and thus would be pointless.)
Fixes#1984.
Wnile it seems quite logical to me that commands use _ as word
separator, while properties use -, I can't really explain the
difference, and it tends to confuse users as well. So always
prefer - as separator for everything.
Using _ still works, and will probably forever. Not doing so would
probably create too much chaos and confusion.
Now it simply changes the options, i.e. what will be requested, instead
of returning M_PROPERTY_UNAVAILABLE.
This is another minor step towards unifying options and properties.
Still a bit weird: it will always return "no" if no file is loaded, and
disregards the option value.
Also replace their implementation with the recently introduced
properties. One significant difference is that audio-channels using OSD
formatting does not print the channel layout. The user can just use the
replacement property instead.
This brings the volume control closer to what is percepted as linear
volume change.
Adjust the --softvol-max default to roughly the old maximum (roughly
doubles the gain).
Now --volume takes an absolute volume, meaning it doesn't depend on
--softvol-max. 0 is still silence, and 100 now always means unchanged
volume. The OSD and the "volume" property are changed accordingly.
Also raise the minimum value of --softvol-max. A value below 100 makes
no sense and breaks the OSD.
Adding a "yes" choice makes the option parser consider this option as a
multi-state flag option, and without argument "yes" is implicitly
selected. "yes" is made an alias for "inf", so it will loop infinitely.
As a negative side effect, the old syntax "-loop inf" does not work
anymore. Since this is ambiguous, the option parser prefers interpreting
the "inf" as filename.
Fixes#1970.
This was called for formal reasons at best. The way it does this is
somewhat dangerous, because if libmpv is unloaded as DLL, this would
attempt to call a dangling function pointer.
(No, we don't want an extra DllMain entrypoint just for win32.)
A rather dumb hack to copy the problematic rectangle textures (mandated
by VDA) into 2D ones.
(This isn't done yet for OpenGL 3.0+. We need to make sure the
performance isn't reduced too much by it.)
The code checking for the type of seeking contained some if else
statements. To improve readability, I decided to refactor those
statements to a switch statement.
Reduce the default tolerance for timestamp jumps from 60 to 15 seconds.
For .ts files, where ts_resets_possible coming from AVFMT_TS_DISCONT is
set, apply a more sophisticated heuristic. It's clear that such a file
wouldn't have a framerate below, say, 23hz. If the demuxer reports a
lower fps, we allow longer PTS jumps.
This should replace long pauses on discontinuities with .ts files with
at most a short stutter.
Of course, all kinds of things could go wrong anyway if the source is
VFR, or FFmpeg's frame rate detection fails in some other way. I haven't
found such a file yet, though.
Commit 10915000 attempted to fix wasting CPU when resyncing and no new
data was actually coming from the demuxer. The fix assumed that at this
point it would have reached the sync point, but since the code attempts
weird incremental decoding, this wasn't actually true. So it broke
seeking in addition to removing the CPU waste.
Try something else. This time, we essentially only wakeup again if
data was read (i.e. audio_decode() returned successfully).
The OSD symbol for seeking to an absolute percentage was always OSD_FFW,
even when it should be OSD_REW. It uses the correct OSD symbols now, by
checking the current position ratio.
Note: The symbol is still incorrectly given when the absolute percentage
is very close to the current position ratio. Fortunately, that's a rare
use case.
This reads the "CUESHEET" tag, and attempts to parse it as .cue data. If
any is found, the cue tracks are added as chapters.
This reuses the parser written for demux_cue.c.
Fixes#1957.
Apparently some A/V receivers do not behave well if "normal" DTS is
passed through using the high bitrate spdif format normally used for
DTS-HD (other receivers are fine with it).
Parse the first packet passed to ad_spdif by decoding it with libavcodec
in order to get the profile. Ignore the --ad-spdif-dtshd if it's not
DTS-HD. (If the codec profile changes midstream, the user is out of
luck. But this is probably an insignificant corner case.)
I thought about parsing the bitstream, but let's not. While it probably
wouldn't be that much effort, we are trying to keep it down on codec
details here - otherwise we could just do our own spdif framing instead
of using libavformat's spdif pseudo-muxer.
Another possibility, using the codec parameters signalled by
libavformat, is disregarded. Our builtin Matroska decoder doesn't do
this, and also we do not want on the demuxer having to decode some
packets in order to retrieve codec params (as libavformat does).
Fixes#1949.
Thsi code path happens during seeking. If video is still being decoded
to get to the first video frame, audio has nothing to do, as it is
synchronized against the first video frame. We only want to wake up if
there's an actual state change.
Fixes#1958.
Only absolute percentage seeking was permitted first. It is now also
possible to seek by relative percentage.
MPSEEK_FACTOR is used as seek_type.
Fixes#1950.
Signed-off-by: wm4 <wm4@nowhere>
It sometimes happens on exit, and it's probably a bad idea. If the
process hangs on exit (possibly due to stupid hardcoded timeouts it's
doing), mpv will also hang now, unfortunately.
It appears some WMs have a problem with out method of setting initial
fullscreen mode. We assume that if the window's _NET_WM_STATE includes
_NET_WM_STATE_FULLSCREEN before mapping the window, the WM will show it
as fullscreen at mapped. EWMH doesn't say anything that this should
work, although one could argue that it's implied.
In any case, since it's not standard behavior without at least some
doubt, it's probably a good idea to try the "old" method as well.
Fortunately, it should be idempotent.
See #1937, #1920.
This is pretty much copy&pasted from Libav commit
a7e0380497306d9723dec8440a4c52e8bf0263cf.
Note that if FFmpeg was not compiled with HEVC DXVA2 support or your
video drivers do not support HEVC, the player will not fallback and
just fail decoding any video. This is because libavcodec appears not
to return an error in this case. The situation is made worse by the
fact that MSYS2 is on an ancient MinGW-w64 release, which does not
have the required headers for HEVC DXVA2 support.
An attempt to get rid of the weird mix of callbacks that take either
struct vo or MPGLCopntext as parameter. This is not perfect, and the
API will probably change a bit until all other code is ported to it.
the main question is how to separate struct vo completely from the
windowing code, which actually needs vo for very little.
In the end, the legacy callbacks will be dropped.
Instead of having separate backends, make use of GLES a flag. This
reduces the number of backends and the resulting annoyances.
Also, nobody cares about using GLES, so there's no backward
compatibility either.
Before this change, Cocoa state was accessed from both the VO and the
Cocoa main thread. This was probably not a good idea. There was some
locking as well as implicit synchronization using the dispatch
mechanism, but it wasn't watertight.
Change this completely. Now Cocoa things are always accessed from the
main thread only. The old mutex falls away, as well as the
vo_cocoa_set_current_context() function, which implicitly used the lock
to coordinate VO accesses. With the new code, the VO thread generally
has to wait for the main thread, while the main thread never waits for
the VO and rarely accesses it. Fortunately, this is rather straight
forward, and most of this is achieved by making vo_cocoa_control() run
on the main thread. The logic of the code does generally not change.
Some aspects are trickier. Apparently we can't access the
NSOpenGLContext from the VO thread, because this object is not thread-
safe. We use some CGLContextObj functions instead, such as for making
the context current and swapping the buffers.
The hardware always decodes to nv12 so using this image format causes less cpu
usage than uyvy (which we are currently using, since Apple examples and other
free software use that). The reduction in cpu usage can add up to quite a bit,
especially for 4k or high fps video.
This needs an accompaning commit in libavcodec.
Arch linux is about to update to lua 5.3.x, but lua 5.2.x will be
provided by package lua52, which contains pkg-config file lua52.pc.
Signed-off-by: wm4 <wm4@nowhere>
Some code always calls vo_event(), even with event==0, which leads to
immediate wakeup, which in turn causes the function to be called again.
This would burn CPU, which was especially noticeable when paused.
Until now, we just blocked SIGPIPE globally. Fix it properly to get away
from it.
MSG_NOSIGNAL should be widely available and is part of the POSIX.1-2008
standard. But it's not available on OSX, because Apple is both evil and
retarded. Thus we continue to ignore the problem on such shitty systems.
Interrupt video timing. This means the Cocoa event loop does not have
to up to 2 video frame durations until redrawing the frame finally has
finished.
We abuse the VO event flags for this. Eventually this should use
wait_vo() or so in the video timing wait function, but for now the
interaction this would require with the code of other VOs/backends
would cause too much of a mess.
Instead of requiring a complicated mechanism to share the entire OpenGL
and renderer state between VO and Cocoa thread just to do the redrawing
during live-resize on the Cocoa thread, let the Cocoa thread wait on the
VO thread. This wil allow some major simplifications and cleanups in the
future.
One problem with this is that it can enter a deadlock whenever the VO
tries to sync with the Cocoa thread. To deal with this, the Cocoa thread
waits with a timeout. This can probably be improved later, though in
general this situation can always happen, unless the Cocoa thread waits
in a reentrant way.
Some other details aren't completely clean either. For example,
pending_events should be accessed atomically. This will also be fixed
later.
Will be used to make video waiting interruptible with Cocoa (see the
following commit).
One worry was that this could cause hangs if the system clock jumps
backwards. Normally we don't support such behavior, because it's
almost impossible to handle it reasonably. E.g. we would have to
change the default clock type for condition variables, which in turn
would require a custom function for creating condition variables,
or so. If the OS even supports different clocks.
But it turns out that this is no issue, because other events seem
to wakeup the wait call anyway, and mpv internal absolute times use
a monotonic clock.
This uses the OpenGL frame interpolation code, which before could be
used by vo_opengl only.
Some effort was made to make it behave like vo_opengl, for the better or
the worse. As a consequence, there is a minor duplication of code and
mechanism. Hopefully this can all be wiped as soon as the VO frame
queue/timing mechanism is cleaned up.
This also attempts to use mpv_opengl_cb_report_flip() (as called by the
API user) to determine the vsync interval. This might need refinement as
well.
(In general, we simply expect the API user to work in vsync-blocking
manner.)
(I have no idea why there are different modes.)
Instead of risking to drop frames too early, give it some margin. Since
there are situations this could deadlock, wait with a timeout. This can
happen if e.g. the API user is refusing to render anything, or if
uninitialization is happening.
There is not much of a reason to have these wrappers around. Use POSIX
standard functions directly, and use a separate utility function to take
care of the timespec calculations. (Course POSIX for using this weird
format for time values.)
As usual, we prefer plain C11 names and semantics, and have to emulate
them if C11 atomics are not available.
For the non-atomic fallback (which is just there to make code compile in
situations the atomic property is not overly important), we require a
gross hack to make the generic macros work without using compiler-
specific extensions.
OpenSSL and GnuTLS are still causing this problem (although FFmpeg could
be blamed as well - but not really). In particular, it was happening to
libmpv users and in cases the pseudo-gui profile is used. This was
because all signal handling is in the terminal code, so if terminal is
disabled, it won't be set. This was obviously a questionable shortcut.
Avoid further problems by always blocking the signal. This is done even
for libmpv, despite our policy of not messing with global state.
Explicitly document this in the libmpv docs. It turns out that a version
bump to 1.17 was forgotten for the addition of MPV_FORMAT_BYTE_ARRAY, so
document that change as part of 1.16.
It's useless, and creates a bogus warning in subprocess-posix.c.
Since I don't know which compilers might have it by default, just change
it to -Wno-redundant-decls.
mp_find_config_file() will print the filename lookup and its result in
verbose mode. This is wanted, but gets inconvenient when it is done for
every playlist entry (for resuming).
Lookup the watch_later subdir only once and cache the result instead.
This drops the logic for loading the resume file from other locations,
which should generally be unnecessary, though might lead to confusion if
the user has mixed old and new config paths (which the user shouldn't).
Also add a mp_find_user_config_file() function for a more
straightforward and reliable way to get actual local configpaths,
instead of possibly global and unwritable locations.
Also, for symmetry, check the resume option in mp_load_playback_resume()
just like mp_check_playlist_resume() does.
Share most of the implementation of config file lookup between
mp_find_all_config_files() and mp_find_config_file().
Also move the check for config path overrides to mp_get_platform_path()
directly. From the point of view of config file lookup, this is a bit
stupid, but on the other hand increases consistency, as user path
resolution exposes the mp_get_platform_path() functionality directly to
the user.
Now among other things panscan can be changed during playback.
Unfortunately, it flickers. The issue is that reconfig() clears the
framebuffer. Removing the clearing shows that the "unused" parts of
the picture are not cleared - even though OSD could render there. As
such, this is a separate issue.
When running with --panscan=1, this could crash - because the current
frame was reduced in size each time the image was redrawn, which would
result in a failed assertion the second time it's drawn.
This creates the window before the first file is loaded. This was
requested a bunch of times, but on the other hand a change to make this
behavior the default was reverted some time ago, because other users
hated it.
Drop mp_chmap_diff() (which is unused too now), and implement
mp_chmap_diffn() in a slightly simpler way. (Too bad there is no
standard function for counting set bits.)
Instead of somehow having 4 different cases with each their own weight,
do it with a single function that decides which channel layout is the
better fallback.
This is simpler, and also introduces new (fixed) semantics. The new test
added to test/chmap_sel.c actually works now. This is a mixed case with
no perfect upmix or downmix, but the better choice is the one which
loses the least channels from the original layout.
One test also changes. If the input is 7.1(wide-side), and the available
layouts are 7.1 and 5.1(side), the latter is now chosen instead of the
former. This makes sense: both layouts contain 6 out of 8 channels from
the original layout, but the 5.1(side) one is smaller. This follows the
general logic. The 7.1 layout has FLC/RLC speakers instead of BL/BR,
and judging by the names, "front left center" is completely different
from "back left". If these should be exchangeable, a separate exception
would have to be added.
Reuse MP_SPEAKER_ID_NA for this. If all mp_chmap entries are set to NA,
the channel layout has special "unknown channel layout" semantics, which
are used to deal with some corner cases.
The client API (libmpv) and encoding (--o) have slightly different
defaults from the command line player. Instead of doing a bunch of calls
to set the options explicitly, use profiles. This is simpler and has the
advantage that they can be listed on command line (instead of possibly
forcing the user to find and read the code to know all the details).
Sometimes, ALSA will return channel layouts with padded channels (NA
speakers). Use them instead of failing.
This still includes the old "braindeath" code to retry with a layout
without NA channels. This might be helpful for performance, and also the
padded channel layout string looks confusing.
To be fair, I have not encountered a case yet which would really need
this, and for which the old "braindeath" code did not fix it.
volatile barely means anything.
The polling is kind of bad too, but relatively harmless as device
opening/closing is a rare event, and the format change is not expected
to take long.
Remove the pointless talloc call too (must have been a leftover
from previous refactoring).
No reason to keep them separate. It's an artifact from the old
ao_coreaudio.c, which kept usage of two different APIs in the same file.
Removes a forward reference too.
This should fix some crashes due to dangling pointers.
The problem was that with_cocoa_lock_on_main_thread() is asynchronous.
It will not wait until it is finished. In the uninit case, this means
the VO could be deallocated and destroyed while cocoa was still running
uninit code.
So simply wait until it is done by using dispatch_sync(). There were
concerns that this could introduce a deadlock by the main thread trying
to wait for something on the VO thread. But from what I can see, this
never happens, and even if it does, it would crash anyway since the VO
is already gone.
One remaining worry is the video_resize_redraw_callback. From what I can
see, it still can mess things up, and will need a more elaborate fix.
Instead of trying to use af_format_conversion_score() (which tries to be
all kinds of clever), just compare the raw bits as a quality measure. Do
this because otherwise, weird formats like padded 24 bit formats will be
excluded, even though they might be the highest precision formats for
some hardware.
This means that for now, the user would have to check whether the format
is usable at all before calling ca_asbd_is_better(). But since this is
currently only used for ao_coreaudio.c and for the physical format, it
doesn't matter.
If coreaudio-exclusive should get PCM support, the best would be to
revert this change, and to add support for 24 bit formats directly.
Some time ago, a mechanism was added for automatically removing PCM-only
filters if the input format is spdif.
This could cause an infinite loop if the AO did not support spdif, but
was falling back to some PCM format. Then this code tried to remove the
last filter, which is a dummy filter for receiving and queuing filter
output. af_remove() simply fails gracefully in this case, so this
happens over and over again.
Fix by explicitly checking whether the filter to remove is a dummy
filter. (af_remove() also fails only if the dummy filters are attempted
to be removed - checking this directly is simpler.)
Move all of the channel map retrieval/negotiation code to a separate
file. This will (probably) be helpful when extending
ao_coreaudio_exclusive.c.
Nothing else changes, other than some minor cosmetics and renaming,
and changing some details for decoupling it from the ao_coreaudio.c
internals.
It appears this is the reason coreaudio-exclusive does not work without
explicitly specifying a device, even if the default device maps to
something passthrough-capable.
Instead of always picking a somehow better format over the previous one,
select a format that is equal to or better the requested format, but is
also reasonably close.
Drop the mFormatID comparison - checking the sample format handles this
already.
Make sure to exclude channel counts that can't be used.
If for example the physical format is set to stereo, the reported
multichannel layout will actually be stereo. It fixes itself only after
the physical format is changed.
Reduces (but likely does not remove) the danger of rounding intermediate
values down to 8 bit. This is important for cscale, or any other
processing that might store raw YUV values in framebuffers.
Fixes#1918.
ao_coreaudio uses AudioUnit - the OSX software mixer. In theory, it
supports multichannel audio just fine. But in practice, this might be
disabled by default, and the user is supposed to select a multichannel
base format in the "Audio MIDI Setup" utility.
This option attempts to change this setting automatically. Some possible
disadvantages and caveats are listed in the manpage additions. It is off
by default, since changing this might be rather bad behavior for a
normal application.
They are not really interesting. At least one user complained about the
noise resulting from use with shell scripts, which connect and
disconnect immediately.
If for example the audio settings are set to 5.1 output, but the
hardware does 8 channels natively (HDMI), the reported channel
layout will have 2 dummy channels. To avoid falling back to stereo,
we have to write audio in this format to the device.
Some audio APIs explicitly require you to add dummy channels. These are
not rendered, and only exist for the sake of the audio API or hardware
strangeness. At least ALSA, Sndio, and CoreAudio seem to have them.
This commit is preparation for using them with ao_coreaudio.
The result is a bit messy. libavresample/libswresample don't have good
API for this; avresample_set_channel_mapping() is pretty useless.
Although in theory you can use it to add and remove channels, you
can't set the channel counts. So we do the ordering ourselves by making
sure the audio data is planar, and by swapping the plane pointers. This
requires lots of messiness to get the conversions in place. Also, the
input reordering is still done with the "old" method, and doesn't
support padded channels - hopefully this will never be needed. (I tried
to come up with cleaner solutions, but compared to my other attempts,
the final commit is not that bad.)
ca_label_to_mp_speaker_id() checked whether the last entry was >= 0, but
actually this condition was never true, and MP_SPEAKER_ID_UNKNOWN0 is
not negative.
This was a mistake, it should definitely be using the device namespace
rather than the file namespace. As it says in the docs, all pipe names
must start with \\.\pipe\
Path expansion (like "~/dir/" in config file) was used inconsistently,
so the cache directory wasn't always created correctly. Fix this by
moving the path expansion from load_file() to its callers.
Since commit 7381db60, strings like "~desktop/" were expanded as
platform-specific paths by mpv. Apparently this similarity to standard
Unix shell expansion caused confusion, so change it to "~~desktop/". The
shell doesn't expand this, so it should be better.
And split the Cocoa and Unix cases. Simplify the Cocoa case slightly by
calling mpv_main directly, instead of passing a function pointer. Also
add a comment explaining why Cocoa needs a special case at all.
This unbreaks compiling command line player and libmpv at the same
time. The problem was that doing so silently disabled the OSX
application thing - but the command line player can not use the
vo_opengl Cocoa backend without it.
The OSX application code is basically dead in libmpv, but it's not
that much code anyway.
If you want a mpv binary that does not create an OSX application
singleton (and creates a menu etc.), you must disable cocoa
completely, as cocoa can't be used anyway in this case.
This now stores caches for multiple ICC profiles, potentially all the
user has ever used. The big use case for this is for users with multiple
monitors. The old logic would mandate recomputing the LUT and discarding
the cache whenever dragging mpv from one screen to another.
This also avoids having to save and check the ICC profile itself, since
the file name already uniquely determines it.
This will essentially make screenshot-tag-colorspace also affect the
"screenshot window" command, where possible.
Unfortunately, it's completely incompatible with icc-profile, due to API
limitations of ffmpeg (we can only give it an enum of well-known
primaries, rather than an actual ICC profile or primaries).
This should take care of the endless complaints about the default
location for screenshots (and will of course create new ones).
If the screenshot-template is set to an absolute path, the directory
won't be used. So this should be reasonably compatible.
So that the user realizes where they come from, or can find them at all.
This was a common complaint, and this is the most lazy solution. Better
suggestions for a default template are welcome.
win32 has a special function for this.
I'm not sure about OSX - it seems ~/Desktop can be hardcoded, and the
OSX GUI actually localizes the _displayed_ path in its UI.
For Unix, there is not much to be done, or is there.
Somewhat less ifdeffery, higher flexibility. Now there are 3 separate
config file resolvers for 3 platforms (unix, win, osx), and they can
still interact with each other somewhat. For example, OSX for now uses
most of Unix, but adds the OSX bundle path.
This can be extended to resolve very specific platform paths, such as
location of the desktop.
Most of the Unix specific code moves to path-unix.c.
The behavior should be the same - if not, it is likely a bug.
(Not sure why it worked without this when I tested the previous
changes.)
Untested, but should be fine. This is equivalent what is done on e.g.
panscan changes.
I think this used to be quite important, because the ancient VfW support
in MPlayer used to output flipped frames. This code has been dead in mpv
for quite some time (because VfW decoders were removed, and the --flip
option was dropped too), so get rid of it.
Currently, the wayland backend needs extra work to avoid drawing more
often than the wayland frame callback allows. (This is not ideal, but
will be fixed at a later time.)
Unify this with the start_frame callback added for cocoa. Some details
change for the better. For example, if a frame is dropped, and a redraw
is done afterwards, the actually correct frame is redrawn, instead
whatever was in the textures from before the dropped frame.
With --idle --force-window, or when started from the bundle, the cocoa
code dropped the first frame. This resulted in a black frame on start
sometimes.
The reason was that the live resizing/redrawing code was invoked, which
simply set skip_swap_buffer to false, blocking redrawing whatever was
going to be rendered next. Normally this is done so that the following
works:
1. vo_opengl draw a frame, releases GL lock
2. live resizing kicks in, redraw the frame
3. vo_opengl wants to call SwapBuffers, drawing a stale buffer
overwritten by the live resizing code
This is solved by setting skip_swap_buffer in 2., and querying it in 3.
Fix this by resetting the skip_swap_buffer at a known good point: when
vo_opengl starts drawing a new frame.
The start_frame function returns bool, so that it can be merged with
is_active in a following commit.
Commit f1746741de changed the drop
logic to have more slack (drop more frames but less frequent) to prevent
drops due to timing jitter when the clip and screen have similar rates.
However, if the clip has higher rate than the screen (or just higher
playback rate), then that policy hurts smoothness since these "chunked
drops" look worse than one frame drop at a time.
This patch restores the old drop logic when the playback frame rate is
higher than ~5% above the screen refresh rate, and solves this issue.
Fixes#1897
Also factor the display size initialization into a separate function.
For some reason this seems to work, although setting the background
color using this 1x1 pixel bitmap does not work. I blame the RPI
beign a terrible piece of hardware with even worse drivers.
This should for now be equivalent; it's merely more explicit and will
be required if we add PCM support.
Note that the property listeners actually tell you what property
exactly changed, but resolving the current listener mess would be too
hard. So check for changes manually.
Useful with some of the following commits.
ca_fill_asbd() should behave exactly as before.
Instead of actually implementing the inverse function of ca_fill_asbd(),
just loop over the (small) list of mpv functions and check if any mpv
equivalent to a given ASBD exists.
kAudioFormatFlagIsSignedInteger implicates that it's only used with
integer formats. The mpv internal flag on the other hand signals the
presence of a sign, and this is set on float formats.
Until now, this probably worked fine, because at least AudioUnit is
ignoring the uncorrect flag.
It's weird that this basically adjusts the contrast between luma and
chroma, and not blackness.
This is more in line with the behavior of libswscale, the vdpau
"procamp" (which mpv doesn't use), and Xv.
This code does not know whether the stream supports reconnecting until
STREAM_CTRL_RECONNECT is called. So the message should be printed after
it. To avoid that reconnects that succeed on the first try go unnoticed,
print a warning on success.
Right now, the default behavior is to pick the numerically lowest screen
ID that overlaps the window in any way - but this means that mpv will
decide to pick an ICC profile in a pretty arbitrary way even if the
window only overlaps another screen by a single pixel.
The new behavior is to query it based on the center of the window
instead.
In the most simple case, this prevents the track ID from changing. One
disadvantage is that if the file fails loading, the track is gone for
good and would have to be re-added explicitly by the user.
Now the rescan_external_files command will by default reselect the audio
and subtitle streams. This should be more intuitive.
Client API users and Lua scripts might break, but can be fixed in a
backward-compatible way by setting the mode explicitly.
Should help with debugging, and might be slightly more userfriendly.
Note that this is called manually in multiple entry-points, instead of
the functions doing the actual work (like mp_remove_track()). This is
done so that exiting the player or calling the sub_reload command won't
print redundant in-between states.
During seeking, and there is momemtarily no new data available yet, the
player will display the seek target as current time. Clamp this time to
the known time range as implied by the start time and the duration of
the file.
This improves behavior especially when seeking in audio files, for which
this for some reason triggers rather often. There were some users
complaining about this.
This makes behavior worse for files with timestamp resets, or
incorrectly reported duration. (The latter is relatively common,
e.g. libavformat shortcomings, or incomplete files.)
We always want to prefer upmix to downmix, as long as it makes sense.
Even if the upmix is not "perfect" (not just adding channels), we want
to prefer the upmix.
Cleanup for commit d3c7fd9d.
We escape only characters below 32, plus " and \. UTF-8 should be apssed
through verbatim. Since char can be signed (and usually is), the check
broke and happened to escape UTF-8 encoded bytes too. This broke UTF-8
completely.
Note that we don't check for broken or invalid UTF-8, such as described
both in the client API and IPC docs.
Fixes#1874.
This will change the format option to "bestaudio/best" instead of
passing the "-x" argument to yt-dl.
Prevents the video still being downloaded in the new mpv versions where
the yt-dl format is set to "best" by default.
As indicated by the added test. In this case, fallback and downmix have
the same score, but fallback happens to give better results. So prefer
fallback over downmix.
(This is probably not a correct solution.)
This could make the player crash on exit if the "sub_reload" command was
used successfully. the reason was that the mpctx->sources array could
have dangling pointers to the unloaded demuxers.
Also fix a memory leak by actually always freeing the per-stream
subtitle decoders (which are a hack to make ordered chapters behave
better).
If the --ytdl-format option is not used, force the "best" format. Do
this because youtube-dl is going to change its default format to one
that will trigger the (partially broken) DASH support in our own code.
Fixes#1867.
vo_opengl (or gl_hwdec_vdpau.c to be specific) calls
mp_vdpau_mixer_render() with video_rect=NULL, which means to use the
full surface. This is incorrect if the surface is actually cropped, as
it can happen with h264. In this case, it was rendering the parts
outside of the image.
Fix it by making this case use the cropped size instead.
Alternative fix for PR #1863.
This was in the "Window" section. It has absolutely nothing to do with
windows. Move it to the "Miscellaneous" section instead. The "--mc"
option, which has a similar function, was already there.
Fixes PNG cover art not showing up immediately (for example when running
with --pause).
libavformat exports embedded cover art as a single packet. For example,
a PNG attachment simply contains the PNG image, which can be sent to the
decoder. Normally you would expect that the PNG decoder would return 1
frame for 1 packet, without any delays. But this stopped working, and it
incurs a 1 frame delay.
This is perfectly legal (even if unexpected), so let our code feed the
decoder packets until we get something back. (In theory feeding the
packet instead of a real flush packet is still somewhat questionable.)
last_av_difference can be MP_NOPTS_VALUE under certain circumstances
(like no video timestamp yet). This triggered the desync message,
because fabs(MP_NOPTS_VALUE) is quite a large value. We don't want to
show a message in this situation.
The build failed because rst2pdf apparently has problems with
page breaks. In this case, the link to the ALSA upmix guide was
causing a page break in an admonition block. My guess is that
rst2pdf screws up when it can’t fill at least one line of text
following a page break, so I worked around this by making that
paragraph a little longer. Seems to do the trick.
I also shortened the URL using GitHub’s service because it was
causing some rather unsightly formatting in the manpage output.
Maybe we should just build HTML instead of a PDF.
Since joystick support was removed and is a difference from mplayer, it
should be included in the document with the mplayer changes.
It will help new users who were using mplayer's joystick support to
seek alternatives when switching to mpv. It will also be helpful for
people that had problems with the joystick support in mplayer (for
example, by incorrectly recognizing other input devices as joystick)
to know that those problems won't persist in mpv.
MP_IMGFIELD_TOP/MP_IMGFIELD_BOTTOM were completely unused, and
MP_IMGFIELD_ORDERED was always set (even though vf_vdpaupp.c strangely
checked for the latter).
This was traditionally needed to silence terminal output from errors
during command line parsing preparsing. Preparsing is done so that
options controlling the terminal and config files are parsed and applied
first, with a second command line parsing pass applying all other
options, _and_ printing error messages for the preparsed ones.
But the hack silencing log output during the preparse pass is actually
not needed anymore, since the terminal is enabled only after preparsing
is finished. update_logging() in main.c does this.
So as long as update_logging() is called before
m_config_preparse_command_line(), this will work.
The options don't change, but they're now declared and used privately by
demux_mkv.c. This also brings with it a minor refactor of the subpreroll
seek handling - merge the code from playloop.c into demux_mkv.c. The
change in demux.c is pretty much equivalent as well.
This change allows forward seeking even if there are no more video
keyframes in forward direction. This helps with files that e.g. encode
cover art as a single video frame (within a _real_ video stream - ffmpeg
seems to like to produce such files). Seeking backwards will still jump
to the nearest video frame, so this improvement has limited use.
The old code didn't do this because of the logic the min_diff variable
followed. Instead of somehow using the timestamp of the last packet read
for min_diff, use the first index entry for it. This actually makes it
fall back to the first/last index entry as the (removed) comment claims.
Note that last_pts is basically random at this point (because the
demuxer can be far ahead of playback position), so this didn't make
sense in the first place.
An approximate measure to make it exit possibly slightly earlier.
Relatively speaking, some time will pass between cancellation and
the cache actually being requested to exit, so it's good if the
cache returns EOF immediately.
The caller can check for cache interruption instead. There's no need to
define special return values and such. It would be rather hard to make
waiting for the condition and stream cancellation atomic too (and
pointless, since the underlying stream will also be "cancelled" and exit
early), so nothing about cancellation being a separate call will change.
This put some effort into distinguishing between two messages to print -
all worthless. Even more so, this kept printing the message, which
doesn't feel overly useful either. (The message will be printed
repeatedly anyway if network recovers for a while and then gets stuck
again.)
All in all, the demuxer cache triggering the buffering state does a
better job here. But don't remove it completely, since knowing that the
network did nothing for a relatively short time is still useful.
It's annoying and unnecessary. They can be manually loaded if really
needed (for things like previewing).
Also remove the #if. It was for suppressing warnings, and we don't need
to be so careful about this in the relatively obscure encoding mode.
Approximate time of video buffered in the demuxer, in seconds. Same as
`demuxer-cache-duration` but returns the last timestamp of bufferred
data in demuxer.
Signed-off-by: wm4 <wm4@nowhere>
If user switched terminals frantically, mpv could get SIGUSR1 twice in a
row, which, up until now, resulted in destroying CRTC twice. This caused
it to segfault. After this fix, double SIGUSR1 should be ignored.
This will be used in the following commit, which adds screenshot_raw.
The reasoning is that this will be better for binding scripting
languages.
One could special-case the screenshot_raw commit and define fixed
semantics for passing through a pointer using the current API, like
formatting a pointer as string. But that would be ridiculous and
unclean.
On EOF, this stopped reporting the actual cache duration, and just
signalled unknown duration. Fix this and keep reporting whatever is left
in the packet queue.
This reverts commit 5438a8b3. The commit doesn't give a good explanation
as to why it is needed, but I guess it was because the reporting was
imperfect (it switched between unknown or 0, and the correct duration).
This also removes a line added in commit 848546f2. The line is
ds->active = false;
The "active" flag basically says that data from this stream is actively
needed, and it's used to calculate the minimum data that can actually be
played (approximately). If this were ignored, a sparse subtitle stream
would set the cache duration to 0s. The commit message adding the line
says "actually does nothing, but in theory it's cleaner". Well, screw
it.
Vapoursynth made an incompatible API change: clips with unknown length
are not supported anymore. In fact, Vapoursynth abort()s the program
(which by the way invalidate all of its claims of API/ABI stability).
So add some nonsense to make it work again.
Remove the old implementation for these properties. It was never very
good, often returned very innaccurate values or just 0, and was static
even if the source was variable bitrate. Replace it with the
implementation of "packet-video-bitrate". Mark the "packet-..."
properties as deprecated. (The effective difference is different
formatting, and returning the raw value in bits instead of kilobits.)
Also extend the documentation a little.
It appears at least some decoders (sipr?) need the
AVCodecContext.bit_rate field set, so this one is still passed through.
This prevents the machine from going to sleep while a video-only stream
is playing. When audio is playing, the audio stack should make this
request for us.
Useful for dealing with libavfilter's terrible graph syntax.
Not strictly backwards compatible (for example "[a[b]" fails now - the
"[" within the quote is interpreted now). But hopefully it's obscure
enough not to warrant any kind of compatibility hacks.
Logging was meant to be silenced only when user uses connector
auto-detection feature. If user supplies connector ID manually, he
should see exact reason why connecting to this specific connector
failed.
If a directory is encountered, replace it with its contents in the
internal playlist.
This is messed into demux_playlist.c, because why not. STREAMTYPE_DIR
could be avoided by unconditonally trying opendir() in demux_playlist.c,
but it seems nicer not to do weird things like calling it on real files.
This does not work on Windows, because msvcrt is retarded.
It's entirely useless, especially now that vo.c handles screenshots in a
generic way, and requires no special VO support. There are some
potential weird use-cases, but actually I've never seen it being used.
The old behavior does not make too much sense after all. If you don't
want to file to be overwritten, the user can check this manually.
This is a change in behavior - let's hope nobody actually relied on it.
libavcodec makes it impossible to distinguish dropped frames (requested
with AVCodecContext.skip_frame), and cases when the decoder simply does
not return a frame by default (such as with VP9, which has invisible
reference frames).
This confuses users when decoding VP9 video. It's basically a cosmetic
issue, so just paint it over by ignoring them if framedropping is
disabled.
It appears youtube-dl sometimes asks for a password on stdin. This won't
work, because mpv already uses the terminal.
(I wonder if this could be simpler, like simply closing FD 0, but let's
not. The FD would be reused by something random.)
We already use 2 screensaver APIs when attempting to disable the
screensaver: XResetScreenSaver() (from xlib) and XScreenSaverSuspend
(from the X11 Screen Saver extension). None of these actually work.
On modern desktop Linux, we are expected to make dbus calls using some
freedesktop-defined protocol (and possibly we'd have to fallback to a
Gnome specific one). At least xscreensaver doesn't respect the "old"
APIs either.
Solve this by running the xdg-screensaver script. It's a terrible, ugly
piece of shit (just read the script if you disagree), but at least it
appears to work everywhere. It's also simpler than involving various
dbus client libraries.
I hope this can replace the --heartbeat-cmd option, and maybe we could
remove our own DPMS/XSS code too.
It seems this choice was never documented. "always" is actually older
than "yes", so just declare it a compatibility value for "yes". (Also
move it before "always" in the C code to make this clear.)
mp_chmap_from_channels_alsa() doesn't always succeed - there are a bunch
of channel counts for which no defined ALSA layout exists. Fallback to
stereo in this case. (Normally, this code path shouldn't happen at all.)
When playing cover art, it conceptually reaches EOF as soon as the image
was put on the VO, causing the EOF message to be repeated every time new
audio was decoded. Just silence the message.
This was mapped to a broken command, so it did nothing but printing an
error message. The intended binding (cycling OSD level) doesn't seem to
useful either, so just drop it.
It was ignored before. Passing an argument makes no sense, and might be
mistaken for some form of --vf-del, so complain about it.
This also affects --af-clr and the vf/af commands.
This is optional, but ensures that linking with -Wl,--as-needed does
not drop the MMAL VC driver. The driver normally "registers" itself
in the library constructor, but since no symbols are explicitly
referenced, the linker could remove it with as-needed enabled.
I tried to find that option by searching for terms like “cover art”
and got nothing. I imagine most users would look for similar terms.
Hope this helps.
It could happen that a lavrresample filter would keep its old output
format when the decoder changed its output format. This simply happened
because the output format was never reset.
Normally, this was not an issue, because lavrresample filters only
inserted for format conversion were removed on format changes. But if
--no-audio-pitch-correction is set and playback speed is changed, then
there is a "permanent" lavrresample filter in the filter chain, which
shows this behavior.
Fix by explicitly resetting output formats for all filters which support
it.
Note: this can crash with libswresample in some cases. I'm not sure if
this is mpv's fault or libswresample's, but since it works with
libavresample, I'm going to assume it's not our's.
The gender specific pronoun is changed, since we shouldn't assume the
gender of the user.
The sentence itself is also changed to be more correct in general.
Not sure why this was so roundabout; probably to keep spam down if the
user's OpenGL drivers are crap (but then just don't enable extended
features), or because the "Disabling..." text was so repetitious.
But there doesn't seem to be a good reason after all. Also, this could
already overflow the fixed size disabled[] array. Just print the
messages directly.
Apparently the standard handles can be set to bogus values on XP. Use
GetFileType to check whether they refer to an actual file/pipe/etc. The
logic used by is_valid_handle() is now pretty similar to what the CRT
uses to check for valid stdio handles.
If mpv is started from Explorer or the Start Menu, it will have no
console and no standard IO handles. In this case, it's fairly safe to
enable the pseudo-gui profile.
Previously, mpv.exe used the --terminal option to decide whether to
attach to the parent process's console, which made it impossible to tell
whether mpv would attach to the console before the config files were
parsed. Instead, make mpv always attach to the console when launched
from the console wrapper (mpv.com) and never attach otherwise. This will
be useful for the next commit, which will use the presence of the
console to decide whether to use the pseudo-gui profile.
This change should also be an improvement in behavior. The old code
would attach to the parent process's console, regardless of whether it
was mpv.com or some other program like cmd.exe. This could be confusing,
since mpv.exe is marked as a Windows GUI program and shouldn't write
text to its parent process's console when launched directly. (See #768.)
Visual Studio does something similar with its devenv.com wrapper.
devenv.exe only attaches to the console when launched from devenv.com.
Add a platform-specific entry-point for Windows. This will allow some
platform-specific initialization to be added without the need for ugly
ifdeffery in main.c.
As an immediate advantage, mpv can now use a unicode entry-point and
convert the command line arguments to UTF-8 before passing them to
mpv_main, so osdep_preinit can be simplified a little bit.
Because gcc (and clang) is a goddamn PITA and unnecessarily warns if
the universal initializer for structs is used (like mp_image x = {})
and the first member of the struct is also a struct, move the w/h
fields to the top.
Remove --keep-open. Switch to --idle=once. This effectively makes the
player quit after end of playback, but still shows the idle screen if it
was started with no files.
Starting the command line player with --no-terminal, the terminal was
sitll initialized. This happened because update_logging() used the
option value before the options were parsed. Fix by moving down the
initialization to before the point where it's actually needed.
They are redundant. They were used by draw_bmp.c only, and only in a
special code path that 1. used fixed image formats, and 2. had image
sized perfectly aligned to chroma boundaries (so computing the chroma
width/height is trivial).
This could help in cases where the DWM (Windows desktop compositor) adds another
layer of bufferring and therefore the SwapBuffers timing could get messed up.
Signed-off-by: wm4 <wm4@nowhere>
on my windows system this allows smoothmotion to work perfectly also in windowed
mode. There's no real right or wrong here, with the the only goal being to
always hit the next vsync. however, on cases where vsync timing is jittery (as
could happen with DWM), this patch tries to aim to the middle of the vsync cycle
to get as least affected as possible by such jitter.
adds 1 vsync interval "slack" before deciding to drop the first frame. it should
help on cases of timing jitter (sleep duration, container timestamps, compositor
vsync timing, etc). once the drop threshold has been crossed, it will keep
dropping until perfect timing alignment. this prevents crossing the drop
threshold back and forth repeatedly and therefore more resilient to frame drops
When setting options like --no-video, ytdl_hook adds the "-x" argument
to youtube-dl, so that bandwith is saved by not downloading the video on
some sites.
Increase the default queue size. This helps with "missed" frames due to
the asynchronous nature of the API. All the other VOs are synchronous,
so if rendering and displaying takes a while, the common code in vo.c
will be blocked until it can continue. But with opengl-cb, vo.c might
immediately push the next ready frame, which causes the current frame
to be dropped _if_ it wasn't rendered yet.
One could fix this by making vo.c wait a while (until the API user calls
the render function, which pulls the frame). But setting the default
queue size to 2 seems much simpler: instead of dropping the frame, it
will be pushed to the API user once the next renderer call finishes.
(This is still a bit strange, and will hopefully be cleaned up when
video scheduling is redone, but for now this appears to deliver
relatively good results.)
Prefer the builtin one again. libavcodec still uses the ASS packet
format that uses inline timestamps, so the packet timestamps are
ignored. This again leads to additional rounding of timestamps, because
the ASS storage format only has 10ms resolution (instead of 1ms
resolution like libass). This again can lead to unintentional overlaps
when converting subtitles. The internal MicroDVD converter avoids this,
because it always uses packet timestamps.
This seems to come up often. I guess '.' vs. ':' for Lua calls is
confusing, and this part of the scripting API is the only one which
requires using it.
The af_add() function has a problem: if the inserted filter returns
AF_DETACH during init, the function will have a dangling pointer. Until
now this was avoided by making sure none of the used filters actually
return AF_DETACH, but it's getting infeasible.
Solve this by requiring passing an unique label to af_add(), which is
then used instead of the pointer.
Silence the usually user-visible warning about unsupported channel maps.
This might be an ALSA bug, but ALSA will never fix this behavior anyway.
(Or maybe it's a feature.)
Log some other information that might be useful.
Only reinit filters if it's actually needed. This is also slightly
easier to understand: if you look at the code, it should now be more
obvious why a reinit is needed (hopefully).
The message log level shouldn't get to decide whether something fails
or not. So replace the fatal error check on the verbose output code
path with a warning.
Avoids a confusing message printed by the vdpau code when taking a
screenshot while using software decoding (because obviously GPU readback
won't work on normal in-memory video frames).
There still might be FFmpeg demuxers which mess up if audio is disabled
(like it happened to the FLV demuxer), but these are bugs and shouldn't
happen.
This matters for png screenshots. We used to hardcode rgb24, but
libavformat's png encoder can do much more. Use the image format list
provided by the encoder, and select the best format from it (according
to the source format).
As a consequence, rgb48 (i.e. 16 bit per component) will be selected if
the source format is e.g. 10 bit yuv. This happens in accordance to
FFmpeg's avcodec_find_best_pix_fmt_of_list() function, which assumes
that 16 bit rgb should be preferred for 10 bit yuv.
This also causes it to print this message in this case:
[ffmpeg] swscaler: full chroma interpolation for destination format 'rgb48be' not yet implemented
I'm not 100% sure whether this is a problem.
Use texture-from-pixmap instead of vaapi's "native" GLX support.
Apparently the latter is unused by other projects. Possibly it's broken
due that, and Intel's inability to provide anything non-broken in
relation to video.
The new code basically uses the X11 output method on a in-memory pixmap,
and maps this pixmap as texture using standard GLX mechanisms. This
requires a lot of X11 and GLX boilerplate, so the code grows. (I don't
know why libva's GLX interop doesn't just do the same under the hood,
instead of bothering the world with their broken/unmaintained "old"
method, whatever it did. I suspect that Intel programmers are just
genuine sadists.)
This change was suggested in issue #1765.
The old GLX support is removed, as it's redundant and broken anyway.
One remaining issue is that the first vaPutSurface() call fails with an
unknown error. It returns -1, which is pretty strange, because vaapi
error codes are normally positive. It happened with the old GLX code
too, but does not happen with vo_vaapi. I couldn't find out why.
We handle picking out font attachments by mime type ourselves in a
higher level, so we really just want to use the mimetype. Also, Matroska
is currently the only code in libavformat which uses the fonts at all,
and we can drop use of the codec IDs completely.
This is a peculiar filter I stumbled upon while playing around with
windows, it removes aliasing almost completely while not ringing at all.
The downside is that it's quite blurry, but at high resolutions it's not
so noticeable.
This merges all of the scaler-related options into a single
configuration struct, and also cleans up the way they're passed through
the code. (For example, the scaler index is no longer threaded through
pass_sample, just the scaler configuration itself, and there's no longer
duplication of the params etc.)
In addition, this commit makes scale-down more principled, and turns it
into a scaler in its own right - so there's no longer an ugly separation
between scale and scale-down in the code.
Finally, the radius stuff has been made more proper - filters always
have a radius now (there's no more radius -1), and get a new .resizable
attribute instead for when it's tunable.
User-visible changes:
1. scale-down has been renamed dscale and now has its own set of config
options (dscale-param1, dscale-radius) etc., instead of reusing
scale-param1 (which was arguably a bug).
2. The default radius is no longer fixed at 3, but instead uses that
filter's preferred radius by default. (Scalers with a default radius
other than 3 include sinc, gaussian, box and triangle)
3. scale-radius etc. now goes down to 0.5, rather than 1.0. 0.5 is the
smallest radius that theoretically makes sense, and indeed it's used
by at least one filter (nearest).
Apart from that, it should just be internal changes only.
Note that this sets up for the refactor discussed in #1720, which would
be to merge scaler and window configurations (include parameters etc.)
into a single, simplified string. In the code, this would now basically
just mean getting rid of all the OPT_FLOATRANGE etc. lines related to
scalers and replacing them by a single function that parses a string and
updates the struct scaler_config as appropriate.
This makes the core much more elegant, reusable, reconfigurable and also
allows us to more easily add aliases for specific configurations.
Furthermore, this lets us apply a generic blur factor / window function
to arbitrary filters, so we can finally "mix and match" in order to
fine-tune windowing functions.
A few notes are in order:
1. The current system for configuring scalers is ugly and rapidly
getting unwieldy. I modified the man page to make it a bit more
bearable, but long-term we have to do something about it; especially
since..
2. There's currently no way to affect the blur factor or parameters of
the window functions themselves. For example, I can't actually
fine-tune the kaiser window's param1, since there's simply no way to
do so in the current API - even though filter_kernels.c supports it
just fine!
3. This removes some lesser used filters (especially those which are
purely window functions to begin with). If anybody asks, you can get
eg. the old behavior of scale=hanning by using
scale=box:scale-window=hanning:scale-radius=1 (and yes, the result is
just as terrible as that sounds - which is why nobody should have
been using them in the first place).
4. This changes the semantics of the "triangle" scaler slightly - it now
has an arbitrary radius. This can possibly produce weird results for
people who were previously using scale-down=triangle, especially if
in combination with scale-radius (for the usual upscaling). The
correct fix for this is to use scale-down=bilinear_slow instead,
which is an alias for triangle at radius 1.
In regards to the last point, in future I want to make it so that
filters have a filter-specific "preferred radius" (for the ones that
are arbitrarily tunable), once the configuration system for filters has
been redesigned (in particular in a way that will let us separate scale
and scale-down cleanly). That way, "triangle" can simply have the
preferred radius of 1 by default, while still being tunable. (Rather
than the default radius being hard-coded to 3 always)
The format doesn't change. Some details are different, though. For
example, it will now accept option values with spaces even if they're
not quoted. (I see no reason why the user should be forced to add
quotes.)
The code is now smaller and should be much easier to extend. It also
can load config from in-memory buffers, which might be helpful in the
future.
read_file() should eventually be replaced with stream_read_complete().
But since the latter function may access options under various
circumstances, and also needs access to the mpv_global struct, there
is a separate implementation for now.
* unify passthrough and pcm exclusive mode format setting/testing
* set passthrough format parameters correctly
* support all of mpv's existing passthrough formats
* automatically test passthrough with exclusive mode and enable
exclusive if it succeeds, even if it was not explictly requested.
this obviates the need for --ao=wasapi,wasapi=exclusive
* if passthrough fails (such as the device doesn't support the
format), fallback to either exclusive pcm or shared mode depending
on what the user specified. Right now this isn't very useful as
it still fails due to the decoder path remainin stuck on spdif.
fixes#1742
Use OPT_CHOICE_C() instead of the custom parser. The functionality is
pretty much equivalent.
(On a side note, it seems --video-stereo-mode can't be removed, because
it controls whether to "reduce" stereo video to mono, which is also the
default. In fact I'm not sure how this should be handled at all.)
The standard say:
NoDisplay means "this application exists, but don't display it in the menus".
Now that --force-window --idle is enabled, we can remove it.
Closes#1755.
libswresample doesn't normalize when remixing to a float format. This
will cause clipping due to float samples being out of the allowed range.
Fortunately this extremely bad default can be changed.
This does not happen with libavresample: it normalizes by default.
Fixes#1752.
This can be set to select a number of default settings that help mpv
pretend that it has a GUI.
I haven't decided yet whether I really want to use the profile mechanism
for this. There are a number of weird details that are not so easy to
handle with profiles, such as disabling pseudo-gui mode again (you can't
unset profiles directly). So this might change. But for now it will do.
There also should be a better way to store builtin profiles.
Unfortunately, the old crappy MPlayer config file parser needs on-disk
files, so just use a bunch of function calls for now.
It simply doesn't work, and is hard to make work. Lua 5.3 is a different
language from 5.1 and 5.2, and is different enough to make adding
support a major issue. Most importantly, 5.3 introduced integer types,
which completely mess up any code which deals with numbers.
I tried to make this a compile time check, but failed. Still at least
try to avoid selecting the 5.3 pkg-config package when the generic "lua"
name is used (why can't Lua upstream just provide an official .pc
file...). Maybe this actually covers all cases.
Fixes#1729 (kind of).
Per-app volume would change the volume across all instances of the same
application, while a private volume control (HAS_PER_APP_VOLUME)
obviously should influence only one instance/audio stream only.
Take advantage of the fact that list_devs is called with a
hotplug_inited ao. Also eliminate unnecessary nested function
abstraction of hotplug_(un)init and list_devs. However, keep list_devs
in ao_wasapi_utils.c since it uses the private functions get_device_id,
get_device_name and exposing these would require including headers for
IMMDevice in ao_wasapi_utils.h.
Create a second copy of the change_notify structure for the hotplug
ao. change_notify->is_hotplug distinguishes the hotplug version from
the regular one monitoring the currently playing ao. Also make the
change notification less verbose now that there might be two of them around.
Rarely used and essentially useless. The only VO for which this was
implemented correctly and for which this did anything was vo_xv, but you
shouldn't use vo_xv anyway (plus it support BT.601 only, plus a vendor
specific extension for BT.709, whose presence this function essentially
reported - use xvinfo instead).
Remove the colorspace-related top-level options, add them to vf_format.
They are rather obscure and not needed often, so it's better to get them
out of the way. In particular, this gets rid of the semi-complicated
logic in command.c (most of which was needed for OSD display and the
direct feedback from the VO). It removes the duplicated color-related
name mappings.
This removes the ability to write the colormatrix and related
properties. Since filters can be changed at runtime, there's no loss of
functionality, except that you can't cycle automatically through the
color constants anymore (but who needs to do this).
This also changes the type of the mp_csp_names and related variables, so
they can directly be used with OPT_CHOICE. This probably ended up a bit
awkward, for the sake of not adding a new option type which would have
used the previous format.
It was "by design" possible to make mpv crash if the parameters didn't
make enough sense, like "format=rgb24:yuv420p". While forcing the format
has some minor (rather questionable) use for debugging, allowing it to
crash is just stupid.
It was already accidentally used unconditionally by command.c.
Apparently this worked well for us, so don't change anything about,
but should it be unavailable, fail at configure time instead of compile
time.
This requires FFmpeg git master for accelerated hardware decoding.
Keep in mind that FFmpeg must be compiled with --enable-mmal. Libav
will also work.
Most things work. Screenshots don't work with accelerated/opaque
decoding (except using full window screenshot mode). Subtitles are
very slow - even simple but huge overlays can cause frame drops.
This always uses fullscreen mode. It uses dispmanx and mmal directly,
and there are no window managers or anything on this level.
vo_opengl also kind of works, but is pretty useless and slow. It can't
use opaque hardware decoding (copy back can be used by forcing the
option --vd=lavc:h264_mmal). Keep in mind that the dispmanx backend
is preferred over the X11 ones in case you're trying on X11; but X11
is even more useless on RPI.
This doesn't correctly reject extended h264 profiles and thus doesn't
fallback to software decoding. The hw supports only up to the high
profile, and will e.g. return garbage for Hi10P video.
This sets a precedent of enabling hw decoding by default, but only
if RPI support is compiled (which most hopefully it will be disabled
on desktop Linux platforms). While it's more or less required to use
hw decoding on the weak RPI, it causes more problems than it solves
on real platforms (Linux has the Intel GPU problem, OSX still has
some cases with broken decoding.) So I can live with this compromise
of having different defaults depending on the platform.
Raspberry Pi 2 is required. This wasn't tested on the original RPI,
though at least decoding itself seems to work (but full playback was
not tested).
There's actually no reason why we should assert. It's unexpected and
"should" not happen, but actually there are several ways to make it
happen.
Still, add a check m_config_get_co(), to avoid matching pseudo-entries
with no name.
Currently, the code just skipped CMS completely. This commit treats them
as sRGB by default, instead.
This also refactors much of the color management code to make it more
generalized and re-usable.
Mktemp on BSDs (at least on OpenBSD) don't have "--tmpdir" option.
Set temporary directory to /tmp by default. Use of ${TMPDIR:=/tmp} allows
overriding temporary directory via environment if needed. (And is indeed needed
in OpenBSD ports infrastructure.)
Signed-off-by: wm4 <wm4@nowhere>
More clearly separate the exclusive and shared mode format discovery.
Make the exclusive mode search more systematic in particular about
channel maps (i.e., use chmap_sel). Assume that the same sample format
/ sample rates work for all channels to narrow the search space.
The code actually uses blocking mode, so opening sound device in non-blocking
mode results in choppy sound. Also, inflating the buffer isn't necessary in
blocking mode, so the function may simply return without doing anything.
This is a minor precaution, because in some cases the number of shader
programs can already hit 10. (chroma merging + separated cscale/scale +
+ sub blending (interpolated) + sub blending (non-interpolated)
+ output (interpolated) + output (non-interpolated) + OSD)
This has a number of user-visible changes:
1. A new flag blend-subtitles (default on for opengl-hq) to control this
behavior.
2. The OSD itself will not be color managed or affected by
gamma controls. To get subtitle CMS/gamma, blend-subtitles must be
used.
3. When enabled, this will make subtitles be cleanly interpolated by
:interpolation, and also dithered etc. (just like the normal output).
Signed-off-by: wm4 <wm4@nowhere>
Bilinear scaling is not a suitable default for something named "hq"; the
whole reason this was done in the past was because cscale used to be
obscenely slow. This is no longer the case, with cscale being nearly
free.
Handling this perfectly with VapourSynth is probably not possible: you
either need to tell it the total number of input frames in advance, or
deliver an infinite stream. With playback, EOF can happen at an
unpredictable point, for which the VapourSynth API has no mechanism at
all. We handle EOF by returning an error to the filter, which will the
filter return all pending frame callbacks.
We still can try to handle it approximately: if the filter requests a
frame past EOF, then send it an error. This seems to work relatively
well with filters which don't request future frames.
With the previous commit, we have no need anymore to check a part of an
extension string (for ignoring a prefix). So check the extension string
properly, instead of just using the broken old strstr() method, which
could accidentally ignore prefixes or suffixes. Do this by extending
the check to whether the extension name is properly delimited by spaces
or string start/end.
Instead of somehow looking for the substring "_swap_control" and trying
to several arbitrary function names, do it cleanly. The old approach has
the problem that it's not very exact, and may even load a pointer to a
function which doesn't exist. (Some GL implementations like Mesa return
function pointers even the functions which don't exist, and calling them
crashes.)
I couldn't find any evidence that glXSwapInterval, wglSwapIntervalSGI,
or wglSwapInterval actually exist, so don't include them. They were
carried over from MPlayer times.
To make diagnostics easier, print a warning in verbose mode if the
function could not be loaded.
Drop support for GL_EXT_framebuffer_object. It has 2 problems: semantics
might be slightly different from the "proper" GL_ARB_framebuffer_object
extension (but is likely completely untested), and also our extension
loader is too dumb to load the same group of function pointers
represented by different extensions only once.
This broke with PulseAudio: when changing some audio filters (like for
playback speed), mixer_reinit_audio() was called - and it overwrote the
volume with whatever mpv thought the volume was before. If the volume
was changed externally before and while mpv was running, this would
reset the volume to the old value.
Fixes#1335.
Pretty messy, which is why it wasn't done at first. (Also, I'd really
like to have simpler syntax and semantics for this damn option, but who
knows when this will happen.)
Fixes#1705.
With a recent cleanup, rar support was stuffed into demux_playlist.c
(because "opening" rar files pretty much just lists archive contents and
adds them to a playlist using a special rar:// protocol, which will
actually access the rar file contents).
Since demux_playlist.c is probed _after_ demux_lavf.c (and should/must
be), libavformat was given the chance to detect DTS streams embedded
within the rar file. This is not really what we want, and a regression
what happened before rar listing was moved to demux_playlist.c.
Fix it by moving the rar listing into its own pseudo-demuxer, and let ir
probe before demux_lavf.c.
(Yes, this feature still has users.)
Why did this exist in the first place? Other than being completely
useless, this even caused some regressions in the past. For example,
there was the case of a laptop exposing its accelerometer as joystick
device, which led to extremely fun things due to the default mappings of
axis movement being mapped to seeking.
I suppose those who really want to use their joystick to control a media
player (???) can configure it as mouse device or so.
We've been prefering the libavcodec mp3 decoder for half a year now.
There is likely no benefit at all for using the libmpg123 one. It's just
a maintenance burden, and tricks users into thinking it's a required
dependency.
Precise seeking requires skipping audio, since the demuxer usually
doesn't seek precisely enough. There is a sanity check that prevents
skipping more than 300 seconds of audio. This still fails with very
large mp3s. For example, with a 1GB sized mp3 with Xing headers, entries
will be 4 MB apart on average, and occasionally much more.
Just bump the limit. I'm not even sure why it was added in the first
place; I suppose it's most important for files with real PTS resets.
When not receiving frame callbacks, we should not draw anything to avoid
blocking the OpenGL renderer. We do this by extending gl context api, by
introducing new optional function 'is_active', that indicates whether
OpenGL renderers should draw or not.
This fixes issue #249.
Define frame callback logic in wayland_common.c
As this should be used by opengl renderer as well.
Preferably drawing should be skipped entierly when no frame callbacks
are received. However, for now only swap buffers is skipped.
Like FFmpeg/Libav do. It seems not all code can actually deal with this
situation, so it's better to shift the special-cases to code which needs
it (possibly OSD code; screenshots of 0x0 windows would just fail).
I guess we don't really care whether this particular function succeeds.
If it fails, it must be completely broken anyway and it would not matter
much to us.
"Non-monotonic" isn't even 100% correct; it's missing "strictly" (for
briefness I guess), and also the message is printed if the PTS jumps
forward. So just print something that is likely a bit easier to
understand.
Unlike other VOs, this rendered OSD even while no VO was created
(because the renderer lives as long as the API user wants). Change this,
and refactor the code so that the OSD object is accessible only while
the VO is created.
(There is a short time where the OSD can still be accessed even after VO
destruction - this is not a race condition, though it's inelegant and
unfortunately unavoidable.)
--af=bs2b:help abort()ed because the default value of the "profile"
option is not represented by any choice. Fix it by adding an "unset"
choice. (It's a bit odd because there's already a "default" choice,
which is not default, but I don't care enough about this filter.)
Fixes#1712.
gl_video_set_options() didn't update it, so the default value set on
initialization was used. Fix by always setting the clear color before
the clear command; it's slightly easier to follow too.
Codecs for hardware acceleration are not blacklisted, but whitelisted.
Also, if this emssage is printed, the codec might not have any hardware
acceleration support in the first place.
Trying to handle such video is almost worthless, but it was requested by
at least 2 users.
If there are no timestamps, enable byte seeking by setting
ts_resets_possible. Use the video FPS (wherever it comes from) and the
audio samplerate for timing. The latter was already done by making the
first packet emit DTS=0; remove this again and do it "properly" in a
higher level.
Normally, the size of an mage plane is assumed to be stride*height. But
in theory, if stride is larger than width*bpp, the last line might not
be padded, simply because it's not necessary. FFmpeg's or mpv's image
allocators always guarantee that this padding exists (it wastes some
insignificant memory for avoiding such subtle issues), but some other
libraries might not.
I suspect one such case might be Xv via vo_xv (see #1698), although my X
server appears to provide full padding. In any case, it can't harm.
There's literally no reason why these functions have to be inline (they
might be performance critical, but then the function call overhead isn't
going to matter at all).
Uninline them and move them to mp_image.c. Drop the header file and fix
all uses of it.
The libavformat rtmp protocol's "timeout" option has two problems:
1) Unlike all other protocols, it's in seconds and not microseconds
2) It enables "listen" mode, which breaks playback
Make the --network-timeout do nothing in the rtmp case.
Fixes#1704.
For some reason there were two points in the code where it warned
against non-monotonic video PTS. The one in video.c triggered on PTS
going backwards or making large jumps forwards, while dec_video.c
triggered on PTS going backwards or PTS not changing. Merge them into a
single check, which warns against all cases.
There was a somewhat obscure optimization in the OSD and subtitle
rendering path: if only the position of the sub-images changed, and not
the actual image data, uploading of the image data could be skipped. In
theory, this could speed up things like scrolling subtitles.
But it turns out that even in the rare cases subtitles have such scrolls
or axis-aligned movement, modern libass rarely signals this kind of
change. Possibly this is because of sub-pixel handling and such, which
break this.
As such, it's a worthless optimization and just introduces additional
complexity and subtle bugs (especially in cases libass does the
opposite: incorrectly signaling a position change only, which happened
before). Remove this optimization, and rename bitmap_pos_id to
change_id.
To handle seeking correctly, we need to flush the filter. libavfilter
does not support flushing, so we destroy and recreate it. We also need
to handle resume-after-EOF, because the mpv audio code sends an EOF
before and after seeking (the latter happens because the player drains
the filter chain in a generic way, which "causes" EOF).
This played e.g. a 1264x722 file as 1264x720. There was some code which
dropped the aspect ratio if the video (in original resolution) wasn't
scaled by more than 4 pixels. Commit 5f3c3f8c introduced this (although
I'm not really sure what the code replaced by it did).
Just remove this "feature".
We now update uniforms every time, so we should try to reduce the number
of uniforms to avoid performance penalties. (Originally, some caching
was planned, but it looks like it would be too complicated to implement
compared to the expected gains.)
OPT_REPLACED can't specify option values or multiple options. Change to
OPT_REMOVED. Also, target-prim doesn't have an srgb option. BT.709 uses
sRGB primaries, so use it instead.
The default scaling was a slight bit too low, which could cause buffer
underruns in some cases.
This should improve the result when using tscale filters other than
oversample. The oversample case should be unaffected.
This adds extra debugging output for buffer underruns, to help track
down possible queueing issues. It also inverts the numberic output for
tscale=oversample to make more sense, without changing the logic.
This moves the color management code out of pass_render_main (which is
now dedicated solely to up/downscaling and hence renamed pass_scale_main)
and into a new function, which gets called from pass_draw_to_screen
instead.
This makes more sense from a logical standpoint, and also means that we
interpolate in linear RGB, before color management - rather than after
it, which is significantly better for color accuracy and probably also
interpolation quality.
This replaces the old smoothmotion code by a more flexible tscale
option, which essentially allows any scaler to be used for interpolating
frames. (The actual "smoothmotion" scaler which behaves identical to the
old code does not currently exist, but it will be re-added in a later commit)
The only odd thing is that larger filters require a larger queue size
offset, which is currently set dynamically as it introduces some issues
when pausing or framestepping. Filters with a lower radius are not
affected as much, so this is identical to the old smoothmotion if the
smoothmotion interpolator is used.
Also the size is now a simple #define that can easily be changed later.
This is done for smoothmotion, which might want to blend more than 4
frames at once, depending on the setting.
Since the gl_rework merge, this started to print some OpenGL errors when
using vdpau hardware decoding with vo_opengl smoothmotion. This happens
because some hwdec unmap_image call were not paired with a map_image
call. Unlike the old vo_opengl, the new code does not do this out of
convenience (it would be a pain to track this exactly). It was triggered
by smoothmotion, because not every rendered frame has actually a new
input video frame (i.e. no map_image call, but it called unmap_image
anyway).
Solve this by handling unmapping differently in the vdpau code. The next
commit will remove the unmap_image callback completely.
Fixes#1687.
This caused complaints because the fps was basically rounded on
microsecond boundaries in the vsync interval (it seemed convenient to
store only the vsync interval). So store the fps as float too, and let
the "display-fps" property return it directly.
Even the lowest supported GL versions have arrays. This test was for
returning arrays from functions, which didn't work in lower GL versions,
but we don't need it anymore.
Previously, mpv would hide the cursor when the autohide timer expired,
even if the window menu was open. This made it difficult to use the menu
with the mouse.
When handling VOCTRL_SET_CURSOR_VISIBILITY, instead of determining
whether to call SetCursor by checking if the cursor is in the client
area, call it based on the parameters to the last WM_SETCURSOR message.
When the window enters "menu mode," it gets a WM_SETCURSOR message with
HIWORD(lParam) set to 0 to indicate that the cursor shouldn't be set.
I think this is what I alwass missed ever since I found the MPlayer
cache options: a way to enable the cache on local files with the default
settings, whatever they are.
Requested change in behavior.
Note that we set the assumed "infinite" display_fps to 1e6, which
conveniently lets vo_get_vsync_interval() return a dummy value of 1,
which can be easily checked against, and still avoids doing math with
float INFs.
I'm not comfortable with VOCTRL_GET_DISPLAY_FPS being called every
frame.
This requires the VO to set VO_EVENT_WIN_STATE if the FPS could have
changed. At least the X11 backend does this.
This adds stuff related to gamma, linear light, sigmoid, BT.2020-CL,
etc, as well as color management. Also adds a new gamma function (gamma22).
This adds new parameters to configure the CMS settings, in particular
letting us target simple colorspaces without requiring usage of a 3DLUT.
This adds smoothmotion. Mostly working, but it's still sensitive to
timing issues. It's based on an actual queue now, but the queue size
is kept small to avoid larger amounts of latency.
Also makes “upscale before blending” the default strategy.
This is justified because the "render after blending" thing doesn't seme
to work consistently any way (introduces stutter due to the way vsync
timing works, or something), so this behavior is a bit closer to master
and makes pausing/unpausing less weird/jumpy.
This adds the remaining scalers, including bicubic_fast, sharpen3,
sharpen5, polar filters and antiringing. Apparently, sharpen3/5 also
consult scale-param1, which was undocumented in master.
This also implements cropping and chroma transformation, plus
rotation/flipping. These are inherently part of the same logic, although
it's a bit rough around the edges in some case, mainly due to the fallback
code paths (for bilinear scaling without indirection).
The basic idea is to use dynamically generated shaders instead of a
single monolithic file + a ton of ifdefs. Instead of having to setup
every aspect of it separately (like compiling shaders, setting uniforms,
perfoming the actual rendering steps, the GLSL parts), we generate the
GLSL on the fly, and perform the rendering at the same time. The GLSL
is regenerated every frame, but the actual compiled OpenGL-level shaders
are cached, which makes it fast again. Almost all logic can be in a
single place.
The new code is significantly more flexible, which allows us to improve
the code clarity, performance and add more features easily.
This commit is incomplete. It drops almost all previous code, and
readds only the most important things (some of them actually buggy).
The next commit will complete it - it's separate to preserve authorship
information.
Starting to get tired of seeing the full config.h in verbose output
every time. Make it slightly more elegant by outputting the list of
satisfied dependencies instead.
Instead of refusing to set properties like "fullscreen" if no VO was
created, always allow it. So if no VO is created, setting the property
merely changes the options (and will be applied once the VO is created).
This is consistent with similar behavior changes to some other
properties.
Improves the behavior reported in #1676.
Also, we shouldn't check the config_ok variable - the VO should do this.
mpctx->filename is obviously not set if no file is playing.
When this code was written, it probably couldn't happen, because the
normal screenshot path fails much earlier in idle mode. But you can
still take screenshots in "full window" mode, and recently the
screenshot code was changed to use the "full window" mode if the normal
path does not work.
If you click on a window that doesn't have a focus, a LeaveNotify
followed by a EnterNotify event can be generated. The former will have
mode set to NotifyGrab, the latter to NotifyUngrab. This will make the
player think the mouse left the window, even though this is not the
case. Ignore these and only react to those with mode set to
NotifyNormal.
Probably fixes#1672, and some other strange issues on some WMs.
In commit 5f8b060e I blindly assumed that the packet sizes were in
pseudo-samples, but they were actually in bytes. Oops.
(The effect was that cutting the audio was a bit less precise than it
can be.)
Also remove the packet size from ad_spdif.c; it didn't actually use it,
and simply takes what the spdif "muxer" returns.
When playback is started after seeking or opening a file, we need to
make sure audio and video line up exactly. This is done by cutting or
padding the audio stream to start on the video PTS.
This does not quite work with spdif: audio is compressed data, within a
spdif frame. There is no way to cut the audio "in between" the frames.
Cutting between the frames would just produce broken spdif packets, and
who knows how receivers will react to this (play noise?). But we still
can cut it in frame boundaries.
Unfortunately, we also insert 0 data for "silence" - we probably
shouldn't do this. Chances are the receiver will switch to PCM or so.
But for now this will have to do.
Note that this could be simplified somewhat, as soon as we work with
frames. See previous commit.
Needed for the next commit. This commit should probably be reverted as
soon as we're working with full audio frames internally, instead of
"flat" FIFOs.
mpv_opengl_cb_render() is supposed to clear the screen with the
background color if there's no video... I think.
It didn't do this, because although uninit() requested gl_video_config()
to be called, this didn't happen, because this function checks whether
the VO is set - and it's unsert after uninit() releases the lock.
Also call the user wakeup callback in this situation, so the user
actually redraws immediately.
Instead of maintaining a private ring buffer, use the generic support
for audio APIs with pull callbacks (internally called AO pull API). This
also fixes latency calculations: instead of just returning the
ringbuffer status, the audio playback state is calculated better and
includes interpolation.
The main reason this wasn't done earlier was mid-stream format
switching. The pull API can now handle it (in a way) by destroying and
recreating the AO. This is a bit brutal, but quite simple. It's untested
in this new AO, though. Some details might not be right, like how ot
restores the old format when reloading.
This could mute a digital passthrough stream by writing zeros. All other
volume values did nothing.
The comment about MPlayer dying hasn't been true in mpv for quite a
while. It's even possible that it's fixed in upstream MPlayer. mpv will
print a scary error message when trying to change volume with spdif, and
continue normally.
If we really want to mute by writing zeros, we should do it in a
separate filter. But I'm not overly fascinated by this approach; is it
even guaranteed receivers will not be confused by a stream of zeros?
The main reason to remove this is that it's in the way of further
cleanups.
Basically, the idea behind the vp parameter is broken - I guess the
intention was to enable rendering to a specific subrectangle of the
target framebuffer, but there's nothing to specify the actual target
rectangle (the VO will still clear e.g. the borders between video and
framebuffer borders).
We're not going to keep the current semantics either with the upcoming
rework of vo_opengl, so declare this for broken. Maybe we can introduce
a function later which does this properly.
Seems appropriate, and will probably avoid performance surprises with
scary architectures which don't have trivial implementations for atomic
loads. (Consider that demux_mkv calls this very often now, and
libavformat demuxers and streams did this for a while now.)
Check async abort notification. libavformat already do something
equivalent.
Before this commit, the demuxer could enter resync mode (and print silly
warning messages) when the stream stopped returning data because of an
abort.
Do not rely on the pointed-to argument to be initialized; VOCTRLs are
supposed to completely overwrite them on success (or not to touch them
on failure).
The currently only caller of VOCTRL_GET_WIN_STATE initializes the value
before calling this, so this is merely about correctness and didn't lead
to any actual bugs.
This causes the cache to be enabled with --cache=auto. It was not done
previously because the small cache size 320k actually led to worse
performance. However, with the current default cache size of 25000kb,
caching notably improves performance.
Add filter parameters to VAAPI deinterlacing filter to actually process
bottom fields instead of deinterlacing top field twice.
Signed-off-by: wm4 <wm4@nowhere>
Some UI elements in OS X – like Launchpad and Dock folders – are implemented
using borderless windows in background demonized applications.
When we use these elements, mpv doesn't stop to be the active application, but
the mpv window leaves keyWindow state while we use the OS controls.
This commit just keeps track of window state to update the cursor visibility
accordingly.
Fixes#513
Handle the failure gracefully, instead of exploding and disabling audio.
Just set the speed back to 1.0.
Also remove the AF_DETACH from af_scaletempo. This actually created a
dangling pointer in af_add(), a tricky consequence of af_add()
reconfiguring the filter chain and the newly added filter using
AF_DETACH. Fortunately the AF_DETACH is not needed (and probably never
worked - it comes from MPlayer times, and MPlayer also disables audio
when trying to change speed with spdif).
Always use af_scaletempo if it's inserted, even if the option
--audio-pitch-correction=no is set.
Make sure all filters are reset on speed change. It's conceivable that
dynamic changes to the filter chain at runtime leave filters around
without resetting their speed parameters.
Also move the code to a separate function.
Make MpvEventsView -signalMousePosition a public method so it can be
called without a compiler warning. Previously, the mouse position would
be reported as (0,0) until the cursor was moved.
mpv would attempt to load ICC profiles several times during VO init
even if no window is displayed. This potentially causes it to load
a profile for a different screen than it is going to be displayed
on, thereby invalidating the profile cache and rebuilding the LUT
every single time.
It would not unload a previously loaded profile when the video
window is moved to a display without an installed profile.
Fix these issues and tweak the log messages a little.
A user reported a webm stream that couldn't be played. The issue was
that this stream 1. was on an unseekable HTTP connection, and 2. had a
SeekHead element (wtf?). The code reading the SeekHead marked the
element as unreadable too early: although you can't seek in the stream,
reading the header elements after the SeekHead read them anyway. Marking
them as unreadable only after the normal header reading fixes this.
(The way the failing stream was setup was pretty retarded: inserting
these SeekHead elements makes absolutely no sense for a stream that
cannot be seeked.)
Fixes#1656.
The global unpack function got moved to table.unpack in Lua 5.2, and
it's only available as the global if 5.2 is built with compatibility
enabled (the default). Lua 5.3 does not build with 5.1 compatibility by
default.
Fixes#1648.
The vaapi equalizer have a custom range, and can have a smaller range
than mpv's normalized video equalizer values. The result is that a vaapi
equalizer value can map to multiple mpv values, so changing a mpv value
by 1 can get "stuck". Fix by remember the mpv value, and returning it if
it still corresponds to the vaapi value.
Really fixes#1647.
(Why am I even bothering with this irredeemable crap?)
If the audio decoder was created, but no audio filter chain created yet
(still trying to decode a first audio frame), setting the "speed"
property could explode. It tried to recreate the filter chain, even
though no format was set yet.
This is inconvenient and should not happen.
Using check_statement() with an empty statement just to check for the
header is quite a hack. Fix check_headers() (so it takes a "use"
parameter), and use it for the checks instead.
main() being called with argc==0 is probably possible. Fix by skipping
the program name early. (I already changed and reverted this once, but
this time we make sure that it's less likely to confuse the skipped argv
with main()'s argv by naming it "options".)
Instead of "vaapi", simply by changing the probe order.
"vaapi" uses the GLX GL interop, which has causing us more problems than
it solved.
Unfortunately this leads also to copying if "--hwdec=auto --vo=vaapi" is
used, even though GLX is not involved in this case - but I don't care
enough to make the probe logic cleverer just for this. You can still get
the zero-copy path with --hwdec=vaapi.
This duplicates the logic which FFmpeg's libavcodec uses. The effects
are unknown, though it's somewhat clear that a single thread doesn't
necessarily saturate a single CPU.
(Eventually we should just let FFmpeg auto-init the thread count, but
for now I prefer it this way, so e.g. verbose mode will print the
thread count.)
Move the command line parsing and some other things to the common init
routine shared between command line player and client API. This means
they're using almost exactly the same code now.
The main intended side effect is that the client API will load mpv.conf;
though still only if config loading is enabled.
(The cplayer still avoids creating an extra thread, passes a command
line, and prints an exit status to the terminal. It also has some
different defaults.)
This gets rid of the need for a second (or more) parameters; instead it
can be all in one parameter. The (now) redundant parameter is still
parsed for compatibility, though.
The way the flags make each other conflict is a bit tricky: they have
overlapping bits, and the option parser disallows setting already set
bits.
MPlayer requires numeric values for input command parameters. mplayer2
also did. mpv changed these to choices using symbolic strings a long
time ago, but left numeric choices for compatibility.
If we're caching a stream with unknown size, and we reach EOF, then
consider the EOF position the file size. Typically makes sense when
reading from a pipe or a http connection that did not send a size.
This automatically sets the gamma option depending on lighting conditions
measured from the computer's ambient light sensor.
sRGB – arguably the “sibling” to BT.709 for still images – has a reference
viewing environment defined in its specification (IEC 61966-2-1:1999, see
http://www.color.org/chardata/rgb/srgb.xalter). According to this data, the
assumed ambient illuminance is 64 lux. This is the illuminance where the gamma
that results from ICC color management is correct.
On the other hand, BT.1886 formalizes that the gamma level for dim environments
to be 2.40, and Apple resources (WWDC12: 2012 Session 523: Best practices for
color management) define the BT.1886 dim at 16 lux.
So the logic we apply is:
* >= 64lux -> 1.961 gamma
* =< 16lux -> 2.400 gamma
* 16lux < x < 64lux -> logaritmic rescale of lux to gamma. The human
perception of illuminance roughly follows a logaritmic scale of lux [1].
[1]: https://msdn.microsoft.com/en-us/library/windows/desktop/dd319008%28v=vs.85%29.aspx
This will be pretty useful to let mpv automatically change VO parameters based
on ambient lighting conditions.
The conversion code and polinomial equation from Apple LMU values to Lux is
taken from Firefox: their license, MPL is GPL compatible and allows
relicensing to GPL (MPL is more liberal).
This reverts commit c8f49be919.
Not needed anymore; fixed in all supported FFmpeg releases. Though I
could not test again, because all sample files are gone (oops).
The af_lavrresample commit made compilation fail on Libav 10, so I think
it's time to require somewhat more recent dependencies.
Libav 11 is the latest release, and FFmpeg 2.4 seems to correspond to
Libav 11. So use these.
Also adjust the configure failure message. Instead of (accidentally)
printing the pkg-config versions twice, print the release version
numbers too. This is helpful, because the release version numbers are
completely different from the pkg-config ones.
I will probably remove some compatibility hacks in the following commits
too.
Although the libraries we use for resampling (libavresample and
libswresample) do not support changing sampelrate on the fly, this makes
it easier to make sure no audio buffers are implicitly dropped. In fact,
this commit adds additional code to drain the resampler explicitly.
Changing speed twice without feeding audio in-between made it crash
with libavresample inc ertain cases (libswresample is fine). This is
probably a libavresample bug. Hopefully this will be fixed, and also I
attempted to workaround the situation that crashes it. (It seems to
point in direction of random memory corruption, though.)
Use the (relatively new) libavformat image format probing functionality,
instead of letting demux_mf guess by file extension and MIME type.
The libavformat support is weird, though. Traditionally, it uses an
absolutely terrible hack to detect images by extension, _and_ (which is
the horrible part) will randomly interpret parts of the filename as
specifiers for matching by number. So something like '%03d' will be
interpreted as placeholder for a frame number. The worst part is that
such character sequences can be perfectly valid and common in http URLs.
This is known as "image2" demuxer. The newer support, which probes by
examining the file header, is split into several format-specific
demuxers with names ending in "_pipe". So we check for such a name
suffix. (At this point we're doing fine-grained hacking around ffmpeg
weirdness, so a clean solution is impossible anyway until upstream
changes.)
It was possible to make the player play local files by putting rar://
links into remote playlists, and some other potentially unsafe things.
Redo the handling of it. Now the rar-redirector (the thing in
demux_playlist.c) sets disable_safety, which makes the player open any
playlist entries returned. This is fine, because it redirects to the
same file anyway (just with different selection/interpretation of the
contents). On the other hand, rar:// itself is now considered fully
unsafe, which means that it is ignored if found in normal playlists.
This warning wasn't overly helpful in the past, and warned against
perfectly fine code. But at least with recent gcc versions, this is the
warning that complains about assignments in if expressions (why???), so
we want to enable it.
Also change all the code this warning complains about for no reason.
Semi-important, because --hwdec=dxva2 outputs NV12, and we really don't
want people to end up with the "old" StretchRect method.
Unfortunately, I couldn't actually get it to work. It seems most D3D
drivers (including the wine D3D implementation) reject D3DFMT_A8L8,
and I could not find any other 2-channel 8 bit Direct3D 9 format. It
seems newer D3D APIs have DXGI_FORMAT_R8G8_UNORM, but there's no way
to get it in D3D9.
Still pushing this; maybe it actually works on some drivers.
This time (there are a lot of times), libswscale randomly ignores
brightness/saturation/contrast settings.
Looking at MPlayer code, it appears the return value of
sws_setColorspaceDetails() signals if changing these settings is
supported at all.
(Nevermind that supporting this feature has almost 0 value, and
obviously eats maintenance time.)
When used with mp.get_screen_size(), mp.get_screen_margins() allows a
Lua script to determine what portion of the mpv window actually has the
video in it.
Breaks vo_opengl by default. I'm hot able to fix this myself, because I
have no clue about the overcomplicated color management logic. Also,
whilethis is apparently caused by commit fbacd5, the following commits
all depend on it, so revert them too.
This reverts the following commits:
e141caa97d653b0dd529729c8b3f64fbacd5de31Fixes#1636.
It is also used for initialization in channel-list setup.
Should fix compilation on FreeBSD, and is more correct
since it is used unconditionally.
Reverts 6445648 .
Just use makeFirstResponder on the mpv events view from client code
if you need the built in keyboard events (this is easier for dealing with view
nesting).
This relies on upstream support in lavc, and will hence basically not
work at all. The intent is to get support for writing this information
into ffmpeg's PNG encoders etc.
We have MP_CSP_TRC defined, but it wasn't being used by practically
anything. This commit adds missing conversion logic, adds it to
mp_image, and moves the auto-guessing logic to where it should be, in
mp_image_params_guess_csp (and out of vo_opengl).
Note that this also fixes a minor bug: csp_prim was not being copied
between mp_image structs if the format was not YUV in both cases, but
this is wrong - the primaries are always relevant.
Refactors an older hack, which for some reason used a more complicated
way. This generates the playlist representing the contents of the rar
file in demux_playlist.c. The pseudo-demuxer could easily be separate
from the the playlist parsers (and in fact there's almost no shared
code), but I don't think this obscure feature deserves a separate file.
Sample files created with:
rar a -v20000k -m0 files.rar file1.mkv file1.mkv
A use of NO_STREAM_ID_FILTER was added to the DVB code recently. While I
have no idea what it's needed for, it makes mpv fail to compile on
FreeBSD 10.1. Add it to the dvb configure check.
In the past it happened quite often that flag options (yes/no) were
changed to choice options (yes/no/some more). The problem with this was
that while flag options don't need a parameter, this wasn't the case
with choice options. A hack was introduced to compensate for this:
setting M_OPT_OPTIONAL_PARAM on the option, and an empty string ("") was
added as choice, so that the choice could be used like a flag. So, for
example, "--mute" would set the choice "".
Fix this by 1. not requiring a parameter if there's a "yes" choice, and
2. redirect an empty parameter to "yes". The effect is that a choice
option with the choices ["yes", "no"] is pretty much equivalent to a
flag option.
Now that we have fast stream switching, we can bump these sizes, as the
queues cause no delay in switching anymore.
Of course, the fast stream switching works for mkv and mp4 only. Other
formats will incur a quite terrible delay especially in network mode,
which this commit changes to 10 seconds. Let's see if someone
complains...
The way I interpreted it, it seemed like this was not default behavior
and could be enabled with --audio-pitch-correction - it should be made
clearer that this is actually *the default behavior*.
This is based on pretty much the same (somewhat naive) logic right now.
I'm not convinced that the extra logic that eg. madVR includes is worth
enough to warrant heavily confusing the logic for it.
This shouldn't slow down the logic at all in any sane shader compiler,
and indeed it doesn't on any shader compiler that I tested.
Note that this currently doesn't affect cscale at all, due to the weird
implementation details of that.
Broken drivers are an issue rather often. Maybe this gives the user an
idea that this could be the reason. (We can't dump much more info on a
80x24 terminal.)
Change test_fbo() so that it checks the FBO lazily, and restructure
check_gl_features() to invoke it only if we know that a FBO will be
needed for a certain enabled feature.
This can avoid strange error messages when using --vo=opengl and the
FBO format does not work. It's also less confusing when reading the
verbose log (users might think that the FBO is actually used, etc.).
This can happen with the "no-colorkey" suboption. Then the code in
xv_draw_colorkey() can be run before vo_x11_config_vo_window(), when
vo_gc is not allocated yet.
Fixes#1629.
This option allows the user to pass non-supported options directly to
youtube-dl, such as "--proxy URL", "--username USERNAME" and
'--password PASSWORD".
There is no sanity checking so it's possible to break things (i.e.
if you pass "--version" mpv exits with random JSON error).
Signed-off-by: wm4 <wm4@nowhere>
If the cache is enabled, the demuxer is closed and opened again (because
currently, the cache can not be enabled atfer data was already read).
The call for opening a new demuxer uses the same params struct, which
references the ctx->uids array. But there is a MP_TARRAY_GROW()
invocation somewhere on the way, which can reallocate the ctx->uids
array, making params.uids a dangling pointer.
This issue probably existed for a longer time, probably since 5cd33853
(slightly more obvious since f50b105d).
Again removes some indirections and extra arguments.
Also replace some memcpy/memmoves with assignments. (Assignments became
possible only later, when reference UIDs were turned into a struct.)
Should behave about the same, but reduces code some duplication with
seeking and reading a header element pointed to by a SeekHead. It also
makes behavior with incomplete files slightly better.
Instead of rendering and upscaling each video frame on every vsync, this
version of the algorithm only draws them once and caches the result,
so the only operation that has to run on every vsync is a cheap linear
interpolation, plus CMS/dithering.
On my machine, this is a huge speedup for 24 Hz content (on a 60 Hz
monitor), up to 120% faster. (The speedup is not quite 250% because of
the overhead that the larger FBOs and CMS provides)
In terms of the implementation, this commit basically swaps
interpolation and upscaling - upscaling is moved to inter_program, and
interpolation is moved to the final_program.
Furthermore, the main bulk of the frame rendering logic (upscaling etc.)
was moved to a separete function, which is called from
gl_video_interpolate_frame only if it's actually necessarily, and
skipped otherwise.
Hopefully, this will really clear up how the thing is supposed to work
(and that it's not SVP, nor MVTools).
I also removed instances of the word "interpolation", since that's a
term that's easily misleading.
Finally, I expanded on smoothmotion-threshold since the purpose/meaning
was a bit confusing.
Update options like --sub-text-margin-y at runtime. These are somewhat
of a problem, because ass_set_selective_style_override() (intentionally)
does not override them. This should be fixed in libass (by providing
additional override modes), but for now this will do.
Also change the signature of mp_ass_configure(), so we can get access to
the track.
Additionally, drop the redundant setting of the style Alignment (it's
overwritten by mp_ass_set_style()).
See #1622 (again).
Overwrite the alignment applied by the OSD style. Additionally, remove
the initialization of the Alignment field in create_ass_track(); the
value is always overwritten by mp_ass_set_style() later.
Fixes#1626.
There was some logic to set certain things on init only. Not sure why
this was done (saving some cheap calculations?) - but since the next
call would override these style settings by applying the usual subtitle
style, I don't think this was intended.
GLES2 randomly does not support the transpose parameter in matrix
uniform calls. So we have to do this manually. Sure it was worth to
mutilate the standard just so all these shitty SoC vendors can safe 3
lines of code.
(Obviously trying to handle all of GLES2 to GL 4.x in a single codebase
was a mistake.)
GLES2 shaders do not have line continuation characters. Abuse the
HAVE_ARRAYS define to exclude code which uses arrays, and which also
happens to cover all code that defines multi-line macros. (So yes, this
is a hack.)
Remove coded_width and coded_height. This was originally added in commit
fd7dde40, when BITMAPINFOHEADER was killed. The separate fields became
redundant in commit e68f4be1. Remove them (nothing passed to the
decoders actually changes with _this_ commit).
Basically abuse the style override mechanism meant for ASS
(mp_ass_set_style()) to update text subtitle styling at runtime too.
This even has the advantage that the style will be overridden, even if
the text subtitle converted (like sd_lavc_conv.c) dares to add a fixed
style in the styles section.
Probably helps with #1622.
This is essentially what it is, and it's a useful for windowing or
downscaling. For upscaling we already have bilinear, no need to cause
extra confusion between biliner and bilinear_slow.
Also made it a bit more well-behaved.
These are EWA-based versions of the keys B/C splines, of which mitchell
is already a member. They are slightly softer and slightly sharper than
mitchell, respectively.
Very easy to define in terms of things we already have.
mitchell, hermite and catmull_rom are all B/C splines and can share the
code which was already written for mitchell. This just redefines them in
terms of that.
This adds a small check for candidates that could potentially be inside
the radius, but aren't necessarily. This speeds up performance by a
negligible amount on my hardware, but it's mainly a prerequisite for a
further change (using a larger true radius).
This echanges the two events hForceFeed/hFeedDone for hResume. This
like the last commit makes things more deterministic.
Importantly, the forcefeed is only done if there is not already a full
buffer yet to be played by the device. This should fix some of the
problems with exclusive mode.
This commit also removes the necessity to have a proxy to the
AudioClient object in the main thread.
fixes#1529
This makes things a bit more deterministic. It ensures that the audio
thread isn't doing anything between IAudioClient_Stop(),
IAudioClient_Reset() and setting the sample_count to 0.
Buffer overfilling on resume is still a problem in exclusive mode (see
next commit).
Previously, this was based on some arbitrary range 1-100, cut off for
no particular reason, and also defined in such a way that higher values
= *less* smoothness. Since it wasn't multiplied by e in the code, the
default had to be 10*e = 28.8539...
Now, it's sane: 1.0 = default, higher = blurrier.
This filter isn't supposed to have a second parameter in the first
place, all literature only uses a single parameter alpha in both places.
The second parameter doesn't even do anything other than adding a
constant factor, which is normalized by the LUT calculation either way.
This is done mainly for consistency, since all of the EWA filters share
similar properties and it's important to distinguish them for
documentation purposes.
No point in duplicating this check all over the place. No point in
really having it in the first place, to be perfectly honest, j1 should
not be THAT badly behaved.
A while ago, we made mpv output the native number of channels by
default, instead of stereo. I assumed this was not wanted for encode
mode.
This commit keeps the assumption, but allows setting the number of audio
output channels at all, instead of always forcing stereo. (Pretty much a
hack.)
The original filter window was design for a radius based on the true
zero, but we always cut it off at our selection of radius either way (by
necessity, due to the square matrix we sample from).
This window is tweaked from the original (true radius) to our actual
cut-off radius, and hence improves the result in a few edge cases. The
main win is the reduction of code complexity, since we no longer need to
know what the true radius actually is.
Commit f54220d9 attempted to improve this, but it got worse. Now there
was a crash when ytdl_hook.lua added external tracks. This happened
because close_unused_demuxers() assumed that sources[0] was the main
demuxer (so that it didn't close it). This assumption failed, because
the ytdl script can add external tracks before the main file is loaded.
The easy fix would have been to check for master_demuxer, and not i==0.
But instead give up on the old idea, make some stricter assumptions how
demuxers and external tracks map, and simplify the code.
Do timeline building (scanning & opening reference files for ordered
chapters, and more) in a thread. As a result, this process can actually
be stopped without having to kill the player.
This is pretty simple: just reuse the demuxer opening thread. We have
to give up on the idea that open_demux_reentrant() is reusable, though.
(Althoughthe timeline readers still need some fixes before they react to
the quit request.)
These functions do blocking work on a separate thread, but wait until
they return. So they are not async or non-blocking. But they do react to
user-input and client API accesses, which makes them reentrant.
Check the scanf() return value, and don't continue if it doesn't find
both numbers (can happen with GLES 1.0). Also, some implementations can
return NULL from glGetString() if something is "broken".
This is a variation of ewa_lanczos that is sinc-windowed instead of
jinc-windowed. Results are pretty similar, but the logic is simpler.
This could potentially replace the ugly ewa_lanczos code.
It's hard to tell, but from comparing stills I think this one has
slightly less ringing than regular ewa_lanczos.
I've reworked pretty much all the logic to correspond to what the theory
actually describes. With this commit, playback is wonderfully smooth on
my machine.
Some of the hacks were not applied if the file format was forced. Commit
37a0c914 moved them to a table, which is checked with normal probing
only.
Fixes#1612 (DVD forces mpeg, which in turn has to export native stream
IDs specifically).
Do some code restructuring on the way. For example, the probescore can
simply be set to the correct initial value, instead of checking whether
it was set at all.
Starting with waf 1.8.6 (in Python 3), the hcode variable isn't a
string, but a byte string.
This commit adds the solution proposed in the upstream waf bug report:
https://code.google.com/p/waf/issues/detail?id=1535
It seems a bit overly verbose, but on the other hand, this solution has
the chance of being most correct/compatible.
Fixes#1604.
Nobody should use an older version. It's perfectly backwards and forward
compatible, so distros have no excuse not to package a recent version.
Older versions lack tons of bug fixes (some of them crashing bugs, and
potentially security relevant).
With love to Debian, which is still on 0.10.2.
Whatever the hell that is. FFmpeg tries to open any files with .bin file
extension with this demuxer (unless it finds a better demuxer), and then
reads the whole damn file, along with spamming dumb crap.
This message will be printed relatively often once EOF is reached. In
some cases this is rather annoying, for example when playing HLS. (With
HLS, the stream is just a playlist file, while libavformat opens actual
media files without mpv's knowledge, so the cache is completely useless
and hits EOF instantly.)
That it retries reading is apparently a good thing: at least local files
can grow, and even after the player got the EOF, playback _could_ be
resumed by basically polling and detecting that there is more data. So
I'm not changing this behavior yet.
Includes some logic for not starting the demuxer thread for fully read
subtitles. (Well, the cache will still waste _lots_ of resources, and
the cache always has to be created, because we don't know whether it'll
be needed _before_ opening the file.)
See #1597.
An attempt to make format-specifics more declarative. (In my opinion,
all of this should be either provided by libavformat, or should not be
needed.)
I'm still leaving many checks with matches_avinputformat_name(), because
they're so specific.
Also useful for the following commit.
Treat an empty string as unset. The fact that the option values can be
NULL is merely weirdness due to how the option parser works (it
unfortunately doesn't initialize string fields to non-NULL).
Add MP_KEY_MOUSE_ENTER to the ignored input if the user has disabled
mouse input. Remove one instance of code duplication, and add a
MP_KEY_IS_MOUSE_MOVE macro to summarize events that are caused by moving
the mouse.
The Matroska timeline code was the only thing which still used the
demuxer.type field. This field explicitly identifies a demuxer
implementation. The purpose of the Matroska timeline code was to reject
files that are not Matroska. But it already forces the Matroska format,
meaning loading will explicitly only use the Matroska demuxer. If the
demuxer can't open the file, no other demuxer will be tried, and thus
checking the field is redundant.
The change in demux_mkv_timeline.c removes the if condition, and
unindents the if body.
Only demux_cue and demux_edl used it. It's a weird field and doesn't
help with anything anymore - by now, it only saves a priv context in the
mentioned demuxers. Reducing the number of confusing things the demuxer
struct has is more important than minimizing the code.
Move the implementation, of which most was in tl_cue.c, to demux_cue.c.
Currently, this is illogical, because tl_cue.c still accesses MPContext.
This is going to change, and then it will be better if everything is in
demux_cue.c. This is only a separate commit to distinguish code movement
and actual work; the next commit will do the actual work.
Instead of accessing MPContext in player/timeline/*, create a separate
context struct, which the timeline loaders fill out. It turns out that
there's not much in the way too big MPContext that these need to access.
One major PITA is managing (and closing) the set of open demuxers. The
problem is that we need a list of all demuxers to make sure no unneeded
streams are enabled.
This adds a callback to the demuxer_desc struct, with the intention of
leaving to to the demuxer to call the right loader, instead of
explicitly checking the demuxer type and dispatching manually in common
code. I also considered making the timeline part of the demuxer state,
but decided against: it's too much of a mess wrt. memory management and
threading, and also doesn't make it clear who owns the child demuxers.
With the struct timeline decoupled from the demuxer state, it's at least
somewhat clear that the child demuxers are independent from the "main"
demuxer.
The actual changes to player/timeline/* are separated in the following
commits, because they're quite verbose. Some artifacts will be removed
later as soon as there's only 1 timeline loading mechanism.
Also effects some other cases.
The real reason for this is for keeping track of which demuxers can be
closed (see following commit). Since I don't want to use reference
counting for this, some sort of simplistic mark-and-sweep is done to
determine whether a demuxer is still needed.
The HLs protocol consists of a "playlist" main file, which mpv downloads
and passes to the HLS demuxer. The HLS demuxer actually requests segment
files containing media data on its own. The packets read from the
demuxer have a source file position set, but it's not from the main
file. This leads to a strange effect: as a last fallback, the player
will calculate the approximate playback position from the file
position/size ratio, and since the main file is tiny, this will always
show 100%. Fix this by resetting the packet file position.
This doesn't affect the case when HLS actually reports a duration.
Now --ass-use-margins doesn't apply to normal subtitles anymore. This is
probably the inverse from the mpv behavior users expected so far, and
thus a breaking change, so rename the option, that the user at least has
a chance to lookup the option and decide whether the new behavior is
wanted or not.
The basic idea here is:
- plain text subtitles should have a certain useful defalt behavior,
like actually using margins
- ASS subtitles should never be broken by default
- ASS subtitles should look and behave like plaintext subtitles if
the --ass-style-override=force option is used
This also subtly changes --sub-scale-with-window and adds the --ass-
scale-with-window option. Since this one isn't so important, don't
bother with compatibility.
You can set in which "corner" the OSD and subtitles are shown. I'd
prefer it a bit more general (so you could set the alignment using
a factor), but the libass API does not provide this.
Until now, they used exactly the same defaults for the styling options.
The defaults were shared, so it was impossible to have different
defaults. Change this. This requires duplicating the full default
struct, even for settings that are the same. The list of options is
still shared, though.
The previous commit effectively fixes the mess caused by 'config' vs.
'mpv.conf', and the hack introduced by commit e01a6dac and extended by
commit db167cd4 isn't needed anymore.
Actually, it's pretty simple to look for multiple filenames at once,
since mp_find_all_config_files() is already a bit "special" anyway.
See #1569. Reverts most of commit db167cd4 (keeps osx-bundle.conf).
A recent behavior change in libavcodec's h264 decoder keeps at least 1
surface even after avcodec_flush_buffers() has been called. We used to
flush the decoder in order to make sure all surfaces are free'd, so that
the hw decoder can be safely uninitialized. This doesn't work anymore.
Fix it by closing the AVCodecContext before the hw decoder is
uninitialized. This is actually simpler and more robust. It seems to be
well-supported too.
Fixes invalid read accesses with vaapi-copy and dxva2-copy. These
destroyed the hwdec API fully on uninit, and could not deal with
surfaces surviving the decoder.
Probably fixes#1587.
The way the AO wakes up the playloop has nothing to do with events;
instead we must query the events on the AO once the playloop was woken
up. Querying the events in every playloop iteration is thus the correct
way to do this.
If the previous subtitle packet is too far back, and the refresh seek
won't pick it up, and the packet never comes again. As a consequence,
the refresh mode was never stopped on the subtitle stream, which caused
all packets to be discarded.
Fix by assuming the file position is monotonically increasing; then it
will resume even if a packet _after_ the intended resume point is
returned. This introduces a new requirement on how the demuxer behaves.
(I'm not sure if mp4 actually satisfies this requirement in all cases.)
Fixes a regression introduced by commit f9f2e1cc.
This commit adds notifications for hot plugging of devices. It also extends
the old behaviour of the `audio-out-detected-device` property which is now
backed by the hotplugging code. This allows clients to be notified when the
actual audio output device changes.
Maybe hotplugging should be supported for ao_coreaudio_exclusive too, but it's
device selection code is a bit fragile.
Apparently there's at least one distro which ships a /etc/mpv/mpv.conf
file (mpv doesn't install such a file). This breaks config files named
'config' located in the user's mpv config directory, because mpv first
loads files named 'config' and then 'mpv.conf'. There is no mechanism
for putting files with different names into the same config path order.
(Even worse, that mpv.conf file only set an option to the default value.
Why do distros always do very stupid things?)
Print a warning on collisions.
Although using 'config' was well-supported, supporting both names is
starting to become messy, so deprecate 'config' and print a warning if
one is found.
At least we will be able to remove the whole mess once 'config' files
are ignored...
This also affects the osx-bundle, which intentionally used these not-so-
optimal semantics. Solve it in a different way. (Unfortunately with an
ifdef - it's not required, but having to explain everyone why mpv tries
to load a osx-bundle.mpv file on Linux and Windows would consume
energy.)
Closes#1569.
Makes all keys documented in XF86keysym.h mappable. This requires the
user to deal with numeric keycodes; no names are queried or exported.
This is an easy way to avoid adding all the hundreds of XF86 keys to
our X11 lookup table and mpv's keycode/name list.
Happens to fix#1581 due to an unfortunate interaction with the way the
VO does not react to commands for a while if a video frame is queued.
Slightly improves other situations as well, if the client spams mpv with
commands during playback.
This removes the delay when switching audio tracks in mkv or mp4 files.
Other formats are not enabled, because it's not clear whether the
demuxers fulfill the requirements listed in demux.h. (Many formats
definitely do not with libavformat.)
Background:
The demuxer packet cache buffers a certain amount of packets. This
includes only packets from selected streams. We discard packets from
other streams for various reasons. This introduces a problem: switching
to a different audio track introduces a delay. The delay is as big as
the demuxer packet cache buffer, because while the file was read ahead
to fill the packet buffer, the process of reading packets also discarded
all packets from the previously not selected audio stream. Once the
remaining packet buffer has been played, new audio packets are available
and you hear audio again.
We could probably just not discard packets from unselected streams. But
this would require additional memory and CPU resources, and also it's
hard to tell when packets from unused streams should be discarded (we
don't want to keep them forever; it'd be a memory leak).
We could also issue a player hr-seek to the current playback position,
which would solve the problem in 1 line of code or so. But this can be
rather slow.
So what we do in this commit instead is: we just seek back to the
position where our current packet buffer starts, and start demuxing from
this position again. This way we can get the "past" packets for the
newly selected stream. For streams which were already selected the
packets are simply discarded until the previous position is reached
again.
That latter part is the hard part. We really want to skip packets
exactly until the position where we left off previously, or we will skip
packets or feed packets to the decoder twice. If we assume that the
demuxer is deterministic (returns exactly the same packets after a seek
to a previous position), then we can try to check whether it's the same
packet as the one at the end of the packet buffer. If it is, we know
that the packet after it is where we left off last time.
Unfortunately, this is not very robust, and maybe it can't be made
robust. Currently we use the demux_packet.pos field as unique packet
ID - which works fine in some scenarios, but will break in arbitrary
ways if the basic requirement to the demuxer (as listed in the demux.h
additions) are broken. Thus, this is enabled only for the internal mkv
demuxer and the libavformat mp4 demuxer.
(libavformat mkv does not work, because the packet positions are not
unique. Probably could be fixed upstream, but it's not clear whether
it's a bug or a feature.)
Until now, some packets could return the same file position if they were
split off from a Matroska-level packet. This was perfectly fine, because
the file position isn't used for anything overly important (it uses it
to estimate playback position if no other information is available). The
following commit will use the demux_packet.pos field as unique ID (as a
simplification), so make the demuxer export more finegrained
information.
Also, the last_filepos field didn't have to be global, at least not
anymore.
Commit e920a00eb assumed that terminate_cocoa_application() actually
would exit. But apparently that is not always the case; e.g. mpv --help
will just hang. The old code had a dummy exit(0), which was apparently
actually called. Fix by explicitly exiting if mpv_main() returns and
terminate_cocoa_application() does nothing.
Requested. See manpage additions.
This also makes the magical loop_times constants slightly saner, but
shouldn't change the semantics of any existing --loop option values.
Setting the input context is always called, both in cplayer and libmpv,
and under HAVE_COCOA. Unsetting the input context was done only the
cplayer uninit call. Also it was under HAVE_COCOA_APPLICATION, so it was
not unset in libmpv (dangling pointer).
The code in main.c calls exit() explicitly, but the code is actually
easier to follow by simply exiting from main() instead. The exit() call
in av_log.c happens only on severely broken builds, so replace it with
abort().
(Shuts up rpmlint warnings.)
This requires jumping through multiple hoops on fire. Since the
PulseAudio API is virtually undocumented, I'm not sure if this is
correct either. We only react to sink events, and only to the NEW/REMOVE
events. CHANGE events are ignored, because PulseAudio fires them far too
often - even if the system is completely idle! If pa_sink_info.name can
change, we're in trouble. pa_sink_info.description is not so important,
but it'd also be a bit un-nice if it can change, and we don't update it.
The weird way how the actual AO and the hotplug context share the same
struct (ao) comes in handy here, although context_success_cb() still had
to be duplicated from success_cb() - the unused argument has a different
type.
Not very important for the command line player; but GUI applications
will want to know about this.
This only adds the internal API; support for specific audio outputs
comes later.
This reuses the ao struct as context for the hotplug event listener,
similar to how the "old" device listing API did. This is probably a bit
unclean and confusing. One argument got reusing it is that otherwise
rewriting parts of ao_pulse would be required (because the PulseAudio
API requires so damn much boilerplate). Another is that --ao-defaults is
applied to the hotplug dummy ao struct, which automatically applies such
defaults even to the hotplug context.
Notification works through the property observation mechanism in the
client API. The notification chain is a bit complicated: the AO notifies
the player, which in turn notifies the clients, which in turn will
actually retrieve the device list. (It still has the advantage that it's
slightly cleaner, since the AO stuff doesn't need to know about client
API issues.)
The weird handling of atomic flags in ao.c is because we still don't
require real atomics from the compiler. Otherwise we'd just use atomic
bitwise operations.
This is a small oversight. The client name (as set on command line
options or, more importantly, the client API) was not set when listing
devices e.g. via the "audio-device-list" property.
Might or might not fix#1578.
Also adjust the log level for an unrelated message.
This reverts commit acc5e8f574.
As expected, some didn't like this. Others won't like this revert.
Whatever.
See #1561.
This should go into mpv 0.8.0 before it's released.
In my opinion the artifacts created by af_scaletempo on extreme slowdown
(50% or so) are too bothersome - but users disagree. So use
af_scaletempo on any speed changes, not just on speedup.
Filters which merely wrap libavfilter (for user-compatibility) like
vf_gradfun had a "lavfi-enable" suboption, which could disable
libavfilter usage. Since none of these filters has an internal
implementation anymore, this was completely useless.
rubberband_available() can return a negative value, which we assigned to
a size_t variable, leading to the frame allocation to fail. This could
spam "Error filtering frame.". (That it spams this instead of exiting
should probably also be considered a bug.)
At least in the realtime mode and in our case, a negative return value
should not have any different meaning from a 0 return value, in
particular because we call rubberband_get_samples_required() or set the
"final" parameter for rubberband_process() to continue/stop processing.
After some testing, I am fairly convinced that these defaults sound
better than the previous settings. This also eliminates some issue
with random crackling and noise.
Also remove the `stretch` option since it has no effect in
realtime mode.
The previous commit on this filter accidentally removed the
RubberBandOptionProcessRealTime option. Without it, the lib prints a
warning and passes the audio through.
Also add the RubberBandOptionSmoothingOn option back. Though for some
reason the output sounds still very wrong.
Remove the confusing crap that allowed a filter using the libavfilter
bridge to be compiled without libavfilter. Instead, compile the wrappers
only if libavfilter is enabled at compile time.
The only filter which still requires it is vf_stereo3d (unfortunately).
Special-case this one. (The whole filter and how it interacts with lavfi
is pure braindeath anyway.)
librubberband exports a big load of options. Normally, the default
settings (whether they're librubberband defaults or our defaults) should
be sufficient, but since I'm not so sure about this, making it
configurable allows others to figure it out for me.
The problem here is that librubberband can buffer an arbitrary amount
of data, but at the same time doesn't provide a way to query how much
data is buffered. So we keep track of this manually, assuming that
librubberband tries to reach the requested time ratio for input and
output (which is probably true).
The disadvantage is that rounding errors could accumulate over time.
(Maybe it should try to round towards keeping the time ratio.)
In theory it could happen that draining on EOF happens incrementally,
and then the unconditional reset could have dropped the remaining
buffered audio.
If "--af=rubberband" is used, librubberband will be used to speed up or
slow down audio with pitch correction.
This still has some problems: the audio delay is not calculated
correctly, so the audio position jitters around by a few milliseconds.
This will probably ruin video timing.
This avoids potentially dropping some small amount of audio data
buffered in filters.
Reinit can be skipped only if the filter is af_scaletempo (which maps to
AF_CONTROL_SET_PLAYBACK_SPEED). The other case using af_lavrresample is
much more complicated due to filter chain politics.
Also, changing speed between 1.0 and something higher typically inserts
or removes the filter, so this obviously requires reinitialization. It
can be prevented by forcing the filter with --af=scaletempo.
Staring at the code a bit, it turns out that changing speed without
losing state is quite easy. The initialization code is big and
complicated, but most of it is specific only to the configured audio
format, not the speed.
Refactor the code so that changing speed at runtime could work. (It's
not actually used yet - the player code still does a complete reinit.
This will be fixed in the next commit.)
The "if (s->speed_tempo == s->speed_pitch)" looks a bit strange, but
does the same thing as the code did before: speed can be changed only if
exactly one flag is set. If both are set or none, speed can't be
changed.
This code skipped initialization if no speed/pitch change was to be
applied.
It also didn't force conversion of the audio to a supported format,
which is probably the most important case in context of compatibility.
With this change applied, af_scaletempo will always force format
conversion.
To make the change less disruptive, make the filter detach if
unconvertable formats are used. Some users use spdif and also have
"af=scaletempo" in their config, so better not completely break this.
In the case the filter was added with the "speed=both" suboption, the
filter also detached itself in this case; but it's an obscure case, so I
don't care about that.
This reverts commit a33b46194c.
It turns out FFmpeg really considers this a bug, and fixed it by making
the decoder output the correct pixel format.
Fixes#1565. Reverts the fix#1528, though it should work fine with
a recent git master FFmpeg.
The intention is that we can test vo_opengl with high bit depth PNGs
better. This throws libswscale completely out of the loop, which before
was needed in order to convert from big endian to little endian.
Also apply a minimal cleanup to fmt-conversion.c (unrelated).
Most things stopped using this field for better support of growing
files. Go through the trouble to repalce the remaining uses, so it can
be removed.
Also move the "streaming" field; saves 4 bytes (wow!).
Fix return types and return values to make them more consistent. Some
reformatting and making code more concise.
In stream_reconnect(), avoid the additional mp_cancel_test() call by
moving the "connection lost" message below the mp_cancel_wait() call,
which effectively leads to the same behavior when the stream was already
canceled. (The goal is not to show the message in this case.)
Merge stream_seek_long() into stream_seek(). It was the only caller.
Always clear the eof flag on seeks.
Reduce access to stream internals in cache.c and stream_lavf.c.
Make it accept "," as separator, instead of only ":". Do this by using
the key-value-list parser. Before this, the option was stored as a
string, with the option parser verifying that the option value as
correct. Now it's stored pre-parsed, although the log levels still
require separate verification and parsing-on-use to some degree (which
is why the msg-level option type doesn't go away).
Because the internal type changes, the client API "native" type also
changes. This could be prevented with some more effort, but I don't
think it's worth it - if MPV_FORMAT_STRING is used, it still works the
same, just with a different separator on read accesses.
Before this, unquoted occurrences of ":" lead to parsing errors. There's
no reason to reject it, especially since the traditional MPlayer syntax
uses ":" as separator. (Which is also the reason why ":" was rejected
before: the parser shares this code for handling splitting/quoting, and
we merely checked explicitly whether the option was split on ",".)
Fixes#1445 without failing in the case where file outside.mkv with N
chapters references inside.mkv with N+1 chapters. Since we don't care
about the number of chapters in inner files (yet; nested chapters would
change this, but also make chapters dynamic anyways), don't check there.
Much cleaner. The difference between timeline_info and
inner_timeline_info is that timeline_info is stuff the top-level cares
about and inner_timeline_info is just for the current chapter.
This introduces a new option linear-scaling, which is now implied by
srgb, icc-profile and sigmoid-upscaling.
Notably, this means (sigmoidized) linear upscaling is now enabled by
default in opengl-hq mode. The impact should be negligible, and there
has been no observation of negative side effects of sigmoidized scaling,
so it feels safe to do so.
Autoload external audio files only if there's at least a video track
(which is not coverart pseudo-video).
Enable external audio file autoloading by default. Now that we actively
avoid doing stupid things like loading an external audio file for an
audio-only file, this should be fine.
Additionally, don't autoload subtitles if a subtitle is played.
Although you currently can't play subtitles without audio or video,
it's disturbing and stupid that the player might load subtitle files
with different extension and then fail.
Giving this such a prominent place is not really appropriate anymore.
Most people seeing this would probably expect a release changelog, not
something about MPlayer.
Since the page still could be useful for former MPlayer users (in
particular to avoid confusion with renamed options etc.), still keep
it in the DOCS directory.
Reindent the whole handle_realaudio() function, and make the surrouding
if block return early instead.
Also contains some cosmetics to the sipr swapping, which hopefully does
not change the semantics, but is untested (the kind of cosmetic changes
everyone loves so much). May the person responsible for sipr rot in
hell. (It was probably done to obfuscate the codec?)
Staring at the code, it doesn't look like the extra code for "normal"
audio is needed. Most of it looks like artifacts from the previous code
structure (much of it was added in the initial commit). I couldn't find
a sample that uses this code path to fully confirm this, though.
I suppose it could lead to subtle changes in behavior in presence of
realvideo files that change aspect radio. With the only sample I had
available, the behavior actually improved (azumi.mkv from the MPlayer
samples FTP; when starting playback in the middle it used the wrong
aspect ratio).
Appears to work, so we can drop some code. For some really odd reason,
the descrambling done on the timestamp requires millisecond units (due
to the "algorithm", not the libavcodec API).
Fixes vp9 missing timestamps. This requires a brand new libavcodec (the
patch for this was just applied to FFmpeg git master).
The timestamp mangling is applied to VP9 only. It'd probably work with
other codecs, but it's not needed. It could break in various ways, so
it has to be explicitly checked for every enabled codec.
Makes it somewhat more uniform, and breaks up the awfully deep nesting.
This implicitly changes multiple small details, rather than only moving
code around. In particular, this computes the packet fields first and
parses them afterwards, which is needed for the next commit.
The compilation database is a JSON file[1] storing all compilation flags. That
is useful for tools using libclang for code completion and error reporting
(for example: YouCompleteMe for vim).
[1]: http://clang.llvm.org/docs/JSONCompilationDatabase.html
Although the use is somewhat questionable, it seems strange that e.g.
--geometry=50% works (and sets the width only), but setting the height
only in a similar manner does not work.
Matroska ordered chapters require all segment files to have the same
layout. If a referenced file has more tracks than the main file or
misses some tracks, the file is invalid.
Also see issue #1553.
This shouldn't exist and for the most part is meant to be used by the
ytdl Lua script, but let's document it anyway. Since the Lua API handles
all the details, it's considered much more "stable" than the raw API,
which is why the raw API wasn't documented.
Looks like a bunch of off-by-one errors.
The track number was mistakenly offset by 1 - this shifted all chapters
by one, and make the first chapter start on the second track (so the
"chapter" property returned -1 in the first track since it was before
the first chapter).
Also, the calculation of the sector destination was messed up. This
comes from commit 32d818f0, where I apparently attempted to calculate
the position to one byte before the section, but unfortunately math
doesn't work this way and it was nonsense. Just drop this idea; while it
may help with seeking (probably...), it also returns slightly different
times. The user shall use hr-seeks if accurate seeking is required.
Hopefully fixes#1560.
Currently, audio packets are always filtered as a whole. Since demux_raw
output a 1 second long packet, this could lead to large delays when
applying softvol volume. It could be fixed by splitting the frames the
decoder outputs before filtering them (like the old filter code used
to), but since this didn't cause any other problems yet, I'm going with
the simpler fix.
Fixes#1558.
Before this change, window creation was delayed until the video was
initialized. This guaranteed that the first window size was that of the
video, so the WM would place it correctly.
Some time ago, it was requested on the IRC channel that --force-window
should not do this. Sometimes, it can take a while until video is
initialized, e.g. when youtube-dl is used (which incurs lots of network
delay). In this case, it's awkward that it takes so long until a window
(any window) is shown.
On the other hand, this can cause incorrect window placement with some
WMs. We simply hope that this won't happen with modern WMs. (Although it
does with the WM I use, crappy old IceWM.)
In ancient times, this was needed because it was not default, and many
VOs had problems with it. But it was always default in mpv, and all VOs
are required to deal with it. Also, running --fixed-vo=no is not useful
and just creates weird corner cases. Get rid of it.
This allows setting these options directly (without going through
properties, or with going through the "options/" property). The
documented restrictions apply to all of these: changes do not get
immediately applied, unlike with corresponding properties.
This is in reaction to #1548.
Apparently CoreGraphics reports the actual refresh rate. DisplayLink can also
query the nominal refresh rate of the display so we use that as fallback
instead of the fugly 60fps hardcode added in aeb1fca0d.
Props to people on https://github.com/glfw/glfw/issues/137
Comment explains why I have been so doubtful at adding this. The Apple docs
say CGDisplayModeGetRefreshRate is supposed to work only for CRTs, but it
doesn't, and actually works for LCD TVs connected over HDMI and external
displays (at least that's what I'm told, I don't have the hardware to test).
Maybe Apple docs are incorrect.
Since AFAIK Apple doesn't want to give us a better API – maybe in the fear we
might be able to actually write some useful software instead of "apps" –
I decided not to care as well and commit this.
This reverts the default behavior introduced in commit 93feffad. Way too
often libavcodec will return RGB data that has an alpha channel as per
pixel format, but actually contains garbage.
On the other hand, this will actually render garbage color values in
e.g. PNG files (for pixels with alpha==0, the color value should be
essentially ignored, which is what the old alpha blend mode did).
This "fixes" #1528, which is probably a decoder bug (or far less likely,
a broken file).
The only reason why cdda:// goes through this wrapper-demuxer is so that
we add chapters to it. Most things related to seeking apply only to
DVD/BD, and in fact broke CDDA sekkability.
Fixes#1555.
Make the lazy gamma initialization less weird, and make the default
value of the "gamma" sub-option 1.0. This means --vo=opengl:help will
list the actual default value.
Also change the lower bound to 0.1 - avoids a division by zero (I don't
know how shaders handle NaN, but it's probably not a good idea to give
them this value).
There was some code accounting for different gamma values for R/G/B.
It's inherited from an old, undocumented MPlayer feature, which was at
some point disabled for convenience by myself (meaning you couldn't
actually set separate gamma because it was removed from the property
interface - mp_csp_copy_equalizer_values() just set them to the same
value). Get rid of these meaningless leftovers.
These commands are counterparts of sub_add/sub_remove/sub_reload which
work for external audio file.
Signed-off-by: wm4 <wm4@nowhere>
(minor simplification)
Basically, the declared option name and the name passed to the
parse_obj_settings_list() must be the same.
Fixes the issue addressed in #1550, but differently.
These were derived from dividing our assumed video gamut (1.961) by some
typical screen values (2.2 for dimly lit and 2.4 for pitch black):
1.961/2.4 = 0.8170833333333334 ~= 0.8
1.961/2.2 = 0.8913636363636364 ~= 0.9
Before this, enabling :gamma in combination with :sigmoid and probably a few
other things results in ugly artifacts because the video isn't clamped until
after the :gamma was applied (or at all, if the cms_matrix is unused).
Previously we let the user use the audio device ID, but this is not persistent
and can change when plugging in new devices. That of course made it quite
worthless for storing it as a user setting for GUIs, or for user scripts.
In theory getting the kAudioDevicePropertyDeviceUID can fail but it doesn't
on any of my devices, so I'm leaving the error reporting quite high and see if
someone complains.
At least the opengl-hq VO allocates additional resources when
downscaling a lot, which is just a waste.
Also see #1547 (although I doubt that this is the cause; if it is,
a real fix will be required).
This is somewhat imperfect, because detection of hw decoding APIs is
mostly done on demand, and often avoided if not necessary. (For example,
we know very well that there are no hw decoders for certain codecs.)
This also requires every hwdec backend to identify itself (see hwdec.h
changes).
This does what it's documented to do.
The implementation reuses the code in mpv_detach_destroy(). Due to the
way async requests currently work, just sending a synchronous dummy
request (like a "ignore" command) would be enough to ensure
synchronization, but this code will continue to work even if this
changes.
The line "ctx->event_mask = 0;" is removed, but it shouldn't be needed.
(If a client is somehow very slow to terminate, this could silence an
annoying queue overflow message, but all in all it does nothing.)
Calling mpv_wait_async_requests() and mpv_wait_event() concurrently is
in theory allowed, so change pthread_cond_signal() to
pthread_cond_broadcast() to avoid missed wakeups.
As requested in issue #1542.
Resizing was happening before reconfig, so src_rect_rot was outdated and
didn't include the rotation. This resulted in corrupted rendering on
initial display, which fixed itself after the first time the window was
somehow resized.
This fixes usage when the bundle is used on the same machine that Python/Cython
was compiled on. It doesn't fix the harder problem of packaging a full Python
installation in the mpv app bundle (and I'm not sure we actually want that).
Fixes#1549
Might fix behavior with mkv files that use ordered chapters and have
cover art tags. In my opinion, this should actually have worked (because
cover art pseudo-tracks are strictly appended), but I don't have a
sample file to test at hand.
Opening the stream and opening the demuxer are both done asynchronously,
meaning the player reacts to client API requests. They also can
potentially take a while. Thus it's better to process outstanding
property changes, so that change events are sent for properties that
were changed during opening.
In particular, this would fix the sending the initial change event. It
was easily missed because MPV_EVENT_FILE_LOADED usually triggered it,
but the actual property could change only later, because audio
initialization really is kind of asynchronous to it.
This probably fixes#1544.
I guess this was supposed to be some sort of optimization, but even
though it probably works, it's pretty meaningless and I couldn't measure
a difference. One special case killed.
This reverts commit 7b3feecbc2.
It's broken, hr-seek never ends at a video position before seek pts.
Not sure what I was thinking, although it did work anyway when
artificially forcing a video frame to display before seek pts.
At least there is _some_ problem if this happens. It would mean that
audio is playing slower than video. Normally, video is synced to audio,
so if audio stops playback completely, video will not advance at all.
But using things like --autosync, it's well possible that this kind of
desync happens.
Move the update_avsync_before_frame() call further down. Moving it
closer to where the time_frame value is used (and which the function
updates) should make the code more readable. With this change, there's
no need anymore to reset the time_frame value on the video reconfig
path.
Move the update_avsync_after_frame() up. Now no meaningful amount of
time passes since the previous get_relative_time() call anymore, and the
second one can be removed.
FFmpeg and Libav have the stupid practice of replacing and deprecating
API symbols on the same day. So with FFmpeg git, this is useless and
will print a compile time warning, while it's required with all stable
releases, and might lead to decoding errors with xvid/avi (apparently).
Add a comment before someone writes a patch and I have to explain it all
over again.
A small simplification. Couldn't be done before, because it was also
used by the OSD code, which required disjoint quads in a single draw
call.
Also mess with the unrelated code in gl_osd.c to simplify it a little
as well.
Now it shows one of:
- "Subtitles hidden" (sub-visibility=no)
- "Subtitles visible" (sub-visibility=yes, sub!=no)
- "Subtitles visible (but no subtitles selected)" (otherwise)
It should be a bit more self-explanatory than before. On the other hand,
I have no clue about UI issues.
This also gets close to what's reasonably possible with the OSD
expansion string syntax, which is why it looks so awful.
Hardware decoding/displaying with vo_opengl is done by replacing the
normal video textures with textures provided by the hardware decoding
API OpenGL interop code. Often, this changes the format (vaglx and vdpau
return RGBA, vda returns packed YUV).
If the format is changed, there was a chance (or at least a higher
potential for bugs) that the shader generation code could be confused by
the mismatch of formats, and would create incorrect conversions.
Simplify this by requiring the hwdec interop driver to set the format it
will return to us. This affects all fields, not just some (done by
replacing the format with the value of the converted_imgfmt field in
init_format), in particular fields like colorlevels.
Currently, no hwdec interop driver does anything sophisticated, and the
win is mostly from the mp_image_params_guess_csp() function, which will
reset fields like colorlevels to expected value if RGBA is used.
Reduces the size of gl_video.c a bit further.
This also uses a separate vertex array object for OSD elements, so the
video one can be simplified slightly.
OSD shader generation is still in gl_video.c, which leads to the strange
additional parameter to mpgl_osd_init(). The issue is that video
parameters influence the OSD shader (????), and also OSD needs to go
through the screen colormanagement.
Useful if we want to reduce the size of gl_video.c further.
To some degree this emulates traditional glDrawArrays() usage. It also
leaves a loophole for avoiding a reupload every time by leaving
ptr==NULL, although this is unused for now.
default_tex_params() and texture_size() are each called only once, so
move inline/reimplement them at the caller.
image_dw/dh were unused. texture_w/h, image_format, and component_bits
were rarely used, and can be replaced. Regroup some other fields.
Rename surface_num to surface_idx, because the former sounded like a
count, and not an index. Move fbosurface_next() closer to its callers
too.
Move the DebugMessageCallback() code to gl_utils.c (also simplify it
by always setting the callback, instead of only when it changes).
This is somewhat messy, because fbotex_init() itself was depending on
some gl_video parameters unrelated to FBO creation (like what scaler was
in use - what the fuck did this check do in this function?), so this
commit does a bit more than moving code around. In particular, the FBO
for the separate scaling intermediate step now always uses GL_NEAREST
sampling, and all FBOs are destroyed/recreated on renderer
reinitialization.
This also moves the function matrix_ortho2d() - trivial enough not to
put it into a separate commit.
Windows Intel drivers seem to reject some (AFAIK) valid GLSL. Make them
happy.
<rossy> GL_RENDERER='Intel(R) HD Graphics 4400'
<rossy> GL_VERSION='3.0.0 - Build 10.18.14.4080'
<rossy> GL_SHADING_LANGUAGE_VERSION='1.30 - Build 10.18.14.4080'
...into its own functions. The central playloop function is still too
big, but looks much cleaner now.
No changes in functionality. The code moved to handle_playback_restart()
is unindented by 1 level and moving it out of the if condition around.
The if condition is inverted and early-exits from the function. Also
some comments are changed.
mpctx->audio_delay always has the same value as opts->audio_delay. (This
was not the case a long time ago, when the audio-delay property didn't
actually write to opts->audio_delay. I think.)
Handles stupid boilerplate OpenGL requires you to handle. It's the same
code as in gl_video.c, although if no VAOs are available, the fallback
code rebinds them on every draw call instead of just once.
gl_common.c contained the function loader (which is big) and additional
utility functions (not so big, but will grow when moving more out of
gl_video.c). Just split them. There are no changes other than some
modifications to comments.
This allows seeking audio between two video frames that are relatively
far away.
The implementation of this is a bit subtle. It pretend the audio
position is different, and the actual PTS adjustment happens in audio.c
with this line:
sync_pts -= mpctx->audio_delay - mpctx->delay;
Effectively this is the same as setting sync_pts to hrseek_pts after
this line, though. (I'm actually not sure if this could be written in a
more straightforward way; probably yes.)
Some files can have audio after video has ended, and playback of the
audio-only remainder is supposed to work just fine.
Seeking is broken-ish though. Not much can be done about this, since
it's the way demuxers work. Also, such files are obscure corner cases.
But enabling hr-seek for audio after video end can improve the situation
a lot.
This helps with issue #1533. The reported also provided a command line
to produce such a file:
ffmpeg -i image.jpg -i audio.flac -threads $(nproc) \
-c:v libvpx -crf 10 -qmin 5 -qmax 55 \
-vf scale=360:-1 -sws_flags lanczos -c:a libvorbis -ac 2 \
-b:a 128K out.webm
Commit acb40644 fixed video with unaligned luma/chroma sizes. It
attempted to disable the fix for videos where it effectively does
nothing (just some minor performance paranoia), but this check was
broken - fix it by not duplicating the logic for this.
PNG uses a different component order from GL_RGBA, so we upload the
surface using the "wrong" order, and then fix it in the shader. This
breaks if a sRGB texture (GL_SRGB) is used: the hardware will not touch
the alpha channel, which means that the B component is not adjusted,
leading to incorrect output.
Just remove the use of sRGB textures completely. It might lead to a
slight slow down when playing RGB with color management enabled, but
with this combination of obscure use case with minor performance impact
it's not a meaningful disadvantage.
Unfortunately this also means that alpha is handled incorrectly with our
own color management, but alpha isn't so important and can be fixed
later. (0.0 and 1.0 are unchanged by the transfer function, so it
"mostly" works.)
Fixes#1530.
This was apparently useful for correct interlaced scaling (although I
don't know anyone who used this). It was rarely used (if at all), had an
inconvenient output format (packed YUV), and now has a better solution
in libavfilter (using the libavfilter "scale" filter via vf_lavfi).
There is no reason to keep this filter any longer.
It's entirely useless. I left it in for a while, because the analog TV
code had a transitional bug that could switch chroma planes, but it was
fixed long ago. It's also available in libavfilter.
Apparently it was completely broken and essentially did nothing. This
was broken sometime in early mpv or mplayer2 times.
Get rid of it. If you _really_ need it, wait until FFmpeg ports it from
MPlayer, which will happen very soon.
This deals with subsampled YUV video that has odd sizes, for example a
5x5 image with 4:2:0 subsampling.
It would be easy to handle if we actually passed separate texture
coordinates for each plane to the shader, but as of now the luma
coordinates are implicitly rescaled to chroma one. If luma and chroma
sizes don't match up, and this is not handled, you'd get a chroma shift
by 1 pixel.
The existing hack worked, but broke separable scaling. This was exposed
by a recent commit which switched to GL_NEAREST sampling for FBOs. The
rendering was accidentally scaled by 1 pixel, because the FBO size used
the original video size, while textures_sizes[0] was set to the padded
texture size (i.e. one pixel larger).
It could be fixed by setting the padded texture size only on the first
shader. But somehow that is annoying, so do something else. Don't pad
textures anymore, and rescale the chroma coordinates in the shader
instead.
Seems like this somehow doesn't work with rectangle textures (and
introduces a chroma shift), but since it's only used when doing VDA
hardware decoding, and the bug occurs only with unaligned video sizes, I
don't care much.
Fixes#1523.
If a file is unseekable (consider e.g. a http server without resume
functionality), but the stream cache is active, the player will enable
seeking anyway. Until know, client API user couldn't know that this
happens, and it has implications on how well seeking will work. So add a
property which exports whether this situation applies.
Fixes#1522.
This was subtly broken by commit a937ba20. Instead of framestepping over
the timeline segment boundary, it would just unpause playback, because
seeking now resets mpctx->step_frames. This was especially apparent when
doing something like "mpv *.jpg --merge-files".
Fix by restoring the step_frames field specifically if the seek is done
for switching segment boundaries. Hopefully the number fields which need
such an exception on seeking won't grow and turn this code into a mess.
With mf://, rather long frame durations are common. By default, one
frame takes 1 second. This causes the if branch changed with this commit
to always being taken, which in turn leads to the player not being woken
up correctly. (As a consequence, it "freezes" by waiting for events that
never come, and moving the mouse cursor over the window will wake it up
again and advance video.)
Obviously, the code should account for how long the video frame takes.
The code is probably still not fully correct, but for now this fixes the
issue at hand.
Fixes#1521.
Make the default value part of the option metadata, instead of doing
this in the screenshot code. Makes more sense with --list-options and
the command.c option metadata properties.
This allows getting the log at all with --no-terminal and without having
to retrieve log messages manually with the client API. The log level is
hardcoded to -v. A higher log level would lead to too much log output
(huge file sizes and latency issues due to waiting on the disk), and
isn't too useful in general anyway. For debugging, the terminal can be
used instead.
The previous default ("no") seemed to be equivalent to "min" in practice
(though it might depend on the website, which is even worse).
Better just select the best stream by default.
Apparently broken files are popular enough that this is still needed.
It's relatively often asked on the IRC channel, and I also found it on
the archwiki mpv page.
The list of aspect ratios is taken from the archwiki page, with "16:10"
removed (because I doubt there are actually files with this aspect
ratio).
This queries the _ICC_PROFILE property on the root window. It also tries
to reload the ICC when it changes, or if the mpv window changes the
monitor. (If multiple monitors are covered, mpv will randomly select one
of them.)
The official spec is a dead link on freedesktop.org, so don't blame me
for any bugs.
Note that this assumes that Xinerama screen numbers match the way mpv
enumerates the xrandr monitors. Although there is some chance that this
matches, it most likely doesn't, and we actually have to do complicated
things to map the screen numbers. If it turns out that this is required,
I will fix it as soon as someone with a suitable setup for testing the
fix reports it.
Merge update_icc_profile() into get_and_update_icc_profile() - there's
no reason anymore to keep them separate. The former is only called by
the latter, and the separation of responsibilities between them is
blurry a best.
Query the ICC profile only if the corresponding feature is actually
enabled. Additionally, change the error behavior of this code. Make
loading failure non-fatal, and distinguish between runtime error and
unimplemented functionality.
Fix a memory leak in gl_lcms.c (although the changes in vo_opengl.c
already take care of this, it's just logical and cleaner).
Seems like several people agree that it's a good filter for downscaling.
Setting this option by default may also prevent people from accidentally
using an unsuitable filter for downscaling by setting "scale" and
without being aware of the impliciations (maybe). On the other hand,
this change is not strictly backwards compatible for the same reasons.
Also, allow disabling this option with scale-down="" (before this, not
setting it was the only way to do this - not possible anymore if it's
set by default). This is what the change in handle_scaler_opt() does.
Somehow, the default radius for filters with variable radius was set in
mp_init_filter(). gl_video.c used NAN as default value for the radius,
which would make the filter use the default radius. Simplify this, and
set the default radius directly in the gl_video options. It also makes
the options easier to understand, because the default value listed in
--vo=opengl:help actually shows the default value.
Remove the function can_use_filter_kernel(), because it doesn't set a
radius if none is set. The function is worthless anyway (something about
making filter_kernels.c reusable to other VOs, and trying to deal with
the possibility that it could provide filters not supported by
vo_opengl.)
Usually, a VO must react to VOCTRL_REDRAW_FRAME in order to redraw the
current screen correctly if video is paused (this is done to update
OSD). But if it's not supported, we can just draw the current image
again in the generic vo.c code.
Unfortunately, this turned out pretty useless, because the VOs which
would benefit from this need to redraw even if there is no image, in
order to draw a black screen in --idle --force-window mode. The way
redrawing is handled in the X11 common code and in vo_x11 and vo_xv is
in the way, and I'm not sure what exactly vo_wayland requires. Other VOs
have a non-trivial implementation of VOCTRL_REDRAW_FRAME, which
(probably) makes redrawing slightly more efficient, e.g. by skipping
texture upload. So for now, no VO uses this new functionality, but since
it's trivial, commit it anyway.
The vo_driver->untimed case is for forcibly disabling redraw for vo_lavc
and vo_image always.
At the time screenshot support was added, images weren't refcounted yet,
so screenshots required specialized implementations in the VOs. But now
we can handle these things much simpler. Also see commit 5bb24980.
If there are VOs in the future which can't do this (e.g. they need to
write to the image passed to vo_driver->draw_image), this still could be
disabled on a per-VO basis etc., so we lose no potential performance
advantages.
For some sites, youtube-dl sends a special user-agent. If we don't send
the same user-agent, the server will reject mpv's connection attempt.
This was observed with trailers.apple.com. Fix it by forcing the
user-agent youtube-dl uses.
Some sites set cookies when doing a website access, and require the
client to provide these cookies when downloading the actual media. This
is needed at least by nicovideo.jp. Fix by adding youtube-dl's cookies
to our request headers.
Both of these require a very recent youtube-dl version (youtube-dl added
the necessary headers a few hours ago). The script still works with
older youtube-dl versions, though.
This was previously done in common code but now it's left to backends. Also
remove the GL4 stuff since requesting a 3_2_Core context creates a 4.1 context
here (wtf).
Use different VOCTRLs for "window" and normal screenshot modes. The
normal one will probably be removed, and replaced by generic code in
vo.c, and this commit is preparation for this. (Doing it the other way
around would be slightly simpler, but I haven't decided yet about the
second one, and touching every VO is needed anyway in order to remove
the unneeded crap. E.g. has_osd has been unused for a long time.)
New command `mouse <x> <y> [<button> [single|double]]` is introduced.
This will update mouse position with given coordinate (`<x>`, `<y>`),
and additionally, send single-click or double-click event if `<button>`
is given.
vo.c queried the VO at initialization whether it wants to be updated on
every display frame, or every video frame. If the smoothmotion option
was changed at runtime, the rendering mode in vo.c wasn't updated.
Just let vo_opengl set the mode directly. Abuse the existing
vo_set_flip_queue_offset() function for this.
Also add a comment suggesting the use of --display-fps to the manpage,
which doesn't have anything to do with the rest of this commit, but is
important to make smoothmotion run well.
Repurpose demuxer->filetype for this. It used to be used to print a
human readable format description; change it to a symbolic format name
and export it as property.
Unfortunately, libavformat has its own weird conventions, which are
reflected through the new property, e.g. the .mp4 case mentioned in the
manpage.
Fixes#1504.
The symlink trick made waf go crazy (deleting source files, getting
tangled up in infinite recursion... I wish I was joking). This means we
still can't build the client API examples in a reasonable way using the
include files of the local repository (instead of globally installed
headers). Not building them at all is better than deleting source files.
Instead, provide some manual instructions how to build each example
(except for the Qt examples, which provide qmake project files).
The logic disabled framedropping if the frame was interpolated (i.e. the
render call is only done to interpolate between the previous frame, and
the frame before that).
It seems doing this wasn't even necessary, and broke framedrop in
smoothmotion mode. In fact, this code did nothing for display with video
fps below display fps. It did prevent the framedrop counter from going
up, though. So change it so that dropped interpolated frames are never
reported. (Doing so can give confusing results, such as dropping 1000s
of frames on slow operations like video start or changing filters.)
The MSDN documentation for IsFormatSupported says a return code of
AUDCLNT_E_UNSUPPORTED_FORMAT means the function "succeeded but the
specified format is not supported in exclusive mode." This seems to
imply that the format is supported in shared mode, and that's what the
old code assumed, however try_format would incorrectly return success
with some drivers.
The remarks section of the documentation contradicts that assumption. It
says that in shared mode, if the audio engine does not support the
caller-specified format or any similar format, ppClosestMatch is set to
NULL and the function returns AUDCLNT_E_UNSUPPORTED_FORMAT. This is the
same as in exclusive mode, so treat AUDCLNT_E_UNSUPPORTED_FORMAT the
same regardless of opt_exclusive. In shared mode, the format selection
code will fall back to the mix format, which should always be supported.
`core-idle` depends on seeking state `mpctx->restart_complete`,
so make `core-idle` notified whenever `seeking` is notified, too.
`paused-for-cache` can be changed on MPV_EVENT_CACHE_UPDATE obviously.
Finally, `MPV_EVENT_PLAYBACK_RESTART` should be notified after
`mpctx->restart_complete` changed.
SmoothMotion is a way to time and blend frames made popular by MadVR. It's
intended behaviour is to remove stuttering caused by mismatches between the
display refresh rate and the video fps, while preserving the video's original
artistic qualities (no soap opera effect). It's supposed to make 24fps video
playback on 60hz monitors as close as possible to a 24hz monitor.
Instead of drawing a frame once once it's pts has passed the vsync time, we
redraw at the display refresh rate, and if we detect the vsync is between two
frames we interpolated them (depending on their position relative to the vsync).
We actually interpolate as few frames as possible to avoid a blur effect as
much as possible. For example, if we were to play back a 1fps video on a 60hz
monitor, we would blend at most on 1 vsync for each frame (while the other 59
vsyncs would be rendered as is).
Frame interpolation is always done before scaling and in linear light when
possible (an ICC profile is used, or :srgb is used).
These aliases were removed in commit 1ec77214. Add a notice to the
manpage how to get these back. Apparently, "lanczos2" and "lanczos3"
were the only interesting aliases possibly used by someone, so the
description is limited to these two.
These are now auto-detected sanely; and enabled whenever it would be a
performance or quality gain (which is pretty much everything except
bilinear/bilinear scaling).
Perhaps notably, with the absence of scale_sep, there's no more way to
use convolution filters on hardware without FBOs, but I don't think
there's hardware in existence that doesn't have FBOs but is still fast
enough to run the fallback (slow) 2D convolution filters, so I don't
think it's a net loss.
This is significantly faster for FBOs on most modern GPUs, although it
did not result in a huge difference for the video source texture on the
sizes I tested. It might be more significant for 1080p or 4K content, so
it's worth revisiting this in the future.
It also renames SAMPLE_BILINEAR to SAMPLE_TRIVIAL to match the
semantics.
This is better even for non-separable. The only exception is when using
bilinear for both lscale and cscale. I've fixed the
documentation/comments to make more sense.
This is not quite the same thing as madVR's antiringing algorithm, but
it essentially does something similar.
Porting madVR's approach to elliptic coordinates will take some amount
of thought.
This fixes compatibility with GLES 2.0 and makes the code a bit neater
in general. It also properly forces indirect scaling for subsampled
video regardless of the lscale setting.
At least the scale_sep_fbo could have been uninitialized or initialized
incorrectly when switching between scalers (e.g. from bilinear to
lanczos). Calling check_resize() should take care of this.
Instead of reading back the image from textures, keep a reference to the
original image, and return that.
The main reason this was done this way was that originally, images
weren't refcounted, and would be deallocated or overwritten as soon as
the VO's draw call returned. But now there isn't really a good reason
for this anymore. One possibly _could_ argue that it was better because
other code could reuse the image sooner (e.g. for the cache), but on the
other hand, the VO runs already on a different thread, and filtering and
decoding each run on other threads too, so this argument probably
wouldn't hold up.
There was a case when we could have rendered to an output surface while
it's still used for display. Not sure why the API doesn't do this
automatically.
Instead of converting the hw surface to an image in the VO, provide a
generic way to convet hw surfaces, and use this in the screenshot code.
It's all relatively straightforward, except vdpau is being terrible. It
needs a huge chunk of new code, because copying back is not simple.
Before this commit, each hw backend had their own specific struct types
for context, and some, like VDA, had none at all. Add a context struct
(mp_hwdec_ctx) that provides a somewhat generic way to pass the hwdec
context around. Some things get slightly better, some slightly more
verbose.
mp_hwdec_info is still around; it's still needed, but is reduced to its
role of handling delayed loading of the hwdec backend.
This also fixes the maximum range to 16.0, which was previously set to
32.0 and incorrectly documented as 8.0. 16 taps should be more than
anybody will ever need, but it's the highest radius that's supported by
all affected filters.
And remove all uses of the VFCAP_CSP_SUPPORTED* constants. This is
supposed to reduce conversions if many filters are used (with many
incompatible pixel formats), and also for preferring the VO's natively
supported pixel formats (as opposed to conversion).
This is worthless by now. Not only do the main VOs not use software
conversion, but also the way vf_lavfi and libavfilter work mostly break
the way the old MPlayer mechanism worked. Other important filters like
vf_vapoursynth do not support "proper" format negotation either.
Part of this was already removed with the vf_scale cleanup from today.
While I'm touching every single VO, also fix the query_format argument
(it's not a FourCC anymore).
I'm still not sure how exactly handling of "lost" devices is supposed
to be handled. In theory, you only have to "reset" the device, instead
of recreating _everything_. But as it is, the code for proper uninit
and for handling the reset is exactly the same, so move it into a
function to reduce code duplication and the danger of potential bugs.
Apparently, extremely crappy graphics drivers don't allow you to use
shaders. Simply disable use of shaders if this happens, and use the
"old" method instead.
One unexpectedly tricky thing is that you need a d3d_device to create
a shader, which in turn requires a window, so the initialization order
changes.
Don't load all the legacy functions (including ancient extensions).
Slightly simplify function loader and context creation, now that legacy
GL doesn't need to be handled. Remove the code for drawing OSD in legacy
mode.
Remove all the header hacks, which were meant for ancient OpenGL headers
which didn't even support things like OpenGL 1.3. Instead, adjust the
GLX check to make sure we get both OpenGL 3x and 2.1 symbols. For win32
and OSX, we assume that the user has the latest headers anyway. For
wayland, we hope that things somehow go right.
Apparently, physically disconnecting the audio device (consider USB
audio) breaks the ALSA device handle forever. It will signal ENODEV.
Fortunately, it's easy for us to handle this, and we can just use
existing mechanisms that will make the playback core close and reopen
the AO. Whether the immediate reopening will actually succeeds really is
ALSA's problem, though.
Simply clamp off the U/V components in the colormatrix, instead of doing
something special in the shader.
Also, since YA8/YA16 gave a plane_bits value of 16/32, and a colormatrix
calculation overflowed with 32, add a component_bits field to the image
format descriptor, which for YA8/YA16 returns 8/16 (the wrong value had
no bad consequences otherwise).
If video output and VO don't support the same format, a conversion
filter needs to be insert. Since a VO can support multiple formats, and
the filter chain also can deal with multiple formats, you basically have
to pick from a huge matrix of possible conversions.
The old MPlayer code had a quite naive algorithm: it first checked
whether any conversion from the list of preferred conversions matched,
and if not, it was falling back on checking a hardcoded list of output
formats (more or less sorted by quality). This had some unintended side-
effects, like not using obvious "replacement" formats, selecting the
wrong colorspace, selecting a bit depth that is too high or too low, and
more.
Use avcodec_find_best_pix_fmt_of_list() provided by FFmpeg instead. This
function was made for this purpose, and should select the "best" format.
Libav provides a similar function, but with a different name - there is
a function with the same name in FFmpeg, but it has different semantics
(I'm not sure if Libav or FFmpeg fucked up here).
This also removes handling of VFCAP_CSP_SUPPORTED vs.
VFCAP_CSP_SUPPORTED_BY_HW, which has no meaning anymore, except possibly
for filter chains with multiple scale filters.
Fixes#1494.
In my opinion, libavformat should be doing this. But a patch handling a
very safe case rejected, so I suppose we have to do it manually. (This
patch was only escaping spaces, which can never work because they break
the basic syntax of the HTTP protocol.)
This commit attempts to do 2 things:
- Try to guess whether libavformat will use the URL for http. This is
not always trivial, because some protocols will recursively pass part
of the user URL to http in some way.
- Try to fix invalid URLs. We fix only the simplest case: only
characters that are never valid are escaped. This excludes invalid
escape codes, which happen with freestanding '%' characters.
Fixes#1495.
Sigh...
The C locale system is incredibly shitty, and if used, breaks basic
string functions. The locale can change the decimal mark from "." to
",", which affects conversion between floats and strings: snprintf() and
strtod() respect the locale decimal mark, and change behavior. (What's
even better, the behavior of these functions can change asynchronously,
if setlocale() is called after threads were started.)
So just check the locale in the client API, and refuse to work if it's
wrong. This also makes the lib print to stderr, which I consider the
lesser evil in this specific situation.
Handles mismatching libavfilter/libavdevice and libavcodec slightly
better.
libavfilter and libavdevice are optional, and thus are checked
separately and at a later point of the build. But if a user system has
at least 2 FFmpeg installations, and one of them lacks libavfilter or
libavdevice, the build script will pick up the libavfilter/libavdevice
package of the "other" FFmpeg installation. The moment waf picks these
up, all include paths will start pointing at the "wrong" FFmpeg, and the
FFmpeg API checks done earlier might be wrong too, leading to obscure
and hard to explain compilation failures.
Just moving the libavfilter/libavdevice checks before the FFmpeg API
checks somewhat deals with this issue. Certainly not a proper solution,
but since the change is harmless, and there is no proper solution, and
the change doesn't actually add anything new, why not.
This function is always available, which is reflected by the fact that
the configure check doesn't actually bother to check for its existence.
Instead, MinGW and Cygwin imply it. The check was probably "needed" when
the priority code was still in a separate source file.
Remove the check, and use _WIN32 for testing for the win32 API (in a
bunch of other places too).
In general, you need to check errno when using strtol(), but as far as I
know, strtol() won't reset errno on success. This has to be done
manually. The code could have failed sporadically if strtol() succeeded,
and errno was already set to one of the checked values.
(This strtol() still isn't fully error checked, but I don't know if it's
intentional, e.g. for parsing a numeric prefix only.)
Before this commit, ao_null was used as last fallback. This doesn't make
too much sense. Why would you decode audio just to discard it? Let audio
initialization fail instead. This also handles the weird but possible
corner-case that ao_null might fail initializing, in which case e.g.
ao_pcm could be autoselected. (This happened once, and had to be fixed
manually.)
This removes the slightly duplicated code for picking the required AO
driver if --audio-device forces one. Now --audio-device reuses the same
code as --ao for this.
As a consequence, ao_alloc_pb() and ao_create() can be merged into
ao_init(). Although the ao_init() argument list, which is already pretty
big, grows by one, it's better than having all these similar sounding
functions around.
Actually, I just wanted to do the change the following commit will do,
but I found this code was more of a mess than it had to be.
libdvdnav is garbage. Seeking by time is incredibly inexact, which is in
part due to the fact that it does not use the DVD seek tables. Instead,
it assumes CBR for certain ranges within the DVD, which makes especially
small seeks unreliable.
I have no good fix for this, other than hacking libdvdnav (I'd rather
prefer to remove mpv DVD support completely than doing this). So here's
a shitty hack that tries to workaround these problems. A basic
observation is that seeking in VLC seems to work quite well; however it
seems to be based on seeking by blocks (unless there is a subtle "trick"
I didn't see in the source code). mpv usually seeks by timestamps, so
this is not an option for us. However, we can pretend we are doing this
in the DVD layer.
The previous commit added a way to pass through relative seeks. This
commit uses the relative seek. STREAM_CTRL_SEEK_TO_TIME is backwards
compatible (there's still dvdread and bluray), so most code is about
extracing the relative seek information and turning it into a block
seek.
(Another way would have been using SEEK_FACTOR stuff, but that would
probably make for a less reliable way to handle this situation.)
Additionally, if a hr-seek is done, add an offset by 10 seconds. As long
as the error done by libdvdnav is not worse, this should help with hr-
seeks - although it makes them much slower.
Pass through the seek flags to the stream layer. The STREAM_CTRL
semantics become a bit awkward, but that's still the least awkward
part about optical disc media.
Make demux_disc.c request relative seeks. Now the player will use
relative seeks if the user sends relative seek commands, and the
demuxer announces it wants these by setting rel_seeks to true. This
change probably changes seek behavior for dvd, dvdnav, bluray, cdda,
and possibly makes seeking useless if the demuxer-cache is set to
a high value.
Will be used in the next commit. (Split to make reverting the next
commit easier.)
Before this, we merely printed a message to the terminal. Now the API
user can determine this properly. This might be important for API users
which somehow maintain complex state, which all has to be invalidated if
(state-changing) events are missing due to an overflow.
This also forces the client API user to empty the event queue, which is
good, because otherwise the event queue would reach the "filled up"
state immediately again due to further asynchronous events being added
to the queue.
Also add some minor improvements to mpv_wait_event() documentation, and
some other minor cosmetic changes.
We still need to send the VO a duration in these cases. Disabling
framedrop has logically absolutely nothing to do with these cases; it
was overlooked in commit 918b06c4.
So we always send the frame duration (or a guess for it), and check
whether framedropping is actually enabled in the VO code. (It would
be cleaner to send framedrop as a flag, but I don't care about that
right now.)
This complains within dvb_strtok_r() that savePtr is uninitialized.
There doesn't seem to be any code path where this can happen though, so
it's probably a false positive. Silence it anyway.
Broke operation with GLSL.
Since 1D texture usage was apparently (and mysteriously) good for speed,
it might be added back, but it's unknown how to do so in a clean way.
The "ontop" and "border" properties already used a common
mp_property_vo_flag() function, and the corresponding VOCTRLs used the
same conventions. "fullscreen" is pretty similar, but was handled
slightly similar. Change how VOCTRL_FULLSCREEN behaves, and use the same
helper function for "fullscreen" as the other flags.
Fixes#1472.
(Maybe these options should have been named --autofit-max and
--autofit-min, but since --autofit-larger already exists, use
--autofit-smaller for symmetry.)
Normally the player doesn't read from unselected streams, so this should
be a no-op. But unfortunately, some broken files can severely confuse
the player, and assign the same demuxer stream to multiple front-end
tracks. Then selecting one of the tracks would deselect the other track,
with the end result that the demuxer stream for the selected track is
deselected. This could happen with mkv files that use the same track
number (which is of course broken). timeline_set_part() sets the tracks
using demuxer_stream_by_demuxer_id(), using the broken non-unique IDs.
The observable effect was that the player never quit, because
demux_read_packet_async() told the caller to wait some longer for new
packets. Fix by returning EOF instead.
Fixes#1481.
This is for the ordered chapters case only. In theory this could have
resulted in initial audio, video or subs missing, although it didn't
happen in practice (because no streams were selected, thus the demuxer
thread didn't actually try to read anything). It's still better to make
this explicit.
Also, timeline_set_part() can be private to loadfile.c.
The last video frame is another case that has a separate code path,
although it's pretty similar to the one in commit 73e5aa87. Fix this
in a different way, which also takes care of the last frame case,
although without context the code becomes slightly more tricky.
As further cleanup, move the decision about framedropping itself to
the same place, so the check in vo.c becomes much simpler. The check
for the vo->driver->encode flag, which is remvoed completely, was
redundant too.
Fixes#1480.
If the program name isn't quoted and the .exe it refers to isn't found,
CreateProcess will add the program arguments to the program name and
continue searching, so for "program arg1 arg2", CreateProcess would try
"program.exe", "program arg1.exe", then "program arg1 arg2.exe". This
behaviour is weird and not really desirable, so prevent it by always
quoting the program name.
When quoting argv[0], escape sequences shouldn't be used. msvcrt, .NET
and CommandLineToArgvW all treat argv[0] literally and end it on the
trailing quote, without processing escape sequences.
The "\\" escape was rendered as "\" on the website. I'm hoping quoting
this in ``...`` will render it correctly.
Also add an example for show_text, which awkwardly does not require
escaping the "\".
If the video format changes (e.g. different frame size), a special code
path is entered to wait until the currently displayed frame is done.
Otherwise, the frame before the change would be destroyed by the
vo_reconfig() call.
This code path didn't respect --untimed; correct this.
Fixes#1475.
__STRICT_ANSI__ disables functions and definitions that aren't in ANSI
C. Unfortunately this includes j1(), which is used by the new
ewa_lanczos code. Cygwin's CFLAGS already unset __STRICT_ANSI__, but it
should be unset for both Cygwin and MinGW.
After finding out more about how video mastering is done in the real
world it dawned upon me why the "hack" we figured out in #534 looks so
much better.
Since mastering studios have historically been using only CRTs, the
practice adopted for backwards compatibility was to simulate CRT
responses even on modern digital monitors, a practice so ubiquitous that
the ITU-R formalized it in R-Rec BT.1886 to be precisely gamma 2.40.
As such, we finally have enough proof to get rid of the option
altogether and just always do that.
The value 1.961 is a rounded version of my experimentally obtained
approximation of the BT.709 curve, which resulted in a value of around
1.9610336. This is the closest average match to the source brightness
while preserving the nonlinear response of the BT.1886 ideal monitor.
For playback in dark environments, it's expected that the gamma shift
should be reproduced by a user controlled setting, up to a maximum of
1.224 (2.4/1.961) for a pitch black environment.
More information:
https://developer.apple.com/library/mac/technotes/tn2257/_index.html
This was forgotten when the option was implemented, and makes this
option work as advertised.
Fixes#1473 (though the default behavior is probably still stupid).
Adds about 7 lines of boilerplate per filter. This could be avoided by
providing a different entrypoint (something like af->filter_inplace),
which would basically mirror the old interface exactly for this kind of
filter. But I feel like it would just be a hack to support all those
old, useless filters better. (The ideal solution would be using a
language that can do closures to provide a compat. wrapper, but
whatever.)
af_bs2b has terribly repetitious code for setting up filter functions
for each format (most of them useless, in addition to bs2b being
useless), so I did something terrible with macros.
af_sinesuppress had commented code for float filtering (maybe it was
broken; it has been commented every since it was added in 2006). Remove
this code.
Just to make sure all filters get the correct format. Together wih the
check in af_add_output_frame(), this asserts that
af->prev->fmt_out == af->fmt_in
This also requires setting the "in" pseudo-filter (s->first) formats
correctly. Before this commit, the fmt_in/fmt_out fields weren't used
for this filter.
Support for taking screenshots when doing hardware decoding needs to be
added later.
This takes the last image queued to the VO, which is logically the image
the player thinks is on screen (so e.g. subtitles will match).
forget_frames() does not clear this, because seeking does not remove the
current image from the screen (until the next one is drawn).
Upon the "DEL" key binding or the "disable-osc" message, the OSC should
stay permanently invisible. This was recently broken (not sure by what),
because other code accidentally reenables it anyway, which resulted in
the OSC appearing again when moving the mouse.
The Qt example already does this. I hoped this was restricted to
QApplication only, but apparently Qt repeated this mistake with
QGuiApplication (QGuiApplication was specifically added for QtQuick at a
much later point, even though QApplication inherits from it).
Commit 5e25a3d2 broke handling of the initial frame (the one decoded
with initial_audio_decode()). It didn't update the pts_offset field,
leading to a shift in timestamps by one audio frame.
Fix by calling the actual decode function in a single place. This
requires slightly more changes than what would be necessary to fix the
bug, but it also somewhat simplifies the data flow.
The percent-pos property normally goes by time, except for file formats
like .ts or .ogg, where you can't trust the timestamps and duration info
to compute the position in the overall files. These use the byte
position and size instead.
When the file position was unavailable (e.g. due to an ongoing seek),
the percent-pos was unknown. Change it to use the time position instead.
In most cases, it's actually accurate enough, and the temporary
unavailability of the property can be annoying, e.g. on the terminal
status line.
The goal is switching the whole audio chain to using refcounted frames.
This brings the architecture closer to FFmpeg, enables better
integration with libavfilter, will reduce useless copying somewhat, and
will probably allow better timestamp tracking.
For now, every filter goes through a semi-awful wrapper in
af_do_filter(), though. This will be fixed step by step, and the wrapper
should eventually be removed. Another thing that will have to be done is
improving the timestamp handling and avoiding extra copies for the AO.
Some of the new code is rather similar to the video filter code (the
core filter code basically just has types replaced). Such code
duplication is normally very unwanted, but in this case there's probably
no other choice. On the other hand, this code is pretty simple (even if
somewhat tricky). Maybe there will be unified filter code in the future,
but this is still far away.
The purpose of this function was to filter only as much audio input as
needed to produce a certain amount of audio output. This could (in
theory) avoid excessive buffering when e.g. changing playback speed with
resampling.
Use of this was already removed in commit 5fd8a1e0. No problems were
experienced, so let's assume this feature is practically worthless.
(Though it's possible that it was quite useful over a decade ago, or in
some cornercases with evil files.)
Having any of these set to 0 makes no sense.
I think some code might still be using 0/0 aspect ratio to signal unset
aspect ratio, but I didn't find it. If there is still code like this, it
should be fixed instead.
Fixes#1467.
For some reason, schedule_resize() can be called with everything set to
0. The code couldn't handle wl->window.aspect set to 0, converting NaNs
to integers. Just work this around.
(I have no idea what I'm doing. This is probably a corner case caused
by my broken-ish wayland setup.)
BSTR_P() passes the string length and start pointer to printf-like
functions. If the lenfth is 0, the pointer can be NULL, but we're
actually still not allowed to pass a NULL pointer in any case.
This is mostly a technically, because nobody in their right mind would
attempt to specifically break such cases. But it's still undefined
behavior, and some libcs might be strict about this.
bstr is a bounded string type, consisting of a pointer and a length
value. If the length is 0, the pointer can be NULL. This is somewhat
logical due to how this abstraction works, but it can leak when
converting to C strings.
talloc_strndup() returns NULL instead of "" in this case, which broke
some other code. Use bstrto0() instead, which is the "proper" function
to convert bstr to char*.
Fixes#1462.
Reading IDs must be checked too. This was basically forgotten in commit
f3a978cd. Also set the *length parameter for ebml_parse_length() in some
error cases, which _really_ should happen.
Fixes#1461.
mpv needs at least an audio or video track to play something. If the
track selection is basically insufficient, the player will immediately
skip to the next file (or quit).
One slightly annoying thing might be that trying to play a subtitle file
will close the VO window, and then go to the next file immediately (so
"mpv 1.mkv 2.srt 3.mkv" would flash the video window when 2.srt is
skipped). Move the check to before the video window is possibly closed.
This is a minor cosmetic issue; one can use --force-window to avoid
closing the video window at all.
Fixes#1459.
Closing the video window sends CLOSE_WIN, which is normally mapped to
the "quit" command. The client API normally disables all key bindings,
and closing the window does nothing. It's simply left to the application
to handle this. This is fine - an embedded window can not be destroyed
by user interaction.
But sometimes, the window might be destroyed anyway, for example because
the containing window is destroyed. If this happens, CLOSE_WIN should
better not be ignored. We can't expect client API users to handle this
specially (by providing their own input.conf), so provide some fallback
for this pseudo key binding. The "quit" command might be too intrusive
(not every client necessarily handles "unexpected" MPV_EVENT_SHUTDOWN),
but I think it's still reasonable.
For some reason, mpv sometimes does not get a MapNotify event with
GtkSocket embedding. This happens maybe 1 out of 10 times. I'm not sure
how this can happen - it certainly shouldn't. Since I was not able to
find the cause, and causes an apparent "deadlock", here's a lazy hack to
fix the misbehavior.
Seems to work with GtkSocket and passing the gtk_socket_get_id() value
via "wid" option to mpv.
One caveat is that using <tab> to move input focus from mpv to GTK does
not work. It seems we would have to interpret <tab> ourselves in this
case. I'm not sure if we really should do this - it would probably
require emulating some other typical conventions too. I'm not sure if an
embedder could do something about this on the toolkit level, but in
theory it would be possible, so leave it as is for now.
Don't use vo_control() for sending VOCTRL_RESET when starting a seek.
This means vo_seek_reset() won't wait until the VO actually processed
VOCTRL_RESET. It happens asynchronously instead.
The impact of this change should be minimal, unless the VO is somehow
too busy (like blocking on vsync).
Remove the "all" special-behavior, and instead interpret trailing "*"
characters. --display-tags=all is replaced by --display-tags=* as a
special-case of the new behavior.
See #1404.
Note that the most straight-forward value for matchlen in the normal
case would be INT_MAX, because it should be using the entire string.
I used keylen+1 instead, because glibc seems to handle this case
incorrectly:
snprintf(buf, sizeof(buf), "%.*s", INT_MAX, "hello");
The result is empty, instead of just containing the string argument.
This might be a glibc bug; it works with other libcs (even MinGW-w64).
Apparently, originally this code was meant to be able to read past the
buffer somewhat, which is why the buffer allocation was padded by 8
byte. This is unclean and confuses valgrind. This probably could have
crashed with certain invalid files too.
Also revert the change added with 10a2f69; it should be not needed
anymore.
We now use threads and other pthread API a lot, and not always we use it
from threads created with pthread_create() (or the main thread). As I
understand, with static linking we would have to use
pthread_win32_thread_attach/detach_np() every time we enter or leave a
foreign thread. We don't do this, and it's not feasible either, so it's
just broken.
This still should work with dynamic pthreads-win32. The MinGW pthread
implementation should be unaffected from all of this.
Make their meaning more exact, and don't pretend that there's a
reasonable definition for "bits-per-pixel". Also make unset fields
unavailable.
average_depth still might be inconsistent: for example, 10 bit 4:2:0 is
identified as 24 bits, but RGB 4:4:4 as 12 bits. So YUV formats
seemingly drop the per-component padding, while RGB formats do not.
Internally it's consistent though: 10 bit YUV components are read as
16 bit, and the padding must be 0 (it's basically like an odd fixed-
point representation, rather than a bitfield).
bpp(bits-per-pixel) and depth(bit-depth for color component) can
be calculated from pixelformat technically but it requires massive
informations to be implemented in client side.
These subproperties are provided for convenience.
Enable asynchronous reading for external files. This excludes subtitle
files (so it's effectively enabled for audio files only), because most
subtitle files are fully read on loading, and running a thread for them
would just cause slowdowns and increase resource usage, without having
any advantages.
In theory, an external file could provide multiple tracks from the same
demuxer, but demux_start_thread() is idempotent, so the code can be
kept simple.
Should help with playing DASH with ytdl_hook.
We still keep the window pointer, because we want to call
QQuickWindow::resetOpenGLState() (which runs on the rendering thread
only). Interesting mess...
We must not try to remap channels with this. Whethever ALSA gives us,
and whatever we do with it, the result will probably be nonsense.
Untested, as I don't have the required hardware.
This avoids issues when upscaling directly in linear light, and is the
recommended way to upscale images according to imagemagick.
The default slope of 6.5 offers a reasonable compromise between
ringing artifacts eliminated and ringing artifacts introduced by
sigmoid-upscaling. Same goes for the default center of 0.75.
The VP9 codec parser has a bug: it doesn't set the data/size pointers
passed to it. As I understand, it must always do this, and in fact, if
it doesn't some libavcodec generic code would be in trouble too.
This helps with #1448, but is not the full fix for it. The codec parser
must be fixed in libavcodec itself.
The previous implementation of opengl-cb kept only latest flipped frame.
This can cause massive frame drops because rendering is done asynchronously
and only the latest frame can be rendered.
This commit introduces frame queue and releated options to opengl-cb.
frame-queue-size: the maximum size of frame queue (1-100, default: 1)
frame-drop-mode: behavior when frame queue is full (pop, clear, default: pop)
The frame queue holds delayed frames and drops frames if the frame queue is
overflowed with next method:
'pop' mode: drops all the oldest frames overflown.
'clear' mode: drops all frames in queue and clear it.
With default options(frame-queue-size=1:frame-drop-mode=pop),
opengl-cb behaves in the same way as previous implementation effectively.
For frame-queue-size > 1, opengl-cb tries to calls update() without waiting
next flip_page() in order to consume queued frames.
Signed-off-by: wm4 <wm4@nowhere>
vo_opengl was crashing since f811348d because it passed NULL for the
events parameter to vo_control. Normally the parameter should not be
NULL, so add a hack to account for this. In particular, we should
handle the events that are returned. For the call in preinit() we
skip this, but it most likely has no meaning anyway, because in this
stage no window is visible yet.
This was always supposed to work.
Just add the option declaration. Normally I'm not a fan of duplicating
such things, but in this case it's (still) harmless.
mpv can be built natively on a Windows machine using MSYS2. Add detailed
instructions on how to build and merge them with the existing
instructions for cross-compilation.
This one avoids use of a FBO. It's less flexible, because it uses works
around the whole QML rendering API. It seems to be the only way to get
OpenGL rendering without any indirections, though.
Parts of this example were insipired by Qt's "Squircle" example.
Also add a README file with a short description of each example, to
reduce the initial confusing.
This affects OSX, where memory profiles are updated e.g. on fullscreen
switches. The profile most likely doesn't change, but the LUT will
be generated and reloaded anyway.
Somewhat of a regression from commit f811348.
Fixes#1439.
If icc-path is set, but the thing is replaced with a memory profile,
then p->icc_path would point to deallocated memory.
Also, the NULL checks are unnecessary.
This used to be required to workaround PulseAudio bugs. Even later, when
the bugs were (partially?) fixed in PulseAudio, I had the feeling the
hacks gave better behavior. On the other hand, I couldn't actually
reproduce any bad behavior without the hacks lately. On top of this, it
seems our hacks sometimes perform much worse than PulseAudio's native
implementation (see #1430).
So disable the hacks by default, but still leave the code and the option
in case it still helps somewhere. Also, being able to blame PulseAudio's
code by using its native API is much easier than trying to debug our own
(mplayer2-derived) hacks.
glXGetProcAddress() is outdated, and as far as I know doesn't give all
the guarantees the "new" ARB function gives.
Probably doesn't matter too much, because until now it always appeared
to work. On the other hand, since this function is (bogusly) used only
on the gl3 code path, it could have happened that users hit this, and
just reverted to vo_opengl_old instead.
The function terminal_in_background() reports whether the player was
backgrounded. In this case, we don't want to annoy the user by still
printing the status to stderr. If no terminal interaction is assumed,
this mechanism is disabled, and stderr is always used. The read_terminal
variable signals this case.
Oddly, just redirecting stderr will disable output to stderr, because
the background check with tcgetpgrp() is done on stderr, but
read_terminal is still true (because that one depends on stdin and
stdout).
Explicitly disable this mechanism if --no-input-terminal is used by
setting read_terminal to true only if terminal input is actually
initialized.
When the given mp_image_params does not match with that of gl_video,
gl_video_config() always calls uninit_video() but calls init_video()
only if valid format is given.
Since uninit_video() does not change image_params of gl_video,
when the same params as the previous one is given to gl_video_config()
after gl_video is unitialized with invalid format, gl_video_config()
never calls init_video().
To prevent this, invalidate image_params of gl_video in uninit_video().
Previously we just forced loading a profile from file, but that has poor
integration for querying the OS / display server for an ICC profile, and
generating profiles on the fly (which we might use in the future for creating
preset 3dluts).
Also changed the previous icc-profile-auto code to use this mechanism, and
moved gl_lcms to be an opaque type with state instead of just providing pure
functions.
This causes more constant reading from the DVB device,
and seems not to hurt the stream-controls.
As the device cache seems to fill up when not
reading from it for several seconds, reading
into the cache fixes desync-issues after
the initial analyzeduration of the demuxer.
Was already possible before by injecting the magic PID
8192 into channels.conf, the flag makes this much more
useable and we also have it documented.
Useful not only for debugging, but also for incomplete
channels.conf (mplayer format...), multi-channel
recording, or channels which do dynamic PID switchng.
full-transponder is also useful for channels which switch PIDs on-the-fly.
ffmpeg can handle this, but it needs the full stream with all PIDs.
There is the STD which contains service-names
(ffmpeg can understand it),
and the EIT which contains the EPG
(ffmpeg does not parse it, but e.g. VLC does).
As we now have more PIDs in general, increase the maximum
number of demuxers we can open. ffmpeg has an internal
limit of 64 PIDs, we are still far from that.
If VDR-channel-config is used, we know the service_id.
Using that, PMT can be extracted from PAT (code for that
inspired by szap-s2 again).
For this, we need to demux PAT with a special filter-setup,
and once PMT was found it can be added to the PID-list.
PMT is only resolved the first time one tunes to a channel,
then stored in the channel-list in memory.
With PMT available, ffmpeg can safely decode the teletext-pids.
I also added a comment explaining the potentially occuring structures for future reference.
For tokenization, a custom strtok_r implementation is used,
inspired by strtok_r as implemented in musl and ffmpeg,
hopefully slightly more readable (av_strtok_r is not available in libav,
and strtok_r is not available everywhere).
Move PID-string parsing into a separate function, reduces code-duplication.
Drop TPID-parsing again since teletext-stream seems not really correctly identified (ffmpeg guessed mp3 in my test...).
Parsing now done using strtok + sscanf instead of sscanfs with fixed numbers of expected fields -
gives more correct checking of PID-overflow, will allow VDR-style PID-field parsing.
The PID-parsing is not really nicely done and also does not yet handle the special VDR-style syntax.
A later commit will move the parsing to a separate function handling that which also checks that the maximum PID-count is not exceeded.
In mplayer-style channels-config, we had a LNB-field used for that.
In old VDR times, the location-field was also containing DISEQc information,
now it does that only indirectly (location => LNB => vdr knows from lnb-config).
We only accept it as this if the field is fully numeric.
Now also "stream ID" (for DVB-S2) and "inversion" are understood.
The parameter-string can also provide information on FEC, rolloff etc. For DVB-S, "auto" which mpv uses by default should be fine, I can also confirm it works. For non-DVB-S cards, it might be useful to also parse this information in case of a vdr-channel list.
As I have no such hardware and thus would have to do it blindly, I added a FIXME.
Mostly complete vdr-channels.conf format documentation is at http://www.vdr-wiki.de/wiki/index.php/Channels.conf (german only).
Still incomplete. Initialize is_dvb_s2 boolean in channel-struct.
We first check whether the channels.conf-line at hand is sscanf'able with a vdr-style pattern. If yes, we assume it is a vdr-channel-config (we check whether sscanf consumed the full line).
The vdr-style config also contains a parameter-string which contains information about polarization + delivery type (e.g. DVB-S2).
With this change, S2-tuning works with a VDR-channel list.
Missing (later commits):
- vdr-parameter-string also contains other information, e.g. invert-flag, needs to be parsed.
- Diseqc-lnb-number is not present in VDR-config (I believe it is handled via the location-parameter + lnb-config there). For backwards compatibility, the location-parameter can be the lnb-number - we should test whether it is an int and assume this in this case.
- VID, AID and TID-lists are extremely ugly in their syntax for VDR. At the moment, only the first number is parsed (and TID fully ignored), needs to be fixed.
It contains now also a boolean which tells whether this is an DVB-S2 channel (not initialized yet - tbd in next commit).
We could also pass through the type of the delivery system (would be more flexible), but the delivery-system-enum is only available with S2API, so this is more backwards-compatible.
If someone has DVB-T2 / DVB-C2 hardware and extends the code to deal with it, this should be changed.
Also allows demuxers to buffer, and we explicitly discard stale QPSK events.
Inspiration taken from the szap-s2 implementation.
S2API is only used if available and for DVB-S cards - it might also be useful to have that for DVB-T2 / DVB-C2 tuning later, but I do not have the HW / no DVB-T2 broadcasting station nearby to test functionality.
This should do no functional changes, only usage of the different API. The S2API is more extensible and a requirement for all the HD-deliveries (e.g. DVB-S2) and since 2.6.28 is the successor of the old API.
Tuning to DVB-S2 channels actually already "works" like this if the delivery is hardcoded in - for a fully working implementation, the channels.conf.sat would need an additional field indicating the delivery type (VDR-type channels.conf have that).
This commit also:
- Cleans up some debug output.
- Moves an unneeded usleep(100000) (only needed for diseqc switching for SAT) into the DVB-S-only part of the implementation.
Define a new macro "DVB_USE_S2API" to be used to switch to S2API if available,
also document what is the main difference in the new S2API.
StreamID-handling is only supported since API 5.2, and some macros
were renamed or only added later with 5.8 (somewhere around kernel 3.6),
so some define-fixups are needed to support these intermediate
kernel header versions.
Commit 87c13de6 added a fallback to video resolution if the subtitle
resolution is unknown. Apparently this fixed some broken files with
vobsubs.
This broke some DVB subtitles. Apparently .ts captures with 1920x1080
video resolution and 720x576 subtitles do exist. The sample at hand had
some streams with 720x576 resolution and no sub resolution set, and some
streams with 1920x1080 resolution and sub resolution set (both against
the same 1920x1080 video). My conclusion is that 720x576 is the only
reasonable fallback for DVB (but I can't be sure).
The fallback is removed for PGS too. I don't know about the PGS case; it
seems the sub resolution must always be set, so it shouldn't matter.
Fixes#1425.
This makes vo_opengl_cb respond to controls like "gamma" and
"brightness". The commit includes an awkward refactor for vo_opengl to
make it easier for vo_opengl_cb.
One problem is a logical race condition. The set of supported controls
depends on the pixelformat, which in turn is set by reconfig(). But the
actual reconfig() call (on the renderer) happens asynchronously on the
renderer thread. At the time it happens, the player most likely already
tried to set some controls for command line options (see init_vo() in
video.c). So setting this command line options will fail most of the
time, though it could randomly succeed. This can't be fixed directly,
because the player can't wait on the renderer thread, because the
renderer thread might already wait on the player.
Although the line count increases, this is better for making sure
everything is handled consistently for all users of the mp_csp_params
stuff.
This also makes sure mp_csp_params is always initialized with
MP_CSP_PARAMS_DEFAULTS (for consistency).
No more equals signs for options that don't take values.
Complete values for options with preset choices.
Complete --no-whatever where applicable.
Fixes#997.
--sub-scale-by-window=no attempts to keep subs always at the same pixel
size.
The implementation is a bit all over the place, because it compensates
already done scaling by an inverse scale factor, but it will probably do
its job.
Fixes#1424. (The semantics and name of --sub-scale-with-window are
kept, and this adds a new option - the name is confusingly similar, but
it's actually analogue to --osd-scale-by-window.)
This adds an "auto" choice to the concurrent-frames suboption, and makes
it the default.
I'm not so sure about making this the default, though. It could lead to
excessive buffering with large CPU counts. But we'll see.
Our own code was introduced when FFmpeg didn't provide this API (or
maybe didn't even have a way to determine the CPU count). But now,
av_cpu_count() is available for all FFmpeg/Libav versions we support,
and there's no reason to have our own code.
libavutil's code seems to be slightly more sophisticated than our's, and
it's possible that the detected CPU count is different on some platforms
after this change.
If there's only 1 chapter, the seeking by chapter (using the chapter
property) will either jump to the chapter point, or quit playback. This
is as designed, but seems like a useless and annoying behavior.
Removes an annoying "No video PTS! Making something up." warning.
Mark it as keyframe, which is needed to prevent strange behavior with
PNG. Also, don't leak the picture data.
Options which take colors accept two variants. The first is "r/g/b/a",
the second is "#AARRGGBB". Since they put alpha at different places,
it's probably better to document the second variant explicitly. (It's a
bit strange that they put alpha in different places, but on the other
hand, it's kind of natural. The second variant should probably be
considered deprecated.)
mpv_opengl_cb_uninit_gl() can be called at any time; but then the
decoder must be destroyed due to complications with hardware decoding.
This is why kill_video() exists. To make things easier, there is the
invariant that while vo_opengl_cb is active, the OpenGL state must
exist. But kill_video() didn't actually destroy the VO; only the video
decoder. This could trigger an assertion (vo_opengl_cb.c:187).
Actually, the video output is always destroyed lazily at a later point
if the decoder is destroyed, but not early enough for out purposes.
Returning the property before the window is mapped could lead to
confusing behavior, and in particular strange differences between
vo_vdpau and vo_opengl. (vo_opengl creates the window right at the
start, while vdpau waits until the first reconfigure event.) It might
even be possible that for vo_opengl random results were returned,
because the hidden window can have different placement than the actual,
final one on initial video reconfig.
Fix this by returning the property only if the window is considered
mapped. command.c handles this case specifically, and makes the property
unavailable, instead of returning an empty list.
There are currently 568 pixel formats (actually fewer, but the namespace
is this big), and for each format elaborate synchronization was done to
call it synchronously on the VO. This is completely unnecessary, and we
can do with just a single call.
This is basically a hack; but apparently a needed one, since many
vapoursynth filters insist on having a FPS set.
We need to apply the FPS override before creating the filters. Also
change some terminal output related to the FPS value.
Most of this is explained in the code comments. This change should
improve performance with vapoursynth, especially if concurrent requests
are used.
This should change nothing if vf_vapoursynth is not in the filter chain,
since non-threaded filters obviously can not asynchronously finish
filtering of frames.
Add an explicit "signal" event type, because the implicit one was
confusing.
Don't rescale the Y axis of the second graph, it was nonsense.
Make the legend for the second graph separate (and cleanup the code
creating the graphs).
* bits instead of bytes
* add valid_bits argument
* just pass in the mp_chmap and get the number and wavext channel map from that
* indicate valid bits in waveformat_to_str
* make appropriate accomodations in try_format
Sort the legend by the used y value of binary events/signals, add a way
to filter branches (although that requires editing the script), and use
the full screen if the second subplot is not used.
We certainly don't use the mplayer configuration dir. The name didn't
matter, but now that it's in user-visible output (as part of config.h
being dumped in verbose mode), it's a bit too strange.
Mostly of cosmetic nature. Move initialization to the same place where
another component (cocoa) will start accessing the input context from a
foreign thread.
It was requested that mpv should print what features etc. have been
enabled at compile time. It can print the configure command line, but it
obviously doesn't include autodetected features.
I tried to think of a nicer way than dumping the config.h as text, but
this was still the simplest way.
I hoped we could always use libavresample, but the FFmpeg project is
being too dickish to enable libavresample by default - which means we
need our libswresample-to-libavresample hack anyway.
Give up, and use the "supported" one of the duplicated libraries when
compiling against FFmpeg (relying on the fact that libswresample won't
be present if compiling against Libav).
While there's no actual need to get rid of these, I want to make sure
nobody actually needs this stuff, and removing it is the best way to
get to know this. We still can revert this commit if it turns out there
is a significant need for this stuff.
The final goal is removing vo_opengl_old entirely. Add a warning, which
basically announces this intention.
The examples simple.c and cocoabasic.m can be compiled without
installing libmpv. But also, they didn't use the correct include path
libmpv programs normally use, so they couldn't be built with a properly
installed system-libmpv. That's pretty bad for examples, which are
supposed to show how to use libmpv correctly.
So do some bullshit that symlinks libmpv to a "mpv" include directory
under the build directory. This name-mismatch is a direct consequence of
the bullshit done in 499a6758 (requested in #539 for dumb reasons). (We
don't want to name the client API headers directory "mpv", because that
would be too unspecific, and clashes with having the mpv binary in the
same directory.)
If you have spaces or other "unusual" characters in your paths, the
build will break, because I couldn't find out where waf hides its
function to escape shell parameters (or a way to invoke programs
without involving the shell). Neither does such a thing to be
documented, nor do they seem to have a clear way to do this in
their code.
This also doesn't compile the Qt examples, because everything becomes
even more terrible from there on.
C++ is the worst language ever, and allows throwing any type, even if it
doesn't make sense. In this case, we were throwing char*, which the
runtime typically treats as opaque, instead of printing it as message if
such an exception was not caught.
Do so by using mp_subprocess(). Although this uses completely different
code on Unix too, you shouldn't notice a difference. A less ncie thing
is that this reserves an entire thread while the command is running
(which wastes some memory for stack, at least). But this is probably
still the simplest way, and the fork() trick is apparently not
implementable with posix_subprocess().
If the stdout or stderr write callback is NULL, then don't redirect this
stream. Preparation for the next commit.
Not sure what to do on Windows; it seems STARTUPINFO doesn't allow
redirection only one of them. So just let them write nothing. For our
intended use-case (next commit), this is probably sensible.
This is only needed for switching video track with `_`, since Cocoa
automatically handles cleaning up the application's presentation options when
quitting the process.
Fixes#1399
Tags keys are case-insensitive. Before commit 8048374a, the casing of
whatever FFmpeg returned was used (it was quite random). But since the
change, the values in --display-tags decides. Consider this an
accidental feature, and make the output nicer by capitalizing
the tag names.
The way we use rectangle textures (required by VDA for no reason) works
onl in OpenGL 3.0 or higher. Below that, the shader will fail to
compile. We could add support for older OpenGL versions, but that would
be a major pain.
This normally doesn't matter; mpv itself always creates OpenGL 3.2
contexts on OSX. But it could matter if a client API user uses
vo_opengl_cb, and gives it a 2.1 context (which OSX also allows you to
create).
This may or may not be useful for client API users.
Fold this API extension into the previous API bump. The previous bump
was only yesterday, so it's ok.
Until now, calling mpv_opengl_cb_uninit_gl() at a "bad moment" could
make the whole thing to explode. The API user was asked to avoid such
situations by calling it only in "good moments". But this was probably a
bit too subtle and could easily be overlooked.
Integrate the approach the qml example uses directly into the
implementation. If the OpenGL context is to be unitialized, forcefully
disable video, and block until this is done.
For some codecs, we need to invoke a codec parser (because libavcodec
will run into trouble otherwise). This was done based on the Matroska
codec field.
But this ignores handling of vfw-muxed files, which use a pseudo-codec
to signal presence of vfw structures, which we must unmangle to get the
real codec. Handle this by rearranging the code.
This fixes at least mp3-in-mkv for vfw-muxed files; typically old files.
Creating a plain Handle() should yield a NULL mpv_handle.
Also, remove the redundant non-const definition of the conversion
operator. At least in this situation it's not needed.
Also, add include guards around qthelper.hpp.
Use queued signals instead of QEvent for the wakeup notification. This
is slightly nicer, and reduces the chance that the event (QEvent::User)
could clash with other code using the same event.
Also switch to modern connect() syntax.
Destruction (e.g. when closing the window) was a bit broken. This commit
fixes some possible crashes, and should make lifetime management
relatively sane. (Still a bit complex, though. Maybe this code should be
moved into a tiny library.)
QtQuick runs the renderer on a separate thread. This thread is rather
loosely connected to the main thread. The loose separation is enforced
by the API, which also makes coordination of initialization and
destruction harder. Throw refcounting at the problem, which fixes it.
The refcounting wrapper introduced in the previous commit is used for
this.
Also contains some general cleanups.
If a filter exists, but has no metadata, just return success. This
allows the user to distinguish between no metadata available, and filter
not inserted.
See #1408.
This message can happen a lot for mkv files which index clusters in the
seekhead (which is also broken non-sense, but that's a different story).
Also remove a duplicate define from matroska.h.
There's no reason why parts of this demuxer would be in a separate
source file. The existence of this code is already somewhat questionable
anyway, so it may as well be dumped into a single file.
Even stranger that demux.c included mf.h for no reason (it was an
artifact from 2002 when the architecture was uncleaner).
This attempts to increase user-friendliness by excluding useless tags.
It should be especially helpful with mp4 files, because the FFmpeg mp4
demuxer adds tons of completely useless information to the metadata.
Fixes#1403.
Until now, these options took effect only at program start. This could
be confusing when e.g. doing "mpv list.m3u --shuffle". Make them always
take effect when a playlist is loaded either via a playlist file, or
with the "loadlist" command.
This message is printed when the audio device advertised a channel map,
but couldn't set it - which is probably a dmix bug (we'll never know,
ALSA doesn't take bug reports).
Print the requested map, so that the user (maybe) can make a connection
when seeing the message and the actually used channel map, which might
be less confusing. Or at least less useless.
Instead of just failing during channel map selection, try to select a close
layout that makes most sense and upmix/downmix to that instead of failing AO
initialization. The heuristic is rather simple, and uses the following steps:
1) If mono is required always prefer stereo to a multichannel upmix.
2) Search for an upmix that is an exact superset of the required channel map.
3) Search for a downmix that is the exact subset of the required channel map.
4) Search for either an upmix or downmix that is the closest (minimum difference
of channels) to the required channel map.
The details of the non-linear transformation from/to BT.2020's constant
luminance system don't really make sense with any other gamma curve,
since changing the gamma curve completely breaks the chroma channels.
Apparently this is what users would expect.
Going the way of least resistance (in terms of messing with this old,
rarely used code), sorting them by some kind of addition timestamp
(called priority in the patch) is the easiest.
Fixes#1390.
On uninitialization, the player will unselect all subtitles, and then
destroy the subtitle decoder. But it didn't correctly remove the
subtitle decoder from the OSD state, so it could happen that it would
access it after the decoder was destroyed.
Could lead to random crashes when switching files often.
Fixes#1389.
There where 3 major errors in the previous code:
1) The kAudioDevicePropertyPreferredChannelLayout selector returns a single
layout not an array.
2) The check for AudioChannelLayout allocation size was wrong (didn't account
for variable sized struct).
3) Didn't query the kAudioDevicePropertyPreferredChannelsForStereo selector
since I didn't know about it's existence.
All of these are fixed.
Might help with #1367
Ever since someone (not me) added some Matroska features which nobody
ever uses (ordered edition or some bullshit), I haven't had a fucking
clue what the fuck is going on in this fucking file. (Still agreed to
it, so it's my fault.)
mplayer2 handled missing chapters correctly (and I suppose in a somewhat
clean/simple manner), but the changed code doesn't. Since I can't even
follow this code because it's so arcanely complicated, just add a hack
that has the same effect.
Essentially, don't make it the mmap() argument, and just add it to the
memory address. This hides tricky things like alignment reequirements
from the user.
Strictly speaking, this is not entirely backwards compatible: this adds
the regression that you can't access past 2 or 4 GB of a file on 32 bit
systems anymore. But I doubt anyone cared about this.
In theory, we could be clever, and just align the offset manually and
pass that to mmap(). This would also be transparent to the user, but
minimally more effort, so this is left as exercise to the reader.
Makes all of overlay_add work on windows/mingw.
Since we now don't explicitly check for mmap() anymore (it's always
present), this also requires us to make af_export.c compile, but I
haven't tested it.
AudioChannelLayout uses a trailing variable sized array so we need to
query CoreAudio for the size of the struct it is going to need (or the
conversion of that particular layout would fail).
Fixes#1366
mpv does 'cascading' configs by overriding options as the config become more
important (bundle -> system level -> user level).
Unfortunately mpv also loads two sets of configs files one after the other [1].
First it looks for 'config', then for 'mpv.conf'. For this reason a mpv.conf in
ANY location will override ANY config files named config (even if the mpv.conf
is in a system path and config in a user path).
[1]: Relevant code in player/configfiles.c
load_all_cfgfiles(mpctx, section, "config");
load_all_cfgfiles(mpctx, section, "mpv.conf");
Fixes: #1361
vo_opengl was originally written against OpenGL 3 core, and it seems
GPUs/drivers supporting this are mostly sane. Later, it was made to work
with OpenGL 2.1 too. Lately we removed the requirement for RG textures,
and look, someone reported a problem with "lesser" Intel GPUs.
This commit does the same in vo_opengl what was added to vo_opengl_old a
long time ago.
Fixes#1383.
If a file (or a demuxer) is broken, seeking close to the end of the file
doesn't work, and seek_to_last_frame() will be called over and over
again, burning CPU for no reason.
Observed with incomplete mp4 files. That this can happen was already
mentioned in commit 090f6cfc, but I guess now I'll do something against
it.
hrseek_lastframe is cleared by reset_playback_state(), so it's only set
if seek_to_last_frame() was called, and no other seek happened since
then. If finding the last frame succeeds, no EOF will happen (unless the
user unpauses, but then it will simply remain at the last frame). If it
fails, then it will return immediately, without retrying.
I noticed that the IPC code does not use MSG_NOSIGNAL or SO_NOSIGPIPE.
The former is "only" POSIX 2008 and also requires switching to sendto(),
while the latter is even less portable.
Not going to bother with this obsolete 80ies crap, just block SIGPIPE,
and instruct client API users to do the same.
Using the IPC with a program, it's not often obvious that a newline must
be sent to terminate a command. Print a warning if the connection is
closed while there is still uninterpreted data in the buffer.
Print the OS reported error if reading/writing the socket fails. Print
an erro if JSON parsing fails.
I considered silencing write errors if the write end is closed (EPIPE),
because a client might send a bunch of commands, and then close the
socket without wanting to read the reply. But then, mpv disconnects
without reading further commands that might still be buffered, so it's
probably a good idea to always print the error.
stream_edl merely makes demux_edl act "special", which checks for the
stream type explicitly and then does something with its URL. If a cache
is added before the stream, it'll try to use the cache's URL (i.e. an
empty string), and will then obviously fail to parse the URL. While this
is slightly stupid, just disabling the entirely useless cache is the
most effective solution.
Fixes#1378.
Using them reduces state change, since now at least vo_opengl_cb has to
setup/break the vertex array bindings on every frame if no VAOs are
available.
This reverts the VAO related change in commit f64665e7.
Originally, this code was written to have full control over the OpenGL
state, rather than having to cooperate with unknown components by being
embeded like vo_opengl_cb is meant to be. As a consequence, it was
thought to be ok to setup a global binding (if the context is below
OpenGL 3.0, which guarantees VAOs).
This could break badly. Fix it by setting up and breaking the bindings
on entry/exit.
snd_pcm_prepare() was not always called, which could result in an
infinite loop.
Whether snd_pcm_prepare() was actually called depended on whether the
device was a hw device (or other characteristics; depending on
snd_pcm_hw_params_can_pause()), and required real suspend (annoying for
testing), so it was somewhat tricky to reproduce without knowing these
things.
GL_ARB_debug_output provides a logging callback, which can be used to
diagnose problems etc. in case the driver supports it. It's enabled only
if the vo_opengl "debug" suboption is set.
The code could as well be in demux.c, but it's better to avoid
accidental clashes with demux_lavf.c.
FFmpeg provides no way yet to map a mime type to a codec, so do it
manually. (It _can_ map a mime type to an "input format", but not a
codec.)
Fixes#1374.
Now it accepts the same renderer arguments as vo_opengl. This also
disables debug checks by default, and reverts the background color
override. Both can now be controlled by the host application.
Most image subtitle formats implicitly terminate the current subtitle
event with the next event (e.g. a new packet read from the demuxer will
instruct the subtitle render to stop display). If the subtitle event is
just trailing, it will be displayed forever. So there's no proper way
of doing this and we just apply an heuristic to avoid annoyances.
The libavcodec PGS decoder sets end_display_time to UINT32_MAX, in an
attempt to signal unknown end time (the API does not allow to signal
this properly, and this was a backwards compatible hack).
While we have no issues with the large value, our code wants to
distinguish between known and unknown end time explicitly.
reset_subtitles() works in mpctx->d_sub[], which is set to NULL before
calling it from uninit_sub(). This fixes resetting the subtitle when
cycling subtitle tracks.
Actually, this was probably a feature, because it's annoying if
subtitles don't show up when cycling them. But it also can have
unintended consequences, so get rid of it.
Whether we have texture_rg doesn't matter much anymore; the scaler
should be fine with this. But on ES 2.0, 1st class arrays are missing,
so even if filterable float textures should be available, it won't work.
Dithering (at least the "fruit" variant) will not work either, because
it uses floats.
Apparently GLES 2 and 3 do not support this. (The implementations I
tested with were derived from desktop OpenGL and were not overly strict
with this.)
This is no problem; just use GL_RGBA and mangle the channels in the
shader.
Also disable direct support for image formats like IMGFMT_RGB555 with
GLES; at least some of them are not supported in this form, and the
formats aren't important anyway.
Commit 0e8fbdbd removed the rg_texture requirement from vo_opengl;
commit 541f6731 changed to a more convenient method. Both commits broke
vo_opengl_old in some ways. vo_opengl_old always requires GL_ALPHA for
single-channel texture, because it draws the OSD without shaders and by
using certain blend modes.
So we need to explicitly distinguish between vo_opengl and vo_opengl_old
in the OSD renderer, and force fixed texture formats for vo_opengl_old.
The other logic is specific to the internals of vo_opengl. (Although it
might be possible to get the same result by playing with the old GL
fixed-function functions in vo_opengl_old. But seems like a waste of
time.)
Fixes#1370.
When setting the ALSA channel map, we never actually set the map we got
from ALSA directly, but convert it to mpv's, and then back to ALSA's.
mpv and ALSA use different conventions for mono, and there is already an
exception for ALSA->mpv, but not mpv->ALSA.
This was only added recently (c1e97161) as an attempt to minimize the
bad impact of channel layout device aliases. But use of these was
removed in commit 49df0132. Now this code does pretty much nothing, and
shouldn't be needed anymore. It does something when using spdif, but
this fallback won't work anyway.
The code in the demuxer etc. was changed to update all metadata/tags at
once, instead of changing each metadata field. As a consequence,
printing of the tags to the terminal was also changed to print
everything on each change.
Some users didn't like this. Add a very primitive way to avoid printing
fields with the same value again if metadata is marked as changed. This
is not always correct (could print unchanged fields anyway), but usually
works.
(In general, a rather roundabout way to reflect a changed title with ICY
streaming...)
Fixes#813 (let's call it a "policy change").
There are probably bugs with GLES support; also, if you somehow get GLES
instead of desktop GL on a desktop computer, something else is probably
wrong. So I see no point in using this automatically. We first need to
find out whether the GLES support works on real hardware, and whether
it is useful at all.
Apparently GLX can do this, so using EGL is not strictly required.
This code tries to create an ES context if creating a desktop GL context
fails. Also, a "x11es" backend for forcing ES (instead of desktop GL) is
added, mostly for testing and debugging.
Before this, missing additional but required functions were ignored.
("Main" functions still made it error out.) But we pretty much expect
that these are present on a given version level, and only an extremely
buggy OpenGL implementation would not do this.
Involve detection of software renderers in the probing properly. Other
VOs could handle probing also more gracefully, and e.g. produce less
noise if an API is unavailable. (Although other than the OpenGL VOs,
only vo_wayland will.)
Now the "sw" suboption for vo_opengl[_old] is strictly speaking not
needed anymore. Doing "--vo=opengl" disables the probing logic, and will
always force it, if possible.
Includes some simplifications as well.
Probably.
The version handling schema is a bit strange (and led to a tricky and
obvious bug), but it's quite similar to what OpenGL does at some places,
so I blame the OpenGL standard.
Rather basic support. Almost nothing works, and even if it does, it's
bound to be inefficient (due to texture upload). This was tested with
the nVidia desktop binary drivers, which provide GLES 2 support only.
However, nVidia is not known to be very strict about OpenGL, and the
driver is very new too, so the vo_opengl code will have bugs too.
This is needed for GLES 2 support. GLES 2 doesn't support
GL_UNPACK_ROW_LENGTH, and we shouldn't even use this constant, since a
GLES implementation could raise an error.
So set it only if neccessary, and leave it in the default state
otherwise. This also smuggles in a ES 2 fallback for glUploadTex(),
and querying an extension relevant for ES 2. For the alignment state
(GL_[UN]PACK_ALIGNMENT) do the same for symmetry. All 4 states
(alignment/rows x pack/unpack) are now assumed to be in their initial
states by default.
Also redo the PixelStorei handling in the function table. I could rebase
this, but look at the commit time.
Previously, this was sort of messed together with desktop OpenGL
loading. Add explicit support for ES, and clean up the mess. Also sort
the function arrays alphabetically (at least most of them).
In some cases, add functions to multiple sections. This keeps the number
of sections down.
Don't check for GL_ARB_vertex_array_object. This feature must be used in
OpenGL 3 core, while it may be not available at all in earlier versions,
and not using it if it's available as extension only has no advantages.
This commit also considers GLES 2, because actual support for it will be
added with a later commit.
Also get rid of some MPGL_VER() uses (see some commits ago).
This was a nice trick to get the mpv colormatrix directly into OpenGL,
because the memory representation happened to match.
Unfortunately, OpenGL ES 2 doesn't have glUniformMatrix4x3fv().
Even more unfortunately, the memory representation is now incompatible.
It would be nice to change it, but that would mean getting into a big
mess.
Really, this doesn't actually matter. It's printed as error only because
it was once thought to be an mostly unneeded fallback, but it turned out
this is still frequently needed, and users are getting confused.
For some reason, when using window embedding, and the window manager is
OpenBox, calling XSetWMNormalHints() before the window is mapped, the
initial window position will be off. It leaves some vertical space,
instead of placing it on the top/left corner. Suspiciously, the vertical
space is as much as a the height of normal window decoration.
I don't know what kind of issue this is. Possibly an OpenBox bug, but
then this happens even if the override-redirect flag is set. (This flag
basically tells the X server to ignore the window manager. Normally we
don't set it.) On other window managers, it works fine. So I don't know
why this is happening.
But this is easy to workaround. XSetWMNormalHints() isn't needed at all
if embedding.
Should fix#1235.
Always include the window position in winrc, even if the window
embedded. This should give the correct positions for things which still
interact with global coordinates, such as the xrandr code.
If GL_RED was not available, we used GL_ALPHA. But this is an
unnecessary complication, and it's easier to use GL_LUMINANCE instead.
With the latter, a texture will return the .r component set, and as long
as the shader doesn't look at the other components, the shader doesn't
need any changes.
Some of the changes added in 0e8fbdbd are now unneeeded.
Also, realign the entire gl_byte_formats_legacy table.
OpenGL 3.0 has the weird situation that it's "hard" to detect whether
legacy GL is really present. Since we pretend that we have OpenGL 3.0 if
we use GLES, it was assuming legacy GL is available.
The idea of using -Werror=format-security comes from MPlayer.
Try to use the compiler flags with any compiler. There's no reason not
to apply them on clang.
Older OSX versions are missing some OpenGL 3 symbols, apparently. At
least there's some precedent in the headers. Just add the symbols
manually for now to fix OSX (on travis-ci) compilation.
Tested with MESA on software emulation. Seems to work well, although the
default FBO format in opengl-hq disables most interesting features. I
have no idea how well it will work on real hardware (or if it does at
all).
Unfortunately, some features, including playback of 10 bit video, are
not supported. Not sure what to do about this.
GLES 2 or 1 do not work.
Remove the readback stuff; it was a useless mess.
Don't test GL_R16 as FBO. The intention was to measure the effective
bitdepth of the texture, except that it was never actually done.
(There's also a OpenGL function which is supposed to retrieve the
bitdepth, but we don't use that either.)
I'm hoping this is generally more compatible, and it works with GLES.
This probably has not much of an effect on desktop GL. It also switches
only the default format for --vo=opengl, not --vo=opengl-hq.
"-hq" already uses GL_RGBA16, though since it's a sized format, the
story is a bit different, and it won't work on GLES either.
Older GLSL dialects as well as GLES3 do not support the following things
in expressions:
- implicit conversions of integer constants to float
- arithmetic of float*vecN
As suggested in issue #1251. I think the main point is that it acts as
a hack to return to the DVD menu when jumping past the last chapter,
because it will reach title-EOF, instead of explicitly jumping to the
next file (which usually exits the player).
This basically reverts commit 8b7418d, except it doesn't include an
off-by-1 error (apparently).
Closes#1251.
Because of the icc-profile-auto option (which was added at a later
point), supporting this would probably be slightly messy: the ICC
profile can spontaneously update, and then it would overwrite the
previously set options.
Don't make icc-profile-auto fatal if unsupported. The "auto" indicates
that it will use whatever it finds, even if it's nothing.
Also add a warning; before this commit, it just refused to initialize
without explanation.
As a mostly unrelated cosmetic change, remove redundant parameters which
had no point anymore.
Probably fixes#1359 (or rather works it around by disallowing it).
Also clarify the statement about what we expect to happen by default.
It's well possible that distros at some point will fix their ALSA
configuration, and e.g. enable the upmix plugin by default.
Padding with spaces is very useless for OSD (because most fonts are
variable width), but it's good when using it on the terminal, e.g. for
reproducing the default terminal status line.
This should work well with most audio APIs, except ALSA. A long-winded
explanation is provided how to make ALSA multichannel output work.
All other AOs should have no such problems. Of course it's possible
that previously unknown issues arise, because I assume that enabling
multichannel audio is actually relatively rare.
This also disables codec downmix by default, which could change the
audio output due to different mixing in the codec and libavresample.
Fixes#1313.
The "old" method (before the ALSA channel map API) used device aliases
like "surround51" to set the channel layout. The "interesting" part was
that these devices usually redirect to a hardware device. This means
playing stereo would lead you to the "default" device (dmix), while e.g.
5.1 to "surround51", which automatically takes care of the fact that
dmix can't do 5.1.
This is pretty much nonsense, though. It shouldn't depend on the damn
input media file whether the player is going to use shared access (dmix)
or exclusive access (direct hw device).
As a consequence, by default ao_alsa will do only what dmix can do. If
the user actually wants multichannel, he has to select a suitable hw
device with --audio-device. From there on, the correct speaker mapping
will be ensured via the channel mapping API.
The change is preparation for making multichannel output the default (as
far as supported by the audio output API). Of the common APIs, only ALSA
messes up beyond repair, so I feel like this change is needed.
On ancient alsa-lib versions, only stereo and mono can be played with
this branch.
dmix reports channel layouts it doesn't support. The rest of the
technical part of the story is in the code comment.
This seems to be the only reasonable way to fallback from trying to
initialize certain devices (like dmix) with multichannel audio. We could
probably add support for such padding channels to our audio chain or to
ao_alsa itself, but this would probably be much more work than this
commit.
What dmix does is probably a bug. I've tried to report it to ALSA. Thay
have a link on their website to a bug tracker, but it's a dead link, and
has been for years. I've posted to alsa-devel, but received no reply.
I'm thus assuming this absolutely retarded behavior is by design, and
nothing will happen to improve upon it.
I'm considering sending Lennart Poettering a "thank you" email, because
with PulseAudio, multichannel audio just works (although some other
things just don't work).
Before this commit, this was defined to trigger undefined behavior. This
was nice because it required less code; but on the other hand, Lua as
well as IPC support had to check these things manually. Do it directly
in the API to avoid code duplication, and to make the API more robust.
(The total code size still grows, though...)
Since all of the failure cases were originally meant to ruin things
forever, there is no way to return error codes. So just print the
errors.
The message was missing a '\n', so it was merged with the next line,
which also typically caused it not to be printed with the colors for
warnings.
Print the full new path in the warning message. (Normally, cfg should
never be NULL, so accounting for this case is just for robustness.)
Obscure feature, and I've never heard of anyone using it.
The anaglyph effects can be reproduced with vf_stereo3d. The only thing
that can't be reproduced with it is "quadbuffer", which requires special
and expensive hardware.
- --lua and --lua-opts change to --script and --script-opts
- 'lua' default script dirs change to 'scripts'
- DOCS updated
- 'lua-settings' dir was _not_ modified
The old lua-based names/dirs still work, but display a warning.
Signed-off-by: wm4 <wm4@nowhere>
This was requested.
It seems libdvdread can't get the duration for titlesets other than the
currently opened title. The data structures contain dangling pointers
for these, and MPlayer works this around by opening every title
separately for the purpose of dumping the title list.
Apparently, we treat different tracks as titles, so returning the number
of titles (which is user-visible as "disc-titles" property) is
completely misguided.
Apparently, libdvdnav always reports a last chapter that points to the
exact end of the title. This is useless for us.
Again, same possible caveats as with the previous commit.
This way, chapter 0 will always point to the start of the title.
At least this is the intention; it's likely that DVDs as well as
libdvdnav do random things that lead to random results.
GNU sed and BSD sed don't share the same options for editing files in-place,
so a workaround is needed.
The most simple way is to use a pure python implementation of applying the
changes.
The --keep-open behavior was recently changed to act only on the last
file due to user requests (see commit 735a9c39). But the old behavior
was useful too, so bring it back as an additional mode.
Fixes#1332 (or rather, should help with it).
Do this by automatically adding the option, if the aliased option name
also has a "no-..." variant.
Could be easier by manually adding "no-..." variants to the option list,
but this seems better because you can't just forget it.
MP_NOPTS_VALUE (basically INT64_MIN) is basically an special timestamp
value that means "unset" or "unknown". Its exact value is internal, and
should never be returned or interpreted by any API.
So return "no" instead (what is also what the parser accepts).
Commit d38bc531 is incorrect: the 50ms queue-ahead value and the flip
queue offset have different functions. The latter is about calling
flip_page in advance, so the change attempted to show video frames 50ms
in advance on all VOs.
The change was for vo_opengl_cb, but that can be handled differently.
These actually are harmless. Even if the data the reader is working on
is essentially random, it's treated like untrusted input data, so there
should be no harm.
But it upsets tools like valgrind.
Probably fixes#1329.
In theory, vo_opengl supports operation without framebuffers. But this
has been broken for a while now (commit cc00b3ff is a contender). It
crashed because it unconditionally called gl->BindFramebuffer() (which
is NULL if framebuffers are missing).
Since this function is actually only called to set the default
framebuffer, the simplest way to deal with this is to provide a dummy
function, insteas of uglifying the code with additional if branches.
The result isn't quite what I imagined, because the A-point is never
marked as a seek point (so you can't jump between A and B), but it's
still slightly better than before.
I think that's expected; mpv shouldn't draw anything while no video is
active. This doesn't blend transparently, though.
Also document the vo_opengl_cb thing.
After being bitten by this, I decided that this mostly unnecessary
requirement sucks.
Allowing this makes it easier to use libmpv, because it can be set after
mpv_initialize(). The latest reasonable time an API user can set this
variable is before actually loading a file.
The previous 2 commits make sure nothing bad can happen if the option is
changed at runtime even if a VO is active. The Cocoa backend should be
fine and doesn't need a change.
This adds API to libmpv that lets host applications use the mpv opengl
renderer. This is a more flexible (and possibly more portable) option to
foreign window embedding (via --wid).
This assumes that methods like context sharing and multithreaded OpenGL
rendering are infeasible, and that a way is needed to integrate it with
an application that uses a single thread to render everything.
Add an example that does this with QtQuick/qml. The example is
relatively lazy, but still shows how relatively simple the integration
is. The FBO indirection could probably be avoided, but would require
more work (and would probably lead to worse QtQuick integration, because
it would have to ignore transformations like rotation).
Because this makes mpv directly use the host application's OpenGL
context, there is no platform specific code involved in mpv, except
for hw decoding interop.
main.qml is derived from some Qt example.
The following things are still missing:
- a way to do better video timing
- expose GL renderer options, allow changing them at runtime
- support for color equalizer controls
- support for screenshots
The most awesome codec, not.
The actual code for svq3 is actually just the part that checks for
MKV_V_QUICKTIME (no other QT-muxed codecs are supported). The rest is
minor refactoring, that actually improves the code in general.
This is just enough to support the 2 svq3-in-mkv sample files I have.
There were complaints that a chapter seek past the last chapter was
quitting the player. Change the behavior to what is expected: the last
frame.
If no chapters are available, this still does nothing.
Not all filter sizes the shaders could handle were in the filter_sizes
list. The shader can handle any multiple of 4 (the sizes 2 and 6 are
special-cased to keep it simple).
Add all possible filter sizes, up to 64. 64 is ridiculously high anyway.
Most of the larger filter sizes are completely useless for upscaling,
but help with the fancy-downscaling option. (Although it would still be
more efficient to use cascaded scalers to handle downscaling better.)
I considered doing something less stupid than the hardcoded array, but
it seems this is still the simplest solution.
Before this commit, the convolution scaler shader functions were pre-
instantiated in the shader file. For every filter size, a corresponding
function (with the filter size as suffix) had to be present.
Change this, and make the C code emit the necessary bits.
This means the shader code is much reduced. (Although hopefully it
doesn't make shader compilation faster - it would require a really dumb
compiler if it spends its time on dead code.)
It also makes it more flexible, which is the main goal.
The DEF_SCALER0 stuff is needed because the C code writes the header of
the shader, at a point where scaler macros are not defined yet.
This was a microoptimization for small filters which need 4 or less
weights per sample point. When I originally wrote this code, using a 1D
texture seemed to give a slight speed gain, but now I couldn't measure
any difference.
Remove this to simplify the code.
There's not much of a reason to have the actual convolution code in a
separate function. Merging them actually simplifies the code a bit, and
gets rid of the repetitious macro invocations to define the functions
for each filter size.
There should be no changes in behavior or output.
Windows uses a heuristic to determine if a window should appear
fullscreen. If the active window's client area covers the whole screen,
the taskbar should move to the bottom of the Z-order, allowing the
window to show through.
Unfortunately, sometimes it doesn't work and the taskbar stays on top of
the "fullscreen" window. ITaskbarList2->MarkFullscreenWindow explicitly
tells the shell that a window wants to be fullscreen, so the taskbar is
always at the bottom of the Z-order while the marked window is active.
This might help with #999. Firefox also uses this interface to fix
fullscreen issues.
This should clearly be impossible, but it seems to happen with ordered
chapters for a user.
Since I can't tell what the actual bug is and it seems impossible to
know the details without downloading possibly huge files, this is
probably the best we can do.
Should at least partially fix#1319.
It feels strange that seeking past EOF with --keep-open actually leaves
the player at a random position. You can't even unpause, because the
demuxer is in the EOF state, and what you see on screen is just what was
around before the seek.
Improve this by attempting to seek to the last video frame if EOF
happens. We explicitly don't do this if EOF was reached normally to
increase robustness (if the VO got a frame since the last seek, it
obviously means we had normal playback before EOF).
If an error happens when trying to find the last frame (such as not
actually finding a last frame because e.g. the demuxer misbehaves), this
will probably turn your CPU into a heater. There is no logic to prevent
reinitiating the last-frame search if the last-frame search reached EOF.
(Pausing usually prevents that EOF is reached again after a successful
last-frame search.)
Fixes#819.
This gives better results with fancy-downscaling. The issue here is that
fancy-downscalign "extends" the filter radius by some amount, which
requires using a larger filter size and shader. Then most of the filter
is "unused", but some filters still return non-0 coefficients, which
create heavy artifacts. Just clamp them off.
I'm not sure if this is the right solution, but at least it's better
than before.
Based on patch by Yuriy Kaminskiy [yumkam gmail].
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37330 b3059339-0415-0410-9bf9-f77b7e298cf2
Signed-off-by: wm4 <wm4@nowhere>
Also replace the weights calculations for 8/12/16 with the generic
weight function definition macro. (The weights 2/4/6 follow slightly
different rules.)
This message was added in commit a0acb6ea. But it showed up in all sorts
of inappropriate contexts, such as when opening m3u from an unseekable
http URL, or playing DVDs. So I guess this didn't work out. Disabling it
again.
m3u files are normally just text files with a list of filenames. Nothing
actually mandates that there is a header. Until now, we've rejected such
files, because there's absolutely no way to detect them.
If nothing else claims the file, the extension is ".m3u", and if the
contents of the file look like text, then load it as m3u playlist. The
text heuristic is pretty cheap, but at least it should prevent trying
to load binary data as playlist. (Which would "work", but result in a
catastrophic user experience.)
Due to the text heuristic, UTF-16/32 files will be rejected (unless they
have a header), but I don't care.
Whether we print it as warning or error doesn't really matter; we
continue anyway. (I don't actually know what the implications of running
in non-blocking mode are; for what's it worth, when I tested with
explicitly changing to non-blocking, it seemed to work fine anyway, so
don't change that part.)
ALSA returns "FL" as channel layout when trying to play mono. mpv and
libavresample don't like this; in particular, using libavresample to
convert stereo to "FL" fails.
If no-block was given, the device would be opened with SND_PCM_NOBLOCK.
Also, after opening, blocking mode was unconditionally enabled anyway
with snd_pcm_nonblock(). Further, if opening with SND_PCM_NOBLOCK
failed, opening was retried without this flag.
This doesn't make any sense to me, and I've never heard of someone using
this suboption. I suspect it has to do with ancient ALSA bugs or API
caveats. Remove it and simplify the code.
ALSA is crap. It's impossible to make multichannel playback just do the
right thing. dmix (the default on most distros) can do stereo only, and
will refuse to play multichannel. On the other hand, if you try like mpv
(and mplayer) to open a multichannel device (like "surround51" etc.),
this will actually open a hardware device, which will either fail if
dmix is active, or block out dmix if opening succeeds.
This commit falls back to "default" (i.e. dmix) if opening a
multichannel device fails, which is a tiny step towards the right
behavior. (Although fixing it fully is impossible.)
This was completely breaking any low-level caching. Change it so that at
least demuxer caching will work.
Do this by using the metadata cache mechanism to funnel through the menu
commands.
For some incomprehensible reason, I had to reorder the events (which
affects their delivery priority), or they would be ignored. Probably
some crap about the event state being cleared before it could be
delivered. I don't give a shit.
All this code sucks. It would probably be better to let discnav.c access
the menu event "queue" directly, and to synchronize access with a mutex,
instead of going through all the caching layers, making things
complicated and slow.
If the user has LEFT/RIGHT/etc. bound in his input.conf, then these were
overriding the menu keys in dvdnav mode.
This hack works because the dvdnav crap happens to be the only user of
MP_INPUT_ON_TOP. If it finds a default key binding in the dvdnav menu
section, it will use that, instead of continuing search and possibly
finding the user key bindings meant for normal playback.
This was required by vf_pp, which was just removed.
vf_dlopen has this stuff in its API. This API is considered stable, so
the related fields are not removed from it. But the fields are always 0
now, so there's no point in keeping the example program around.
vf_pullup.c did some extremely awkward passthrough of this information,
but didn't actually use it.
This is an ancient filter, and we assume it's not useful anymore.
If you really want this, it's still available in libavfilter (e.g. via
--vf=lavfi=[pp...]). The disadvantage is that mpv doesn't pass through
QP information to libavfilter. (This was probably the reason vf_pp still
was part of mpv - it was slightly easier to pass QP internally.)
This wasn't done before because there was no advantage in "abstracting"
it. This changed, and putting this into its own files is better than
messing it into gl_common.c/h.
Same as with the previous commits.
In theory, vdpau/x11 GL interop doesn't assume GLX. It could use EGL as
well. But since it's always GLX in practice, so we're fine with this.
Remove the gl_hwdec.mpgl field - it's unused now.
Basically, don't access the vo field.
There's also no reason anymore to access MPGLContext. We still need to
access loaded GL functions though, so add a field for that to gl_hwdec.
Untested.
This fixes using the mpv:// custom protocol on Yosemite were apparently
we receive an url which is automatically urlencoded by the system.
/cc mpv-player/stable
This is simply not allowed, and doing it triggered an assertion. It's
still not allowed, because the terminal and related functionality is a
global resource, and there doesn't seem to be a sane way to manage the
signal handlers.
But be a bit nicer, and just the terminal if it's already in use.
Note that terminal _output_ happens anyway. This becomes usable with
this commit. To facilitate logging-only usage further, also explicitly
disable terminal input, so that "terminal=yes" can be used for logging
without much interference with other things. (It'll still overwrite some
signal handlers, though.)
Always set the viewport on entry. The way the viewport is tracked is a
bit complicated in my opinion, and in fact it doesn't even reduce the
number of GL calls. Setting it on entry is actually redundant if video
covers the screen fully, because the handle_pass() unconditionally sets
it anyway, but avoiding it would complicate the cases gl->Clear() is
actually needed.
Add a fbo argument to gl_video_render_frame(). This allows you to render
into a FBO rather than the default framebuffer. It will be useful for
providing an API to render on an external GL context. (If that will
actually be added.)
Seems like a waste not to print this.
Anyone with enough technical knowledge to have use for the exact error
can map the number back to the GL symbol, so don't bother to convert it
to a symbol.
So the OSC will still appear when using --no-input-default-bindings. It
also means it may override a user's mouse_move binding, but I guess
users who don't want the OSC should just use the --no-osc option.
The player thinks an error happened because no audio or video was played
after finishing the file, but this obviously makes no sense with stream
dumping. (error_playing follows the client API convention that negative
values are errors.)
Channel amp otpions were not copied correctly: it copied the size of a
pointer to struct chmap, not the struct itself. Since mp_chmap is
currently 9 bytes, this meant the last channel entry was not copied
correctly on 64 bit systems, leading to very strange failures. It could
be triggered especially when using the client API, because the client
API always copies options on access (mpv command line options tend to
work directly on options).
This could trigger an assertion when using ao_alsa or ao_coreaudio. The
code was simply assuming the number of channel maps was bounded
statically (which was true at first in both AOs).
Fix by using dynamic memory allocation. It needs to be explicitly
enabled by the AOs by setting a temp context, because otherwise the
memory couldn't be freed. (Or at least this seems to be the most elegant
solution.)
Fixes#1306.
All of these are already the defaults.
One exception is glDepthMask(), which is enabled by default. But if the
framebuffer has no depth buffer anyway, it shouldn't make a difference.
Before it used whatever was in ao->format and changed the bits even
though this might have nothing to do with the actual WAVEFORMAT
negotiated with WASAPI.
For example, if the initial ao->format was a float and we had set the
WAVEFORMAT to s24, this would create a non-existent float24 format.
Worse, it might put an u16 into ao->format when WAVEFORMAT described s16.
WASAPI doesn't support unsigned at all as far as I can tell.
If EOF is reached after reading a line, the EOF flag is set. This was a
problem for the m3u code, which checked for EOF _after_ reading a line,
which will discard the last line read.
Also fix a typo in an unrelated part of the file.
By now, input.conf is actually just a small part of input handling.
Rename the section to something else ("command interface" was the
first reasonable thing that came to mind).
Also fix a minor typo further down.
libxkbcommon keysyms are the same as X11 keysyms (sans prefix),
so I simply copied the missing subsection from x11_common.c.
Signed-off-by: Sergey Kvachonok <ravenexp@gmail.com>
Signed-off-by: wm4 <wm4@nowhere>
I don't know why this done; most likely it had no real reason.
Remove it because it breaks "refresh seeks" to the same position.
(Although the refresh seeks mpv sometimes does were fine.)
Yep, Lua is so crappy that the stdlib doesn't provide anything like
this.
Repurposes the undocumented mp.format_table() function and moves it to
mp.utils.
MS Windows doesn't allow windows larger than the screen, so we include
a hack to make the window smaller. This hack recenters the window (what
else would it do?).
It didn't account for the virtual offset of the current screen, and it
was reported that it forces the window to the first screen.
Should fix#1292.
Ordered chapter EOF was handled as special-case of ending the last
segment. This broke --kee-open, because it set AT_END_OF_FILE in an
"inconvenient" place (after checking for --keep-open, and before the
code that exits playback if EOF is reached).
We don't actually need to handle the last segment specially. Instead, we
remain in the same segment if it ends. The normal playback logic will
recognize EOF, because the end of the segment "cuts off" the file.
Now timeline_set_from_time() never "fails", and we can remove the old
segment EOF handling code in mp_seek().
All of this is basically due to how MPlayer's codecs.conf worked. It
had a demuxer-interface based an AVI, using FourCCs and data structures
found in AVI. FourCCs were used to map streams to decoders. For codecs
not supported by AVI, "MPlayer internal" FourCCs were made up.
codec_tags.c is there to bridge demuxers written against the old API to
the mpv one. By now, only demux_mkv.c needs this (because demux_mkv is
the only serious demuxer left - preferably, we should use libavformat
for mkv too, but I can't see this happening any time soon, because
libavformat _still_ doesn't support segment linking). But the codec
tables are full of weird stuff automatically extracted from the old
codecs.conf tables. Most of it isn't needed for mkv.
Remove all custom tags, readd those used by demux_mkv.c internally
(see vinfo and mkv_audio_tag tables). The rest is handled by the
tables provided by libavformat, which includes AVI and QT tags.
this involved inverting the logic of find_formats, enumerate_devies
and wasapi_fill_VistaBlob. The latter two were trivial as their return
values were not actually checked (to be fixed in a later
commit).
Before these definitions were incorrectly guarded by and #ifdef
but since they aren't macros, this would never be true so that
if they were ever added to mingw headers we would have problems.
rename KSDATAFORMAT constants with the same mp prefix for consistency.
also use DEFINE_GUID rather than defining the bare structure
It was more complicated than necessary.
The behavior changes slightly. Now it might pass through extradata when
it didn't before (hopefully harmless), and doesn't fail with an error if
extradata is not available, even though it's needed (harmless, will fail
either way).
The if branch has a weak check to test whether the codec_id is the short
ID, and handles the long IDs in the else branch. The long IDs are all
longer than 12 bytes long, so hardcoding the string offset to get the
trailing part of the name makes sense. But the if condition checks for
another thing, which could get the else branch run even if the codec_id
is short.
Fix the bogus control flow and check if the codec_id is long enough. One
of these checks could be considered redundant, but include them both for
defensive coding.
I suspect this is what is happening in github issue #1265 (at least
partially).
If D3DFMT_A8 is not available, fall back to RGBA. This is less efficient
in general, so we normally want to avoid it.
Running "sub_add file.srt auto" during hook execution automatically
selected the first added track. This happened because all tracks added
with sub_add are marked as "external", and external subtitles are always
selected by default.
Fix this by negating the "external" flag when autoselecting subtitles
during loading. The no_default flag exists for this purpose; it was
probably added for libquvi originally, where we had the same issue.
This is a somewhat obscure situation, and happens only if audio starts
again after it has ended (in particular can happens with files where
audio starts later). It doesn't matter much whether audio starts
immediately or some milliseconds later, so simplify it.
When playing paused, the amount of decoded audio is limited to a small
amount (1 sample), because we don't write any audio to the AO when
paused. The small amount could trigger the case of the wanted audio
being too far in the future in the PTS sync code, which set the audio
status to STATUS_DRAINING, which in turn triggered the EOF code in the
next iteration. This was ok, but unfortunately, this triggered another
retry in order to check resuming from EOF by setting the status to
STATUS_SYNCING, which in turn lead to the busy loop by alternating
between the 2 states. So don't try resyncing while paused.
Since the PTS syncing code also calls ao_reset(), this could cause the
pulseaudio daemon to consume some CPU time as well.
This was caused by commit 33b57f55. Before that, the playloop was merely
run more often, but didn't cause any problems.
Fixes#1288.
...because everything is terrible.
strerror() is not documented as having to be thread-safe by POSIX and
C11. (Which is pretty much bullshit, because both mandate threads and
some form of thread-local storage - so there's no excuse why
implementation couldn't implement this in a thread-safe way. Especially
with C11 this is ridiculous, because there is no way to use threads and
convert error numbers to strings at the same time!)
Since we heavily use threads now, we should avoid unsafe functions like
strerror().
strerror_r() is in POSIX, but GNU/glibc deliberately fucks it up and
gives the function different semantics than the POSIX one. It's a bit of
work to convince this piece of shit to expose the POSIX standard
function, and not the messed up GNU one.
strerror_l() is also in POSIX, but only since the 2008 standard, and
thus is not widespread.
The solution is using avlibc (libavutil, by its official name), which
handles the unportable details for us, mostly. We avoid some pain.
Always create the context in mpgl_init(), instead of doing it when
mpgl_config_window() is called the first time. This is a small step
towards cleaning up the GL backend interface, and adding other things
like perhaps GLES support, or a callback-driven backend for libmpv.
this currently uses a sketchy but apparently working workaround,
which will be removed once the neccessary changes in youtube-dl
are implemented
Fixes#1277
This seems safer: otherwise, opening the AO could randomly fail if the
audio formats happens to be not float.
Unfortunately, this only works if the user does not select a device.
Since ALSA devices are arbitrary strings, including plugins with complex
parameters, it's not trivial or maybe even impossible to edit the string
in a way the "plug" plugin is added.
With --audio-device, it would be safe for users to select either
"default" or one of the "plughw" devices. Everything else seems
questionable.
Use the ALSA channel map API for querying and selecting supported
channel maps.
Since we (probably?) want to be compatible with ALSA versions before the
change, we still try to select the device name by channel map, and open
that device. There's no way to negotiate a channel map before opening,
so we're stuck with this approach. Fortunately, it seems these devices
allow selecting and setting any other supported channel layout, so maybe
this is not an issue at all. In particular, this avoids selecting the
default (dmix) device, which can only do stereo.
Most code is based on Martin Herkt <lachs0r@srsfckn.biz>'s alsa_ng
branch, with heavy modifications.
Don't crash if no fallback channel layout could be found (caller can't
handle NULL return from select_chmap()). Apparently this could never
actually happen, though.
Don't treat snd_pcm_hw_params_set_periods_near() failure as fatal error.
Same deal as with snd_pcm_hw_params_set_buffer_time_near().
Actually free channel maps returned by snd_pcm_get_chmap().
Adjust some messages.
Currently, --ytdl is off by default, but even if this is changed, never
enable it by default for the client API. It would be inappropriate to
start an intrusive external subprocess behind the host application's
back.
No functional changes.
ALSA_PCM_NEW_HW_PARAMS_API was a pre-ALSA 1.0.0 thing and does nothing
with modern ALSA. It stopped being necessary about 10 years ago.
3 functions are moved to avoid forward references.
Makeshift-solution for working around certain fontconfig issues.
With --use-text-osd=no, libass and fontconfig won't be initialized, and
fontconfig won't block everything with scanning for fonts.
If ALSA reports a channel map, and it looks like it makes sense (i.e.
could be converted to mpv channel map, and the channel count matches),
then use that instead of the channel map we are assuming.
This is based on code written by lachs0r (alsa_ng branch).
We don't actually want to test all possible combinations; we just want
to make sure that each thing (e.g. linux/osx, ffmpeg/libav) is tested
once.
Exclude Linux + ffmpeg-stable, because ffmpeg-stable is already tested
on OSX.
Exclude clang on Linux, because OSX needs clang, but Coverity (running
on Linux) needs gcc - so we use gcc only on Linux.
I also wanted to reduce the matrix to a single configuration when
running Coverity, but apparently this is not possible.
(See travis-ci/travis-ci#1975.)
For the purpose of running Coverity correctly.
Although I'm not sure how well this works. gcc won't work on OSX, and
also I'm not sure if Coverity will act up if the build matrix has more
than 1 configuration (will it submit multiple scans?).
Simpler, and leaves the decision to repeat or not fully to the script
(instead of requiring the user to care about it when remapping a script
binding).
Use a fixed size array for the client name, which also limits the client
name in size. Sanitize the client name string, and replace characters
that are not in [A-Za-z0-9] with '_'.
Otherwise, mouse button bindings added by mp.add_key_binding() would be
ignored.
It's possible that this "breaks" some older scripts using undocumented
Lua script functions, but it should be safe otherwise.
Fixes#1283.
The subprocess code was already split into fairly general functions,
separate from the Lua code. It's getting pretty big though, especially
the Windows-specific parts, so move it into its own files.
Normally, when creating a process with inherited handles on Windows, the
process inherits all inheritable handles from the parent, including ones
that were created on other threads. This can cause a race condition,
where unintended handles are copied into the new process, preventing
them from being closed correctly while the process is running. The only
way to prevent this on Windows XP was to serialise the creation of all
inheritable handles, which is clearly unacceptable for libmpv.
Windows Vista solves this problem by allowing programs to specify
exactly which handles are inherited, so do that on Vista and up.
See http://blogs.msdn.com/b/oldnewthing/archive/2011/12/16/10248328.aspx
The CREATE_NO_WINDOW flag is used to prevent the subprocess from
creating an empty console window when mpv is not running in a console.
When mpv is running in a console, it causes the subprocess to detach
itself, and prevents it from seeing Ctrl+C events, so it hangs around in
the background after mpv is killed.
Fix this by only specifying CREATE_NO_WINDOW when mpv is not attached to
a console. When it is attached to a console, subprocesses will
automatically inherit the console and correctly receive Ctrl+C events.
I'm not sure if this is necessary, but it can't hurt, and it's what
you're supposed to do before leaving the stack frame that contains the
OVERLAPPED object and the buffer. If there is no pending I/O, CancelIo
will do nothing and GetOverlappedResult will silently fail.
The caller set up the "start" pointer array using the number of planes,
the encode() function used the number of channels. This copied
uninitialized values for packed formats, which makes Coverity warn.
From what I understand the division is to align the dimension of the
value from seconds to milliseconds. Hard to tell whether the "rounding"
was intentional or not; I'm tipping on "not".
Found by Coverity.
Silences a Coverity warning.
Also, drop the assert(); although it should be pretty much guaranteed
that things happen this way, it's still a bit fuzzy.
In addition to the messed-up expression, the endianness was also
inverted. The code reads big endian by default.
It "worked" by coincidence, but for little endian, codepoints outside of
latin1 were broken.
The broken expression was found by Coverity.
In all of these situations, NULL is logically not allowed, making the
checks redundant.
Coverity complained about accessing the pointers before checking them
for NULL later.
The fact that it's a generic command prefix that is parsed even when
using the client API is a bit unclean (because this flag makes sense
for actual key-bindings only), but it's less code this way.
Does the same thing as the drop_buffers command. When implementing that
command, it turned out that resetting the higher level playback state
was more effective for achieving smooth recovery.
Untested; I don't even have any DVDs or DVD images with multiple angles.
This command was actually requested on IRC ages ago, but I forgot about
it.
The main purpose is that the decoding state can be reset without issuing
a seek, in particular in situations where you can't seek.
This restarts decoding from the middle of the packet stream; since it
discards the packet buffer intentionally, and the decoder will typically
not output "incomplete" frames until it has recovered, it can skip a
large amount of data.
It doesn't clear the byte stream cache - I'm not sure if it should.
Avoids a crash if OpenSSL tries to write to a broken connection with
write().
Obviously OpenSSL really should use send() with MSG_NOSIGNAL, but for
some reason it doesn't. This should probably be considered an OpenSSL
bug, but since in this case we "own" the process, there is no harm in
ignoring the signal.
This is not done with libmpv, because as a library we don't want to mess
with global state. It's also not done if terminal handling is disabled -
this is a bit arbitrary, but I don't care much.
It's passed with the '--format' option to youtube-dl.
If it isn't set, we don't pass '--format best' so that youtube-dl can
use the options from its configuration file.
Signed-off-by: wm4 <wm4@nowhere>
This sub-option was turned into a flag when the sub-option parser was
changed to the generic one (probably accidentally). Turn it into a
proper choice-option.
Also, adjust what the options do. Though none of this probably makes
much sense; the default should work, and if it doesn't, the GPU/driver
is probably beyond help.
Instead of threads, use overlapped (asynchronous) I/O to read from both
stdout and stderr. Like in d0643fa, stdout and stderr could be closed at
different times, so a sparse_wait function is added to wrap
WaitForMultipleObjects and skip NULL handles.
When the audio thread fails to properly init, it signals failure
to the main thread, AND THEN starts to clean up. For this to work,
ao_init callback must not return until the thread's cleanup is finished.
This is correctly handled in the ao_uninit callback by waiting for
the thread to exit, so just call that to clean up the main thread.
I have no idea why I didn't do this in the first place.
Probably needs to be polished a bit more. Also, might require a key
binding that can set/clear the loop points in a more intuitive way.
For now, something like this can be put into input.conf to use it:
ctrl+y set ab-loop-a ${time-pos} # set A
ctrl+x set ab-loop-b ${time-pos} # set B
ctrl+c set ab-loop-a no # clear (mostly)
Fixes#1241.
Due to the current code structure, the "current" entry and the entry
which is playing can be different. This is probably silly, but still
try to mark the entries correctly.
Refs #1260.
This actually doesn't even write/return the new sub-property, because
I dislike the idea of dumping that field for every single playlist
entry, even though it's "needed" only for one.
Fixes#1260.
dsound was set as default, because there were some hard to fix problems
with wasapi. These problems were probably fixed now, so let's try with
wasapi as default again.
Even with change notifications, there are still (rare) cases when the
feed thread gets AUDCLIENT_DEVICE_INVALIDATED. So handle failures in
thread_feed by requesting ao_reload.
on changes to PKEY_AudioEngine_DeviceFormat, device status, and default device.
call ao_reload directly in the change_notify "methods".
this requires keeping a device enumerator around for the duration of
execution, rather than just for initially querying devices
Before, failures, particularly in the thread loop init, could lead to a
bad state for the duration of mpvs execution. Make sure that
everything that was initialized gets properly and safely
uninitialized.
Now that the code for stderr and stdout does exactly the same things,
and the specialization is in the callbacks, this is blatantly
duplicated.
Also, define a typedef for those callbacks to reduce the verbosity.
Do a minimal check on data read with stream_peek(). This could help with
probing from unseekable streams in some situations. (We could check the
entire EBML and Matroska headers, but probably not worth the trouble. We
could also seek back to the start, which demux.c doesn't do, but which
would work usually - also not worth the trouble.)
libass won't use embedded fonts, unless ass_set_fonts() (called by
mp_ass_configure_fonts()) is called. However, we call this function when
the ASS_Renderer is initialized, which is long before the .ass file is
actually loaded. (I'm not sure why it tries to keep 1 ASS_Renderer, but
it always did this.)
Fix by calling mp_ass_configure_fonts() after loading them. This also
means this function will be called multiple times - hopefully this is
harmless (it will reinit fontconfig every time, though).
While we're at it, also initialize the ASS_Renderer lazily.
Fixes#1244.
Make the changes started in commit c827ae5f more eloborate, and provide
an option to control the amount of data read before the seek-target. To
achieve this, rewrite the loop that finds the lowest still acceptable
target cluster. It is now searched by time instead of file position. The
behavior (both with and without preroll option) may be different from
before this change, although it shouldn't be worse.
The change demux_mkv_read_cues() fixes a bug: when seeking after playing
normally, the code would erroneously assume that durations are set. This
doesn't happen if the first operation after loading was a seek instead
of playback.
This might be interesting for GUIs and such.
It's probably still a little bit insufficient. For example, the filter
and audio/video output lists are not available through this.
When mpv is backgrounded initially (via & in the shell), do no longer
change terminal settings on startup. This fixes broken local echo after
launching a backgrounded mpv.
This was removed in commit 480f82fa. This caused the cache display not
to update while paused, because the update_cache() function is never
called in the thread (now I remember why the extra call was "needed").
The old implementation intentionally run update_cache() only before
waiting on a mutex, with no further checks for the condition variable.
In theory, this is strictly not sane, but since it was just for the
retrieval of the very fuzzy cache status, it was ok. Now we want to call
update_cache() outside of the mutex though - which means that in order
to avoid missed wakeups, a proper condition has to be used.
Remove the extra vf_chain.output field - there's absolutely no need for
it, because there is always a last filter which will buffer the output.
For some reason, vf_chain.last was never set, which we now need to fix
too.
The purpose of temporarily setting stop_play was to make the audio
uninit code to explicitly drain audio if needed. This was the only way
to do it before ao_drain() was made a separate function; now we can just
do it explicitly instead.
We absolutely need to clear the AO reference in the mixer.
The audio_status must be changed to a state where no code assumes that
the AO is available. (It's allowed to do this blindly.)
When initialization failed, vo_lavc may cause an irrecoverable state in
the ffmpeg-related structs. Therefore, we reject additional
initialization attempts at least until we know a better way to clean up
the mess.
ao_lavc currently cannot be initialized more than once, yet it's good to
do consistent changes there as well.
Also, clean up uninit-after-failure handling to be less spammy.
The mp_audio_from_avframe() function requires the AVFrame to be
refcounted, and merely increases its refcount while referencing the same
data. For non-refcounted frames, it simply did nothing and potentially
would make the caller pass around a frame with dangling pointers.
(libavcodec should always return refcounted frames, but it's not clear
what other code does; and also the function should simply work, instead
of having weird requirements on its arguments.)
Revert commit 24e52f66; even though the old beheavior doesn't make sense
(as the commit message assured), it turns out that this works better:
typically, it means preroll will start from the previous video key frame
(the video CUE index will contain clusters with video key frames only),
which often coincides with subtitle changes. Thus the old behavior is
actually better.
Change the code that uses CueDuration elements. Instead of merely
checking whether preroll should be done, find the first cluster that
needs to be read to get all subtitle packets. (The intention is to
compensate for the enlarged preroll cluster-range due to reverting
commit 24e52f66.)
The previous fix breaks another obscure case: if the second vf_sub adds
margins, the image is accidentally not extended, which would return in
an assertion failure when returning the bogus image.
This rewrites the audio decode loop to some degree. Audio filters don't
do refcounted frames yet, so af.c contains a hacky "emulation".
Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of
estimating how much audio we need to filter, we always filter full
frames. Maybe this should be adjusted later: in case filtering increases
the volume of the audio data, we should try not to buffer too much
filter output by reducing the input that is fed at once.
For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it
doesn't seem worth the trouble.
Use a pseudo-filter when changing speed with resampling, instead of
somehow changing a samplerate somewhere. This uses the same underlying
mechanism, but is a bit more structured and cleaner. It also makes some
of the following changes easier.
Since we now always use filters to change audio speed, move most of the
work set_playback_speed() does to recreate_audio_filters().
A helper to allocate refcounted audio frames from a pool. This will
replace the static buffer many audio filters use (af->data), because
such static buffers are incompatible with refcounting.
Causes the player to reload the demuxer and to relist the found
streams. Probably slightly dangerous/broken, because the demuxer
thread and possibly even the decoders will keep reading data from
the new title before the new demuxer takes over.
Fixes#1250.
A first step towards refcounted audio frames.
Amazingly, the API just does what we want, and the code becomes
simpler. We will need to NIH allocation from a pool, though.
If the audio callback suddenly stops, and the AO provides no "reset"
callback, then reset() could deadlock by waiting on the audio callback
forever.
The waiting was needed to enter a consistent state, where the audio
callback guarantees it won't access the ringbuffer. This in turn is
needed because mp_ring_reset() is not concurrency-safe.
This active waiting is unavoidable. But the way it was implemented, the
audio callback had to call ao_read_data() at least once when reset() is
called. Fix this by making ao_read_data() set a flag upon entering and
leaving, which basically turns p->state into some sort of spinlock.
The audio callback actually never needs to spin, because there are only
2 states: playing audio, or playing silence. This might be a bit
surprising, because usually atomic_compare_exchange_strong() requires a
retry-loop idiom for correct operation.
This commit is needed because ao_wasapi can (or will in the future)
randomly stop the audio callback in certain corner cases. Then the
player would hang forever in reset().
As usual, we use C11 semantics, and emulate it if <stdatomic.h> is not
available.
It's a bit messy with __sync_val_compare_and_swap(). We assume it has
"strong" semantics (it can't fail sporadically), but I'm not sure if
this is really the case. On the other hand, weak semantics don't seem to
be possible, since the builtin can't distinguish between the two failure
cases that could occur. Also, to match the C11 interface, use of gcc
builtins is unavoidable. Add a check to the build system to make sure
the compiler supports them (although I don't think there's any compiler
which supports __sync_*, but not these extensions).
Needed for the following commit.
This is what you would expect. Before this commit, each
ao_request_reload() call would just queue a reload command, and then
recreate the AO for the number of times the function was called.
Instead of sending a command, introduce some sort of event retrieval
mechanism. At least for the reload case, use atomics, because we're too
lazy to setup an extra mutex.
This commit fixes a "cosmetic" user interface issue. Instead of
displaying the interpolated seek time on OSD, show the actual audio
time.
This is rather silly: when seeking in audio-only mode, it takes some
iterations until audio is "ready", but on the other hand, the audio
state machine is rather fickle, and fixing this cosmetic issue would be
intrusive. So just add a hack that paints over the ugly behavior as
perceived by the user. Probably the lesser evil.
It doesn't happen if video is enabled, because that mode sets the
current time immediately to video PTS. (Audio has to be synced to video,
so the code is a bit more complex.)
Fixes#1233.
This reverts commit d859549424.
Going to apply the alternative fix through PR #1256, which came just
some seconds after pushing the reverted commit. The reverted commit
was reported as not actually working.
Found by clang sanitizer. Casting unsigned integers to signed integers
with same size has implementation defined behavior (it's even allowed to
crash), but it seems reasonable to expect that reasonable
implementations do a complement of 2 "conversion".
Following the discussion in #1253.
The events won't be removed for a while, though. (Or maybe never, unless
we run out of bits for the uint64_t event mask.)
This is not a real change (the events still work, and the alternative
mechanisms were established a few API revisions earlier), but for the
sake of notifying API users, update DOCS/client-api-changes.rst.
The values compared here happen to be of unsigned enum types - but the
test is not supposed to break if we somehow force the enum to signed, or
if the compiler happens to use a signed type (as far as I remember, the
exact integer type the compiler can use is implementation-defined).
The main need I see for this is with libmpv - it would be confusing if
some application showed up as "mpv" on whateverthehell PulseAudio uses
it for (generally it does show up on various PA GUI tools).
This silences the warning:
video/out/gl_video.c:1091:51: runtime error: division by zero
when running with clang -fsanitize=undefined. Division by zero is legal
according to IEEE, but I guess clang doesn't care about standard. While
triggering this warning isn't actually avoided in all cases, it's
avoided in the common case and also makes people shut up about it.
Call VOCTRL_GET_DISPLAY_NAMES it when the property is
requested. The vo should return the names of the displays that the mpv
window is covering. For example, with x11 vos, xrandr names LVDS1,
HDMI1, etc.
XRRGetOutputInfo contains a "name" element which corresponds to to the
display names given to the user by the "xrandr" command line
utility. Copy it into the xrandr_display struct for each display.
On VOCTRL_GET_DISPLAY_NAMES, send a copy of the names
of the displays spanned by the mpv window on.
The intention is to avoid using the timeout-based fallback.
There's some minor hope that this will help with OpenBSD (see #1239),
although it probably won't.
Some chance that this will cause trouble with obscure OSS
implementations or emulations.
If calling ao->driver->wait() fails, we need to fallback to timeout-
based waiting. But it could be that at this point, the mutex was already
released (and then re-acquired). So we need to recheck the condition in
order to avoid missed wakeups.
This probably wasn't an actually occurring problem, but still could
cause a small race-condition window if the dynamic fallback is actually
used.
This considered only index entries that were for the same track ID as
the track used for seeking. This doesn't make much sense for preroll;
it'll just possibly skip clusters, and select an earlier cluster.
One possible negative side-effect is that the preroll might be too tight
now, and miss subtitle packets more often.
The demuxer has a hack to seek to the cluster before the target cluster
in order to "catch" subtitle lines that start before the seek target,
but overlap with the video after the seek target.
Avoid this hack if the cue index indicates that there are no overlapping
subtitle packets that can be caught by seeking to the previous cluster.
Nothing is done with them yet. This is preparation for the following
commit.
CueRelativePosition isn't even saved anywhere, because I don't intend to
use it. (Too messy for no gain.)
Instead of indexing only 1 packet per cluster (which is enough for
working seeking), add every packet to the index.
Since on seek, we go through every single index entry, this probably
makes seeking slower. On the other hand, this code is used for files
without index only (e.g. incomplete files), so it probably doesn't
matter much.
Preparation for the following commits.
update_subtitle() already uees playback_pts to make subtitles work
better in no-audio mode. Using get_current_time() usually gets
playback_pts, but also has the advantage that it will use the seek
target time during seeks. This will result in multiple sub_seek commands
doing the right thing (at least as long as they're far enough apart so
that seeking is actually initiated when the second command is run).
If no packets are queued, the readahead time is obviously 0.
If the end time is smaller than the start time, the problem is probably
that audio and video start at slightly different times - report 0 in
this case too.
Do this because seeing "???" as readahead time is a bit annoying.
Instead, use the native-endian alias, and switch the wayland format
depending on the target platform's endian.
This drops support for swapped-endian formats, but I think that is ok.
Not only are the affected formats rather ancient and backwards, but
using swapped formats probably does not make any sense for performance
either.
Untested.
These formats are still supported; you just can't reference them via a
defined constants directly. They are now handled via the generic
passthrough.
(If you want to use such a format, you either have to add the entry
back, or use AV_PIX_FMT_* directly.)
This is a rather radical change: instead of maintaining a whitelist of
FFmpeg formats we support, we automatically support all formats.
In general, a format which doesn't have an explicit IMGFMT_* name will
be converted to a known format through libswscale, or will be handled
by code which can treat pixel formats in a generic way using the pixel
format description, like vo_opengl.
AV_PIX_FMT_UYYVYY411 is a special-case. It's packed YUV with chroma
subsampling by 4 in both directions. Its component order is documented
as "Cb Y0 Y1 Cr Y2 Y3", meaning there's one UV sample for 4 Y samples.
This means each pixel uses 1.5 bytes (4 pixels have 1 UV sample, so
4 bytes + 2 bytes). FFmpeg can actually handle this format with its
generic mechanism in an extremely awkward way, but it doesn't work for
us. Blacklist it, and hope no similar formats will be added in the
future.
Currently, the AV_PIX_FMT_*s allowed are limited to a numeric value of
500. More is not allowed, and there are some fixed size arrays that need
to contain any possible format (look for IMGFMT_END dependencies).
We could have this simpler by replacing IMGFMT_* with AV_PIX_FMT_*
through the whole codebase. But for now, this is better, because we
can compensate for formats missing in Libav or older FFmpeg versions,
like AV_PIX_FMT_RGB0 and others.
FFmpeg has only a AV_PIX_FMT_FLAG_BE flag, not a LE one, which causes
problems for us: we want to have the LE flag too, so code can actually
detect whether a format is non-native endian. Basically, we want to
reconstruct the LE/BE suffix all AV_PIX_FMT_*s have.
Doing this is hard due to the (messed up) way AVPixFmtDescriptor works.
The worst is AV_PIX_FMT_RGB444: this group of formats describe an
endian-independent access (since no component actually spans 2 bytes,
you only need byte accesses with a fixed offset), so we have to go
through some pain.
This influences the demuxer readahead display. If a stream has reached
EOF, we want to ignore it for the purpose of this calculation.
Note that if a stream contains no packets, it still should cause the
value 0s to be displayed (unless it's EOF), because that's just the
actual situation.
This was relying on the fact that timestamps will always be numerically
larger than MP_NOPTS_VALUE, but the trick didn't actually work for
MP_PTS_MIN. Be a bit more sincere, and don't rely on this anymore. This
fixes the comparison, and avoids the readahead amount displaying as
"???" in some situations (since one of the values was NOPTS).
In this case, we didn't find any new packets for this stream, even
though we've read ahead as much as possible. (If reading ahead in this
case, the "Too many packets in the demuxer packet queues" error is
normally printed.)
If we do consider this an underrun, handle_pause_on_low_cache() will
pause and show the "buffering" state, which is not useful.
Could also happen on very bad interleaving.
This mechanism was introduced for Opus, and allows correct skipping of
"preroll" data, as well as discarding trailing audio if the file's
length isn't a multiple of the audio frame size.
Not sure how to handle seeking. I don't understand the purpose of the
SeekPreRoll element.
This was tested with correctness_trimming_nobeeps.opus, remuxed to mka
with mkvmerge v7.2.0. It seems to be correct, although the reported file
duration is incorrect (maybe a mkvmerge issue).
Add a generic mechanism to the VO to relay "extra" events from VO to
player. Use it to notify the core of window resizes, which in turn will
be used to mark all affected properties ("window-scale" in this case) as
changed.
(I refrained from hacking this as internal command into input_ctx, or to
poll the state change, etc. - but in the end, maybe it would be best to
actually pass the client API context directly to the places where events
can happen.)
Add a comment about this to avoid confusing users of this function. The
parameter is essentially unused, but exists so that we don't need to
add extra APIs if the need for it arises.
Instead of defining a separate data structure in the core.
For some odd reason, demux_chapter exported the chapter time in
nano-seconds. Change that to the usual timestamps (rename the field
to make any code relying on this to fail compilation), and also remove
the unused chapter end time.
Note that you can't pass .cue or .edl files to it, at least not yet.
Requested in context of allowing to specify custom chapters. For that
to work well, we probably need to add some sort of chapter metadata
pseudo-demuxer.
Using the --playlist option is no longer recommended.
A while ago, mpv rewrote all playlist parsers and added some minimal
security mechanisms (like not allowing local file access or unsafe
protocols in remote playlists). Further, mpv can load playlists by
passing them as normal file arguments, without the option.
Now, --playlist is needed only in these situations:
1) loading plaintext files
2) disabling additional security mechanisms
(e.g. using a remote playlist to play local files)
NSDisableScreenUpdates came to hunt me in the end and when mpv was paused, it
did wait for a frame that never came (because of interaction with the live
resizing code)!
This was shown only if decoder-framedropping was enabled, and only if at
least 50 frames were dropped by it. Since drop_frame_cnt used to mean
"number of late frames", this code made sense, but this is not the case
anymore: drop_frame_cnt can be even 0, all while video gets hopelessly
behind audio.
One problem with this is that short desync spikes (which usually can
probably dealt with) will also cause this message to be shown. If it
gets triggered too often, the code will need to be adjusted.
This shouldn't use the host's 'ar' when building static libs. It only
worked until now because Linux 'ar' is usually built with PE support.
Couldn't confirm whether it works, because this dumb crap is just
broken when cross-compiling to mingw.
Thanks to STREAM_CTRL_HAS_AVSEEK, we actually know whether CTRL_AVSEEK
is implemented at all, and we can avoid a blocking wait on the cache if
demux_lavf sends CTRL_AVSEEK even if it won't wait. I'm hoping this
can't currently happen, but why hope if we can explicitly prevent it.
It'll make us more robust against future changes in libavformat.
For example, if --force-window is used, and video is switched off during
playback, then you need to redecide the rendering method to get subs
displayed correctly.
Do this by moving the state setup code into a function, and call it on
every frame.
Apparently this can "sometimes" return an error. In my opinion, this
should never return an error: neither the semantics of the function,
nor the ALSA documentation or ALSA sample code seem to indicate that
a failure is to be expected. I'm not perfectly sure about this though
(I blame ALSA being a weird, big, underdocumented API).
Since it causes problems for some users, and since there is really no
reason why we should abort on such an error, turn it into a warning.
Fixes#1231.
If you played e.g. an audio-only file and something bad happened that
interrupted playback, the exit message could say "No files played".
This was awkward, so show a different message in this case.
Also overhaul how the exit status is reported in order to make this
easier. This includes things such as not reporting a playback error
when loading playlists (playlists contain no video or audio, which
was considered an error).
Not sure if I'm happy with this, but for now it seems like a slight
improvement.
Basically, this will mark the demuxer as seekable with rtmp* and mmsh
protocols. These protocols have network-level time seeking, and whether
you can seek on the byte level does not matter.
Until now, seeking was typically only enabled because of the cache, and
a (nonsensical) warning was shown accordingly.
It still could happen that the server doesn't actually support thse
requests (or simply rejects them), so this is somewhat imperfect.
I'm not sure if this could be done in libavformat instead. Probably not,
because libavformat doesn't seem to have any mechanism for trying one
protocol and reverting (or redirecting) to another one if needed.
This commit is sort of a hack too, because it redirects the URL by
pretending the http:// link is a playlist containing the mmsh:// link.
The list of mime types is borrowed from MPlayer (which has completely
different code to handle this).
Apparently this is needed for correct 3D mode subtitles. In general,
it seems you need to duplicate the whole "GUI", so it's done for all
OSD elements.
This doesn't handle the "duplication" of the mouse pointer. Instead,
the mouse can be used for the top/left field only. Also, it's possible
that we should "compress" the OSD in the direction it's duplicated, but
I don't know about that.
Fixes#1124, at least partially.
This is probably what libmpv users want; and it also improves error
reporting (or we'd have to add a way to communicate such mid-playback
failures as events).
This was probably done incorrectly in cases when the currently selected
channel had no data. I'm not sure if this codepath is functional at all,
though. Maybe not.
Untested due to lack of DVB hardware.
Using magic integer values was an attempt to keep the API less verbose.
But it was probably not a good idea.
Reason 1 (restart) is not made explicit, because it is not used anymore
starting with the previous commit. For ABI compatibility, the value is
left as a hole in the enum.
Use the codepath that is normally used for DVD/BD title switching and
DVB channel switching. Removes some extra artifacts from the client API:
now MPV_EVENT_END_FILE will never be called on reloads (and neither is
MPV_EVENT_START_FILE).
This was originally done for DVD/BD/DVB, where the start position could
be something different from 0, and seeking back to 0 would mess it up
completely.
Since we're not quite sure that these streams are unseekable, we can
simplify this somewhat, and also make sure we also start at 0 for normal
files. Helps a little bit with the following edition reloading commit.
The only reason for mpv_wait_event() not being thread-safe is that it
returns a pointer to a mpv_event struct member in the mpv_handle
context, which in turn is done for ABI-safety (user doesn't allocate or
manage the struct), and to avoid the need additional memory management.
Some users were interpreting this incorrectly.
Without --force-window, this is called on every iteration or so, and
calling uninit_video_out() sends the video-reconfig event. Avoid sending
redundant events.
Fixes#1225 (using an alternative patch).
In interlaced modes, we output fields, not complete frames, so the
framerate doubles.
The method to calculate this was borrowed from xrandr code.
Hopefully fixes#1224.
At least on my machine, reading back the frame with system memcpy is
slower than just using software rendering. Use the optimized gpu_memcpy
from LAV to speed things up.
Pretty much a fringe-feature, but also it's awkward if something appears
on the terminal with no indication for the source.
This is made quite awkward by the fact that stderr and stdout could be
closed at different times, and that poll() doesn't accept "holes" in its
FD list. Invalid (.e.g negative) FDs just make it return immediately, as
required by the standard. So sparse_poll() takes care of the messy
details.
What was the purpose of that? Probably none.
Also simplify another thing: if we get the cancel signal through FD,
there's no reason to check it separately.
We need to manually define the flag since we are using a separate identifier
for each of the Lua checks. This was done before 9b45b48 by the composed check
with a define_key (see waftools/checks/generic.py).
The pkg-config check was the only one to not redefine a define key because Waf
already does that automatically when we call the generated function with the
same identifier as the generator function. Now if they are called with two
different arguments we will get two different definitions.
Fixes#1218
No development activity (or even any sign of life) for almost a year.
A replacement based on youtube-dl will probably be provided before the
next mpv release. Ask on the IRC channel if you want to test.
Simplify the Lua check too: libquvi linking against a different Lua
version than mpv was a frequent issue, but with libquvi gone, no
direct dependency uses Lua, and such a clash is rather unlikely.
The immediate mode (which is the default) uses a tiny ringbuffer and
doesn't grab timestamps. This leads to quite bad behavior due to the
fact that there's an additional buffer between playloop and TV code (the
demuxer thread, which doesn't exist in MPlayer).
Always grab the timestamps and use a decently-sized buffer. I still have
no clue what I'm doing, and hacked it until it appeared to work. Report
regressions if you experience any.
I can't believe how shitty this (MPlayer-derived) code is. Maybe it
should be fixed or be replaced with using libavdevice, but that doesn't
seem worth the effort.
Anyway, for now reduce the time it's blocking to wait for new frames
from 10ms to 1ms, because 10ms might be a bit too tight: it could
deliver the frame up to 10ms late - now it's only up to 1ms. (And yes,
it does that instead of using condition variables. It also abuses
volatile variables as atomics. It's hilarious.)
stream_rar.c peeks the first few bytes when trying to open, which means
that opening any stream reads at least 2KB of data (internal buffer
size) on opening. This broke --stream-dump, which saved only the data
following this initial buffer.
Hack it around by writing the current buffer to the capture file too,
and move stream_capture_write() above stream_set_capture_file() for this
purpose.
Cleaner solutions might include: handling the terrible rar thing
differently, or using the "proper" stream API for dumping. (The latter
is not done, because --stream-dump shares code with the --stream-capture
misfeature.)
Fixes#1215.
Apparently if resizing a NSWindow from a secondary thread Cocoa will
automatically protect itself using NSViewHierarchyLock and in our case,
cause a deadlock.
Fixes#1210
Although this is fine when the stream cache is active (which caches
these and returns the result immediately), it seems cleaner not to
rely on this detail.
Remove the update_cache() call from demux_thread(), because it's sort
of in the way. I forgot why it exists, and there's probably no good
reason for it to exist anyway.
It's needed for some obscure feature in combination with .rar reading.
However, it's unconditionally used by the subtitle loader code, so take
care of not blocking the main thread unnecessarily.
(Untested.)
So a client API user can know when a window is created or destroyed.
Also might be useful for the OSC: it could disable itself if video is
disabled.
Before this commit, there were only indirect ways of detecting this.
Most things should be allowed to access the client API unconditionally
(for example for sending events), so move destroying the client API
down. Also, mp_uninit_ipc() should happen before the point at which all
clients are shutdown, or there will be a small time window in which new
clients can be created after destroying them all.
Wether and when the text of a button should be squeezed when it
gets too long can now be configured in the layout:
lo.button.maxchars = <number>
nil = no squeezing (default)
If the button text has more than <maxchars> characters, it will
be squeezed to the estimated width of <maxchars>.
Doing that doesn't make sense anyway: it's meant for interactive input,
and if the output of the player is not on the terminal, how will you
interact with it?
It was also quite in the way when trying to read verbose output with
e.g. less while the player was running, because the player would grab
half of all input meant for less (simply because stdin is still
connected to the terminal).
Remove the now redundant special-casing of pipe input.
The player was supposed to exit playback if both video and audio failed
to initialize (or if one of the streams was not selected when the other
stream failed). This didn't work; for one this check was missing from
one of the failure paths. And more importantly, both checked the
current_track array incorrectly.
Fix these issues, and move the failure handling code into a common
function.
CC: @mpv-player/stable
Some rationale for the documented/suggested behavior:
It's not really clear what to do with invalid UTF-8, since JSON simply
can't transport this information. Maybe you could transfer such strings
as byte arrays, but that would be very verbose and inconvenient, and
would pose the problem that it's hard to distinguish between strings
encoded in this way and actual arrays.
There are many other ways how this could be handled. For example, you
could replace invalid sequences with '?'. Or you could do it like
Python, and use certain reserved unicode codepoints to "tunnel" through
invalid bytes.
Which of these works really depends on the application. And since this
can be done entirely on the byte level (invalid UTF-8 sequences can
appear only in strings in our case), it's best to leave this to the
receiver.
Assume mpv.exe is located in $mpv_exe_dir, then config files were
preferably loaded from "$mpv_exe_dir/mpv". This was mostly traditional,
and inherited from MPlayer times.
Reverse the config path priority order, and prefer $CSIDL_APPDATA/mpv as
main config path. This also fixes behavior when writing watch_later
configs, and mpv is installed in a not-writable path.
It's possible that this will cause regressions for some users, if the
change in preference suddenly prefers stale config files (which may
happen to longer around in the appdata config dir) over the user's
proper config.
Also explicitly document the behavior.
Since the list associated with --audio-device is supposed to enable
simple user-selection, it doesn't make much sense to include overly
special things like ao_pcm or ao_null in the list. Specifically,
ao_pcm is harmful, because it will just dump all audio to a file
named audiodump.wav in the current working directory. The user can't
choose the filename (it can be customized, but not through this
option), and the working directory might be essentially random,
especially if this is used from a GUI.
Exclude "strange" entries. We reuse the fact that there's already a
simple list ordered by auto-probe priority in order to avoid having to
add an additional flag. This is also why coreaudio_exclusive was moved
above ao_null: ao_null ends auto-probing and marks the start of
"special" outputs, which don't show up on the device, but we want
coreaudio_exclusive to be selectable (I think).
Move it above ao_null, so that it can be selected during auto-probing
(even if it's only last). I see no reason why it should not be included,
and it makes the following commit slightly more elegant. (See
explanations there.)
Because Lua is so terrible, it's easy to confuse temporary values pushed
to the Lua stack with arguments if the arguments are checked after that.
Add a hack that should fix this.
The behavior of reverse cycling (with the "!reverse" magic value) was a
bit weird and acted with a "delay". This was because the command set the
value the _next_ command should use. Change this and make each command
invocation select and use the next command directly. This requires an
"uninitialized" special index in the counter, but that is no problem at
all.
Due to the way video-rotate currently works, the state will be
automatically updated once new video is decoded. So the filter chain
doesn't need to be reinitialized automatically, but there is a need to
trigger the video instant refresh code path instead.
Also move the support function closer to an annoying similar yet
different function. They probably can be unified next time major changes
are done to this code.
Allows properly changing/updating the cursor state. Useful for client
API window embedding, because the host application may not want the mpv
window to grab mouse input, and this has to manually handle the cursor.
Changing the cursor of foreign windows is usually not sane.
It might make sense to allow changing the cursor icon, but that would be
much more complicated, so I won't add it unless someone actually
requests it.
Apparently using the stream index is the best way to refer to the same
streams across multiple FFmpeg-using programs, even if the stream index
itself is rarely meaningful in any way.
For Matroska, there are some possible problems, depending how FFmpeg
actually adds streams. Normally they seem to match though.
Getting subtitle scaling and positioning right even if there are video
filters, which completely change the image (like cropping), doesn't seem
to have a single, correct solution. To some degree, the results are
arbitrary, so we may as well do what is most useful to the user.
In this case, if the PGS resolution aspect ratio and the video output
aspect ratio mismatch, letter-box it, instead of stretching the subs
over the video frame. (This will require additional fixes, should it
turn out that there are PGS subtitles which are stretched by design.)
Fixes#1205.
It assumed that any >\"< sequence was an escape for >"<, but that is not
the case with JSON such as >{"ducks":"\\"}<. In this case, the second
>\< is obviously not starting an escape.
This isn't quite as robust as idet.sh as the default detection
interval is only 4 seconds vs 35 for idet.sh. idet.sh can have such a
large sample time since it turns off the vo and uses --untimed, which
is currently not possible from lua.
It turns out the glibc people are very clever and return an error if the
thread name exceeds the maximum supported kernel length, instead of
truncating the name. So everyone has to hardcode the currently allowed
Linux kernel name length limit, even if it gets extended later.
Also the Lua script filenames could get too long; use the client name
instead.
Another strange thing is that on Linux, unrelated threads "inherit" the
name by the thread they were created. This leads to random thread names,
because there's not necessarily a strong relation between these threads
(e.g. script command leads to filter recreation -> the filter's threads
are tagged with the script's thread name). Unfortunate.
On win32, open() is a function-like macro. The line of code changed
with this commit accidentally expanded the macro. Prevent this macro
expansion. Not sure why that happened now. Since as far as I remember
system functions can be defined as macros, this affects in theory not
only win32.
Especially with other components (libavcodec, OSX stuff), the thread
list can get quite populated. Setting the thread name helps when
debugging.
Since this is not portable, we check the OS variants in waf configure.
old-configure just gets a special-case for glibc, since doing a full
check here would probably be a waste of effort.
Windows doesn't have unix domain sockets, and can't handle sockets and
pipes in an uniform way. Only the libwaio fallback code is available,
which doesn't do JSON.
This is not realy obvious, so I assume this is a helpful hint.
Although the usefulness of such an approach is probably influenced by
the fact that the player might send events that arrive in the short
window while the socket is connected.
The ipc_thread can exit any time, and will free the mp_ipc_ctx when
doing this, leaving a dangling pointer. This was somewhat handled in the
original commit by setting mpctx->ipc_ctx to NULL when the thread
exited, but that was still a race condition.
Handle it by freeing most things after joining the ipc_thread. This
means some resources will not be freed until player exit, but that
should be ok (it's an exceptional error situation).
Also, actually close the pipe FDs in mp_init_ipc() on another error
path.
Thanks to the recently introduced mp_lua_PITA(), this is "simple" now.
It fixes leaks on Lua errors. The hack to avoid stack overflows
manually isn't needed anymore, and the Lua error handler will take
care of this.
The JSON parser was introduced for the IPC protocol, but I guess it's
useful here too.
The motivation for this commit is the same as with 8e4fa5fc (again).
Because 1) Lua is terrible, and 2) popen() is terrible. Unfortunately,
since Unix is also terrible, this turned out more complicated than I
hoped. As a consequence and to avoid that this code has to be maintained
forever, add a disclaimer that any function in Lua's utils module can
disappear any time. The complexity seems a bit ridiculous, especially
for a feature so far removed from actual video playback, so if it turns
out that we don't really need this function, it will be dropped again.
The motivation for this commit is the same as with 8e4fa5fc.
Note that there is an "#ifndef __GLIBC__". The GNU people are very
special people and thought it'd be convenient to actually declare
"environ", even though the POSIX people, which are also very special
people, state that no header declares this and that the user has to
declare this manually. Since the GNU people overtook the Unix world with
their very clever "embrace, extend, extinguish" strategy, but not 100%,
and trying to build without _GNU_SOURCE is hopeless; but since there
might be Unix environments which support _GNU_SOURCE features partially,
this means that in practice "environ" will be randomly declared or not
declared by system headers. Also, gcc was written by very clever people
too, and prints a warning if an external variable is declared twice (I
didn't check, but I suppose redeclaring is legal C, and not even the gcc
people are clever enough to only warn against a definitely not legal C
construct, although sometimes they do this), ...and since we at mpv hate
compiler warnings, we seek to silence them all. Adding a configure test
just for a warning seems too radical, so we special-case this against
__GLIBC__, which is hopefully not defined on other libcs, especially not
libcs which don't implement all aspects of _GNU_SOURCE, and redefine
"environ" on systems even if the headers define it already (because they
support _GNU_SOURCE - as I mentioned before, the clever GNU people wrote
software THAT portable that other libcs just gave up and implemented
parts of _GNU_SOURCE, although probably not all), which means that
compiling mpv will print a warning about "environ" being redefined, but
at least this won't happen on my system, so all is fine. However, should
someone complain about this warning, I will force whoever complained
about this warning to read this ENTIRE commit message, and if possible,
will also force them to eat a printed-out copy of the GNU Manifesto, and
if that is not enough, maybe this person could even be forced to
convince the very clever POSIX people of not doing crap like this:
having the user to manually declare somewhat central symbols - but I
doubt it's possible, because the POSIX people are too far gone and only
care about maintaining compatibility with old versions of AIX and HP-UX.
Oh, also, this code contains some subtle and obvious issues, but writing
about this is not fun.
Using the Lua API is a big PITA because it uses longjmp() error
handling. That is, a Lua API function could any time raise an error and
longjmp() to a lower part of the stack. This kind of "exception
handling" is completely foreign to C, and there are no proper ways to
clean up the "skipped" stack frames.
Other than avoiding such situations entirely, the only way to deal with
this is using Lua "userdata", which is basically a malloc'ed data block
managed by the Lua GC, and which can have a destructor function
associated (__gc metamethod).
This requires an awful lot of code (because the Lua API is just so
terrible), so I avoided this utnil now. But it looks like this will make
some of the following commits much easier, so here we go.
After removing synchronous libdispatch calls, this looks like it doesn't
deadlock anymore. I also experimented with pthread_mutex_trylock liek wm4
suggested, but it leads to some annoying black flickering. I will fallback to
that only if some new deadlocks are discovered.
It's kind of obvious, since the protocol by design has to allow you to
read (loadfile) and write (screenshot_to) random files, but better
make it explicit so that nobody accidentally does something insecure.
As I understand, otherwise, the code will try to destroy the same
window again in the cleanup part of the gui_thread(), which makes no
sense and is potentially dangerous.
mp_stat() instead of stat() was used in the normal code (i.e. even
on Unix), because MinGW-w64 has an unbelievable macro-mess in place,
which prevents solving this elegantly.
Add some dirty workarounds to hide mp_stat() from the normal code
properly. This now requires replacing all functions that use the
struct stat type. This includes fstat, lstat, fstatat, and possibly
others. (mpv currently uses stat and fstat only.)
Now requires newest libass git. Since this feature wasn't part of a
libass release yet, I'm not bothering making the mpv code compatible
with as how it was previously implemented (it will just be disabled
with any older libass).
CC: @mpv-player/stable (because mpv-build uses libass git, and this
breaks the feature)
It possibly goes to sleep without actually starting to decode audio.
Possibly fixes a problem with --no-osc --no-video reported on IRC.
CC: @mpv-player/stable
The previous commit was actually incorrect, and the change had
absolutely no effect. The two formats are (fortunately) the same. I'm
probably too tired.
This would have been wrong for hw decoders which pass us NV12 or NV21.
The format the GL shader filter chain gets is stored in p->image_desc,
while p->image_format still contains the "real" input format (which in
case of hw decoding is an opsque hw accel format). Since no hw decoder
did this, this is really just a theoretical fix and doesn't fix any
actual bugs.
This was probably commented as an oversight. Since the subtitle renderer
was uninitialized on reinitialization anyway, this had no negative
consequences, except a memory on exit.
A vague idea to get something similar what libquvi did.
Undocumented because it might change a lot, or even be removed. To give
an idea what it does, a Lua script could do the following:
-- type ID priority
mp.commandv("hook_add", "on_load", 0, 0)
mp.register_script_message("hook_run", function(param, param2)
-- param is "0", the user-chosen ID from the hook_add command
-- param2 is the magic value that has to be passed to finish
-- the hook
mp.resume_all()
-- do something, maybe set options that are reset on end:
mp.set_property("file-local-options/name", "value")
-- or change the URL that's being opened:
local url = mp.get_property("stream-open-filename")
mp.set_property("stream-open-filename", url .. ".png")
-- let the player (or the next script) continue
mp.commandv("hook_ack", param2)
end)
Normally, we pass libavformat demuxers a wrapped mpv stream. But in some
cases, such as HLS and RTSP, we let libavformat open the stream itself.
In these cases, set typical network properties like useragent according
to the mpv options.
(We still don't set it for the cases where libavformat opens other
streams on its own, e.g. when opening the companion .sub file for .idx
files - not sure if we maybe should always set these options.)
Fixes opening some streams.
This means the HLS playlist will be opened twice, but that's not much of
a problem, considering it's pretty small, and HLS will make many other
http accesses anyway.
OSD cycling attempted to remove the current message by setting an empty
message with duration 0. Duration 0 tripped up a corner case causing no
OSD to be displayed (until the next message was set), so exclude this
explicitly.
Apparently there's an use for this; see #1178.
I won't redocument obscure FFmpeg features, so add a hint to the
manpage that some protocols are documented in FFmpeg instead.
This could produce an extra frame, because reaching the maximum merely
signals the playloop to exit, without strictly enforcing the limit.
Fixes#1181.
CC: @mpv-player/stable
This provides some helper functions and classes for C++/Qt. As the top
of qthelper.hpp says, this is built on top of the client API, and is a
mere helper provided for convenience.
Maybe this should be a separate library, but on the other hand I don't
see much of a point in that. It's also header-only, but C++ people like
such things. This makes it easier for us, because we don't need to care
about ABI compatibility.
The client API doesn't change, but bump it so that those who are using
this header can declare a proper dependency.
Instead of letting the window-scale property return the old value until
X11 actually executed the resize, just set the new assumed internal
window size immediately. This avoids a "lag" between setting and reading
the window-scale property, like OSD controls typically do.
Remove the additional calls from vo_x11_highlevel_resize() - they're
pointless and slightly wrong, and resize events will take care of
updating these things correctly anyway.
Fixes#1176.
("window-scale" works via VOCTRL_[S|G]ET_UNFS_WINDOW_SIZE.)
In theory, vsscript should be doing it, but it's not there yet, neither
did there seem to be any interest in making it flexible enough to handle
more than 1 scripting language.
This allows mpv's view to take key and send events to mpv's core.
To set key status correctly, clients must call -[NSWindow selectNextKeyView:]
during reconfig on the main thread. All is 'documented' in the cocoabasic
example.
If someone knows a better way to handle giving key to the embedded view,
let me know!
Objective-C categories need special linker flags from the user when statically
linking (-ObjC LDFLAG), so make everyone's life simpler and remove them.
Showed "Volume: (unavailable)%". That was dumb.
The message string is now a bit convoluted; mostly because the property
expand syntax can't do "if-else", just "if".
CC: @mpv-player/stable
This does nothing good. This reverts a change made over a year ago - I
don't remember why this was originally done this way.
The main problem is that even if the volume option is set (something
like "--volume=75"), the volume property will always return "100" until
audio is initialized. If audio is uninitialized again, the volume
property will remain frozen at its last value.
This can hang if the window was destroyed externally (or that's what I
suspect happens), and we somehow didn't receive the DestroyNotify event.
I'm not sure why we wouldn't receive this event (since it should just be
in the xlib event queue), but on the other hand there's no real need to
wait for window destruction.
This essentially reverts 97fc74e2.
Some VS filters will requests frames from their parent filters while
they're initialized. Thy do this in a blocking manner, and
initialization will not succeed until the frame request is satisfied.
This deadlocked mpv, because we can feed frames to the filter only after
initialization is finished.
Return an error instead of deadlocking.
Note that we (probably) can handle frames being requested during init
fine, as long as the requests don't block initialization. But we can
distinguish this situation, and a simple test seems to indicate VS
usually doesn't do this.
See #1168.
Allows passing native types as arguments.
Also some minor doc improvements, including giving some (natural)
improvements to mpv_free_node_contents().
Note: mpv_command_node_async() is completely untested.
playlist-pos is 0-based, but everyone thinks it's 1-based. Stupid crap.
(The "not the same as MPlayer" refers to a playlist manipulation command
that doesn't exist anymore in mpv.)
CC: @mpv-player/stable
Manually setting can break things forever, because it puts the VO cursor
state out of sync with the remembered state by handle_cursor_autohide().
Use the normal autohide code during idle mode too instead. (Originally
the idea was to make the cursor always visible in idle mode, but not so
important.)
Regression since e1e8b07c. Fixes#1166.
CC: @mpv-player/stable
While conceptually this sink stuff in PulseAudio does just the right
thing, actually listing the sinks is unbelievable complicated. Not only
is the idea that listing them should happen asynchronously completely
bullshit (who the fuck runs the PulseAudio server on a separate
computer), but the way this is done is full of bullshit too. Why
separate callbacks for each device? Why this obtuse mainloop shit?
Especially the mainloop shit makes it actively worse than doing things
manually with pthread primitives, and the reason for that (different
mainloop implementations for GUIs?) is laughable too. It's like they
chose the most complicated API possible just because they attempted
to "abstract" basic mechanisms in order to handle "everything". While
I don't claim to design the best APIs, this API is fucking terrible
without any excuse. (End of rant.)
All the dumb crap in pa_init_boilerplate() is needed to talk to the
audio server at all. Might also fix some subtle bugs in the init code
(which is strange, because the original file was contributed by the
devil himself).
Seems logical. For some reason, the player allows deselecting both audio
and video stream without quitting (a deliberate feature of which I have
no idea why it was added years ago), so this is needed.
This reverts commit 45c8b97efb.
Some else complained (github issue #1163).
The feature requested in #1148 will be implemented differently in
the following commit.
The one in msg.c was mistakenly removed with commit e99a37f6.
I didn't actually test the change in ao_sndio.c (but obviously "ap"
shouldn't be static).
Don't wait after the audio thread has pushed the remaining audio to the
AO. Avoids hard hangs if the heuristic fails completely (could still
happen if get_delay returns absurd values).
CC: @mpv-player/stable
Since the internal AO driver API has no proper way to determine EOF, we
need to guess by querying get_delay. But some AOs (e.g. ao_pulse with
no-latency-hacks set) may never reach 0, maybe because they naively add
the latency to the buffer level. In this case our heuristic can break.
Fix by always using the delay to estimate the EOF time. It's not even
that important - it's mostly used to avoid blocking draining. So this
should be ok.
CC: @mpv-player/stable (maybe)
The event monitor is used to get keyboard events when there is no window, but
since it is a global monitor to the current process, we don't want it in a
library setting.
After @frau's split of macosx_events from macosx_application, `is_cplayer' is
not needed anymore. At the moment only global events such as Media Keys and
Apple Remote work, because the VO-level ones were hardcoded to be disabled.
(that will be fix in a later commit ).
Worryingly wrong. Fixes#1162.
Also fix another issue (window title was set anyway), which was why I
didn't notice this and testing it seemed to be fine.
Unfortunately, ALSA is particularly bad with this, because mpv has to
add all sorts of magic crap to the device name to make things work. The
device selection overrides this, so explicitly selecting devices will
most likely break your audio. This has yet to be solved.
For cover art, we pretend that the video stream is infinite, but also
stop decoding once we have an image on the VO (this seems advantageous
for the case when strange filters are inserted or the VO image gets
lost). Since a while ago, the video chain started decoding 2 images
though ("Non-monotonic video pts: 0.000000 <= 0.000000"), which is
annoying and wasteful.
Improve this by handling a certain corner case at initialization, which
will decode a second image while the first one is still stuck in the
filter chain. Also, just in case there are filters which buffer a lot,
also force EOF filtering (which means we tell the filters to flush
buffered frames).
CC: @mpv-player/stable
This was added with commit 3cbd79b3, but it turns out this
unintentionally enables "real" pausing when seeking while buffering. It
was done for ensuring correct state of the "cache-buffering-state"
property, but it also turns out that this was unneeded (another variable
that is reset when seeking happens to take care of this).
Maybe using strings for log levels was a mistake (too broad and too
impractical), so I'm adding numeric log level at least for the receiver
side. This makes it easier to map mpv log levels to other logging
systems.
I'm still too stingy to add a function to set the log level by a numeric
value, though.
The numeric values are not directly mapped to the internal mpv values,
because then almost every file in mpv would have to include the client
API header.
Coalesce this into API version 1.6, since 1.6 was bumped just yesterday.
For the purpose of demonstration.
Also make the windows larger. I'm not exactly sure how Qt determines
the default window sizes, but here they are a bit tiny, so force them
larger.
The API could return partial lines, meaning the message could stop
in the middle of a line, and the next message would have the rest of
it (or just the next part of it). This was a pain for the user, so do
the nasty task of buffering the lines ourselves.
Now only complete lines are sent. To make things even easier for the
API user, don't put multiple lines into a single event, but split them.
The terminal output code needed something similar (inserting a prefix
header on start of each line). To avoid code duplication, this commit
refactors the terminal output so that lines are split in a single
place.
The status line is a bit special; for example it uses special control
codes by design and is not terminated with a newline character in order
to update it on the terminal without scrolling. It's not helpful for
client API users either, and would require special-casing them
(emulating aspects of a terminal?). Also, the status line code is
explicitly disabled on osd.c unless the --terminal option is enabled,
so there was no good way to even enable the status line for the API.
Just pretend that the status line does not exist as far as the client
API is concerned. It won't be sent as MPV_EVENT_LOG_MESSAGE.
So client API users don't have to worry about this specifically.
Also document the overflow case. (Not sure if we really need to do
this; maybe it'd be better not to, since this just adds more noise
to the docs.)
stream provides a read buffer (so even something like stream_read_char()
is very fast). This means the stream reads ahead by a few KBs, and
implies that the internal position (s->pos, which would match e.g. the
file position in stream_file.c), and the external position
(stream_tell()) can be different. stream_tell() shows how these are
related.
When dropping buffers, which happens on byte-level discontinuities with
a bunch of streams (including DVB), we should not change the position as
seen by the demuxer. On the other hand, the internal position is not
really meaningful, since these streams aren't seekable anyway. So just
change the code such that stream_drop_buffers() doesn't change the
demuxer visible position.
I'm hoping that this will fix a few problems with DVB. (Also see
previous commit.)
Document the "normal" behavior (if MPV_FORMAT_NONE is not used) first,
and then introduce MPV_FORMAT_NONE as exception.
The actual semantics didn't change in mpv; this is only clarification.
Whether you consider the semantics weird or not depends on your use
case, but I suppose it's a bit confusing anyway. At this point, we keep
MPV_EVENT_PAUSE/UNPAUSE for compatibility only.
Make the "core-idle" property somewhat more useful in this context.
bstr.start can be NULL when bstr.len is 0, so don't call memcmp or
strncasecmp if that's the case. Passing NULL to string functions is
invalid C, even when the length is 0, and it causes Windows to raise an
invalid parameter error.
Should fix#1155
For segment linking (this mechanism matches file extensions to avoid
opening files which are most likely not Matroska files in order to speed
up scanning).
Now any action that stops playback of a file (even playlist navigation)
will save the position. Normal EOF is of course excluded from this, as
well as commands that just reload the current file.
The option name is now slightly off, although you could argue what the
word "quit" means.
Fixes#1148 (or at least this is how I understood it).
This function is available starting with PulseAudio 2.0, while we only
require 1.0. This broke compilation on Ubuntu 12.04.5 LTS.
Use our own function to calculate the buffer size, which is actually
simpler and needs slightly less code.
Hopefully fixes#1154.
CC: @mpv-player/stable
Run opening the stream and opening the demuxer in a separate thread.
This should remove the last code paths in which the player can normally
get blocked on network.
When the stream is opened, the player will still react to input and so
on. Commands to abort opening can also be handled properly, instead of
using some of the old hacks in input.c. The only thing the user can
really do is aborting loading by navigating the playlist or quitting.
Whether playback abort works depends on the stream implementation; with
normal network, this will depend on what libavformat (via "interrupt"
callback) does.
Some pain is caused by DVD/BD/DVB. These want to reload the demuxer
sometimes. DVB wants it in order to discard old, inactive streams.
DVD/BD for the same reason, and also for reloading stream languages
and similar metadata. This means the stream and the demuxer have to
be loaded separately.
One minor detail is that we now need to copy all global options. This
wasn't really needed before, because the options were accessed on
opening only, but since opening is now on a separate thread, this
obviously becomes a necessity.
This is the first of a series of commits that will change the Cocoa way in a
way that is easily embeddable inside parent views. To reach that point common
code must avoid referencing the parent NSWindow since that could be the host
application's window.
This is just temporary code but is a good base for future work (and baby
steps are required for these changes). The 'final destination' is embedding
the video view into any NSView but that requires some more work (the mechanism
will be the same: pass the view's pointer casted to int64_t through -wid).
For instance we will need to remove as much usage of the window instance
as possible, and use nil guards where not possible. For this reason I will
remove stuff like the mission control fullscreen feature (it's a cute feature
but annoying to support and quite limited, go make your GUIs), and a way to
lookup the current screen directly from the NSView absolute coordinates
(this is needed for ICC detection mostly, and reporting back the screen to
mpv's core).
Moreover the current view.m will need to be separated into 2 views: the actual
video view that will be embedded, and a parent view that will not be embedded
and will be responsibile for tracking events.
This could be dangerous because we initialize the window asynchronously and
return immediately from config, but since the OpenGL context is already
created, this seems to work correctly and doesn't cause weird deadlock cases.
This is for users who don't like changes. I'm hoping it will make the
process of cleaning up key bindings less bumpy.
It should be mentioned in the release notes of the next release.
It was more complicated than it had to be: the audio thread already
determines whether audio has ended, so we can use that. Remove the
separate logic for draining.
Commit 957097 attempted to use PA_STREAM_FAIL_ON_SUSPEND to make
ao_pulse exit if the stream was started suspended.
Unfortunately, PA_STREAM_FAIL_ON_SUSPEND is active even during playback.
If you pause mpv, pulseaudio will close the actual audio device after a
while (or something like this), and unpausing won't work. Instead, it
will spam "Entity killed" error messages.
Undo this change and check for suspended audio manually during init.
CC: @mpv-player/stable
This doesn't look to be needed anymore. Fullscreening with both the NSView
and the NSWindow API works correctly. I guess this was forgotten in from older
code which changed presentation options directly for going fullscreen.
--x11-netwm=yes now forces NetWM fullscreen, while --x11-netwm=auto
(detect whether NetWM fullsctreen support is available) is the old
behavior and still the default.
See #888.
Unfortunately using dispatch_sync for synchronization turned out to be really
bad for us. It caused a wide array of race conditions, deadlocks, etc.
Moving to a very simple mutex. It's not clear to me how to do liveresizing
with this, for now it just flickers with is unacceptable (maybe I'll draw
black instead).
This should fix all the threading cocoa bugs. Reopen if it's not the case!
Fixes#751Fixes#1129
Also recreate ASS_Library on every file played. This means we can move
the code out of main.c as well.
Recreating the ASS_Library object has no disadvantages, because it
literally stores only the message callback, the (per-file) font
attachment as byte arrays, and the set of style overrides. Hopefully
this thing can be removed from the libass API entirely at some point.
The only reason why the player core creates the ASS_Renderer, instead
of the subtitle renderer, is because we want to cache the loaded fonts
across ordered chapter transitions, so this probably still has to stay
around for now.
When the VO was moved it its own thread, responsibility for redrawing
was given to the VO thread itself. So if there was a condition that
indicated that redrawing was required, like expose events or certain
VOCTRLs, the VO thread was redrawing itself.
This worked fine, but there are some corner cases where this works
rather badly. E.g. if I fullscreen the player and hit panscan controls
with mpv's default autorepeat rate, playback stops. This happens because
the VO redraws itself after every panscan change command. Running each
(repeated) command takes so long due to redrawing and (involuntary)
waiting on vsync, that it never leaves the input processing loop while
the key is held down. I suspect that in my case, redrawing in fullscreen
mode just gets slow enough that it takes 2 vsyncs instead of 1 on
average, and the processing time gets larger than the autorepeat delay.
Fix this by taking redraw control from the VO, and instead let the
playloop issue a "real" redraw command to the VO if needed. This
basically reverts redraw handling to what it was before moving the VO to
a thread.
CC: @mpv-player/stable
Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The
main use of this was that you could pass a bitmask of these flags to
uninit_player(). Except in some situations where you wanted to
uninitialize nearly everything, this wasn't really useful. Moreover, it
was quite annoying that subsystems had most of the code in a specific
file, but the uninit code in loadfile.c (because that's where
uninit_player() was implemented).
Simplify all this. Remove the flags; e.g. instead of testing for the
INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code
to separate functions, e.g. uninit_audio_out().
Sometimes, ao_pulse starts in suspended mode, which means playback is
essentially paused in pulseaudio. This gives the impression that mpv is
hanging, since it times video against the audio playback progress, and
audio never makes progress in this state.
I'm not sure if this will help - possibly it does with mixed
pulseaudio/alsa setups. However, if the alsa setup has the pulseaudio
plugin, alsa will hang too. But there's still a chance we get less
blame for pulseaudio messes.
This gets rid of this warning:
Could not update timestamps for skipped samples.
This required an API addition to FFmpeg (otherwise it would instead
doing arithmetic on the timestamps itself), so whether it works depends
on the FFmpeg version.
For the sake of libmpv. Might make things much easier for the user,
especially on Windows. On the other hand, it's a bit sketchy that a
command exists that makes the player access arbitrary memory regions.
(But do note that input commands are not meant to be "secure" and never
were - for example, there's the "run" command, which obviously allows
running random shell commands.)
Somewhat more flexible: now there's a separate overlay struct, and you
don't need to coerce all state into struct sub_bitmap. Also, removing
the previous mapping (munmap call) is now all in one place, the
replace_overlay function.
Makes the next commit easier to implement.
Use percent-pos instead, which is exactly the same, except with the
range 0.0-100.0.
I'm not sure how this got there; it was probably introduced and then
removed again as percent-pos got more precise.
CC: @mpv-player/stable
Another fallout resulting from the changes whether or not to wait for
mapping the window. In this case, it obviously makes no sense to wait
for mapping, because the root window is always mapped. Mapping will
never happen, and it would wait forever.
Fixes#1139.
CC: @mpv-player/stable
The messages "Audio: no audio" and "Video: no video" could be printed
twice each if initializing them failed. Prevent his silliness.
CC: @mpv-player/stable
Apparently this is what users want. When playing with normal speed,
nothing is done. When playing slower than normal, resampling is used
instead, because scaletempo (which does the pitch correction) adds
too many artifacts.
Although the "af" command already could do this, it seems it's better
to introduce a lower level mechanism for now. This avoids some messy
issues, since that code would recursive call reinit_audio_chain().
To be used by the next commit.
There's no real reason why audio_init_filter() should exist. Just use
af_init or af_reinit directly. (We lose a useless message; the same
information is printed in a quite close place with more details.)
Requires less code, and the way the filter chain is marked as having
failed to initialize allows just switching off audio instead of
crashing if trying to insert a volume filter in mixer.c fails, and
recreating the old filter chain fails too.
Simple fix for issue #1137.
Since all sub-bitmaps are packed on a larger texture, there's still a
"fall off" on the border due to the linear scaling. This could be
fixed by constraining each sub-bitmap to its own texture, or by
clamping on the shader level, but I don't care for now.
At least on kwin, we decide to proceed without waiting for the window
being mapped (due to the frame exts hack, see commit 8c002b79). But that
leaves us with a window size of 0x0, which causes VdpOutputSurfaceCreate
to fail. This prints some warnings, although vo_vdpau recovers later and
this has no other bad consequences.
Do the following things to deal with this:
- set the "known" window size to the suggested window size before the
window is even created
- allow calling XGetGeometry on the window even if the window is not
mapped yet (this should work just fine)
- make the output surface minimum size 1x1
Strictly speaking, only one of these would be required to make the
warning disappear, but they're all valid changes and increase robustness
and correctness. At no point we use a window size of 0x0 as magic value
for "unset" or unknown size, so keeping it unset has no purpose anyway.
CC: @mpv-player/stable
Commit 50e131b43e happened to make it work for DVD (because the higher
bits of the ID are masked in the DVD case), but failed for Bluray. This
probably fixes it, although I don't have a sample to multiple streams to
confirm it really does it right.
CC: @mpv-player/stable
This would play some silence in case video was slower than audio. If
framedropping is already enabled, there's no other way to keep A/V
sync, short of changing audio playback speed (which would give worse
results). The --audiodrop option inserted silence if there was more
than 500ms desync.
This worked somewhat, but I think it was a silly idea after all. Whether
the playback experience is really bad or slightly worse doesn't really
matter. There also was a subtle bug with PTS handling, that apparently
caused A/V desync anyway at ridiculous playback speeds.
Just remove this feature; nobody is going to use it anyway.
When embedding, if the parent window is destroyed, it will cause mpv's
window to be destroyed as well. Since WM_USER wakeups are sent to the
window, destroying the window will prevent wakeups and cause uninit to
hang.
Fix this by quitting the event loop on WM_DESTROY. Events should only be
processed for the lifetime of the window, from CreateWindowEx to
WM_DESTROY. After the event loop is finished, mp_dispatch_queue_process
can handle any remaining requests.
Might matter when libavformat tries to do tiny seekbacks in an
unseekable stream, and the seekback buffer isn't large enough. In this
case, seeking would fail, and would drop the current buffer. The
seekback would end up dropping future data.
This change probably doesn't have any observable effects. libavformat
normally has its own stream buffer, and demux_mkv.c tries carefully
never to seek back.
This makes no sense to use with DVD/BD/DVB and some others, and these
streams happen to be unseekable.
Also, other kinds of unseekable streams (like reading from pipe) should
work, but will exhibit sketchy behavior if they need to seek. So just
disable it, and leave these problems to the memory cache (cache.c).
This code meant to flush demuxer internal buffers by doing a byte seek
to the current position. In theory this shouldn't drop any stream data.
However, if the stream positions mismatch, then avio_seek() (called by
av_seek_frame()) stops being a no-op, and might for example read some
data to skip to the seek target. (This can happen if the distance is
less than SHORT_SEEK_THRESHOLD.)
The positions get out of sync because we drop data at one point (which
is what we _want_ to do). Strictly speaking, the AVIOContext flushing is
done incorrectly, becuase pb->pos points to the start of the buffer, not
the current position. So we have to increment pb->pos by the buffered
amount.
Since there are other weird reasons why the positions might go out of
sync (such as stream_dvd.c dropping buffers itself), and they don't
necessarily need to be in sync in the first place unless AVIOContext has
nothing buffered internally, just use the sledgehammer approach and
correct the position manually.
Also run av_seek_frame() after this. Currently, it shouldn't read
anything, but who knows how that might change with future libavformat
development.
This whole change didn't have any observable effect for me, but I'm
hoping it fixes a reported problem.
When flushing the AVIOContext, make sure it can't seek back to discarded
data. buf_ptr is just the current read position, while buf_end - buffer
is the actual buffer size. Since mpegts.c is littered with seek calls,
it might be that the ability to seek could read
Mark the stream (which the demuxer uses) as not seekable. The cache can
enable seeking again (this behavior is sometimes useful for other
things). I think this should have had no bad influence in theory, since
seeking BD/DVD first does the "real" seek, then flushes libavformat and
reads new packets.
The intention was to adjust the window size to video size. It never
worked well; it prevented the user to make the window smaller. For
unknown reason it stopped resizing properly as well.
This is just a cheap example, and I don't intend to fight with Qt, so
replace the "demonstration" behavior by something slightly lamer.
Commit 64b7811c tried to do the "right thing" with respect to whether
keyboard input should be enabled or not. It turns out that X11 does
something stupid by design. All modern toolkits work around this native
X11 behavior, but embedding breaks these workarounds.
The only way to handle this correctly is the XEmbed protocol. It needs
to be supported by the toolkit, and probably also some mpv support. But
Qt has inconsistent support for it. In Qt 4, a X11 specific embedding
widget was needed. Qt 5.0 doesn't support it at all. Qt 5.1 apparently
supports it via QWindow, but if it really does, I couldn't get it to
work.
So add a hack instead. The new --input-x11-keyboard option controls
whether mpv should enable keyboard input on the X11 window or not. In
the command line player, it's enabled by default, but in libmpv it's
disabled.
This hack has the same problem as all previous embedding had: move the
mouse outside of the window, and you don't get keyboard input anymore.
Likewise, mpv will steal all keyboard input from the parent application
as long as the mouse is inside of the mpv window.
Also see issue #1090.
We inserted these filters with fixed parameters, which was ok. But this
also didn't change image parameters for the filters down the filter
chain and the VO. For example, if rotation by 90° was requested by the
file, we would insert a filter and rotate the video, but the VO would
still receive image parameters that direct rotation by 90°.
This wasn't a problem, but it could become one.
Fix this by letting the filters automatically pick up the image params.
The image params are reset on application. (We could probably also
always try to apply and reset image params in a filter, instead of
having special "auto" parameters. This would probably work, and video.c
would insert a "rotate=0" filter. But I'm afraid this would be confusing
and the current solution is cosmetically slightly nicer.)
Unfortunately, the vf_stereo3d.c change turned out a big mess, but once
the "internal" filter is fully replaced with libavfilter, most of this
can be radically simplified.
Some filters exists only to create a specific lavfi graph. Allow these
filters to reset the graph exactly on reconfig, and allow them to modify
some image parameters too. Also make vf_lw_update_graph() behave like
vf_lw_set_graph() - they had a subtitle difference with filter==NULL.
Useful for the following commit.
Originally, all options were copied to ensure that input_ctx remins
thread-safe, even if options are changed asynchronously. But this got
a bit inconsistent. Copy them automatically and reduce some weirdness.
Until now, creating the input_ctx was delayed until the command line
and config files were parsed. Separate creation and loading so that
input_ctx is available from start.
This should make it possible to simplify some things. For example,
some complications with Cocoa were apparently only because input_ctx
was available only "later". (Although I'm not sure if this is still
relevant, or if the Cocoa code should even be organized this way.)
libsndio has absolutely no mechanism to discard already written audio
(other than SIGKILLing the sound server). sio_stop() will always block
until all audio is played. This is a legitimate design bug.
In theory, we could just not stop it at all, so if the player is e.g.
paused, the remaining audio would be played. When resuming, we would
have to do something to ensure get_delay() returns the right value. But
I couldn't get it to work in all cases.
get_delay needs to report the current audio buffer status. It's
important for A/V sync that this information is current, but functions
which update it were called on play() or get_space() calls only.
This was in bytes, but it's more convenient to use samples (or frames;
in any case the smallest unit of audio that includes all channels).
Remove the ao->bps line too; it will be set after init() returns.
This warning makes absolutely no sense. Passing an empty string to
printf-like functions is perfectly fine. In the OSD case, it just sets
an empty message, practically clearing the OSD.
Makes it behave slightly better for VP9. This is also the behavior
libavformat has.
Also while we're at it, don't set duration except for the first packet.
Normally we don't use the duration except for subtitles (which are never
parsed or "laced"), so this should make no observable difference.
Similar as the previous commits.
Most of the code is actually copied from the stream_dvdnav.c code, but
I'd rather prefer to duplicate it, than to entangle them. The latter
would probably result in terrible things in a few years.
Same hack as with stream_dvd.c.
VIDEO_TS.IFO files are now opened via stream_dvdnav.c. Directories
containing a VIDEO_TS.IFO or VIDEO_TS/VIDEO_TS.IFO file are also
opened with it.
stream_dvd.c includes a pseudo-protocol that recognizes .IFO files, and
plays them using libdvdread. This was relatively lazy, and could perhaps
easily trigger with files that just had the .ifo extension.
Make the checks stricter, and even probe the file header. Apparently the
first bytes in an .ifo file are always "DVDVIDEO-VTS", so check for
this.
Refuse to load the main "video_ts.ifo". The plan is to use stream_dvdnav
for it.
This also removes at least 1 memory leak.
set_osd_bar_chapters() always cleared the OSD bar stops, even if the
current bar was not the seek bar. Obviously it should leave the state of
the bar alone in this case.
Also change the function control flow so that we can drop one
indentation level, and do the equivalent change for the other OSD bar
functions.
Eliminate the remains of the OSD message stack. Another simplification
comes from the fact that we do not need to care about time going
backwards (we always use a monotonic time source, and wrapping time
values are practically impossible). What this code was pretty trivial,
and by now unnecessarily roundabout.
Merge get_osd_msg() into update_osd_msg(), and add_osd_msg() into
set_osd_msg_va().
There's no need to update OSD messages and the terminal status if nobody
is going to see it. Since the player doesn't block on video display
anymore, this update happens to often and probably burns slightly more
CPU than necessary. (OSD redrawing is handled separately, so it's just
mostly useless text processing and such.)
Change it so that it's updated only on every video frame or all 50ms
(whatever comes first).
For VO OSD, we could in theory try to lock to the OSD redraw heuristic
or the display refresh rate, but that's more complicated and doesn't
work for the terminal status.
When using --force-window (and no video or cover art), this heuristic
prevents any redrawing during seeking. It should be applied only if
there is any form of video.
This makes subtitle display somewhat work if no video is displayed, but
a VO window exists (--force-window or cover art display).
The main problem with normal subtitle display is that it's locked to
video: it uses the video PTS as reference, and the subtitles advance
only if a new video frame is displayed. In audio-only mode on the other
hand, no video frame is ever displayed (or only 1 in the cover art
case). You would need a workaround to adjust the subtitle PTS, and you
would have to decide with what frequency to update the display. In
general, there is no "right" display FPS for subtitles. Some formats
(ASS) have animations parameterized by time, and any refresh rate could
be used.
Sidestep these problems by enabling the text OSD-based subtitle
mechanism. This is similar to --no-sub-ass, and updates and renders
subtitles with plain OSD. It has some caveats: no bitmap subs, somewhat
incorrect timing, no formatting. Timing in particular is a bit strange
and depends how often the audio output asks for new data, or other
events that happen to wakeup the playloop.
This was once central, but now it's almost unused. Only vf_divtc still
uses it for extremely weird and incomprehensible reasons. The use in
stream.c is trivial. Replace these, and remove mpbswap.h.
stream_cdda's output format is linked to demux_raw's default audio
format, and at least we don't care enough to provide a separate
mechanism to let stream_cdda explicitly set the format, so they must
match.
Judging from the existing code, it looks like CDDA always outputs little
endian. stream_cdda.c changed this back to native endian (what demux_raw
expects). Just make them both little endian. This requires less code,
and also having a raw demuxer's behavior depend on the endianness of the
machine isn't very sane anyway.
See previous commits. This finally replaces directly reading the file
data into a struct with reading them manually. In theory this is more
portable (no alignment issues and other things). For the most part,
it's nice seeing this gone.
MPlayer traditionally did this because it made sense: the most important
formats (avi, asf/wmv) used Microsoft formats, and many important
decoders (win32 binary codecs) also did. But the world has changed, and
I've always wanted to get rid of this thing from the codebase.
demux_mkv.c internally still uses it, because, guess what, Matroska has
a VfW muxing mode, which uses these data structures natively.
Let codec_tags.c do the messy mapping.
In theory we could simplify further by makign demux_mkv.c directly use
codec names instead of the MPlayer-inherited "internal FourCC" business,
but I'd rather not touch this - it would just break things.
For a while, we used this to transfer PCM from demuxer to the filter
chain. We had a special "codec" that mapped what MPlayer used to do
(MPlayer passes the AF sample format over an extra field to ad_pcm,
which specially interprets it).
Do this by providing a mp_set_pcm_codec() function, which describes a
sample format in a generic way, and sets the appropriate demuxer header
fields so that libavcodec interprets it correctly. We use the fact that
libavcodec has separate PCM decoders for each format. These are
systematically named, so we can easily map them.
This has the advantage that we can change the audio filter chain as we
like, without losing features from the "rawaudio" demuxer. In fact, this
commit also gets rid of the audio filter chain formats completely.
Instead have an explicit list of PCM formats. (We could even just have
the user pass libavcodec PCM decoder names directly, but that would be
annoying in other ways.)
Digital pass-through was probably broken. Possibly fix it (no way to
test). This also should make the logic slightly saner.
Fortunately, it's unlikely that anyone who uses OSS has a spdif setup.
Commit 5b5a3d0c broke this. The really funny thing is that this code was
actually always under "#if BYTE_ORDER == BIG_ENDIAN". The breaking
commit just edited this code slightly, but it must have failed to
compile on big endian long before (since over 1 year ago, commit d3fb58).
Should be able to pass-through AC3, DTS, and others.
It seems PulseAudio wants players to fallback to PCM on certain events
signaled by the server, but we don't implement that. There's not much
documentation available anyway.
Before this commit, there was AF_FORMAT_AC3 (the original spdif format,
used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS
and DTS-HD), which was handled as some sort of superset for
AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used
IEC61937-framing, but still was handled as something "separate".
Technically, all of them are pretty similar, but may use different
bitrates. Since digital passthrough pretends to be PCM (just with
special headers that wrap digital packets), this is easily detectable by
the higher samplerate or higher number of channels, so I don't know why
you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs.
AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is
just a mess.
Simplify this by handling all these formats the same way.
AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3).
All AOs just accept all spdif formats now - whether that works or not is
not really clear (seems inconsistent due to earlier attempts to make
DTS-HD work). But on the other hand, enabling spdif requires manual user
interaction, so it doesn't matter much if initialization fails in
slightly less graceful ways if it can't work at all.
At a later point, we will support passthrough with ao_pulse. It seems
the PulseAudio API wants to know the codec type (or maybe not - feeding
it DTS while telling it it's AC3 works), add separate formats for each
codecs. While this reminds of the earlier chaos, it's stricter, and most
code just uses AF_FORMAT_IS_IEC61937().
Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to
include special formats, so that it always describes the fundamental
sample format type. This also ensures valid AF formats are never 0 (this
was probably broken in one of the earlier commits from today).
This code tried to play with the format bits, and potentially could
create invalid formats, or reinterpret obscure formats in unexpected
ways.
Also there was an abort() call if the winapi or mpv used a format with
unexpected bit-width. This could probably easily happen; for example,
mpv supports at least one 64 bit format. And what would happen on 8 bit
formats anyway?
Untested.
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.
From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.
This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
IEC 61937 frames should always be little endian (little endian 16 bit
words). I don't see any apparent need why the audio chain should handle
swapped-endian formats.
It could be that some audio outputs might want them (especially on big
endian architectures). On the other hand, it's not clear how that works
on these architectures, and it's not even known whether the current code
works on big endian at all. If something should break, and it should
turn out that swapped-endian spdif is needed on any platform/AO,
swapping still could be done in-place within the affected AO, and
there's no need for the additional complexity in the rest of the player.
Note that af_lavcac3enc outputs big endian spdif frames for unknown
reasons. Normally, the resulting data is just pulled through an auto-
inserted conversion filter and turned into little endian. Maybe this was
done as a trick so that the code didn't have to byte-swap the actual
audio frame. In any case, just make it output little endian frames.
All of this is untested, because I have no receiver hardware.
Until now, we always required the playback core to decode a new frame to
get more output from the filter. That seems to be completely
unnecessary, because filtered results may arrive before that.
Add a filter_out callback, and restructure the code such that it can
return any filtered frames, or block if it hasn't read at least one
frame.
In the worst case, it still can happen that bursts of input requests and
output requests happen. (This commit tries to reduce burst-like
behavior, but it's not entirely possible due to the indeterministic
nature of VS threading.)
This is a similar change as with 95bb0bb6.
E.g. --loop-file=2 will play the file 3 times (one time normally, and 2
repeats).
Minor syntax issue: "--loop-file 5" won't work, you have to use
"--loop-file=5". This is because "--loop-file" still has to work for
compatibility, so the "old" syntax with a space between option name and
value can't work.
libavcodec/libavformat now handles gapless audio better. In theory, this
could be implemented with ad_mpg123 too, but since libavformat strips
metadata from mp3 files and passes pure mp3 packets to the decoders
only, this can't work by itself. Instead, the player must pass this
metadata separately. libav* do this relatively transparently over packet
"side data" (attached to AVPacket).
It might also be possible to let libmpg123 handles all this by
implementing it as demuxer that outputs PCM, but that would have other
problems, and I think it's better to make libavformat work correctly.
libmpg123 can still be used with '--ad=mpg123:mp3'.
Also see issue #1101.
We generally want 2 things:
1. minimal wakeups for decoding each frame
2. minimal number of frames decoded on continuous seeking
Commit 35810cb8 changed this a bit, and fixed 1. But it broke 2., and
now it decodes 2 frames instead of 1 when you keep seeking (arrow key
held down or such). This made seeking appear slower.
Fix this by making the logic more explicit. In particular, call the
filters only if we actually try to get a new frame.
When playing with --no-audio and all other distractions disabled (like
OSC), it still wakes up 2 times per frame - but the second time is
merely because the VO didn't accept the new frame yet.
Be less annoying, print the actual OSD level instead of something
meaningless, but still clear the OSD if OSD level 0 (no OSD) is set.
Remove the special handling for terminal OSD, that was just dumb.
This means that if a property not listed in property_osd_display[] is
changed, it will be shown on the OSD as "name: ${name}".
Properties that are listed in property_osd_display[] and have osd_name
not set stay invisible by default. This is used for "pause" and
"fullscreen", which (like before this commit) are not shown by default,
because it would be annoying.
The defaults still can be changed with command prefixes (osd-msg,
no-osd, others).
Probably not many user-visible changes. One notable change is that the
terminal OSD code for OSD bar fallback handling is removed with no
replacement. Instead, terminal OSD gets the same text message as normal
OSD. For volume, this is ok, because the text message is reasonable.
Other properties will look worse, but could be adjusted, and there are
in fact no other such properties that would be useful in audio-only
mode.
The fallback message for seeking falls away as well, but that message
was useless anyway - the terminal status line provides all information
anyway.
I believe the show_property_osd() code is now much easier to follow.
If no VO was open, these options couldn't be changed or even queried.
Although these properties are nearly useless if no VO exists, there's
actually no good reason to forbid querying or setting them. Also, even
if the VO is created, it doesn't mean the VO window was created.
Why bother?
Also, since now some properties could be mapped to non-existing options,
but mp_property_generic_option() is used, deal with this case and return
a not-found error code.
If there's a command that uses the OSD by default, then always print the
associated message (or a fallback made of name + value), even if the
command has an associated OSD bar.
This means volume, gamma, panscan, etc. all show both a message and a
OSD bar.
Also, add a '%' to the volume message. The extra_msg thing is not needed
anymore.
See issue #1103.
It's just confusing; users are encouraged to edit input.conf instead
(changing the argument to the "add" command).
Update input.conf to keep the old behavior.
When pausing after a frame was just dropped, we're logically at the
dropped frame, and thus should redraw the dropped frame. This was
implemented, but didn't work after unpausing for the second time,
because of a minor logic bug.
For incomprehensible reasons, AV_PIX_FMT_GRAY8 (and some others) have a
palette. This literally makes no sense and this issue has bitten us
before, but it is how it is.
This also caused a crash with vo_direct3d: this mapped a texture as
IMGFMT_Y8 (i.e. AV_PIX_FMT_GRAY8), and when copying this, it tried to
copy the non-existent palette.
Fixes#1113.
vo_vdpau uses its own framedrop code, mostly for historic reasons. It
has some tricky heuristics, of which I'm not sure how they work, or if
they have any effect at all, but in any case, I want to keep this code
for now. One day it might get fully ported to the vo.c framedrop code,
or just removed.
But improve its interaction with the user-visible framedrop controls.
Make --framedrop actually enable and disable the vo_vdpau framedrop
code, and increment the number of dropped frames correctly.
The code path for other VOs should be equivalent. The vo_vdpau behavior
should, except for the improvements mentioned above, be mostly
equivalent as well. One minor change is that frames "shown" during
preemption are always count as dropped.
Remove the statement from the manpage that vo_vdpau is the default; this
hasn't been the case for a while.
vc->vsync_interval and vsync_interval should be the same value, but
actually vc->vsync_interval was updated after vsync_interval was
initialized. This was probably not intended. Fix this by removing the
duplicate local variable. There were probably no bad effects.
When compiling semaphore_osx.c on win32, the following error happened:
/usr/i686-w64-mingw32/include/semaphore.h:160:6: error: unknown type name 'mode_t'
This is because this system header references symbols that are not
not defined anywhere. This is clearly a bug in pthreads-w32, but has
been known and unfixed since 2012, so add a hack to fix it.
We build semaphore_osx.c this way because it saves us an extra configure
check. On win32, Linux, etc. it's empty and contains
"#include <semaphore.h>" only.
Should fix#1108.
Be less clever, and restore the volume state even with AOs like pulse,
which have per-application audio.
Before this commit we didn't do this, because the volume is global (even
if per-application), so the volume will persist between invocations. But
to me it looks like always restoring is less tricky and makes for easier
to understand semantics.
Also, don't always unmute on exit. Unmuting was done even with ao_pulse,
and interfered with user expectations (see #1107).
This might annoy some users, because mpv will change the volume all the
time. We will see.
Fixes#1107.
Follow up to previous commit.
This is probably confusing from a user point of view, since this field
shouldn't show up normally anymore. (Before this commit, it could show
up sporadically when a slow operation was performed during playback,
such as switching fullscreen.)
Normally, feeding a packet to the decoder should always return a frame
_if_ we received a frame before. So while we can't know exactly whether
a frame was dropped, at least the normal case is easily detectable.
This means we display something closer to the actual framedrop count,
instead of a bad guess.
This is the "old" framedropping mode (derived from MPlayer). At least in
the mplayer2/mpv source base, it stopped working properly years ago (or
maybe it never worked properly). For one, it depends on the video
framerate, which assume constant framerate. Another problem was that it
could lead to freezing video display: video could get so much behind
that it couldn't recover from framedrop.
Make some small changes to improve this.
Don't use the current audio position to check how much we are behind.
Instead, use the last known A/V difference. last_av_difference is
updated only when a video frame is scheduled for display. This means we
can keep stop dropping once we're done catching up, even if video is
technically still behind. What helps us here that this forces a video
frame to be displayed after a while. Likewise, we reset the
dropped_frames count only when scheduling a new frame for display as
well.
Some inspiration was taken from earlier work by xnor (see issue #620),
although the implementation turned out quite different.
This still uses the demuxer-reported (possibly broken) FPS value. It
also doesn't account for filters changing FPS. We can't do much about
this, because without decoding _and_ filtering, we just can't know how
long a frame is. In theory, you could derive that from the raw packet
timestamps and the filter chain contents, but actually doing this is
too involved. Fortunately, the main thing the FPS affects is actually
the displayed framedrop count.
Sometimes, --af=hrtf produces heavy artifacts or silence. It's possible
that this commit fixes these issues. My theory is that usually, the
uninitialized coefficients quickly converge to sane values as more audio
is filtered, which would explain why there are often artifacts on init,
with normal playback after that. It's also possible that sometimes, the
uninitialized values were NaN or inf, so that the artifacts (or silence)
would never go away.
Fix this by initializing the coefficients to 0. I'm not sure if this is
correct, but certainly better than before.
See issue #1104.
Let us set a different rate and delay.
Needed for the following commit where we set rate and delay reported by weston.
But only if the option native-keyrepeat is set.
Uses the new mechanism introduced in the previous commit.
Depending on the actual filter, this distributes CPU load more evenly
over time, although it probably doesn't matter.
Consider a filter which turns 1 frame into 2 frames (such as an
deinterlacer). Until now, we forced filters to produce all output frames
at once. This was done for simplicity.
Change the filter API such that a filter can produce frames
incrementally.
Rename video_decode_and_filter to video_filter, and add a new
video_decode_and_filter function. This function now calls the decoder.
This is done so that we can check filters a second time after decoding,
which avoids a useless playloop iteration.
(This and the previous commits are really just microoptimizations, which
simply reduce the number of times the playloop has to recheck
everything.)
Move the check to a function. Run the check a second time after
decoding/filtering. This second check is strictly speaking redundant
(which is why it wasn't done until now), but it avoids a useless
playloop iteration.
Move this code below the code that "shifts" the newly filtered frame.
This allows us to skip a useless playloop iteration later, because
obviously we need to filter a new frame after the previous frame has
been "shifted", and not before that.
Until now, you could override only level 3 with --osd-status-msg. Extend
this, add add --osd-msg1 to --osd-msg3 (one for each OSD level). OSD
level 0 always means disable OSD, so that isn't included.
--osd-msg3 corresponds to --osd-status-msg, but they're not exactly the
same. To allow more customization, --osd-msgN do not include the OSD
symbol. The symbol can be manually added with "${osd-sym-cc}". We keep
the "old" option for some short-term compatibility.
--osd-msg1 should be particularly useful; for example you could do:
--osd-msg1='${?pause==yes:${osd-sym-cc}}'
to display a "paused" symbol when paused, and nothing during normal
playback. (Although admittedly, the syntax is quite a bit of work.)
We don't allow this by default, because it would be silly if random
external data (like filenames or file tags) could accidentally trigger
them.
Add a property that magically disables this ASS tag escaping.
Note that malicious input could still disable ASS tag escaping by
itself. This would be annoying but harmless.
Pausing/unpausing while the audio device can't be reopened, and then
unpausing again when the device is finally reopened, can hang the
player for a while.
This happens because p->prepause_samples grows without bounds each
time the player is unpaused while the device is lost. On unpause,
ao_oss plays prepause_samples of silence to compensate for A/V timing
issues due to the partially lost buffer (we can't pause the device at
an arbitrary sample position, and the current period will be lost).
This in turn will make the player appear to be frozen if too much
audio is queued. (Normally, play() must never block, but here it
happens because more data is written than get_space() reports. A
better implementation would never let prepause_samples grow larger
than the period size.)
The unbounded growth happens because get_space() always returns that
the device can be written while the device is lost. So limit it to
200ms. (A better implementation would limit it to the period size.)
Also see #1080.
There's no reason to let the core wait until the frame is done
displaying. In practice, the core normally didn't need this additional
wakeup, and the VO was quick enough to fetch the new frame, before the
core even attempted to queue a new frame. But it wasn't entirely clean,
and the correct wakeup handling might matter in some cases.
With default settings, this allows you to hit the 100% mark (with
default --softvol-max in the middle) even if you've reached min or max
volume before. This is because 50 is not divisible by 3 (old default)
but by 2 (new default).
Not really sure why there still can be issues with higher --softvol-max
and --volstep=1, but this is where I stop caring.
If --write-filename-in-watch-later-config is used, and the filename
contains newline characters (as generally allowed on Unix), then the
newline will be written to the resume file literally, and the parts
after the newline character are interpreted as options.
This is possibly security relevant.
Change newline characters (and in fact any other special characters)
to '_'.
Reported as #1099 (this commit is a reimplementation of the proposed
pull request).
CC: @mpv-player/stable
Apparently making ESC exit fullscreen mode is the more popular
convention compared to ESC quitting the program.
It was also concluded that ESC should do nothing when the windows is
already in normal state.
See discussion in #973.
The filter output size can be 0. Due to how filtering works, this is
nothing unusual, but avresample_convert() will return 0. The same case
is already handling with "normal" resampling (this commit fixes the
reordering code).
Additionally, don't use an assert(). avresample_convert() failing is
unusual, but might also happen due to e.g. internal out of memory
conditions, so we shouldn't just crash on it.
Curiously observed with --ao=oss --audio-channels=5.1 when changing
speed.
Apparently NetBSD users want/need this (see issue #1080).
In order not to break playback, we need at least to emulate get_delay().
We do this approximately by using the system clock.
Also, always close the audio device on reset. Reopen it on play only. If
we can't reopen it, don't retry until after the next time reset or
resume is called, to avoid spam and unexpectedly "stealing" back the
audio device.
Also do something about framestepping causing audio desync.
The context struct had an audio_buf_info field, but there's no reason
why this would be needed. It's a tiny struct, and it isn't permanent
state. It's always returned by SNDCTL_DSP_GETOSPACE. Keeping this as
field is just confusing, so get rid of it.
The code for reopening the audio device was separate, and duplicated
some of the "real" open code. This was very badly done, and major
required parts of initialization were skipped. Fix this by removing
the code duplication. This consists mainly of moving the code for
opening the device to a separate function, and adding some changes
to handle format changes gracefully. (We can't change the audio
format on the fly, but we can at least not explode and play noise
when that happens.)
As a minor change, actually always use SNDCTL_DSP_RESET when closing
the audio device. We don't want to wait until the rest of the buffer
is played.
Also, don't use strerror() when printing the error message that
reopening failed, simply because reopen_device() takes care of this,
and also errno might be clobbered at this point.
I have no idea whether this is true, because there literally doesn't
seem to exist documentation for SNDCTL_DSP_RESET. But at least on
Linux' OSS emulation, it is true. Also, it would be quite insane if
it would be needed.
It seems on NetBSD SNDCTL_DSP_RESET exists, but using it for pausing
is not feasible. We still use it to discard the audio buffer when
closing the audio device.
Some window managers can prevent mapping of a window as a feature. i3
can put new windows on a certain workspace (with "assign"), so if mpv is
started on a different workspace, the window will never be mapped.
mpv currently waits until the window is mapped (blocking almost all of
the player), in order to avoid race conditions regarding the window
size. We don't want to remove this, but on the other hand we also don't
want to block the player forever in these situations.
So what we need is a way to know when the window manager is "done" with
processing the map request. Unfortunately, there doesn't seem to be a
standard way for this. So, instead we could do some arbitrary
communication with the WM, that may act as "barrier" after map request
and the "immediate" mapping of the window. If the window is not mapped
after this barrier, it means the window manager decided to delay the
mapping indefinitely. Use the _NET_REQUEST_FRAME_EXTENTS message as such
a barrier. WMs supporting this message must set the _NET_FRAME_EXTENTS
property on the mpv window, and we receive a PropertyNotify event. If
that happens, we always continue and cancel waiting for the MapNotify
event.
I don't know if this is sane or if there's a better mechanism. Also,
this works only for WMs which support this message, which are not many.
But at least it appears to work on i3. It may reintroduce flickering on
fullscreen with other WMs, though.
This is pretty imperfect, but it's just a demo.
The main purpose is clarifying how and when to get the video size.
In theory, retrieving the properties this way has a race condition:
after reading dwidth, the video could resize again. But the worst that
can happen are mismatching dwidth/dheight values, and the
MPV_EVENT_VIDEO_RECONFIG event would be immediately received again,
which would fix the mismatch. You could read the full video-out-params
property to absolutely avoid it, but it's not worth the trouble.
Merges pull request #1094, with some minor changes. mpv expects IEEE,
and IEEE allows divisions by 0 for floats, so these shouldn't actually
be a problem, but do it anyway for the sake of clang.
Signed-off-by: wm4 <wm4@nowhere>
Use libwaio to read from pipes (stdin or named pipes) on Windows. This
liberates us from nasty issues, such as pipes (as created by most
programs) not being possible to read in a non-blocking or event-driven
way. Although it would be possible to do that in a somewhat sane way
on Vista+, it's still not easy, and on XP it's especially hard. libwaio
handles these things for us.
Move pipe.c to pipe-unix.c, and remove Windows specific things. Also
adjust the input.c code to make this work cleanly.
If you send the "quit" or "stop" command with the client API, it will
now attempt to kill network I/O immediately (same as normal input in the
previous commits).
Refine the ugly hack from the previous commit, and let the "quit"
command and some others abort playback immediately. For
playlist_next/playlist_prev, still use the old hack, because we can't
know if they would stop playback or not.
This mechanism originates from MPlayer's way of dealing with blocking
network, but it's still useful. On opening and closing, mpv waits for
network synchronously, and also some obscure commands and use-cases can
lead to such blocking. In these situations, the stream is asynchronously
forced to stop by "interrupting" it.
The old design interrupting I/O was a bit broken: polling with a
callback, instead of actively interrupting it. Change the direction of
this. There is no callback anymore, and the player calls
mp_cancel_trigger() to force the stream to return.
libavformat (via stream_lavf.c) has the old broken design, and fixing it
would require fixing libavformat, which won't happen so quickly. So we
have to keep that part. But everything above the stream layer is
prepared for a better design, and more sophisticated methods than
mp_cancel_test() could be easily introduced.
There's still one problem: commands are still run in the central
playback loop, which we assume can block on I/O in the worst case.
That's not a problem yet, because we simply mark some commands as being
able to stop playback of the current file ("quit" etc.), so input.c
could abort playback as soon as such a command is queued. But there are
also commands abort playback only conditionally, and the logic for that
is in the playback core and thus "unreachable". For example,
"playlist_next" aborts playback only if there's a next file. We don't
want it to always abort playback.
As a quite ugly hack, abort playback only if at least 2 abort commands
are queued - this pretty much happens only if the core is frozen and
doesn't react to input.
This means they get special handling for asynchronously aborting
playback, even if the player is "stuck".
Also document "stop". It seems somewhat useful for client API users
(although that will be implemented properly only in the following
commits.)
The memcpy() is actually not enough: the types are incompatible, and no
memcpy, union, etc. will change that. (Although no real compiler will
ever break this.) Attempt to make this theoretically correct by actually
using a struct pointer. It's not the same struct type, but supposedly
it's ok, because all struct pointers always have the same size and
representation in standard C.
Idle mode went to sleep too early, e.g. just pressing "ESC" did nothing,
until the next event happened. This was because it directly went to
sleep after processing commands. What we should do instead is rechecking
all state after processing commands, redraw OSD, and then go to sleep.
This also fixes some strange OSD-related behavior.
Also move some other code around to separate idle mode initialization
from the normal run loop.
Not sure why this was originally added as autorepeated. It makes no
sense, because switching between choices should never autorepeat. (For
the normal "add"/"cycle" commands, autorepeat is usually enabled, but
command.c tries to disable it specifically for choice properties.)
Mismatching units in timeout calculation.
Also, as a near-cosmetic change, explicitly wake up the core on the
right time. Currently this does nothing, because the core is woken up
anyway - but it will matter with the next commit.
Replace select() usage with poll() (and reduce code duplication).
Also, while we're at it, drop --disable-audio-select, since it has the
wrong name anyway. And I have doubts that this is needed anywhere. If
it is, it should probably fallback to doing the right thing by default,
instead of requiring the user to do it manually. Since nobody has done
that yet, and since this configure option has been part of MPlayer ever
since ao_oss was added, it's probably safe to say it's not needed.
The '#ifdef SNDCTL_DSP_GETOSPACE' was pointless, since it's already used
unconditionally in another place.
1. Separate buffer and temporary file handling from the vo to make maintenance
and reading code easier
2. Skip resizing as much as possible if back buffer is still busy.
3. Detach and mark osd buffers for deletion if we want to redraw them and they
are still busy. This could be a possible case for the video buffers as
well. Maybe better than double buffering.
All the above steps made it possible to have resizing without any artifacts
even for subtitles. Also fixes dozen of bugs only I knew, like broken subtitles
for rgb565 buffers. I can now sleep at night again.
OSX is POSIX conformant, but it's a sad joke: it provides the
<semaphore.h> prototype as the standard demands, but they're empty
wrappers, and all functions just return ENOSYS.
Emulate them similar to how osdep/io.h emulate filesystem functions on
Windows. By including the header, working sem_* functions become
available.
To make it async-signal safe, use a pipe for wakeup (write() is AS-safe,
but mutexes can't be). Actually I'm not sure anymore if we really need
AS-safety, but for now the emulation can do it.
On Linux, the system provides a far more efficient and robust
implementation. We definitely want to avoid using the emulation if
possible, so this code is active on OSX only. For convenience we always
build the source file though, even if the implementation is disabled and
no actual code is generated.
(Linux provides working semaphores, but is formally not POSIX
conformant. On OSX it's the opposite. Is POSIX a complete joke?)
I'm not quite sure what we should actually do (maybe read input
commands?), but interpreting input as terminal key sequences is
definitely weird. So just do nothing.
Do terminal input with a thread, instead of using the central select()
loop. This also changes some details how SIGTERM is handled.
Part of my crusade against mp_input_add_fd().
An attempt at fixing #1070. Apparently something goes wrong if the
video size is equal to the screen size. Since the window decorations
add to the window size, it must actually be larger than the screen.
Actually I don't know what exactly is going wrong, but since this
commit also slightly improves the behavior otherwise, it's a win
anyway.
Try to keep the window size strictly below screen size, even accounting
for window decorations. Size it down and center the window so that it
fits (by either touching the left/right or top/bottom screen borders).
I haven't found any information on what is the maximum allowed size and
position of a window so that it doesn't collide with the task bar, so
assume that we can use the entire screen, minus 1 pixel to avoid
triggering fullscreen semantics (if that is even possible).
The purpose is making accessing the current playlist entry saner when
commands are executed during initialization, termination, or after
playlist navigation commands.
For example, the "playlist_remove current" command will invalidate
playlist->current - but some things still access the playlist entry even
on uninit. Until now, checking stop_play implicitly took care of it, so
it worked, but it was still messy.
Introduce the mpctx->playing field, which points to the current playlist
entry, even if the entry was removed and/or the playlist's current entry
was moved (e.g. due to playlist navigation).
To handle legacy commands, string replacement is used; the modified
string is returned by parse_cmd_str(), but it also frees all temporary
memory, which includes the replaced string.
Closes#1075.
Continues commit 348dfd93. Replace other places where input was manually
fetched with common code.
demux_was_interrupted() was a weird function; I'm not entirely sure
about its original purpose, but now we can just replace it with simpler
code as well. One difference is that we always look at the command
queue, rather than just when cache initialization failed. Also, instead
of discarding all but quit/playlist commands (aka abort command), run
all commands. This could possibly lead to unwanted side-effects, like
just ignoring commands that have no effect (consider pressing 'f' for
fullscreen right on start: since the window is not created yet, it would
get discarded). But playlist navigation still works as intended, and
some if not all these problems already existed before that in some
forms, so it should be ok.
reinit_window_state() will set VO_EVENT_RESIZE when it runs, so we
don't need to set it manually depending on the VOCTRL.
Probably avoids duplicated resize events. I don't expect this actually
fixes anything, but might help spotting other bugs easier (if there
are any).
This makes the player wait until each script is loaded. Do this to give
the script a chance to setup all its event handlers. It might also be
useful to allow a script to change options that matter for playback.
While waiting for a script to be loaded, the player actually accepts
input. This is needed because the scripts can execute player commands
anyway while they are being "loaded". The player won't react to most
commands though: it can't quit or navigate the playlist in this state.
For deciding whether a script is finally loaded, we use a cheap hack: if
mpv_wait_event() is called, it's considered loaded. Let's hope this is
good enough. I think it's better than introducing explicit API for this.
Although I'm sure this will turn out as too simplistic some time in the
future, the same would probably happen with a more explicit API.
Expose the central event handling functions explicitly, so that other
parts of the player can use them.
No functional changes. Preparation for the next commit.
This was kept in the codebase because it is slightly faster than --vo=opengl
on really old Intel cards (from the GMA era). Time to kill it, and let it rest.
Fixes#1061
With e.g --start=-3 --audio-buffer=10 the decoder entered EOF state
before the initial sync was finished, entered STATUS_EOF, and just
started playing audio from a random position.
This doesn't handle seeking outside of the file, which is a different
case. E.g. --start=30:00 with audio and video enabled in a file shorter
than 30:00 will play a random last part of audio. This could perhaps be
fixed by using the hr-seek target for cutting audio, instead of the
video PTS, but that would be kind of intrusive, so don't do it for now.
The simpler solution, assuming audio EOF on video EOF, wouldn't work,
because we allow audio to start before video, or to last after video.
Improve the logic how the audio thread decides how to wait until the AO
is ready for new data. The previous commit makes some of this easier,
although it turned out that it wasn't required, and we still can handle
AOs with bad get_space implementation (although the new code prints an
error message, and it might fail in obscure situations).
The new code is pretty similar to the old one, and the main thing that
changes is that complicated conditions are tweaked. AO waiting is now
used better (mainly instead of max>0, r>0 is used). Whether to wakeup
is reevaluated every time, instead of somehow doing the wrong thing
and compensating for it with a flag.
This fixes the specific situation when the device buffer is full, and
we don't want to buffer more data. In the old code, this wasn't handled
correctly: the AO went to sleep forever, because it prevented proper
wakeup by the AO driver, and as consequence never asked the core for new
data. Commit 4fa3ffeb was a hack-fix against this, and now that we have
a proper solution, this hack is removed as well.
Also make the refill threshold consistent and always use 1/4 of the
buffer. (The threshold is used for situations when an AO doesn't
support proper waiting or chunked processing.)
This commit will probably cause a bunch of regressions again.
Round get_space() results in the same way play() rounds the input size.
Some audio APIs do this for various reasons.
This affects only "push" based AOs. Some of these need no change,
because they either do it already right (like ao_openal), or they seem
not to have any such requirements (like ao_pulse).
Needed for the following commit.
"#" starts a comment, so the # key needs to be handled specially.
MPlayer has the same issue, but its input.conf is wrong. Or at least I
think it's wrong; looking at the MPlayer code it's doubtful they somehow
special-case and handle this.
Remove the unnecessary indirection through ao fields.
Also fix the inverted result of AOCONTROL_HAS_TEMP_VOLUME. Hopefully the
change is equivalent. But actually, it looks like the old code did it
wrong.
This is a deadlock caused by a lock order issue: sub/osd.c locks the OSD
first, then the subtitle decoder lock. player/sub.c does the reverse.
Fix this by discussing away the requirement for locking (see below),
which allows us to drop the broken sub lock. sub_get_text() still
acquires and releases the sub decoder lock, but it's not held at the
same time as the OSD lock anymore, so it should be fine.
Originally, the sub lock was acquired because sub_get_text() returns a
pointer to a mutable string. We simply declare that it's ok to call it
unlocked, as long as only 1 thread accesses it, which works out fine in
this case.
With --gapless-audio=no, changing from one file to the next apparently
made it hang, until the player was woken up by unrelated events like
input. The reason was that the AO doesn't notify the player of EOF
properly. the played was querying ao_eof_reached(), and then just went
to sleep, without anything waking it up.
Make it event-based: the AO wakes up the playloop if the EOF state
changes.
We could have fixed this in a simpler way by synchronously draining the
AO in these cases. But I think proper event handling is preferable.
Fixes: #1069
CC: @mpv-player/stable (perhaps)
It seems hrtf works in 48khz only - and if that wasn't the input, the
filter just exited with an error. Make it request the 48khz instead. The
player will insert a resampling filter.
Not sure why it wasn't done like this in the first place.
The Windows version of tmpfile is actually pretty broken. It tries to
create the file in the root directory of the current drive, which means
on Vista and up, it normally fails due to insufficient permissions.
Replace it with a version that uses GetTempPath.
Also remove the Windows-specific note about automatic deletion of the
cache file. FILE_FLAG_DELETE_ON_CLOSE is available in NT, and it should
be pretty reliable.
Somehow, there was a larger misunderstanding in the code: ao_buffer
does not need to be preserved over audio reinit for proper support of
gapless audio. The actual AO internal buffer takes care of this.
In fact, preserving ao_buffer just breaks audio resync. In the ordered
chapter case, end_pts is used, which means not all audio data in the
buffer is played, thus some data is left over when audio decoding
resumes on the next segment. This triggers some code that aborts resync
if there's "audio decoded" (ao_buffer contains something), but no PTS
is known (nothing was actually decoded yet).
Simplify, and always bind the output buffer to the decoder.
CC: @mpv-player/stable (maybe)
The oldest supported FFmpeg release doesn't provide
av_vdpau_alloc_context(). With these versions, the application has no
other choice than to hard code the size of AVVDPAUContext. (On the other
hand, there's av_alloc_vdpaucontext(), which does the same thing, but is
FFmpeg specific - not sure if it was available early enough, so I'm not
touching it.)
Newer FFmpeg and Libav releases require you to call this function, for
ABI compatibility reasons. It's the typcal lakc of foresight that make
FFmpeg APIs terrible. mpv successfully pretended that this crap didn't
exist (ABI compat. is near impossible to reach anyway) - but it appears
newer developments in Libav change the function from initializing the
struct with all-zeros to something else, and mpv vdpau decoding would
stop working as soon as this new work is relewased.
So, add a configure test (sigh).
CC: @mpv-player/stable
This wraps waf's find_program in our own check boilerplate code so that it
can be used in the declarative dependencies section of the wscript.
Can be used like this:
}, {
'name': 'sed',
'desc': 'sed program',
'func': check_program('sed', 'SED'),
}, {
First argument is the program name, and the second is the waf variable name
where the program path will be stored. In this example we will be able to
refer to sed with ${{SED}} when creating waf Tasks in wscript_build.
/cc @giselher: I think you need this for wayland-scanner.
Don't worry, your ~/.config/... paths are safe. This merely removes
handling of $XDG_CONFIG_DIRS for global paths.
Maybe there is a better solution for this, like still including the
"traditional" config dir. But I will leave the fine reading of this
(crappy) spec and fixing the code accordingly to someone else. So, if
anyone has interest in getting this behavior back, you will have to
write a patch. This patch should _also_ not break expected behavior.
Fixes#1060.
The last demuxed file position (demuxer->filepos) is used to estimate
the total playback percentage in files with possible timestamp resets
(like MPEG-PS). Until know, reading from any stream set this position
freely. This makes the position jump around.
Fix this by allowing icnreasing file position only. Reset it on seeking.
With crazy formats, this still could go wrong, but there's only so much
you can do.
HLS streams as demuxed by libavformat have no track title metadata. So
show the HLS bitrate if no title is set. Could be useless or annoying,
so it's a bit controversial, I guess.
--hls-bitrate=min/max lets you select the min or max bitrate. That's it.
Something more sophisticated might be possible, but is probably not even
worth the effort.
This catches a few cases which basically call:
m_property_strdup_ro(..., ..., NULL)
which would return NULL strings. This should generally be avoided
(although it's allowed due to reasons), and it seems most callers
actually intend this to mean M_PROPERTY_UNAVAILABLE.
Because that might be a bad idea.
Note that remote playlists still can use any protocol marked with
is_safe and is_network, because the case of http-hosted playlists
containing URLs using other streaming protocols is not unusual.
Loading a playlist with --playlist from a sub-directory added the
playlist's base path twice: one time in the playlist demuxer, and then
again in playlist_parse_file(). The latter function is used only for
--playlist, so it worked when loading the playlist directly.
(This is probably a mess-up when the MPlayer playlist parsers were
replaced with newer code.)
CC: @mpv-player/stable
The event was copied early, and wasn't released if it was rejected
instead of being added to the event queue. Fix by copying the event at a
point when it's certainly added to the event queue.
The dup_event_data() function is merely moved.
Until now, you had to use --load-unsafe-playlists or --playlist to get
playlists loaded. Change this and always load playlists by default.
This still attempts to reject unsafe URLs. For example, trying to invoke
libavdevice pseudo-demuxer is explicitly prevented. Local paths and any
http links (and some more) are always allowed.
Probably no observable effect, but it's more correct. Setting audio to
EOF could have bad effects otherwise (anywhere the player logic for
example decides whether EOF was reached, and such).
The audio/video sync code in player/audio.c calls ao_reset() each time
audio decoding is entered, but the player is paused, and there would be
more than 1 sample to skip to make audio start match with video start.
This caused a wakeup feedback loop with push.c.
CC: @mpv-player/stable
When embedding a X window, it's hard to control whether it receives
mouse/keyboard input or not. It seems the X protocol itself makes this
hard (basically due to the outdated design mismatching with modern
toolkits), and we have to take care of these things explicitly.
Simply do this by manually querying and using the parent window event
flags.
This restores some MPlayer behavior (it doesn't add back exactly the
same code, but it's very similar).
This probably has some potential to interfere with libmpv embedding, so
bump the client API minor.
CC: @mpv-player/stable (if applied, client-api-changes.rst has to be
adjusted to include the 0.5.2 release)
In particular, use the note markup. The issue about rounded timestamps
is mostly with respect to Matroska (which usually rounds them to
milliseconds), which somewhat adds to the reliability issue.
This inserts an automatic conversion filter if a Matroska file is marked
as 3D (StereoMode element). The basic idea is similar to video rotation
and colorspace handling: the 3D mode is added as a property to the video
params. Depending on this property, a video filter can be inserted.
As of this commit, extending mp_image_params is actually completely
unnecessary - but the idea is that it will make it easier to integrate
with VOs supporting stereo 3D mogrification. Although vo_opengl does
support some stereo rendering, it didn't support the mode my sample file
used, so I'll leave that part for later.
Not that most mappings from Matroska mode to vf_stereo3d mode are
probably wrong, and some are missing.
Assuming that Matroska modes, and vf_stereo3d in modes, and out modes
are all the same might be an oversimplification - we'll see.
See issue #1045.
Since we have to be portable, our options for creating temporary files
are somewhat limited. tmpfile() happens to be available everywhere, so
use that. This function doesn't allow having a "visible" filename or
location, so we use the magic string "TMP" for this.
A (hopefully) temporary hack to make stream switching delays tolerable.
It's not clear how this should be handled (either executing a precise
seek on track switching, or always enabling all streams), so get this
issue out of the way for now by picking a rather low value.
In the else branch pict_type is always 3, so pict_type != 3 is always
false. (Note that I have no idea of what it was supposed to do and it is
just an equivalent of the old behaviour.)
The player didn't quit when seeking past EOF in audio-only mode while
paused. The only case when we don't want to quit is when the last video
frame is displayed while paused.
This logic was probably broken a while ago, but I'm not exactly sure.
CC: @mpv-player/stable
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.
The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
Just so I can move this file without modifying its contents in the next
commit.
compat/compiler.h is to be moved to osdep/ with the next commit, so add
a dummy header.
Don't reconnect to the cache (since the cached stream already handles
reconnection). This is necessary, because since commit 0b428e44 the
"streaming" field (which also controls whether attempting to reconnect
makes sense at all) is inherited to the cache stream wrapper.
Also, let the stream reset its own position on reconnect. This removes
some assumptions and messy handling from the reconnect function.
Make sure the cache is dropped on reconnect. This takes care of
readjusting the stream position if necessary. (Also drop the cache on
DVB channel switching commands.)
Add a mechanism to the client API code, which allows the player core to
query whether a client API event is needed at all. Use it for the cache
update.
In this case, this is probably a pure microoptimization; but the
mechanism will be useful for other things too.
Don't attempt to resync after speed changes. Note that most other cases
of audio reinit (like switching tracks etc.) still resync, but other
code paths take care of setting the audio_status accordingly.
This restores the old behavior of not trying to fix audio desync, which
was probably changed with commit 261506e3.
Note that the code as of now wasn't even entirely correct, since the A/V
sync values are slightly shifted. The dsync depends on the audio buffer
size, so a larger buffer size will show more extreme desync. Also see
mplayer2 commit 213a224e, which should fixed this - it was not merged
into mpv, because it disabled audio for too long, resulting in a worse
user experience. This is similar to the issue this commit attempts to
fix.
Fixes: #1042 (probably)
CC: @mpv-player-stable
Regression since commit f14722a4. For some reason, this worked on
nvidia, but rightfully failed on mesa.
At least in C, the ## operator indeed needs two macro arguments, and
you can't just concatenate with non-arguments.
This change will most likely fix it.
CC: @bjin
Remove the hardcoded wait time of 2 seconds. Instead, adjust the wait
time each time we unpause: if downloading the data took longer than its
estimated playback time, increase the amount of data we wait for. If
it's shorter, decrease it.
The +/- is supposed to avoid oscillating between two values if the
elapsed time and the wait time are similar. It's not sure if this
actually helps with anything, but it can't harm.
Use the "native" underrun detection, instead of guessing by a low cache
duration. The new underrun detection (which was added with the original
commit) might have the problem that it's easy for the playloop to miss
the underrun event. The underrun is actually not stored as state, so if
the demuxer thread adds a new packet before the playloop happens to see
the state, it's as if it never happened. On the other hand, this means
that network was fast enough, so it should be just fine.
Also, should it happen that we don't know the cached range (the
ts_duration < 0 case), just wait until the demuxer goes idle (i.e.
read_packet() decides to stop). This pretty much should affect broken or
unusual files only, and there might be various things that could go
wrong. But it's more robust in the normal case: this situation also
happens when no packets have been read yet, and we don't want to
consider this as reason to resume playback.
The cache percentage was useless. It showed how much of the total stream
cache was in use, but since the cache size is something huge and
unrelated to the bitrate or network speed, the information content of
the percentage was rather low.
Replace this with printing the duration of the demuxer-cached data, and
the size of the stream cache in KB.
I'm not completely sure about the formatting; suggestions are welcome.
Note that it's not easy to know how much playback time the stream cache
covers, so it's always in bytes.
The "buffering" logic was active even if the stream cache was disabled.
This is contrary to what the manpage says. It also breaks playback
because of another bug: the demuxer cache is smaller than 2 seconds,
and thus the resume condition never becomes true.
Explicitly run this code only if the stream cache is enabled. Also, fix
the underlying problem of the breakage, and resume when the demuxer
thread stops reading in any case, not just on EOF.
Broken by previous commit. Unbreaks playback of local files.
Add the --cache-secs option, which literally overrides the value of
--demuxer-readahead-secs if the stream cache is active. The default
value is very high (10 seconds), which means it can act as network
cache.
Remove the old behavior of trying to pause once the byte cache runs
low. Instead, do something similar wit the demuxer cache. The nice
thing is that we can guess how many seconds of video it has cached,
and we can make better decisions. But for now, apply a relatively
naive heuristic: if the cache is below 0.5 secs, pause, and wait
until at least 2 secs are available.
Note that due to timestamp reordering, the estimated cached duration
of video might be inaccurate, depending on the file format. If the
file format has DTS, it's easy, otherwise the duration will seemingly
jump back and forth.
The purpose of the unconditional pthread_cond_signal() when reading
cached DEMUXER_CTRLs and STREAM_CTRLs was apparently to update the
stream cache state. Otherwise, the cached fields would never be updated
when the stream is e.g. paused.
The same could be said about other CTRLs, but these aren't as important,
since they are normally updated while reading packet data.
In order to reduce wakeups, make this logic explicit.
Add a new parameter 'p' to gaussian filter. The new formula used
a different base taken from fmtconv plugin, so that the
new parameter is exactly same as the one used in Avisynth and
Vapoursynth.
The new default value is 2 / log(2) * 10, with the default value it
conforms to the original kernel taken from vector-agg.
Add two new options, make it possible for user to set the radius
for some of the filters with no fixed radius.
Also add three new filters with the new radius parameter supported.
When video format changes, the frame before the frame with the new
format sets video_status briefly to STATUS_DRAINING. This caused the
code to handle the EOF case to kick in, which just pauses the player
when trying to step past the last frame. As a result, trying to
framestep over format changes resulted in pausing the player.
Fix by testing against the correct status.
"Shift+X" didn't actually map any key, as opposed to "Shift+x". This is
because shift usually changes the case of a character, so a plain
printable character like "X" simply can never be combined with shift.
But this is not very intuitive. Always remove the shift code from
printable characters. Also, for ASCII, actually apply the case mapping
to uppercase characters if combined with shift. Doing this for unicode
in general would be nice, but that would require lookup tables. In
general, we don't know anyway what character a key produces when
combined with shift - it could be anything, and depends on the keyboard
layout.
The little lua snippet at #488 as well as the actual implementation
seems to indicate that not expanding properties is indeed the correct
behavior. Document that.
Signed-off-by: wm4 <wm4@nowhere>
This shouldn't change anything functionally.
Change the A/V desync message. --framedrop is enabled by default now, so
the text must be changed a little. I've never heard of audio outputs
messing up A/V sync recently, so remove that part.
Remove the unused ao_pts field.
Reorder 2 A/V sync related expressions so that they look the same.
Abandon the "old" infrastructure for --input-file (mp_input_add_fd(),
select() loop, non-blocking reads). Replace it with something that
starts a reader thread, using blocking input.
This is for the sake of Windows. Windows is a truly insane operating
system, and there's not even a way to read a pipe in a non-blocking
way, or to wait for new input in an interruptible way (like with
poll()). And unfortunately, some want to use pipe to send input to
mpv. There are probably (slightly) better IPC mechanisms available
on Windows, but for the sake of platform uniformity, make this work
again for now.
On Vista+, CancelIoEx() could probably be used. But there's no way on
XP. Also, that function doesn't work on wine, making development
harder. We could forcibly terminate the thread, which might work, but
is unsafe. So what we do is starting a thread, and if we don't want
the pipe input anymore, we just abandon the thread. The thread might
remain blocked forever, but if we exit the process, the kernel will
forcibly kill it. On Unix, just use poll() to handle this.
Unfortunately the code is pretty crappy, but it's ok, because it's late
and I wanted to stop working on this an hour ago.
Tested on wine; might not work on a real Windows.
For --input-test, print messages on terminal by default.
Raise message level for enabling input sections, because the OSC makes
this very extremely annoying.
This is a simplification, because it lets us use the AVPacket
functions, instead of handling the details manually.
It also allows the libavcodec rawvideo decoder to use reference
counting, so it doesn't have to memcpy() the full image data. The change
in av_common.c enables this.
This change is somewhat risky, because we rely on the following AVPacket
implementation details and assumptions:
- av_packet_ref() doesn't access the input padding, and just copies the
data. By the API, AVPacket is always padded, and we violate this. The
lavc implementation would have to go out of its way to make this a
real problem, though.
- We hope that the way we make the AVPacket refcountable in av_common.c
is actually supported API-usage. It's hard to tell whether it is.
Of course we still use our own "old" demux_packet struct, just so that
libav* API usage is somewhat isolated.
If a packet is appended to a stream, and there were already packets
queued, nothing about the state changed, as far as the user (i.e. the
player) is concerned. Thus no wakeup is needed.
The pthread_cond_signal() call following this is not interesting - it
will simply be a NOP if there are actually no waiters.
It makes no sense to set the packet duration, because libavcodec doesn't
know the timebase. And in fact, no subtitle decoder accesses the packet
duration, except text subtitle converters, which are not relevant here.
So this code did nothing - drop it.
Also fix a blatantly incorrect comment.
If duration<0, it means the duration is unknown. Disable framedropping,
because end_time makes no sense in this case.
Also, strictly never drop the first frame.
This fixes weird behavior with the cover-art case (for the 100th time).
Commit 846257da introduced an accidental feature: if you kept seeking
(so playback never really resumes), the audio would never be played.
This was nice, but commit 4c25b000 accidentally removed it again (due
to the video_next_pts being earlier available than it used to be, so
audio could be played before the player executed the next queued seek).
Implicitly reintroduce the old behavior again by not decoding a second
video frame immediately. Usually, the second frame is used to compute
the frame duration needed to for accurate framedropping, but since the
first frame after a seek is never dropped, we don't need this.
Now the video code will queue the new frame to the VO immediately, and
since fill_audio_out_buffers() is called in the playloop before
write_video() and execute_queued_seek(), it never gets the chance to
enter STATUS_READY, and seeks will be silent.
This also has a nice side-effect: since the second frame is not decoded
and filtered, seeking becomes slightly faster (back to the same level
as with framedrop disabled).
It seems this still sometimes plays a period of audio when keeping a
seek key down. In my tests, this appeared to happen because the seek
finished before the next key repeat was sent.
In theory, timestamps can be negative, so we shouldn't just return -1
as special value.
Remove the separate code for clearing decode buffers; use the same code
that is used for normal seek reset.
Commit 5afc025c broke this. The reason is that mpctx->delay is updated
when a new video frame is added. This value is also needed to resync
audio, but it will be for the wrong PTS. They must be consistent with
each other, and if they aren't, initial sync will be off by N video
frames, which results at least in worse user experience.
This can be reproduced by for example heavily switching between normal
and 2x speed, or similar.
Fix by readding the video_next_pts field (keeping its use minimal,
instead of reverting the commit that removed it).
If video reaches EOF, and audio is also EOF (or is otherwise not
meaningful, like audio disabled), then the playback position was briefly
set to 0. Fix this by not trying to use a bogus audio PTS.
CC: @mpv-player/stable (maybe)
This simplifies the code, and fixes an odd bug: the second-last frame
was displayed for a very short duration if framedrop was enabled. The
reason was that basically the time difference between second-last and
last frame were skipped, because at this point EOF was already
signaled. Also see commit b0959488 for a similar issue in the
same code.
This removes the messiness of the next_frame 2-frame queue, and
strictly runs the "new frame" code when a frame is moved to the first
position of the queue, instead of somehow messing with return codes.
This also merges update_video() into video_output_image().
No functional changes. init_vo() is now needed a bit further down, and
moving it keeps definition and use close. adjust_sync() will be used by
a function further up in one of the following commits.
Add two new script environment variables 'video_in_dw' and
'video_in_dh', representing the display resolution of video. Along
with video resolution, sample ratio aspect can be calculated in
scripts.
Currently it's impossible to change sample ratio aspect with single
vapoursynth filter since '_SARNum' and '_SARDen' frame properties
from output clip will be ignored. A following 'dsize' filter is
necessary for this purpose.
This is independent of terminfo/termcap, and supports more keys.
Originally, the goal was just extending the set of supported key
sequences, but since the terminfo stuff actually makes this much harder,
and since it's a big blob of bloated legacy crap, just drop it. Instead,
use hardcoded tables.
It's pretty easy to get on the same level as the old code (with fewer
LOC), and we avoid additional error situations, such as mallocs which
could fail (the old code just ignores malloc failures). We also try to
support some xterm escape sequences, which are in relatively widespread
use. (I'm not sure about the urxvt ones.)
Trying to deal with xterm shift/ctrl/alt modifiers is probably a bit
overcomplicated, and only deals with prefixes - xterm randomly uses
prefix sequences for some keys, and suffixes for others (what the heck).
Additionally, try to drop unknown escape codes. This basically relies
on a trick: in almost 100% of all situations, a read() call will
actually return complete sequences (possibly because of pipe semantics
and atomic writes from the terminal emulator?), so it's easy to drop
unknown sequences. This prevents that they trigger random key bindings
as the code interprets the part after ESC as normal keys.
This also drops the use of terminfo for sending smkx/rmkx. It seems
even vt100 (to which virtually everything non-legacy is reasonably
compatible with) supports the codes we hardcode, so it should be fine.
This commit actually changes only the code if terminfo/termcap are not
found. The next commit will make this code default.
Surprisingly, WaitFor* works on console handles. We can simply run the
code for reading the console in a thread, and don't have to worry about
crazy win32 crap in the rest of the player's input code anymore.
This also fixes the issue that you couldn't unpause the player from the
terminal, because the player would stop polling for input.
We already redirect all terminal output through our own wrappers (for
the sake of UTF-8), so we might as well use it to handle ANSI escape
codes.
This also changes behavior on UNIX: we don't retrieve some escape codes
per terminfo anymore, and just hardcode them. Every terminal should
understand them.
The advantage is that we can pretend to have a real terminal in the
normal player code, and Windows atrocities are locked away in glue
code.
The original intention was probably to avoid unnecessarily high numbers
of wakeups. Change it to wait at most 25% of buffer time instead of 75%
until refilling. Might help with the dsound problems in issue #1024, but
I don't know if success is guaranteed.
So talking to a certain Intel dev, it sounded like modern VA-API drivers
are reasonable thread-safe. But apparently that is not the case. Not at
all. So add approximate locking around all vaapi API calls.
The problem appeared once we moved decoding and display to different
threads. That means the "vaapi-copy" mode was unaffected, but decoding
with vo_vaapi or vo_opengl lead to random crashes.
Untested on real Intel hardware. With the vdpau emulation, it seems to
work fine - but actually it worked fine even before this commit, because
vdpau was written and designed not by morons, but competent people
(vdpau is guaranteed to be fully thread-safe).
There is some probability that this commit doesn't fix things entirely.
One problem is that locking might not be complete. For one, libavcodec
_also_ accesses vaapi, so we have to rely on our own guesses how and
when lavc uses vaapi (since we disable multithreading when doing hw
decoding, our guess should be relatively good, but it's still a lavc
implementation detail). One other reason that this commit might not
help is Intel's amazing potential to fuckup anything that is good and
holy.
This could be used by VO implementations to report a recent vsync time
to the generic VO code, which in turn will use it and the display FPS
to estimate at which point in time the next vsync will happen.
This uses glXGetVideoSyncSGI() to check how many vsyncs happened since
the last flip_page() call. It allows checking a pattern of vsync
increments of at most 2 elements. For example, to check ~24 fps playback
on a ~60 Hz monitor, this can be used:
--vo=opengl:check-pattern=[3-2]:waitvsync
Whether the reported results are accurate or just plain wrong may depend
on the driver and if the waitvsync sub-option is used. There are no
guarantees.
This option is undocumented, and may be removed again in the near or
distant future.
For debugging (drawing fun plots with TOOLS/stats-conv.py).
Also move last_flip under the correct comment: it's not protected by the
lock, and can be accessed by the VO thread only.
Playing with high framedrop could make it run out of surfaces. In
theory, we wouldn't need an additional surface, if we could just clear
the vo_vaapi internal surface - but doing so would probably be a pain,
so I don't care.
Only reports the most recently entered output if the window is displayed on
2 or more outputs. Should be changed to the lowest fps of all outputs the
window is visible. Until no one complains this will have to wait.
Look for the VO framedropping for more information on this topic.
After a new file is loaded, playback never starts instantly. Rather, it
takes some playloop iterations until initial audio and video have been
decoded, and the outputs have been (lazily) initialized. This means you
will get status line updates between the messages that inform of the
initialized outputs. This is a bit annoying and clutters the terminal
output needlessly.
Fix this by never printing the status line before playback isn't fully
initialized. Do this by reusing the --term-playing-msg code (which
prints a message once playback is initialized). This also makes sure the
status line _is_ shown during playback restart when doing seeks.
It's possible that the change will make the output more confusing if for
some reason is stuck forever initializing playback, but that seems like
an obscure corner case that never happens, so forget about it.
print_status() is called at a later point anyway (and before sleeping),
so this code has little effect. This code was added in commit a4f7a3df5,
and I can't observe any problems with idle mode anymore.
Now print_status() is called from a single place only, within osd.c.
If the Xrandr configuration changes, re-read it. So if you change
display modes or screen configuration, it will update the framedrop
refresh rate accordingly.
This passes the rootwin to XRRSelectInput(), which may or may not be
allowed. But it works, and the documentation (which is worse than used
toilet paper, great job Xorg) doesn't forbid it, or in fact say anything
about what the window parameter is even used for.
Nvidia's vdpau implementation is pretty good, but other factors make it
much less attractive for use as default VO. For example, Mesa often has
low quality drivers (mess up things with the presentation queue and the
vdpau API time source). Intel ruins things completely, and we're likely
to run on emulation via OpenGL. Compositing has unknown effects (to me
anyway), but appears to reduce the vdpau advantages.
One important reason to prefer vo_vdpau was that it could do proper
framedropping. Framedropping got fixed for the other VOs, so this reason
is going away.
This works only on X11, and only if the refresh rate changes due to the
window being moved to another screen (detected by us). It doesn't
include system screen reconfiguration yet.
This calls VOCTRL_GET_DISPLAY_FPS on every frame, which makes me uneasy.
It means extra thread communication with the win32 and Cocoa backends.
On the other hand, a frame doesn't happen _that_ often, and the
communication should still be pretty cheap and fast, so it's probably
ok.
Also needs some extra fuzz for vo_vdpau.c, because that does everything
differently.
This is always included in the Xorg development headers. Strictly
speaking it's not necessarily available with other X implementations,
but these are hopefully all dead.
Drop use of the ancient XF86VM, and use the slightly less ancient Xrandr
extension to retrieve the refresh rate. Xrandr has the advantage that it
supports multiple monitors (at least the modern version of it).
For now, we don't attempt any dynamic reconfiguration. We don't request
and listen to Xrandr events, and we don't notify the VO code of changes
in the refresh rate. (The later works by assuming that X coordinates map
directly to Xrandr coordinates, which probably is wrong with compositing
window manager, at least if these use complicated transformations. But I
know of no API to handle this.)
It would be nice to drop use of the Xinerama extension too, but
unfortunately, at least one EWMH feature uses Xinerama screen numbers,
and I don't know how that maps to Xrandr outputs.
--demuxer-readahead-secs now controls how much the demuxer should
readahead by an amount of seconds. This is based on the raw packet
timestamps. It's not always very exact. For example, h264 in Matroska
does not store any linear timestamps (only PTS values which are going
to be reordered by the decoder), so this heuristic is usually off by
several hundred milliseconds.
The decision whether to readahead is basically OR-ed with the other
--demuxer-readahead-packets options. Change the manpage descriptions
to subtly convey these semantics.
Since the display FPS is currently detected on X11 only (and even there
it's known to be wrong on certain setups), it seems like a good idea to
make this user-configurable.
I'm not sure about the merit, though it does print nice numbers if debug
output is enabled.
Basically, this tries to achieve similar results as the glFinish()
business, but again it entirely depends on the drivers whether this
does anything meaningful, or whether it's actively harmful.
It seems that at least on nvidia systems with composting disabled, we
can get it to block deterministically on the actual vsync event, which
should improve framedropping.
This mostly uses the same idea as with vo_vdpau.c, but much simplified.
On X11, it tries to get the display framerate with XF86VM, and limits
the frequency of new video frames against it. Note that this is an old
extension, and is confirmed not to work correctly with multi-monitor
setups. But we're using it because it was already around (it is also
used by vo_vdpau).
This attempts to predict the next vsync event by using the time of the
last frame and the display FPS. Even if that goes completely wrong,
the results are still relatively good.
On other systems, or if the X11 code doesn't return a display FPS, a
framerate of 1000 is assumed. This is infinite for all practical
purposes, and means that only frames which are definitely too late are
dropped. This probably has worse results, but is still useful.
"--framedrop=yes" is basically replaced with "--framedrop=decoder". The
old framedropping mode is kept around, and should perhaps be improved.
Dropping on the decoder level is still useful if decoding itself is too
slow.
Apparently users prefer this behavior.
It was used for subtitles too, so move the code to calculate the video
offset into a separate function. Seeking also needs to be fixed.
Fixes#1018.
The OSD is marked as changed, but the core isn't notified and this
doesn't immediately wakeup. (Possibly the OSD code should wakeup the
core instead, but maybe that woudl be overkill.)
Observed when using "mp.use_suspend = false" in the OSC, and pausing,
and moving the mouse pointer out of the window. The last part of the
fade remained visible for longer than intended.
Code reorganized to make layouts exchangeable
alternative test layout can be tested with
layout=slimbox
in the OSC config
timers are now used to properly animate the fade out when the
player is paused
duplicate seeks are discarded again
sub_reset() was called on cycling subtitle tracks and on seeking. Since
we don't want that subtitles disppear on cycling, sd_lavc.c didn't clear
its internal subtitle queue on reset, which meant that seeking with PGS
subtitles could leave the subtitle on screen (PGS subtitles usually
don't have a duration set).
Call it only on seeking, so we can also strictly clear the subtitle
queue in sd_lavc.
(This still can go very wrong if you disable a subtitle, seek, and
enable it again - for example, if used with libavformat that uses "SSA"
style demuxed ASS subtitle packets. That shouldn't happen with newer
libavformat versions, and the user can "correct" it anyway by executing
a seek while the subtitle is selected.)
Until recently, vo_opengl could be accessed from a single thread only,
due to the OpenGL API context being thread-specific. This issue doesn't
exist anymore, because VOs run on their own thread. This means we can
simply lock/unlock the playloop instead of doing something complicated
to get the playloop thread to execute our code.
It's not true anymore that the core will stop replying for 50ms
(waiting for video) without calling this function. Simplify the
documentation accordingly. Accessing properties that go through
the VO still have this problem, though.
The previous commit made the completion script always return non-zero, even when
a match is found. This explicitly sets the return value to zero whenever a match
is found but defaults to non-zero in case nothing is matched.
Returning a non-zero value signals to the zsh completion system that no matches
were added by the script so that it can try the user-defined matchers (e.g.
those defined with matcher-list).
Fixes#1008.
This ran adjust_sync() on every playloop iteration, instead of every
newly decoded frame. It seems this was idempotent in the common case,
but the code was originally designed to be run once only, so restore
that.
Otherwise vdp_video_mixer_destroy() would later fail when called on an invalid
video mixer handle. With mesa r600 vdpau driver, this would cause a segfault.
Since the 'syms' tool is shipped in waf's extras, when using system waf the
default tool overrides our own. Force our syms tool by providing the tooldir.
Fixes#1006
Xlib is not thread-safe. Or actually it is, but it's an incomprehensible
hack that was added later, and which needs to be acitvated manually
(this makes no sense). And it appears that the vdpau accesses X from the
decoder thread if GLX interop is used (and not in any other situations -
this doesn't make too much sense either).
So, just call the magic function that enables Xlib thread-safety.
The previous commit broke these things, and fixing them is separate in
this commit in order to reduce the volume of changes.
Move the image queue from the VO to the playback core. The image queue
is a remnant of the old way how vdpau was implemented, and increasingly
became more and more an artifact. In the end, it did only one thing:
computing the duration of the current frame. This was done by taking the
PTS difference between the current and the future frame. We keep this,
but by moving it out of the VO, we don't have to special-case format
changes anymore. This simplifies the code a lot.
Since we need the queue to compute the duration only, a queue size
larger than 2 makes no sense, and we can hardcode that.
Also change how the last frame is handled. The last frame is a bit of a
problem, because video timing works by showing one frame after another,
which makes it a special case. Make the VO provide a function to notify
us when the frame is done, instead. The frame duration is used for that.
This is not perfect. For example, changing playback speed during the
last frame doesn't update the end time. Pausing will not stop the clock
that times the last frame. But I don't think this matters for such a
corner case.
The VO is run inside its own thread. It also does most of video timing.
The playloop hands the image data and a realtime timestamp to the VO,
and the VO does the rest.
In particular, this allows the playloop to do other things, instead of
blocking for video redraw. But if anything accesses the VO during video
timing, it will block.
This also fixes vo_sdl.c event handling; but that is only a side-effect,
since reimplementing the broken way would require more effort.
Also drop --softsleep. In theory, this option helps if the kernel's
sleeping mechanism is too inaccurate for video timing. In practice, I
haven't ever encountered a situation where it helps, and it just burns
CPU cycles. On the other hand it's probably actively harmful, because
it prevents the libavcodec decoder threads from doing real work.
Side note:
Originally, I intended that multiple frames can be queued to the VO. But
this is not done, due to problems with OSD and other certain features.
OSD in particular is simply designed in a way that it can be neither
timed nor copied, so you do have to render it into the video frame
before you can draw the next frame. (Subtitles have no such restriction.
sd_lavc was even updated to fix this.) It seems the right solution to
queuing multiple VO frames is rendering on VO-backed framebuffers, like
vo_vdpau.c does. This requires VO driver support, and is out of scope
of this commit.
As consequence, the VO has a queue size of 1. The existing video queue
is just needed to compute frame duration, and will be moved out in the
next commit.
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Martin Herkt