The problem here is likely ao_alsa specific and has the same symptons as
what the previous commit fixed (audio not playing when the file changes
but the details are a bit different here and the sample rate does not
matter. When using gapless audio (the default), the core player
immediately marks the audio status as EOF after it starts draining and
allows the remaining audio buffers to play while it marches on. This
works fine. When not using gapless audio, it doesn't immediately set EOF
and instead waits for the audio to finish playing before it does
anything else. The problem is that ao_is_playing is always true so the
core waits forever thinking audio is still playing when it actually
isn't.
ao_play_data is what is in charge of setting the mp_pcm_state with it
calling out to the backend for additional help. Unfortunately, this
doesn't work for alsa because it's too dumb to signal the desired states
in this edge case so we have to help it a bit. The main thing to notice
here is that even though we can get EOF from a frame, there can still be
additional valid samples that compe after it. So we can't just
immediately quit after EOF is seen. The approach here is to simply save
if we got eof sometime in the past, wait until we get no more samples,
mark state.playing as false and then jump over to the eof code. This
will set p->playing to false as desired which allows the core code to
set EOF and finally we can go through the reset logic and actually play
audio for the next file.
There is a condition here that is supposed to wait until the ao is
finished playing before it goes through the rest of this function and
actually starts the reinit of the audio chain. This is not needed
however and actually causes a bug. uninit_audio_out already takes the
gapless audio case into account and drains the audio before preceding.
There is no reason to bail here. Additionally, returning early here
actually breaks ao_alsa. ao_alsa is pretty dumb and not able to cope
with changing samplerates. It explictly needs to stop the PCM and then
start it again for it to work. The old code here ensured the ao would
never reinit itself and then would wait forever trying to play samples
with no actual data. Other AOs (pipewire and pulse) seem to be able to
cope with this and reinit themselves in other ways (didn't look in
detail). This change makes no difference for them. Note that in many
cases, this appeared to work on ALSA because there is a high likelihood
that your default audio device will choose a 48 kHz sampling rate and
thus your entire playlist is resampled to 48 kHz which avoids the
bug.
Fixes#15477.
Previous fixes in bcc3d72 make subtitles on videos with non-square pixel
aspect ratios appear incorrect. This can happen on subtitles for SD
Blu-rays where the video resolution is 720x480 but the aspect ratio is
4:3. The scaling still applies when the video frame is cropped with
filters so the previous fix will still work.
Fixes#15415.
console is drawn below the OSC. This looks bad only with layout=box
because there is no user-data/osc/margins so it overlaps with it. Fix it
by using z -1 with box layout.
Follow up to 86d24b0 and 9c22d6b.
This is still used for caclulating a/v sync and delay remaining even
for initial samples, so we should always update it to the actual vsync
for those initial samples so we have something to work with at least.
And if we receive bogus values, also reset it to 0 along with
prev_vsync.
Not having base_vsync set to _some_ value completely breaks
vsync_skip_detection, and mpv stays stuck in a permanent mistimed state
where every frame is marked as delayed and never recovers from it.
Introduced a new `fadein` option to control the fade-in effect for the OSC.
The default value is `no`, which disables fade-in. This option allows users
to enable a fade-in effect when the OSC appears. Updated documentation
accordingly.
These are easier to read with _ between words. Also screenx/y could be
assumed to be display coordinates, so rename them to osd_w/h. They are
not named osd_width/osd_height to avoid Redefined local LSP warnings
with the ones in the upper scope.
Update flags are often updated, so make space for expand them. Reorder
all values to make it easier to add new ones, until there is no more
space.
Co-authored-by: Guido Cella <guido@guidocella.xyz>
With --ignore-path-in-watch-later-config the basename of files is used
for watch later files, but since 1d640c9887 this was checking the wrong
variable to determine if the file is a URL and thus also taking the
basename of URLs, when URLs should use the full path regardless of this
option.
This is already done in WndProc, but the flag was not set if we exited
event loop and were processing mpv dispatch messages in fallback loop.
This fixes the case when Window is destroyed externally or otherwise
event loop fails for whatever reason. Calling vo_w32_uninit does call
DestroyWindow() through dispatch, which in turn calls WndProc, so we
need to guard recursive call.
If a hook fails to invoke or a client has been removed during hook
processing, attempt to continue with the next hook if possible.
Before this commit, any failure during hook processing would cause the
hook processing to stop.
This issue can be reproduced when auto_profiles.lua registers an on_load
hook and exits afterward because no profiles are registered. In such a
state, the next hook would fail and not invoke.
There are many scenarios where a hook client may be broken,
unresponsive, or disabled, so it makes sense to continue processing
subsequent hooks.
The regression we observed was caused by e2284fba1 and 5dc404741, which
made auto_profiles.lua register hooks and then exit, leaving stale hooks
registered. However, the root cause is not limited to the
auto_profiles.lua case.
Fixes: #15465
* Early exit if there is no useful data in the AVReplayGain
structure (FFmpeg does check that gain of at least one thing is
not INT32_MIN, but leaves the peak unchecked so it can be zero).
* Less depth in the if structure.
This simplifies the code, not requiring a loop.
Moves the getter definition somewhat upwards to allow
for its usage in a function that was defined before it.
This allows to select DVD/Blu-Ray title easily. Titles are listed as
editions with their duration and number.
I wanted to include Angles also in this selection, but currently Angles
are not that well supported, so let's stick with titles and leave the
rest for the future changes.
We might migrate to lavf demuxer for DVD/Blu-Ray in the future, the mpv
implementation is rotted anyway.
Fixes#14586
This ignores --video-align-{x,y} when the video is smaller than the
window in the respective direction.
After zooming in, panning and zooming out, this is useful to recenter
the video in the window.
Unlike doing this by observing osd-dimensions in a script, this is done
before rerendering, so you don't see the image being rendered uncentered
for an instant after zooming out, before being rerendered centered.
Also update --video-align docs while at it.
This doesn't work well with --video-pan-{x,y} because you can move the
output rectangle far away from the image, and when zooming out pan is
abruptly reset to the center. It doesn't feel like natural like zooming
out after changing --video-align-{x,y}. So this commit doesn't set pan
to 0. Also this leaves a way to move scaled images within the window
even with --video-recenter.
The first AVPacket method has been deprecated in FFmpeg, and only
the HEVC decoder would read this information from there. The native
AV1 decoder as well as the libdav1d wrapper only support the newer
propagation method via codec context's side data. {ad,sd,vd}_lavc
call mp_set_avctx_codec_headers which copies the side data from
codecpar to codec context, so no changes are required on the
receiving end.
This process has been deprecated since avformat 60.15.100 and
since avcodec 60.30.101 AV_PKT_DATA_DOVI_CONF has been read from
AVCodecContext's coded_side_data. Additionally, the HEVC decoder
is the only one which currently reads this side data from packets,
the native AV1 decoder as well as the libdav1d wrapper instead only
support the newer propagation method via codec context's side data.
How this currently is supposed to propagate is:
* demux_lavf copies the codec parameters into sh_stream's lav_codecpar.
* {ad,sd,vd}_lavc call mp_set_avctx_codec_headers that calls
avcodec_parameters_to_context, which then sets global side data
to the codec context.
As this logic is already in place, so no additional changes are required
for things to work and the deprecated function call can just be
cleaned up.
Ref: FFmpeg/FFmpeg@5432d2aaca
Ref: FFmpeg/FFmpeg@804be7f9e3
Ref: FFmpeg/FFmpeg@12e5116872
This avoids inserting the characters in front of the cursor again when
completing to text already in front of the cursor.
This is important after 2f271a92de made Enter automatically insert the
first completion.
