ao-volume is represented in the code with a `struct ao_control_vol_t`
which contains volumes for two channels, left and right.
However the code implementing this property in command.c never treats
these values individually. They are always averaged together.
On the other hand the code in the AOs handling these values also has to
handle the case where *not* exactly two channels are handled.
So let's remove the `struct ao_control_vol_t` and replace it with a
simple float.
This makes the semantics clear to AO authors and allows us to drop some code from the AOs and command.c.
In debug mode the macro causes an assertion failure.
In release mode it works differently and tells the compiler that it can
assume the codepath will never execute. For this reason I was conversative
in replacing it, e.g. in mpv-internal code that exhausts all valid values
of an enum or when a condition is clear from directly preceding code.
[motivation]
Seeking on MacOS appears to be lagged when users connect
to wireless audio output (airpods for example).
This commit attempts to fixmpv-player/mpv#10270
[observation]
1. When using other media player (VLC to be exact) simultaneously,
the lagging on seek disappear. We could guess that the AudioDevice
is on some sort of "warm-up" state.
See mpv-player/mpv#9243 for detailed description.
2. `AudioOutputUnitStart` takes significant longer time after each seek
or pause/play when using wireless output devices compares to wired devices.
[rationale]
After investigate codes in ao_coreaudio.c, it appears that the the `stop`
function was used as `ao_driver.reset` function. Therefore every seek
and pause would call `AudioOutputUnitStop`.
It turns out that `ao_driver.reset` function is used in `ao_reset`.
And `ao_reset` function is used to clean up the state of current `ao`
so I think `AudioUnitReset` is more proper than `AudioOutputUnitStop`
under this semantics.
Since ao_coreaudio use pull base mechanism, audio playback behaviors
upon pause/seek could be handled by callback function
(streaming silence when paused) so there is no need to stop AudioUnit when resetting.
Therefore using `AudioUnitReset` as `ao_driver.reset` looks proper.
Additionally, after using proper reset, the AudioUnit that represents
hardware I/O devices doesn't need to be restart everytime seek/pause actions happen.
Restarting wireless devices simply takes longer in MacOS which is
the root cause of lagging observed by users when they seek or pause/play media.
[method]
Use `AudioUnitReset` for ao_driver.reset.
When a pull AO reaches reaches EOF then ao_read_data() will set
p->playing = false.
Because the ao is marked as not playing ao_set_pause(true) will not
reset the AO.
This keeps the output stream unintentionally open.
Fixes#9835
This allows us to more easily see the datapath from mpv to pipewire.
We know how often the callbacks are triggered, how big the buffers are
and how much data mpv provides to pipewire.
This allows the core of mpv to know about issues in the AO.
Otherwise playback will just freeze as no more data callbacks are sent
by PipeWire.
Also it allows mpv to try to reconnect the AO or find another, working
AO.
We want to add more logic to the stream event handler.
This logic should not be triggered during normal stream shutdown, so we
remove the listener beforehand.
The AO is feature-complete now.
As PipeWire also provides compatibility with PulseAudio, ALSA and Jack
we should put it before those for the autodetection to work.
The pure presence of PipeWire does not mean that it is actually driving
the audio session. For example it could only be meant for video.
Currently there is no proper API to detect this (see [0]), so we check
for the presence of audio sinks.
As soon as a proper API exists, we should use that.
[0] https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1835
Specifying the id of the target node during stream connect is
deprecated. Instead the property target.object should be used to link
by target serial or name. Using the name allows us to drop a bunch of
custom code.
Previously we would only call list_devs() on available AOs if an AO
*did not* have a hotplug_init() callback or for the first one that *did*
have it.
This is problematic when multiple fully functional hotplug-capable AOs
are available.
The second one would not be able to contribute discovered devices.
This problem prevents ao_pipewire from introducing full hotplug support
with hotplug_init().
When a platform has multiple valid AOs that can provide hotplug events
we should try to use the one that also provides playback.
Concretely this will help when introducing hotplug support for
ao_pipewire.
Currently ao_pulse is probed by ao_hotplug_get_device_list() before
ao_pipewire and on the common setups where both AOs could work pulse
will be selected for hotplug handling.
This means that hotplug_init() of ao_pipewire will never be called and
list_devs() has to do its own initialization.
But if ao_pulse is non-functional or not compiled-in suddenly
ao_pipewire *must* implement hotplug_init() for hotplugging events to
work for all.
Also if the hotplug ao_pulse connects to a PulseAudio instance that is
not emulated by the same PipeWire instance as the playback ao_pipewire
the hotplug events are useless.
uau did some investigation and noticed that we do not send a wakeup
event when we encounter end-of-stream in ao_read_data(), in contrast to
the equivalent logic for push AOs in ao_play_data().
Inserting that wakeup fixes the original problem of lack of
reinitialization on a format change without the problems we saw with
the previous attempted fix.
Fixes#10566
The error description in #10545 could indicate that we are overflowing
we are corrupting the buffer metadata ourselves through out-of-bound
writes.
This check is also present in pw-cat so it seems to be expected for
b->requested to exceed the actual available buffer space.
Potential fix for #10545
pw_core_disconnect frees the core, so accessing it afterward to
destroy the context is not allowed.
Instead, just destroy the context, the first thing it does is disconnect
all cores for us.
mpv only remembers volume for two channels.
Always apply the same volume to all channels in case of
non-stereo layout similarly to ao_pulse.
Don't try to do anything smart when averaging volumes,
normally they are equal anyway.
Pass channel volumes to `pw_stream_set_control` as array.
This is correct calling conventions and prevents
right channel muting every time ao-volume property is changed.
Terminate `pw_stream_set_control` calls with 0.
Changes:
* fixed hangups in the loop function and in some other cases
* refactoring according to @michaelforney's recommendations in #8314
* a few minor and/or cosmetic changes
* ability to build ao_sndio using meson
Changes:
- rewrite to use new internal MPV API;
- code refactoring;
- fix buffers size calculations;
- buffer set to auto;
- reset() - clean/reinit device only after errors;
The AO provides a way for mpv to directly submit audio to the PipeWire
audio server.
Doing this directly instead of going through the various compatibility
layers provided by PipeWire has the following advantages:
* It reduces complexity of going through the compatibility layers
* It allows a richer integration between mpv and PipeWire
(for example for metadata)
* Some users report issues with the compatibility layers that to not
occur with the native AO
For now the AO is ordered after all the other relevant AOs, so it will
most probably not be picked up by default.
This is for the following reasons:
* Currently it is not possible to detect if the PipeWire daemon that mpv
connects to is actually driving the system audio.
(https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1835)
* It gives the AO time to stabilize before it is used by everyone.
Based-on-patch-by: Oschowa <oschowa@web.de>
Based-on-patch-by: Andreas Kempf <aakempf@gmail.com>
Helped-by: Ivan <etircopyhdot@gmail.com>
Ever instance of m_obj_list is a constant and for all of them, the field
is true. Just remove the field all together.
Signed-off-by: Emil Velikov <emil.l.velikov@gmail.com>
This fixes a mismatch between configure working and build time
failing with Linux + OSSv4, enabling compilation on Debian based
Linux systems with the oss4-dev package.
Fixes#9378
Changes:
- code refactored;
- mixer options removed;
- new mpv sound API used;
- add sound devices detect (mpv --audio-device=help will show all available devices);
- only OSSv4 supported now;
Tested on FreeBSD 12.2 amd64.
This makes the behavior of all control messages consistent,
fixing an inconsistency that has been with us since
4d8266c739 - which is the initial
rework of the polyaudio AO into the pulseaudio AO.
Muting the stream also directly triggers an update to the OSD.
When not waiting for the command completion this read of the mute
property may read the old state. A stale read.
Note that this somehow was not triggered on native Pulseaudio, but it is
an issue on Pipewire.
See https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/868
Set pcm state to SND_PCM_STATE_XRUN in case -EPIPE is received,
and handle this state as per the usual logic.
This way snd_pcm_prepare gets called, and the loop continued.
Inspired by a patch posted by malc_ on #mpv.
--audio-stream-silence is a shitty feature compensating for awful
consumer garbage, that mutes PCM at first to check whether it's
compressed audio, using formats advocated and owned by malicious patent
troll companies (who spend more money on their lawyers than paying any
technicians), wrapped in a wasteful way to make it constant bitrate
using a standard whose text is not freely available, and only rude users
want it. This feature has been carelessly broken, because it's
complicated and stupid. What would Jesus do? If not getting an aneurysm,
or pushing over tables with expensive A/V receivers on top of them, he'd
probably fix the feature. So let's take inspiration from Jesus Christ
himself, and do something as dumb as wasting some of our limited
lifetime on this incredibly stupid fucking shit.
This is tricky, because state changes like end-of-audio are supposed to
be driven by the AO driver, while playing silence precludes this. But it
seems code paths for "untimed" AOs can be reused.
But there are still problems. For example, underruns will just happen
normally (and stop audio streaming), because we don't have a separate
heuristic to check whether the buffer is "low enough" (as a consequence
of a network stall, but before the audio output itself underruns).
Create a central function which pumps data through the filter. This also
might fix bogus use of the filter API on flushing. (The filter is just
used for convenience, but I guess the overall result is still simpler.)
It is now the AO's responsibility to handle period size alignment. The
ao->period_size alignment field is unused as of the recent audio
refactor commit. Remove it.
It turns out that ao_alsa shows extremely inefficient behavior as a
consequence of the removal of period size aligned writes in the
mentioned refactor commit. This is because it could get into a state
where it repeatedly wrote single samples (as small as 1 sample), and
starved the rest of the player as a consequence. Too bad. Explicitly
align the size in ao_alsa. Other AOs, which need this, should do the
same.
One reason why it broke so badly with ao_alsa was that it retried the
write() even if all reported space could be written. So stop doing that
too. Retry the write only if we somehow wrote less.
I'm not sure about ao_pulse.
This replaces the two buffers (ao_chain.ao_buffer in the core, and
buffer_state.buffers in the AO) with a single queue. Instead of having a
byte based buffer, the queue is simply a list of audio frames, as output
by the decoder. This should make dataflow simpler and reduce copying.
It also attempts to simplify fill_audio_out_buffers(), the function I
always hated most, because it's full of subtle and buggy logic.
Unfortunately, I got assaulted by corner cases, dumb features (attempt
at seamless looping, really?), and other crap, so it got pretty
complicated again. fill_audio_out_buffers() is still full of subtle and
buggy logic. Maybe it got worse. On the other hand, maybe there really
is some progress. Who knows.
Originally, the data flow parts was meant to be in f_output_chain, but
due to tricky interactions with the playloop code, it's now in the dummy
filter in audio.c.
At least this improves the way the audio PTS is passed to the encoder in
encoding mode. Now it attempts to pass frames directly, along with the
pts, which should minimize timestamp problems. But to be honest, encoder
mode is one big kludge that shouldn't exist in this way.
This commit should be considered pre-alpha code. There are lots of bugs
still hiding.
Previously get_state() would keep setting the cork status
while paused, but it only does for that after underflows now.
Correct this oversight by creating the stream corked for start()
to uncork it at a later time.
fixes#8026
FFmpeg expects those fields to be set on the AVFrame when
encoding audio, not doing so will cause the avcodec_send_frame
call to return EINVAL (at least in recent builds).
When get_state() corks the stream after an underrun happens
priv->playing is incorrectly reset to true, which can cause the
player to miss the underrun entirely. Stop resetting priv->playing
during corking (but not uncorking) to fix this.
The underflow callback introduced in d27ad96 can be called
when the buffer is still full, causing playback to never
resume afterwards since get_state() reports free_samples == 0.
Fix this by fully resetting on underrun, which flushes
the stream and ensures free buffer space.
fixes#7874
