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 has been a long standing annoyance - ffmpeg is removing
sizeof(AVPacket) from the API which means you cannot stack-allocate
AVPacket anymore. However, that is something we take advantage of
because we use short-lived AVPackets to bridge from native mpv packets
in our main decoding paths.
We don't think that switching these to `av_packet_alloc` is desirable,
given the cost of heap allocation, so this change takes a different
approach - allocating a single packet in the relevant context and
reusing it over and over.
That's fairly straight-forward, with the main caveat being that
re-initialising the packet is unintuitive. There is no function that
does exactly what we need (what `av_init_packet` did). The closest is
`av_packet_unref`, which additionally frees buffers and side-data.
However, we don't copy those things - we just assign them in from our
own packet, so we have to explicitly clear the pointers before calling
`av_packet_unref`. But at least we can make a wrapper function for
that.
The weirdest part of the change is the handling of the vtt subtitle
conversion. This requires two packets, so I had to pre-allocate two in
the context struct. That sounds excessive, but if allocating the
primary packet is too expensive, then allocating the secondary one for
vtt subtitles must also be too expensive.
This change is not conditional as heap allocated AVPackets were
available for years and years before the deprecation.
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.
When af_scaletempo2.c:process() detects a format change, it goes back
through mp_scaletempo2_init() to reinitialize everything. However,
mp_scaletempo2.input_buffer is not necessarily reallocated due to a
check in af_scaletempo2_internals.c:resize_input_buffer(). This is a
problem if the number of audio channels increases, since without
reallocating, the buffer for the new channel(s) will at best point to
NULL, and at worst uninitialized memory.
Since resize_input_buffer() is only called from two places, pull size
check out into mp_scaletempo2_fill_input_buffer(). This allows each
caller to decide whether they want to resize or not. We could be
smarter about when to reallocate, but that would add a lot of machinery
for a case I don't expect to be hit often in practice.
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