This commit adds a state `hw_paused` for pull-based AO.
`driver->set_paused(false)` is only called if `hw_paused` is true.
`hw_paused` is cleared after `ao_reset`, so `set_paused` will
not be called after a reset; instead, `driver->start()` will
be called, which properly starts the AO.
Same as ffmpeg uses. Such big values does not make sense probably, but
let's not overflow values and maybe one day it will be useful.
Fixes signed integer overflow.
This problem does not exist with --demuxer=lavf. --demuxer=mkv just never
signals EOF for the problematic sample, so it needs to be fixed there, not
in AO.
This reverts commit 0cfd52074b.
6863eefc3d handled this situation by using
an atomic variable to express the state for which the wakeup is caused
by AO control, and the dispatch queue is only processed at this state.
However, this can cause permanent lockup of the player core when the
following happens:
- AO control sets the thread state to WASAPI_THREAD_DISPATCH, and
sets the wakeup handle.
- WASAPI thread reads the WASAPI_THREAD_DISPATCH state and processes
the dispatch queue.
- Another AO control happens. A dispatch item is enqueued, and the
state stays at WASAPI_THREAD_DISPATCH.
- WASAPI thread resets the thread state to WASAPI_THREAD_FEED since
the state has not changed.
- WaitForSingleObject() returns in the WASAPI thread, sees this state,
and does not process the dispatch queue.
- The player core locks permanently because it is waiting for the dispatch
to be processed.
This has been experimentally verified on a system under high contention:
The easiest way to trigger this lockup is to continuously hold down "i",
which rapidly issues AO get volume/mute controls.
To properly handle this, use separate handles for system and user wakeup
requests. Only feed audio when woke up by system and only process the
dispatch queue when woke up by user.
Fixes: 6863eefc3d
This allows users to set buffer duration in exclusive mode. We have
been using the default device period as the buffer size and it is
robust enough in most cases. However, on some devices there are
horrible glitches after a stream reset. Unfortunately, the issue is not
consistently reproducible, but using a smaller buffer size (e.g., the
minimum device period) seems to resolve the problem.
Fixes#13715.
"playthread" is a confusing name which doesn't describe what it really
is. Rename it to ao_thread, and ao_wakeup_playthread to ao_wakeup,
in the same style as VO threads. This makes call stack function names
less confusing.
A figure from pipewire documentation:
```
stream time domain graph time domain
/-----------------------\/-----------------------------\
queue +-+ +-+ +-----------+ +--------+
----> | | | |->| converter | -> graph -> | kernel | -> speaker
<---- +-+ +-+ +-----------+ +--------+
dequeue buffers \-------------------/\--------/
graph internal
latency latency
\--------/\-------------/\-----------------------------/
queued buffered delay
```
We calculate `end_time` in the following steps:
1. get current timestamp in mpv
```
int64_t end_time = mp_time_ns();
```
2. add duration of samples to enqueue
```
end_time += MP_TIME_S_TO_NS(nframes) / ao->samplerate;
```
3. add delay of the pipewire graph
```
end_time += MP_TIME_S_TO_NS(time.delay) * time.rate.num / time.rate.denom;
```
4. add duration of queued and buffered samples.
```
end_time += MP_TIME_S_TO_NS(time.queued) / ao->samplerate;
end_time += MP_TIME_S_TO_NS(time.buffered) / ao->samplerate;
```
New in this commit. `time.queued` is usually zero as `SPA_PARAM_BUFFERS_buffers`
is default to 1; however it is not always.
`time.buffered` is non-zero if there is a resampler involved.
5. add elapsed duration from when `time` is captured
```
end_time -= pw_stream_get_nsec(p->stream) - time.now;
```
New in this commit. `time` is captured at `time.now`.
From then, time has passed so we need to exclude the elapsed time,
by calculating the diff of `pw_stream_get_nsec()` and `time.now`.
`hotplug_cb` was registered only in `hotplug_init()`.
This commit make it registered in `init()` as well,
so that the ao can listen for latency change
in playback.
`buf` contains a `struct spa_data` for each channel.
Therefore the number of channels does not matter to calculate the frame capacity of one `struct spa_data`.
In practice this shouldn't make a difference as `b->requested` would reduce nframes even more.
During AO init, snd_pcm_open() is called, which calls snd_config_update()
to allocate a global config node and stores it in the snd_config global
variable. This is never freed on uninit.
Fix this by freeing the global config node on uninit.
The device latency may change during hotplugging.
This commit updates p->hw_latency_ns each time
hotplug_cb is called so that it can reflect
updated device latency.
As far as I can tell PulseAudio introduced a bug in 16.0
where if a stream is (un)paused too often the reported latency
will momentarily spike by 3000% or more. Apparently in certain cases
just pausing once and waiting can also cause this.
Save the remaining users of PA the trouble of debugging the various
obscure issues that can arise from this (desync is a harmless example)
by enabling the latency hack code again.
ref: <https://github.com/mpv-player/mpv/issues/12057>
<https://github.com/mpv-player/mpv/issues/10333>
