Commit 39f7f83 changed ao_driver.reset to use AudioUnitReset instead of
AudioOutputUnitStop. The problem with calling AudioOutputUnitStop was
that AudioOutputUnitStart takes a significant amount of time after a
stop when a wireless audio device is being used. This resulted in
lagging that was noticeable to users during seeking and short
pause/resume cycles. Switching to AudioUnitReset eliminated this
lagging.
However with the switch to AudioUnitReset the macOS daemon coreaudiod
continued to consume CPU time and did not release a powerd assertion
that it created on behalf of mpv, preventing macOS from sleeping.
This commit will change ao_coreaudio.reset to call AudioOutputUnitStop
after a delay if playback has not resumed. This preserves the faster
restart of playback for seeking and short pause/resume cycles and avoids
preventing sleep and needless CPU consumption.
Fixes#11617
The code changes were authored by @orion1vi and @lhc70000.
Co-authored-by: Collider LI <lhc199652@gmail.com>
Currently, the softvol gain control attempts to clip floating point ao
formats within -1 and +1. However, this is "optimized out" at unity gain,
where no clipping is applied. This results in inconsistent behavior when
the source audio is already out of -1 and +1 range, where a gain of 0.99
results in clipping, but not at exactly 1.
Since a big advantage of floating point audio data is that they do not
lose information through out-of-range data because the ao sink can apply
suitable negative gain to prevent clipping before converting them to
integer formats, clipping should not be performed on these data.
Fix this by removing the existing clipping behavior. It now results in
a simple multiplication, which faciliates compiler auto-vectorization
of this operation over audio data.
Currently, running AO control wakes up the WASAPI renderer thread in the
`WASAPI_THREAD_FEED` state, where `thread_feed` will be called. However,
it seems that in recent Windows versions (tested on Windows 10 build
19044.3930 and Windows 11 build 22631.3007) we can't know if it is safe
to feed more audio data in event-driven exclusive mode:
- `IAudioClient_GetCurrentPadding` always returns `bufferFrameCount`,
even if *NO* data has ever been written. This means we don't know how
much free space we have that is available for writing. This is not the
case in shared mode, where the return value correctly reflects the
size of data waiting to be processed. As a sidenote, MS did not
document the precise definition of the return value for an
event-driven, exclusive stream [1].
- `IAudioRenderClient_GetBuffer` never fails. We can call it for 10
times in a roll, each time requesting an entire buffer (the unit at
which data is exchanged in exclusive mode using event-driven
buffering; there are 2 such buffers) and get a successful return code
everytime. In shared mode, we get `AUDCLNT_E_BUFFER_TOO_LARGE` if we
request a buffer larger than that currently available.
As a result, `thread_feed` will always write `bufferFrameCount` frames
of audio in exclusive mode. There will therefore be glitches each time
`thread_control` is called due to the subsequent `thread_feed`
overwriting frames yet to be processed. Also, an irreversible error is
accumulated to `sample_count` as long as there is no AO reset, leading
to eventual, unbounded A/V desync.
As a fix to the issue, add a dedicated state for dispatch queue
processing so that `thread_feed` is only called when signaled by the OS.
The buffer checks in `thread_feed` that use `GetCurrentPadding` in
exclusive mode are kept in case there are older versions where the two
APIs behave differently.
Closes#12615.
[1] https://learn.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient-getcurrentpadding
Deprecated upstream 1cc24d7495
We need to reallocate the context here because `avcodec_free_context`
also frees the context, and we want to reuse the context with some
reconfig.
As mentioned in [0] the suffix "_locked" would have been the appropriate
naming in line with similar uses inside mpv.
See `mp_abort_recheck_locked()`, `mp_abort_trigger_locked()`,
`retrigger_locked()`, `wakeup_locked()`...
[0] https://github.com/mpv-player/mpv/pull/12811#discussion_r1477518525
Fix DTS passthrough playback of 44.1 khz content. Also, take into account that there are some DTS variants with a samplerate of 96khz (e.g. DTS 24/96), somehow they are recognized wrongly as 48khz by the code. Don´t rely on this "bug", do it correctly. Now every samplerate above 44.1Khz is correctly treated as 48khz, and 44.1khz files are treated as 44.1khz for bitstreaming.
Stopping output implies that it can't be paused anymore.
This is consistent with the documented API in internal.h as well
as the behavior of other AOs.
resolves#13267
In commit c09245cdf2
long-path support was enabled for mpv without actually
making sure that there was no code left that used the
old limit (260 Unicode chars) for buffer sizes.
This commit fixes all but one case.
- Don't define _GNU_SOURCE on Windows, no need
- Define WIN32_LEAN_AND_MEAN to strip some unneded headers from
windows.h
- Define NOMINMAX and _USE_MATH_DEFINES as they are common for Windows
headers
We prefer to fail fast rather than degrade in unpredictable ways.
The example in sub/ is particularly egregious because the code just
skips the work it's meant to do when an allocation fails.
I'd like some names to be more descriptive, but to work with 15 chars
limit we have to make some sacrifice.
Also because of the limit, remove the `mpv/` prefix and prioritize
actuall thread name.
It was found that this causes issues with at least ao_coreaudio,
essentially revealing a way bigger issue:
Some AOs don't check for 0 and/or have no way to deal with short writes.
Someone will have to figure out a fix later but get rid of the direct
cause for now.
This reverts commit ae908a70ce.
ao_read_data() is used by pull AOs potentially from threads managed by
external libraries. These threads can be sensitive to blocking.
For example the pipewire ao is using a realtime thread for the
callbacks.
