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.
since i was going to fix the include order of stdatomic, might as well
sort the surrouding includes in accordance with the project's coding
style.
some headers can sometime require specific include order. standard
library headers usually don't. but mpv might "hack into" the standard
headers (e.g pthreads) so that complicates things a bit more.
hopefully nothing breaks. if it does, the style guide is to blame.
replace it with <stdatomic.h> and replace the mp_atomic_* typedefs with
explicit _Atomic qualified types.
also add missing config.h includes on some files.
Pull AOs work off of a callback that relies on mpv's internal timer. So
like with the related video changes, convert all of these to nanoseconds
instead. In many cases, the underlying audio API does actually provide
nanosecond resolution as well.
There's a lot of wild 1e6, 1000, etc. lying around in the code. A macro
is much easier to read and understand at a glance. Add some helpers for
this. We don't need to convert everything now but there's some simple
things that can be done so they are included in this commit.
Why a bigger search-interval is required:
scaletempo2 doesn't do a good job when the signal contains frequencies
less then 1/search_interval. With a search interval of 30ms that means
anything below 33.333Hz sounds bad.
Depending on the genre it's very for music to contain frequencies down
to 30Hz, and sometimes even a little bit below that. Therefore a higher
default value is needed to handle such cases.
Based on that an argument can be made for a value of 50, as that should
work down to 20Hz, or something even higher because movies sometimes
have some infrasonic content.
However the downside of big search intervals is increased CPU usage and
intelligibility at higher speeds, as it effectively leads to parts of
the audio being skipped.
A value of 40 can handle frequencies down to 25Hz, enough for all music
except very rare edge cases, while still providing decent
intelligibility.
Why a smaller window-size is required:
Large values reduce intelligibility at high speeds and therefore small
values are preferred.
However when values get too small it starts to sound weird
(similar to librubberband).
In my testing a value of 10 already works well, but adding a small
safety margin seems like a good idea, especially since it made no
noticeable difference to intelligibility, which is why 12 was chosen.
Linux and macOS already use nanosecond resolution for their sleep
functions. It was just being converted from microseconds before. Since
we have mp_time_ns now, go ahead and bump the precision here. The timer
for windows uses some timeBeginPeriod thing which I'm not sure what it
does really but whatever just convert the units to ms like they were
doing before. There's really no reason to keep the mp_sleep_us helper
around. A multiplication by 1000 is trivial and underlying OS clocks
have nanosecond precision.
Dudemanguy
sfan5
der richter
nanahi
sunpenghao
llyyr
Thomas Weißschuh
Kacper Michajłow
Alex Mitzsch
leetoburrito
Guido Cella
Umar Getagazov
NRK
Christoph Heinrich