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.
This is the most supported in standard layout, if we request more it
tends to fallback to stereo instead. Also channels mask is 32-bit and it
can get truncated.
A bit different from the OPT_REPLACED/OPT_REMOVED ones in that the
options still possibly do something but they have a deprecation
message. Most of these are old and have no real usage. The only
potentially controversial ones are the removal of --oaffset and
--ovoffset which were deprecated years ago and seemingly have no real
replacement. There's a cryptic message about --audio-delay but who
knows. The less encoding mode code we have, the better so just chuck
it.
Avoid generating too much audio after EOF.
Note: This often has no effect, because less audio is produced than
required.
Usually this comes to effect with the userspeed filter at high speed
(4x) and going back to 1x speed to remove the filter.
After the final input packet, the filter padded with silence to allow
one more iteration. That was not enough to process the final frames.
Continue padding the end of `input_buffer` with silence until the final
frames have been processed.
Implementation: Instead of padding when adding final samples, pad before
running WSOLA iteration. Count number of added silent frames and
remaining input frames for time keeping.
This changes the emitted pts values from the start of the search block
to the center of the search block. Change initial `output_time`
accordingly. Initial `search_block_index` is irrelevant, because it's
overwritten before the first iteration.
Using the `output_time` removes the rounding of `search_block_index`,
which also fixes the <20 microsecond gaps in timestamps between output
packets.
Rationale:
The variance in audio position was in the range `0..search-interval`.
With this change, the range is
(- search-interval / 2)..(search-interval / 2)`
which ensures lower maximum offset.
Target block can be anywhere in the previous search-block, varying by
`search-interval` while the filter is active. This resulted in constant
audio offset when returning to 1x playback speed.
- Move the search block to the target block to sync up exactly.
- Drop old frames to minimize input_buffer usage.
The internal time update function involved multiple problems:
- Time was updated after WSOLA iteration. The means speed was updated
one iteration later than it could be.
- The update functions caused spikes of too many or too few samples
advanced, leading to audio glitches on speed changes.
- The inconsistent updates made it very difficult to produce gapless
audio packets.
- The `output_time` update function involved complicated feedback:
`search_block_index` influenced how many frames from `input_buffer`
are retained, which influenced how much `output_time` is changed,
which influenced `search_block_index`.
With these changes:
- Time is updated before WSOLA iterations. Speed changes are effective
instantly.
- There are no spikes in playback speed during speed changes.
- No significant gaps are introduced in output packets.
- The time update function becomes (function calls omitted for brevity)
output_time += ola_hop_size * playback_rate
Functions received a `playback_rate` parameter to check how many samples
are needed before iteration. Internal state is only updated when the
iteration is actually run, so the speed is allowed to change until
enough data is received.
The first WSOLA iteration overlapped audio with whatever was in the
`wsola_output` buffer. This was either silence (if not run before), or
old frames (if switching to 1x and back to a different speed).
Track the state of the output buffer and memcpy the whole window for the
first iteration instead.
