The mplayer1/2/mpv CoreAudio audio output historically contained both usage
of AUHAL APIs (these go through the CoreAudio audio server) and the Device
based APIs (used only for output of compressed formats in exclusive mode).
The latter is a very unwieldy and low level API and pretty much forces us to
write a lot of code for little workr. Also with the widespread of HDMI, the
actual need for outputting compressed audio directly to the device is getting
lower (it was very useful with S/PDIF for bandwidth constraints not allowing
a number if channels transmitted in LPCM).
Considering how invasive it is (uses hog/exclusive mode), the new AO
(`ao_coreaudio_device`) is not going to be autoprobed but the user will have
to select it.
Something like "char *s = ...; isdigit(s[0]);" triggers undefined
behavior, because char can be signed, and thus s[0] can be a negative
value. The is*() functions require unsigned char _or_ EOF. EOF is a
special value outside of unsigned char range, thus the argument to the
is*() functions can't be a char.
This undefined behavior can actually trigger crashes if the
implementation of these functions e.g. uses lookup tables, which are
then indexed with out-of-range values.
Replace all <ctype.h> uses with our own custom mp_is*() functions added
with misc/ctype.h. As a bonus, these functions are locale-independent.
(Although currently, we _require_ C locale for other reasons.)
rgain is not an additive value. It's a multiplier/gain.
Previous behaviour produced negative level values in some cases
(when rgain < 1.0) which caused volume to be louder when its value
was lowered.
CC: @mpv-player/stable
Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com>
Signed-off-by: wm4 <wm4@nowhere>
While I'm not very fond of "const", it's important for declarations
(it decides whether a symbol is emitted in a read-only or read/write
section). Fix all these cases, so we have writeable global data only
when we really need.
So the device buffer can be refilled quickly. Fixes dropouts in certain
cases: if all data is moved from the soft buffer to the audio device
buffer, the waiting code thinks it has to enter the mode in which it
waits for new data from the decoder. This doesn't work, because the
get_space() logic tries to keep the total buffer size down. get_space()
will return 0 (or a very low value) because the device buffer is full,
and the decoder can't refill the soft buffer. But this means if the AO
buffer runs out, the device buffer can't be refilled from the soft
buffer. I guess this mess happened because the code is trying to deal
with both AOs with proper event handling, and AOs with arbitrary
behavior.
Unfortunately this increases latency, as the total buffered audio
becomes larger. There are other ways to fix this again, but not today.
Fixes#818.
Apparently this can happen. So actually only return from waiting if ALSA
excplicitly signals that new output is available, or if we are woken up
externally.
This did not flush remaining audio in the buffer correctly (in case an
AO has an internal block size). So we have to make the audio feed thread
to write the remaining audio, and wait until it's done.
Checking the avoid_ao_wait variable should be enough to be sure that all
data that can be written was written to the AO driver.
This code handles buggy AOs (even if all AOs are bug-free, it's good for
robustness). Move handling of it to the AO feed thread. Now this check
doesn't require magic numbers and does exactly what's it supposed to do.
Until now, we've always calculated a timeout based on a heuristic when
to refill the audio buffers. Allow AOs to do it completely event-based
by providing wait and wakeup callbacks.
This also shuffles around the heuristic used for other AOs, and there is
a minor possibility that behavior slightly changes in real-world cases.
But in general it should be much more robust now.
ao_pulse.c now makes use of event-based waiting. It already did before,
but the code for time-based waiting was also involved. This commit also
removes one awkward artifact of the PulseAudio API out of the generic
code: the callback asking for more data can be reentrant, and thus
requires a separate lock for waiting (or a recursive mutex).
There were subtle and minor race conditions in the pull.c code, and AOs
using it (jack, portaudio, sdl, wasapi). Attempt to remove these.
There was at least a race condition in the ao_reset() implementation:
mp_ring_reset() was called concurrently to the audio callback. While the
ringbuffer uses atomics to allow concurrent access, the reset function
wasn't concurrency-safe (and can't easily be made to).
Fix this by stopping the audio callback before doing a reset. After
that, we can do anything without needing synchronization. The callback
is resumed when resuming playback at a later point.
Don't call driver->pause, and make driver->resume and driver->reset
start/stop the audio callback. In the initial state, the audio callback
must be disabled.
JackAudio of course is different. Maybe there is no way to suspend the
audio callback without "disconnecting" it (what jack_deactivate() would
do), so I'm not trying my luck, and implemented a really bad hack doing
active waiting until we get the audio callback into a state where it
won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we
can be sure that the callback doesn't access the ringbuffer or anything
else anymore. Since both sched_yield() and pthread_yield() apparently
are not always available, use mp_sleep_us(1) to avoid burning CPU during
active waiting.
The ao_jack.c change also removes a race condition: apparently we didn't
initialize _all_ ao fields before starting the audio callback.
In ao_wasapi.c, I'm not sure whether reset really waits for the audio
callback to return. Kovensky says it's not guaranteed, so disable the
reset callback - for now the behavior of ao_wasapi.c is like with
ao_jack.c, and active waiting is used to deal with the audio callback.
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
The i_bps members of the sh_audio and dev_video structs are mostly used
for displaying the average audio and video bitrates. Keeping them in
bits-per-second avoids truncating them to bytes-per-second and changing
them back lateron.
In my opinion, we shouldn't use atomics at all, but ok.
This switches the mpv code to use C11 stdatomic.h, and for compilers
that don't support stdatomic.h yet, we emulate the subset used by mpv
using the builtins commonly provided by gcc and clang.
This supersedes an earlier similar attempt by Kovensky. That attempt
unfortunately relied on a big copypasted freebsd header (which also
depended on much more highly compiler-specific functionality, defined
reserved symbols, etc.), so it had to be NIH'ed.
Some issues:
- C11 says default initialization of atomics "produces a valid state",
but it's not sure whether the stored value is really 0. But we rely on
this.
- I'm pretty sure our use of the __atomic... builtins is/was incorrect.
We don't use atomic load/store intrinsics, and access stuff directly.
- Our wrapper actually does stricter typechecking than the stdatomic.h
implementation by gcc 4.9. We make the atomic types incompatible with
normal types by wrapping them into structs. (The FreeBSD wrapper does
the same.)
- I couldn't test on MinGW.
Use the time as returned by mp_time_us() for mpthread_cond_timedwait(),
instead of calculating the struct timespec value based on a timeout.
This (probably) makes it easier to wait for a specific deadline.
This didn't quite work. The main issue was that get_space tries to be
clever to reduce overall buffering, so it will cause the playloop to
decode and queue only as much audio as is needed to refill the AO in
reasonable time. Also, even if ignoring the problem, the logic of the
previous commit was slightly broken. (This required a few retries,
because I couldn't reproduce the issue on my own machine.)
When the audio buffer went low, but could not be refilled yet, it could
happen that the AO playback thread and the decode thread could enter a
wakeup feedback loop, causing up to 100% CPU usage doing nothing. This
happened because the decoder thread would wake up the AO thread when
writing 0 bytes of newly decoded data, and the AO thread in reaction
wakes up the decoder thread after writing 0 bytes to the AO buffer.
Fix this by waking up the decoder thread only if data was actually
played or queued. (This will still cause some redundant wakeups, but
will eventually settle down, reducing CPU usage close to ideal.)
I don't think this is really a very good idea because it is conceptually
incorrect but other prominent multimedia programs use this approach
(VLC and xbmc), and it seems to make the conversion more robust in certain
cases.
For example it has been reported, that configuring a receiver that can output
7.1 to output 5.1, will make CoreAudio report 8 channel descriptions, and the
last 2 descriptions will be tagged kAudioChannelLabel_Unknown.
Fixes#737
The code was falling back to the full waveext chmap_sel when less than 2
channels were detected. This new code is slightly more correct since it only
fills the chmap_sel with the stereo or mono chmap in the fallback case.
CoreAudio supports 3 kinds of layouts: bitmap based, tag based, and speaker
description based (using either channel labels or positional data).
Previously we tried to convert everything to bitmap based channel layouts,
but it turns out description based ones are the most generic and there are
built-in CoreAudio APIs to perform the conversion in this direction.
Moreover description based layouts support waveext extensions (like SDL and
SDR), and are easier to map to mp_chmaps.
Changing --softvol-max and then resuming would change the volume level
on resume to something different than the original volume. This is
because the user volume setting is always between 0-100, and 100
corresponds to --softvol-max gain.
Avoid that changing -softvol-max and resuming an older file could lead
to a too loud volume level by refusing to restore if --softvol-max
changed.
