Use mimalloc for Ladybird-owned allocations without overriding malloc().
Route kmalloc(), kcalloc(), krealloc(), and kfree() through mimalloc,
and put the embedded Rust crates on the same allocator via a shared
shim in AK/kmalloc.cpp.
This also lets us drop kfree_sized(), since it no longer used its size
argument. StringData, Utf16StringData, JS object storage, Rust error
strings, and the CoreAudio playback helpers can all free their AK-backed
storage with plain kfree().
Sanitizer builds still use the system allocator. LeakSanitizer does not
reliably trace references stored in mimalloc-managed AK containers, so
static caches and other long-lived roots can look leaked. Pass the old
size into the Rust realloc shim so aligned fallback reallocations can
move posix_memalign-backed blocks safely.
Static builds still need a little linker help. macOS app binaries need
the Rust allocator entry points forced in from liblagom-ak.a, while
static ELF links can pull in identical allocator shim definitions from
multiple Rust staticlibs. Keep the Apple -u flags and allow those
duplicate shim symbols for LibJS and LibRegex links on Linux and BSD.
Audio output on macOS was consuming Core Audio resources until the
PlaybackStream creation took well over the timeout for some tests.
This was observed in media-source-buffered.html, where it would time
out due to the long-running callback on the main thread to create the
PlaybackStream for AudioMixingSink.
However, the AudioUnit init should definitely not be blocking the main
thread, so I've added a FIXME there.
This allows us to avoid returning a PlaybackStream in cases where the
async initialization fails.
This is a step towards more graceful fallbacks when audio fails in
AudioMixingSink.
Instead of specifying the sample rate, channel count/map, etc. to the
PlaybackStream, we'll now use the output device's sample specification
whenever possible. If necessary, the stream will fall back to sane
default.
This hugely simplifies AudioMixingSink, since it no longer has to take
care of reinitializing the stream with a new sample specification when
it encounters a track with a higher sample rate or more channels. We
wouldn't be likely to benefit from this anyway, since it turns out that
at least Windows's virtual surround doesn't work through WASAPI at all,
and WASAPI likely wouldn't support downmixing.
This commit breaks playback of audio files that don't match the system
default audio device's sample rate and channel count. The next commit
introduces a converter into the pipeline to allow mixing of any sample
specification.
We were already assuming that our streams were using floats, we may as
well hardcode this. If we ever encounter a platform API that doesn't
support or convert from float, we can always bring this back.
Also, since we don't support big-endian systems, remove that check in
PulseAudioWrappers.
We should be fine to just fall back to zero or the last returned value
if we encounter an error in PlaybackStream::total_time_played(), and
this also simplifies the using code.
The SharedSingleProducerCircularQueue used here has dubious value, This
queue is used to pass commands to the audio thread, such as play/pause/
seek/volume change/etc. We can make do with a simple locked vector, as
we were blocking to enqueue tasks anyways. We can also use an atomic
bool to tell the audio thread when it needs to take a lock on the task
queue, to keep the thread lock-free most of the time.
Instead of having to duplicate the audio stream backend conditions, just
define PlaybackStream::create in each audio backend implementation file.
We provide a weak definition in PlaybackStream.cpp as the fallback.
