Includes a set of wav files of different frequencies, these are
each loaded and then written to a temporary file, checking that
the meta-data is correctly read and that the output matches the input.
This commit un-deprecates DeprecatedString, and repurposes it as a byte
string.
As the null state has already been removed, there are no other
particularly hairy blockers in repurposing this type as a byte string
(what it _really_ is).
This commit is auto-generated:
$ xs=$(ack -l \bDeprecatedString\b\|deprecated_string AK Userland \
Meta Ports Ladybird Tests Kernel)
$ perl -pie 's/\bDeprecatedString\b/ByteString/g;
s/deprecated_string/byte_string/g' $xs
$ clang-format --style=file -i \
$(git diff --name-only | grep \.cpp\|\.h)
$ gn format $(git ls-files '*.gn' '*.gni')
https://developer.apple.com/documentation/audiounit
Apple has a number of audio frameworks we could use. This uses the Audio
Unit framework, as it gives us most control over the rendering of the
audio frames (such as being able to quickly pause / discard buffers).
From some reading, we could implement niceties such as fading playback
in and out while seeking over a short (10ms) period. This patch does not
implement such fancy features though.
Co-Authored-By: Andrew Kaster <akaster@serenityos.org>
This implementation is very naive compared to the PulseAudio one.
Instead of using a callback implemented by the audio server connection
to push audio to the buffer, we have to poll on a timer to check when
we need to push the audio buffers. Implementing cross-process condition
variables into the audio queue class could allow us to avoid polling,
which may prove beneficial to CPU usage.
Audio timestamps will be accurate to the number of samples available,
but will count in increments of about 100ms and run ahead of the actual
audio being pushed to the device by the server.
Buffer underruns are completely ignored for now as well, since the
`AudioServer` has no way to know how many samples are actually written
in a single audio buffer.
This will ensure that we don't leak any memory while playing back
audio.
There is an expectation value in the test that is only set to true when
PulseAudio is present for the moment. When any new implementation is
added for other libraries/platforms, we should hopefully get a CI
failure due to unexpected success in creating the `PlaybackStream`.
To ensure that we clean up our PulseAudio connection whenever audio
output is not needed, add `PulseAudioContext::weak_instance()` to allow
us to check whether an instance exists without creating one.
This removes a lot of duplicated stream creation code from the plugins,
and also simplifies the way that the appropriate plugin is found. This
mirrors the ImageDecoderPlugin design and necessitates new sniffing
methods on the loaders.
The deallocation of the test cases at the very end happens through a
NonnullRefPtr<TestCase>, meaning the deallocation will assume the wrong
object size and trip up ASAN. Therefore, we cannot use a TestCase
subclass.
I also took this opportunity and made use of the new LoaderError
formatter.
Before, some loader plugins implemented their own buffering (FLAC&MP3),
some didn't require any (WAV), and some didn't buffer at all (QOA). This
meant that in practice, while you could load arbitrary amounts of
samples from some loader plugins, you couldn't do that with some others.
Also, it was ill-defined how many samples you would actually get back
from a get_more_samples call.
This commit fixes that by introducing a layer of abstraction between the
loader and its plugins (because that's the whole point of having the
extra class!). The plugins now only implement a load_chunks() function,
which is much simpler to implement and allows plugins to play fast and
loose with what they actually return. Basically, they can return many
chunks of samples, where one chunk is simply a convenient block of
samples to load. In fact, some loaders such as FLAC and QOA have
separate internal functions for loading exactly one chunk. The loaders
*should* load as many chunks as necessary for the sample count to be
reached or surpassed (the latter simplifies loading loops in the
implementations, since you don't need to know how large your next chunk
is going to be; a problem for e.g. FLAC). If a plugin has no problems
returning data of arbitrary size (currently WAV), it can return a single
chunk that exactly (or roughly) matches the requested sample count. If a
plugin is at the stream end, it can also return less samples than was
requested! The loader can handle all of these cases and may call into
load_chunk multiple times. If the plugin returns an empty chunk list (or
only empty chunks; again, they can play fast and loose), the loader
takes that as a stream end signal. Otherwise, the loader will always
return exactly as many samples as the user requested. Buffering is
handled by the loader, allowing any underlying plugin to deal with any
weird sample count requirement the user throws at it (looking at you,
SoundPlayer!).
This (not accidentally!) makes QOA work in SoundPlayer.
Note that this still keeps the old behaviour of putting things in std by
default on serenity so the tools can be happy, but if USING_AK_GLOBALLY
is unset, AK behaves like a good citizen and doesn't try to put things
in the ::std namespace.
std::nothrow_t and its friends get to stay because I'm being told that
compilers assume things about them and I can't yeet them into a
different namespace...for now.
We have a new, improved string type coming up in AK (OOM aware, no null
state), and while it's going to use UTF-8, the name UTF8String is a
mouthful - so let's free up the String name by renaming the existing
class.
Making the old one have an annoying name will hopefully also help with
quick adoption :^)
This now prepares all the needed (fallible) components before actually
constructing a LoaderPlugin object, so we are no longer filling them in
at an arbitrary later point in time.
The FLAC "spec tests", or rather the test suite by xiph that exercises
weird FLAC features and edge cases, can be found at
https://github.com/ietf-wg-cellar/flac-test-files and is a good
challenge for our FLAC decoder to become more spec compliant. Running
these tests is similar to LibWasm spec tests, you need to pass
INCLUDE_FLAC_SPEC_TESTS to CMake.
As of integrating these tests, 23 out of 63 fail. :yakplus:
