Previously, we were sending Buffers to the server whenever we had new
audio data for it. This meant that for every audio enqueue action, we
needed to create a new shared memory anonymous buffer, send that
buffer's file descriptor over IPC (+recfd on the other side) and then
map the buffer into the audio server's memory to be able to play it.
This was fine for sending large chunks of audio data, like when playing
existing audio files. However, in the future we want to move to
real-time audio in some applications like Piano. This means that the
size of buffers that are sent need to be very small, as just the size of
a buffer itself is part of the audio latency. If we were to try
real-time audio with the existing system, we would run into problems
really quickly. Dealing with a continuous stream of new anonymous files
like the current audio system is rather expensive, as we need Kernel
help in multiple places. Additionally, every enqueue incurs an IPC call,
which are not optimized for >1000 calls/second (which would be needed
for real-time audio with buffer sizes of ~40 samples). So a fundamental
change in how we handle audio sending in userspace is necessary.
This commit moves the audio sending system onto a shared single producer
circular queue (SSPCQ) (introduced with one of the previous commits).
This queue is intended to live in shared memory and be accessed by
multiple processes at the same time. It was specifically written to
support the audio sending case, so e.g. it only supports a single
producer (the audio client). Now, audio sending follows these general
steps:
- The audio client connects to the audio server.
- The audio client creates a SSPCQ in shared memory.
- The audio client sends the SSPCQ's file descriptor to the audio server
with the set_buffer() IPC call.
- The audio server receives the SSPCQ and maps it.
- The audio client signals start of playback with start_playback().
- At the same time:
- The audio client writes its audio data into the shared-memory queue.
- The audio server reads audio data from the shared-memory queue(s).
Both sides have additional before-queue/after-queue buffers, depending
on the exact application.
- Pausing playback is just an IPC call, nothing happens to the buffer
except that the server stops reading from it until playback is
resumed.
- Muting has nothing to do with whether audio data is read or not.
- When the connection closes, the queues are unmapped on both sides.
This should already improve audio playback performance in a bunch of
places.
Implementation & commit notes:
- Audio loaders don't create LegacyBuffers anymore. LegacyBuffer is kept
for WavLoader, see previous commit message.
- Most intra-process audio data passing is done with FixedArray<Sample>
or Vector<Sample>.
- Improvements to most audio-enqueuing applications. (If necessary I can
try to extract some of the aplay improvements.)
- New APIs on LibAudio/ClientConnection which allows non-realtime
applications to enqueue audio in big chunks like before.
- Removal of status APIs from the audio server connection for
information that can be directly obtained from the shared queue.
- Split the pause playback API into two APIs with more intuitive names.
I know this is a large commit, and you can kinda tell from the commit
message. It's basically impossible to break this up without hacks, so
please forgive me. These are some of the best changes to the audio
subsystem and I hope that that makes up for this :yaktangle: commit.
:yakring:
With the following change in how we send audio, the old Buffer type is
not really needed anymore. However, moving WavLoader to the new system
is a bit more involved and out of the scope of this PR. Therefore, we
need to keep Buffer around, but to make it clear that it's the old
buffer type which will be removed soon, we rename it to LegacyBuffer.
Most of the users will be gone after the next commit anyways.
A mistake I've repeatedly made is along these lines:
```c++
auto nread = TRY(source_file->read(buffer));
TRY(destination_file->write(buffer));
```
It's a little clunky to have to create a Bytes or StringView from the
buffer's data pointer and the nread, and easy to forget and just use
the buffer. So, this patch changes the read() function to return a
Bytes of the data that were just read.
The other read_foo() methods will be modified in the same way in
subsequent commits.
Fixes#13687
`static const` variables can be computed and initialized at run-time
during initialization or the first time a function is called. Change
them to `static constexpr` to ensure they are computed at
compile-time.
This allows some removal of `strlen` because the length of the
`StringView` can be used which is pre-computed at compile-time.
Of course, Buffer is going to be removed very soon, but much of the
WavLoader behavior still depends on it. Therefore, this intermediary
API will allow adopting the Loader infrastructure without digging too
deep into the WavLoader legacy code. That's for later :^)
The Buffer files had contained both the ResampleHelper and the
sample format utilities. Because the Buffer class (and its file) is
going to be deleted soon, this commit separates those two things into
their own files.
The data is taken straight from the ISO/IEC 11172-3 standard. For
decoding efficiency the tables are transformed into trees at compile
time using a constexpr approach. That way no runtime initialization is
necessary and decoding can be faster than searching through tables.
The old FIXME asserting that Core::AnonymousBuffer cannot be invalid
or zero-sized is no longer accurate. Add a default constructor for
Audio::Buffer that has all invalid state instead of going to the OS to
allocate a 1 sample buffer for the "no more samples" states in the WAV
and FLAC plugins.
Apologies for the enormous commit, but I don't see a way to split this
up nicely. In the vast majority of cases it's a simple change. A few
extra places can use TRY instead of manual error checking though. :^)
This change unfortunately cannot be atomically made without a single
commit changing everything.
Most of the important changes are in LibIPC/Connection.cpp,
LibIPC/ServerConnection.cpp and LibCore/LocalServer.cpp.
The notable changes are:
- IPCCompiler now generates the decode and decode_message functions such
that they take a Core::Stream::LocalSocket instead of the socket fd.
- IPC::Decoder now uses the receive_fd method of LocalSocket instead of
doing system calls directly on the fd.
- IPC::ConnectionBase and related classes now use the Stream API
functions.
- IPC::ServerConnection no longer constructs the socket itself; instead,
a convenience macro, IPC_CLIENT_CONNECTION, is used in place of
C_OBJECT and will generate a static try_create factory function for
the ServerConnection subclass. The subclass is now responsible for
passing the socket constructed in this function to its
ServerConnection base; the socket is passed as the first argument to
the constructor (as a NonnullOwnPtr<Core::Stream::LocalServer>) before
any other arguments.
- The functionality regarding taking over sockets from SystemServer has
been moved to LibIPC/SystemServerTakeover.cpp. The Core::LocalSocket
implementation of this functionality hasn't been deleted due to my
intention of removing this class in the near future and to reduce
noise on this (already quite noisy) PR.
The format of these names is "Full Abbreviation (.fileformat)". For
example: "FLAC (.flac)", "RIFF WAVE (.wav)", "MPEG Layer III (.mp3)",
"Vorbis (.ogg)" The reasoning is that the container and therefore the
file ending may differ significantly from the actual format, and the
format should be given as unambiguously as possible and necessary.
FixedArray now doesn't expose any infallible constructors anymore.
Rather, it exposes fallible methods. Therefore, it can be used for
OOM-safe code.
This commit also converts the rest of the system to use the new API.
However, as an example, VMObject can't take advantage of this yet,
as we would have to endow VMObject with a fallible static
construction method, which would require a very fundamental change
to VMObject's whole inheritance hierarchy.
Previously, FlacLoader would read the data for each frame into a
separate vector, which are then combined via extend() in the end. This
incurs an avoidable copy per frame. By having the next_frame() function
write into a given Span, there's only one vector allocated per call to
get_more_samples().
This increases performance by at least 100% realtime, as measured by
abench, from about 1200%-1300% to (usually) 1400% on complex test files.
The 'muted' methods referred to the 'main mix muted' but it wasn't
really clear from the name. This change will be useful because in the
next commit, a 'self muted' state will be added to each audio client
connection.
As long as possible, entire decoded frame sample vectors are moved into
the output vector, leading to up to 20% speedups by avoiding memmoves on
take_first.
Previously, a libc-like out-of-line error information was used in the
loader and its plugins. Now, all functions that may fail to do their job
return some sort of Result. The universally-used error type ist the new
LoaderError, which can contain information about the general error
category (such as file format, I/O, unimplemented features), an error
description, and location information, such as file index or sample
index.
Additionally, the loader plugins try to do as little work as possible in
their constructors. Right after being constructed, a user should call
initialize() and check the errors returned from there. (This is done
transparently by Loader itself.) If a constructor caused an error, the
call to initialize should check and return it immediately.
This opportunity was used to rework a lot of the internal error
propagation in both loader classes, especially FlacLoader. Therefore, a
couple of other refactorings may have sneaked in as well.
The adoption of LibAudio users is minimal. Piano's adoption is not
important, as the code will receive major refactoring in the near future
anyways. SoundPlayer's adoption is also less important, as changes to
refactor it are in the works as well. aplay's adoption is the best and
may serve as an example for other users. It also includes new buffering
behavior.
Buffer also gets some attention, making it OOM-safe and thereby also
propagating its errors to the user.
This consists of two changes: First, a utility function create_empty
allows the user to quickly create an empty buffer. Second, most creation
functions now return a NonnullRefPtr, as their failure causes a VERIFY
crash anyways.
Decoding the residual in FLAC subframes is by far the most I/O-heavy
operation in FLAC decoding, as the residual data makes up the majority
of subframe data in LPC subframes. As the residual consists of many
Rice-encoded numbers with different bit sizes for differently large
numbers, the residual decoder frequently reads only one or two bytes at
a time. As we use a normal FileInputStream, that directly translates to
many calls to the read() syscall. We can see that the I/O overhead while
FLAC decoding is quite large, and much time is spent in the read()
syscall's kernel code.
This is optimized by using a Buffered<FileInputStream> instead, leading
to 4K blocks being read at once and a large reduction in I/O overhead.
Benchmarking with the new abench utility gives a 15-20% speedup on
identical files, usually pushing FLAC decoding to 10-15x realtime speed
on common sample rates.
Some nuances in the FLAC loading code can do well with an explanation,
as these non-obvious insights are often the result of long and painful
debugging and nobody should touch the affected code without careful
deliberation.
(Of course, secretly I just want people to maintain my loader code.)
:^)
This fixes all current code smells, bugs and issues reported by
SonarCloud static analysis. Other issues are almost exclusively false
positives. This makes much code clearer, and some minor benefits in
performance or bug evasion may be gained.
This little functional change uses the most common algorithm for panning
audio, known as constant power panning. It makes it so that the total
output power (not directly the sample value, i.e. the peak) stays the
same no matter how the audio is panned.
"Frame" is an MPEG term, which is not only unintuitive but also
overloaded with different meaning by other codecs (e.g. FLAC).
Therefore, use the standard term Sample for the central audio structure.
The class is also extracted to its own file, because it's becoming quite
large. Bundling these two changes means not distributing similar
modifications (changing names and paths) across commits.
Co-authored-by: kleines Filmröllchen <malu.bertsch@gmail.com>
The conversion from a linear scale (how we think about audio) to a
logarithmic scale (how audio actually works) will be useful for other
operations, so let's extract it to its own utility function. Its inverse
will also allow reversible operations to be written more easily.
