This is a sensible separation of concerns that mirrors the WindowServer
IPC split. On the one hand, there is the "normal" audio interface, used
for clients that play audio, which is the primary service of
AudioServer. On the other hand, there is the management interface,
which, like the WindowManager endpoint, provides higher-level control
over clients and the server itself.
The reasoning for this split are manifold, as mentioned we are mirroring
the WindowServer split. Another indication to the sensibility of the
split is that no single audio client used the APIs of both interfaces.
Also, useless audio queues are no longer created for managing clients
(since those don't even exist, just like there's no window backing
bitmap for window managing clients), eliminating any bugs that may occur
there as they have in the past.
Implementation-wise, we just move all the APIs and implementations from
the old AudioServer into the AudioManagerServer (and respective clients,
of course). There is one point of duplication, namely the hardware
sample rate. This will be fixed in combination with per-client sample
rate, eliminating client-side resampling and the related update bugs.
For now, we keep one legacy API to simplify the transition.
The new AudioManagerServer also gains a hardware sample rate change
callback to have exact symmetry on the main server parameters (getter,
setter, and callback).
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 :^)
Each of these strings would previously rely on StringView's char const*
constructor overload, which would call __builtin_strlen on the string.
Since we now have operator ""sv, we can replace these with much simpler
versions. This opens the door to being able to remove
StringView(char const*).
No functional changes.
The file is now renamed to Queue.h, and the Resampler APIs with
LegacyBuffer are also removed. These changes look large because nobody
actually needs Buffer.h (or Queue.h). It was mostly transitive
dependencies on the massive list of includes in that header, which are
now almost all gone. Instead, we include common things like Sample.h
directly, which should give faster compile times as very few files
actually need Queue.h.
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:
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 '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.
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.
Executing `asctl set r 96000` no longer results in weird sample rates
being set on the audio devices. SB16 checks for a sample rate between 1
and 44100 Hz, while AC97 implements double-rate support which allows
sample rates between 8kHz and 96kHZ.
The new asctl (audio server control) utility expands on avol with a
completely new command line interface (documented in the man page) that
supports retrieving and setting all exposed audio server settings, like
volume and sample rate. This is currently the only user-facing way of
changing the sample rate.
