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.
If the underlying parameter is logarithmic, the slider respects that and
switches to a logarithmic display. Currently, the used base is e, and
we'll have to see in practice if 2 or 10 might be better. The parameters
that make use of this, as can be seen in the previous commit, are all of
the time dependent parameters such as the synth envelope parameters, as
with these, usually fine-grained control at small time scales and
coarser control at large time scales is desired.
This was a good opportunity to refactor the slider step count into a
constant.
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.
Adds the ability to add a track and cycle through the
tracks from player widget. Also displays the current track
being played or edited in a dropdown that allows
for quick track selection.
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.
With this change, System::foo() becomes Core::System::foo().
Since LibCore builds on other systems than SerenityOS, we now have to
make sure that wrappers work with just a standard C library underneath.
Almost all synthesizer code in Piano is removed in favor of the LibDSP
reimplementation.
This causes some issues that mainly have to do with the way Piano
currently handles talking to LibDSP. Additionally, the sampler is gone
for now and will be reintroduced with future work.
The processor parameter values are displayed with two decimal places by
default. However, when these values become very large and exceed about 7
text symbols, the text is too long to fit the label and it'll simply not
show up. This commit fixes that by disabling the decimal place for such
large values, which allows us to show values up to 9,999,999, be it
only at integer precision.
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.
"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>
Derivatives of Core::Object should be constructed through
ClassName::construct(), to avoid handling ref-counted objects with
refcount zero. Fixing the visibility means that misuses like this are
more difficult.
This commit is separate from the other Applications/Libraries changes
because it required additional adaption of the code. Note that the old
code did precisely what these changes try to prevent: Create and handle
a ref-counted object with a refcount of zero.
Derivatives of Core::Object should be constructed through
ClassName::construct(), to avoid handling ref-counted objects with
refcount zero. Fixing the visibility means that misuses like this are
more difficult.
Across the entire audio system, audio now works in 0-1 terms instead of
0-100 as before. Therefore, volume is now a double instead of an int.
The master volume of the AudioServer changes smoothly through a
FadingProperty, preventing clicks. Finally, volume computations are done
with logarithmic scaling, which is more natural for the human ear.
Note that this could be 4-5 different commits, but as they change each
other's code all the time, it makes no sense to split them up.
And also try_create<T> => try_make_ref_counted<T>.
A global "create" was a bit much. The new name matches make<T> better,
which we've used for making single-owner objects since forever.
This is the first step in transitioning Piano to a full LibDSP backend.
For now, the delay effect is replaced with a (mostly identical)
implementation in LibDSP.
The new ProcessorParameterSlider attaches to a LibDSP::Processor's
range parameter (LibDSP::ProcessorRangeParameter) and changes it
automatically. It also has the ability to update an external GUI::Label.
This is used for the three delay parameters and it will become useful
for auto-generating UI for Processors.
As Piano will later move to the RollNote defintions of LibDSP, it's a
good idea to already insert velocity and pitch support, even though it's
currently not used.
1) The Sound Player visualizer couldn't deal with small sample buffers,
which occur on low sample rates. Now, it simply doesn't update its
buffer, meaning the display is broken on low sample rates. I'm not too
familiar with the visualizer to figure out a proper fix for now, but
this mitigates the issue (and "normal" sample rates still work).
2) Piano wouldn't buffer enough samples for small sample rates, so the
sample count per buffer is now increased to 2^12, introducing minor
amounts of (acceptable) lag.
All audio applications (aplay, Piano, Sound Player) respect the ability
of the system to have theoretically any sample rate. Therefore, they
resample their own audio into the system sample rate.
LibAudio previously had its loaders resample their own audio, even
though they expose their sample rate. This is now changed. The loaders
output audio data in their file's sample rate, which the user has to
query and resample appropriately. Resampling code from Buffer, WavLoader
and FlacLoader is removed.
Note that these applications only check the sample rate at startup,
which is reasonable (the user has to restart applications when changing
the sample rate). Fully dynamic adaptation could both lead to errors and
will require another IPC interface. This seems to be enough for now.
This allows for typing [8] instead of [8, 8, 8, 8] to specify the same
margin on all edges, for example. The constructors follow CSS' style of
specifying margins. The added constructors are:
- Margins(int all): Sets the same margin on all edges.
- Margins(int vertical, int horizontal): Sets the first argument to top
and bottom margins, and the second argument to left and right margins.
- Margins(int top, int vertical, int bottom): Sets the first argument to
the top margin, the second argument to the left and right margins,
and the third argument to the bottom margin.
Applications previously had to create a GUI::Menubar object, add menus
to it, and then call GUI::Window::set_menubar().
This patch introduces GUI::Window::add_menu() which creates the menubar
automatically and adds items to it. Application code becomes slightly
simpler as a result. :^)
