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 only major functional change is that the Track now needs to know
whether it's active or not, in order to listen to the keyboard (or not).
There are some bugs exposed/created by this, mainly:
* KeysWidget sometimes shows phantom notes. Those do not actually exist
as far as debugging has revealed and do not play in the synth.
* The keyboard can lock up Piano when rapidly pressing keys. This
appears to be a HashMap bug; I invested significant time in bugfixing
but got nowhere.
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:
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.
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.
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.
Piano is an old application that predates AudioServer. For this reason,
it was architected to directly talk to the soundcard via the /dev/audio
device. This caused multiple problems including simultaneous playback
issues, no ability to change volume/mute for Piano and more.
This change moves Piano to use AudioServer like any well-behaved audio
application :^) The track processing and IPC communication is moved to
the main thread because IPC doesn't like multi-threading. For this, the
new AudioPlayerLoop class is utilized that should evolve into the
DSP->AudioServer interface in the future.
Because Piano's CPU utilization has gotten so low (about 3-6%), the UI
update loop is switched back to render at exactly 60fps.
This is an important commit on the road to #6528.
This patch implements a couple of enhancements to the synthesizer
engine:
* Each track has a volume control.
* The input and tooltips for all controls are improved.
* The noise channel is pitched, which allows for basic drum synthesis.
SPDX License Identifiers are a more compact / standardized
way of representing file license information.
See: https://spdx.dev/resources/use/#identifiers
This was done with the `ambr` search and replace tool.
ambr --no-parent-ignore --key-from-file --rep-from-file key.txt rep.txt *
Because it's what it really is. A frame is composed of 1 or more samples, in
the case of SerenityOS 2 (stereo). This will make it less confusing for
future mantainability.
