This change was a long time in the making ever since we obtained sample
rate awareness in the system. Now, each client has its own sample rate,
accessible via new IPC APIs, and the device sample rate is only
accessible via the management interface. AudioServer takes care of
resampling client streams into the device sample rate. Therefore, the
main improvement introduced with this commit is full responsiveness to
sample rate changes; all open audio programs will continue to play at
correct speed with the audio resampled to the new device rate.
The immediate benefits are manifold:
- Gets rid of the legacy hardware sample rate IPC message in the
non-managing client
- Removes duplicate resampling and sample index rescaling code
everywhere
- Avoids potential sample index scaling bugs in SoundPlayer (which have
happened many times before) and fixes a sample index scaling bug in
aplay
- Removes several FIXMEs
- Reduces amount of sample copying in all applications (especially
Piano, where this is critical), improving performance
- Reduces number of resampling users, making future API changes (which
will need to happen for correct resampling to be implemented) easier
I also threw in a simple race condition fix for Piano's audio player
loop.
Previously when pausing or stoping a track in SoundPlayer,
the visualizations would glitch out. This was caused by them
being updated with the same buffer over and over.
With this patch a pause action will freeze the visualization at the
point of the pause and a stop action will reset the visualization
so it displays nothing until a track is started.
The current sample count is already reported like that, so this fixes a
mismatch between current and total. In the future, we should look into
abstracting this away properly instead of requiring the user to think of
converting it manually everywhere.
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 :^)
When the visualization is set to "Album Cover", the player will now try
to load the embedded image. On failure, it defaults to a "Cover" image
file in the directory.
In Player::play_file_path, file_name_changed now needs to be executed
after that the loader have been set, to get the correct image.
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.
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 fix syncs up the AudioPlayer's internal state for showing
playlist information with the AudioPlayer's GUI. Before, if the
AudioPlayer was opened with a playlist file (.m3u or .m3u8) it would
automatically show the playlist information in the GUI and set the
loop mode to playlist, but the menu options would be unchecked. In
order to hide the playlist information, the menu option would then
have to be toggled twice -- once on and again off.
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.
The path returned by GUI:FilePicker is stored on the stack when the
callback is executed. The player only stored a StringView to the path
however it should take ownership of the path instead since the path is
accessed even after the file menu open action has returned.
The shuffling algorithm uses a naïve bloom filter to provide random
uniformity, avoiding items that were recently played. With 32 bits,
double hashing, and an error rate of ~10%, this bloom filter should
be able to hold around ~16 keys, which should be sufficient to give the
illusion of fairness to the shuffling algorithm.
This avoids having to shuffle the playlist itself (user might have
spent quite a bit of time to sort them, so it's not a good idea to mess
with it), or having to create a proxy model that shuffles (that could
potentially use quite a bit of memory).
This is a first pass at refactoring SoundPlayer so that the View widget
is decoupled from the player itself.
In doing so, this fixed a couple of issues, including possibly
inconsistent states (e.g. player could be paused and stopped at the
same time).
With the change, Player actually controls the show, and calls methods
overriden by its subclasses to perform actions, such as update the Seek
bar; the hard work of massaging the raw data is done by the Player
class, so subclasses don't need to reimplement any of these things.
This also removes some copies of playlist management code that happened
to be copied+pasted inside callbacks of buttons -- it now lives inside
a neatly packaged Playlist class, and the Player only asks for the next
song to play.
In addition, the menu bar has been slightly rearranged.
