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 :^)
This doesn't have any immediate uses, but this adapts the code a bit
more to `Core::Stream` conventions (as most functions there use
NonnullOwnPtr to handle streams) and it makes it a bit clearer that this
pointer isn't actually supposed to be null. In fact, MP3LoaderPlugin
and FlacLoaderPlugin apparently forgot to check for that completely
before starting to decode data.
This now prepares all the needed (fallible) components before actually
constructing a LoaderPlugin object, so we are no longer filling them in
at an arbitrary later point in time.
`Bytes` is very slim, so the memory and/or performance gains from
passing it by reference isn't that big, and it passing it by value is
more compatible with xvalues, which is handy for things like
`::try_create(buffer.bytes())`.
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.
This was changed with a recent move to MutexLocker, but the exact
previous behavior is safer as it holds the lock for the minimum amount
of time in both locations. We don't want to introduce these kinds of
subtle bugs.
If this is not done, the event loop pointer will be initialized to
exploded MALLOC_SCRUB_BYTEs and the null pointer check at destruction
time will fail, causing a crash any time an audio client without a
started enqueuer thread exits. With this change, we correctly skip
quitting the event loop both when it was never started (if the enqueuer
thread never ran) as well as if it already exited (if the enqueuer
thread exited fast enough) without additional logic for the two very
different cases.
Otherwise, we end up propagating those dependencies into targets that
link against that library, which creates unnecessary link-time
dependencies.
Also included are changes to readd now missing dependencies to tools
that actually need them.
Also do this for Shell.
This greatly simplifies the CMakeLists in Lagom, replacing many glob
patterns with a big list of libraries. There are still a few special
libraries that need some help to conform to the pattern, like LibELF and
LibWebView.
It also lets us remove essentially all of the Serenity or Lagom binary
directory detection logic from code generators, as now both projects
directories enter the generator logic from the same place.
8-bit PCM samples are unsigned, at least in WAV, so after rescaling them
to the correct range we also need to center them around 0. This fix
should make 8-bit WAVs have the correct volume of double of what it was
before, and also future-proof for all other unsigned PCM sample formats
we may encounter.
This commit does three things atomically:
- switch over Core::Account+SystemServer+LoginServer to sid based socket
names.
- change socket names with %uid to %sid.
- add/update necessary pledges and unveils.
Userland: Switch over servers to sid based sockets
Userland: Properly pledge and unveil for sid based sockets
This patch allows to insert "%uid" in `IPC_CLIENT_CONNECTION`
declaration and in SystemServer's ini files. This pattern is replaced
then replaced by the UID of the owner of the service. It opens a path
for seamlessly managed, per-user portal.
The audio enqueuer thread goes to sleep when there is no more audio data
present, and through normal Core::EventLoop events it can be woken up.
However, that waking up only happens when the thread is not currently
running, so that the wake-up events don't queue up and cause weirdness.
The atomic variable responsible for keeping track of whether the thread
is active can lead to a racy deadlock however, where the audio enqueuer
thread will never wake up again despite there being audio data to
enqueue. Consider this scenario:
- Main thread calls into async_enqueue. It detects that according to the
atomic variable, the other thread is still running, skipping the event
queue wake.
- Enqueuer thread has just finished playing the last chunk of audio and
detects that there is no audio left. It enters the if block with the
dbgln "Reached end of provided audio data..."
- Main thread enqueues audio, making the user sample queue non-empty.
- Enqueuer thread does not check this condition again, instead setting
the atomic variable to indicate that it is not running. It exits into
an event loop sleep.
- Main thread exits async_enqueue. The calling audio enqueuing system
(see e.g. Piano, but all of them function similarly) will wait until
the enqueuer thread has played enough samples before async_enqueue is
called again. However, since the enqueuer thread will never play any
audio, this condition is never fulfilled and audio playback deadlocks
This commit fixes that by allowing the event loop to not enqueue an
event that already exists, therefore overloading the audio enqueuer
event loop by at maximum one message in weird situations. We entirely
get rid of the atomic variable and the race condition is prevented.
This will conflict with apps that don't use this thread and it also
creates unnecessary overhead for non-enqueuing clients like AudioApplet.
Use the new Thread is_started info to start the thread only if necessary
(on first call to async_enqueue).
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.
Error::from_string_literal now takes direct char const*s, while
Error::from_string_view does what Error::from_string_literal used to do:
taking StringViews. This change will remove the need to insert `sv`
after error strings when returning string literal errors once
StringView(char const*) is removed.
No functional changes.
This has been overkill from the start, and it has been bugging me for a
long time. With this change, we're probably a bit slower on most
platforms but save huge amounts of space with all in-memory sample
datastructures.
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.
This completely removes WavLoader's dependency on LegacyBuffer: We
directly create the result sample container and write into it. I took
this opportunity to rewrite most of the sample reading functions as a
single templated function, which combined with the better error handling
makes this "ported" code super concise.
This makes the code much more readable and concise, reduces the size of
the WavLoader class itself, moves almost all fallible initialization out
of the constructor and should provide better error handling in general.
Also, a lot of now-unnecessary imports are removed.
* All clang-tidy warnings fixed except read_header cognitive complexity
* Use size_t in more places
* Replace #define's with constexpr constants
* Some variable renaming for readability
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
