Commit graph

7 commits

Author SHA1 Message Date
Tim Schumacher
d43a7eae54 LibCore: Rename File to DeprecatedFile
As usual, this removes many unused includes and moves used includes
further down the chain.
2023-02-13 00:50:07 +00:00
Andrew Kaster
a492e2018d Userland: Silence warnings from ElapsedTimer::elapsed() type change
We changed elapsed() to return i64 instead of int as that's what
AK::Time::to_milliseconds() returns, causing a bunch of implicit lossy
conversions in callers. Clean those up with a mix of type changes and
casts.
2023-01-07 14:51:04 +01:00
sin-ack
3f3f45580a Everywhere: Add sv suffix to strings relying on StringView(char const*)
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.
2022-07-12 23:11:35 +02:00
sin-ack
60f6bc902b Userland: Convert command line arguments to String/StringView
StringView was used where possible. Some utilities still use libc
functions which expect null-terminated strings, so String objects were
used there instead.
2022-07-12 23:11:35 +02:00
kleines Filmröllchen
49b087f3cd LibAudio+Userland: Use new audio queue in client-server communication
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:
2022-04-21 13:55:00 +02:00
Brian Gianforcaro
af3751e4dd Utilities: Use default execpromises parameter to pledge(..) 2022-04-03 17:13:51 -07:00
kleines Filmröllchen
9c40311622 Utilites: Add abench utility
abench (audio benchmark) is an audio benchmarking utility that allows
testing decoder performance.
2021-11-28 13:33:51 -08:00