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:
Previously netstat would print the whole line of an ip address or
resolved hostname. If the hostname was longer than the address column
length, it would push following columns into disaligned output.
This sets the default behavior to truncate any IP address or symbolic
hostname that is larger than the maximum address column size to provide
cleaner output. In the event the user wishes to see the whole address
name, they can then pass the wide option that will output as wide as
necessary to print the whole name.
It's a bit cleaner to just rely on AK/IPv4Address' ability to determine
the validity of the given input. If a valid IP address is not returned,
then input will be processed as a hostname.
This mainly does two things,
1. Removes spaces after commas
2. Elides "0x" and leading zeros in most contexts
Remaining differences are:
1. objdump always has memory size annotations
We lack these and probably have some annotations wrong
2. Boolean check names
We use jump-zero, while objdump uses jump-equal for example
3. We sometimes add "00 00" symbols, which objdump elides
4. We always demangle (This is a good thing)
5. We always resolve relocations (This is a good thing)
6. We seem to detect some symbols differently/incorrectly
This prints 7 instruction bytes per line, which is enough for most
x86-64 instructions (rex+opcode+mod/rm+imm32) and is also what
objdump uses.
Co-authored-by: Simon Wanner <skyrising@pvpctutorials.de>
