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Work on AudioServer

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The center of this is now an ABuffer class in LibAudio.
ABuffer contains ASample, which has two channels (left/right) in
floating point for mixing purposes, in 44100hz.

This means that the loaders (AWavLoader in this case) needs to do some
manipulation to get things in the right format, but that we don't need
to care after format loading is done.

While we're at it, do some correctness fixes. PCM data is unsigned if
it's 8 bit, but 16 bit is signed. And /dev/audio also wants signed 16
bit audio, so give it what it wants.

On top of this, AudioServer now accepts requests to play a buffer.
The IPC mechanism here is pretty much a 1:1 copy-paste from
LibGUI/WindowServer. It can be generalized more in the future, but for
now I want to get AudioServer working decently first :)

Additionally, add a little "aplay" tool to load and play a WAV file. It
will break with large WAVs (run out of memory, heh...) but it's a start.

Future work needs to make AudioServer block buffer submission from
clients until it has played the buffer they are requesting to play.
This commit is contained in:
Robin Burchell 2019-07-15 12:54:52 +02:00 • committed by Andreas Kling
parent 3db9706e57
commit 2df6f0e87f
19 changed files with 873 additions and 141 deletions
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37
Servers/AudioServer/ASEventLoop.cpp Normal file
View file

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#include "ASEventLoop.h"
#include "ASClientConnection.h"
#include <sys/types.h>
#include <sys/socket.h>
#include <unistd.h>
ASEventLoop::ASEventLoop()
{
unlink("/tmp/asportal");
sockaddr_un address;
address.sun_family = AF_LOCAL;
strcpy(address.sun_path, "/tmp/asportal");
int rc = bind(m_server_sock.fd(), (const sockaddr*)&address, sizeof(address));
ASSERT(rc == 0);
rc = listen(m_server_sock.fd(), 5);
ASSERT(rc == 0);
m_server_notifier = make<CNotifier>(m_server_sock.fd(), CNotifier::Read);
m_server_notifier->on_ready_to_read = [this] { drain_server(); };
}
void ASEventLoop::drain_server()
{
sockaddr_un address;
socklen_t address_size = sizeof(address);
int client_fd = accept(m_server_sock.fd(), (sockaddr*)&address, &address_size);
if (client_fd < 0) {
dbgprintf("AudioServer: accept() failed: %s\n", strerror(errno));
} else {
dbgprintf("AudioServer: accept()ed client %d\n", client_fd);
static int s_next_client_id = 0;
new ASClientConnection(client_fd, s_next_client_id++, m_mixer);
}
}