ladybird/Userland/Libraries/LibAudio/ConnectionToServer.cpp
Zaggy1024 bb156f8133 LibAudio: Add a Serenity implementation of PlaybackStream
This implementation is very naive compared to the PulseAudio one.

Instead of using a callback implemented by the audio server connection
to push audio to the buffer, we have to poll on a timer to check when
we need to push the audio buffers. Implementing cross-process condition
variables into the audio queue class could allow us to avoid polling,
which may prove beneficial to CPU usage.

Audio timestamps will be accurate to the number of samples available,
but will count in increments of about 100ms and run ahead of the actual
audio being pushed to the device by the server.

Buffer underruns are completely ignored for now as well, since the
`AudioServer` has no way to know how many samples are actually written
in a single audio buffer.
2023-08-29 11:31:10 +02:00

165 lines
5.1 KiB
C++

/*
* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
* Copyright (c) 2022, kleines Filmröllchen <filmroellchen@serenityos.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include <AK/Atomic.h>
#include <AK/Debug.h>
#include <AK/Format.h>
#include <AK/OwnPtr.h>
#include <AK/Time.h>
#include <AK/Types.h>
#include <LibAudio/ConnectionToServer.h>
#include <LibAudio/Queue.h>
#include <LibAudio/UserSampleQueue.h>
#include <LibCore/Event.h>
#include <LibThreading/Mutex.h>
#include <Userland/Services/AudioServer/AudioClientEndpoint.h>
#include <sched.h>
#include <time.h>
namespace Audio {
ConnectionToServer::ConnectionToServer(NonnullOwnPtr<Core::LocalSocket> socket)
: IPC::ConnectionToServer<AudioClientEndpoint, AudioServerEndpoint>(*this, move(socket))
, m_buffer(make<AudioQueue>(MUST(AudioQueue::create())))
, m_user_queue(make<UserSampleQueue>())
, m_background_audio_enqueuer(Threading::Thread::construct([this]() {
// All the background thread does is run an event loop.
Core::EventLoop enqueuer_loop;
m_enqueuer_loop = &enqueuer_loop;
enqueuer_loop.exec();
{
Threading::MutexLocker const locker(m_enqueuer_loop_destruction);
m_enqueuer_loop = nullptr;
}
return (intptr_t) nullptr;
}))
{
update_good_sleep_time();
async_pause_playback();
set_buffer(*m_buffer);
}
ConnectionToServer::~ConnectionToServer()
{
die();
}
void ConnectionToServer::die()
{
{
Threading::MutexLocker const locker(m_enqueuer_loop_destruction);
// We're sometimes getting here after the other thread has already exited and its event loop does no longer exist.
if (m_enqueuer_loop != nullptr) {
m_enqueuer_loop->wake();
m_enqueuer_loop->quit(0);
}
}
if (m_background_audio_enqueuer->is_started())
(void)m_background_audio_enqueuer->join();
}
ErrorOr<void> ConnectionToServer::async_enqueue(FixedArray<Sample>&& samples)
{
if (!m_background_audio_enqueuer->is_started()) {
m_background_audio_enqueuer->start();
// Wait until the enqueuer has constructed its loop. A pseudo-spinlock is fine since this happens as soon as the other thread gets scheduled.
while (!m_enqueuer_loop)
usleep(1);
TRY(m_background_audio_enqueuer->set_priority(THREAD_PRIORITY_MAX));
}
m_user_queue->append(move(samples));
// Wake the background thread to make sure it starts enqueuing audio.
m_enqueuer_loop->post_event(*this, make<Core::CustomEvent>(0));
m_enqueuer_loop->wake();
async_start_playback();
return {};
}
void ConnectionToServer::clear_client_buffer()
{
m_user_queue->clear();
}
void ConnectionToServer::update_good_sleep_time()
{
auto sample_rate = static_cast<double>(get_self_sample_rate());
auto buffer_play_time_ns = 1'000'000'000.0 / (sample_rate / static_cast<double>(AUDIO_BUFFER_SIZE));
// A factor of 1 should be good for now.
m_good_sleep_time = Duration::from_nanoseconds(static_cast<unsigned>(buffer_play_time_ns)).to_timespec();
}
void ConnectionToServer::set_self_sample_rate(u32 sample_rate)
{
IPC::ConnectionToServer<AudioClientEndpoint, AudioServerEndpoint>::set_self_sample_rate(sample_rate);
update_good_sleep_time();
}
// Non-realtime audio writing loop
void ConnectionToServer::custom_event(Core::CustomEvent&)
{
Array<Sample, AUDIO_BUFFER_SIZE> next_chunk;
while (true) {
if (m_user_queue->is_empty()) {
dbgln_if(AUDIO_DEBUG, "Reached end of provided audio data, going to sleep");
break;
}
auto available_samples = min(AUDIO_BUFFER_SIZE, m_user_queue->size());
for (size_t i = 0; i < available_samples; ++i)
next_chunk[i] = (*m_user_queue)[i];
m_user_queue->discard_samples(available_samples);
// FIXME: Could we receive interrupts in a good non-IPC way instead?
auto result = m_buffer->blocking_enqueue(next_chunk, [this]() {
nanosleep(&m_good_sleep_time, nullptr);
});
if (result.is_error())
dbgln("Error while writing samples to shared buffer: {}", result.error());
}
}
ErrorOr<void, AudioQueue::QueueStatus> ConnectionToServer::realtime_enqueue(Array<Sample, AUDIO_BUFFER_SIZE> samples)
{
return m_buffer->enqueue(samples);
}
ErrorOr<void> ConnectionToServer::blocking_realtime_enqueue(Array<Sample, AUDIO_BUFFER_SIZE> samples, Function<void()> wait_function)
{
return m_buffer->blocking_enqueue(samples, move(wait_function));
}
unsigned ConnectionToServer::total_played_samples() const
{
return m_buffer->weak_tail() * AUDIO_BUFFER_SIZE;
}
unsigned ConnectionToServer::remaining_samples()
{
return static_cast<unsigned>(m_user_queue->remaining_samples());
}
size_t ConnectionToServer::remaining_buffers() const
{
return m_buffer->size() - m_buffer->weak_remaining_capacity();
}
bool ConnectionToServer::can_enqueue() const
{
return m_buffer->can_enqueue();
}
void ConnectionToServer::client_volume_changed(double volume)
{
if (on_client_volume_change)
on_client_volume_change(volume);
}
}