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serenity/Userland/Services/AudioServer/Mixer.cpp
Jelle Raaijmakers 5c64686666 Kernel+AudioServer: Use interrupts for Intel HDA audio buffer completion
We used to not care about stopping an audio output stream for Intel HDA
since AudioServer would continuously send new buffers to play. Since
707f5ac150ef858760eb9faa52b9ba80c50c4262 however, that has changed.

Intel HDA now uses interrupts to detect when each buffer was completed
by the device, and uses a simple heuristic to detect whether a buffer
underrun has occurred so it can stop the output stream.

This was tested on Qemu's Intel HDA (Linux x86_64) and a bare metal MSI
Starship/Matisse HD Audio Controller.
2023-07-04 00:05:34 +02:00

195 lines
6.4 KiB
C++

/*
* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
* Copyright (c) 2021, kleines Filmröllchen <filmroellchen@serenityos.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "Mixer.h"
#include <AK/Array.h>
#include <AK/Format.h>
#include <AK/MemoryStream.h>
#include <AK/NumericLimits.h>
#include <AudioServer/ConnectionFromClient.h>
#include <AudioServer/ConnectionFromManagerClient.h>
#include <AudioServer/Mixer.h>
#include <LibCore/ConfigFile.h>
#include <LibCore/Timer.h>
#include <pthread.h>
#include <sys/ioctl.h>
namespace AudioServer {
Mixer::Mixer(NonnullRefPtr<Core::ConfigFile> config, NonnullOwnPtr<Core::File> device)
: m_device(move(device))
, m_sound_thread(Threading::Thread::construct(
[this] {
mix();
return 0;
},
"AudioServer[mixer]"sv))
, m_config(move(config))
{
m_muted = m_config->read_bool_entry("Master", "Mute", false);
m_main_volume = static_cast<double>(m_config->read_num_entry("Master", "Volume", 100)) / 100.0;
m_sound_thread->start();
}
NonnullRefPtr<ClientAudioStream> Mixer::create_queue(ConnectionFromClient& client)
{
auto queue = adopt_ref(*new ClientAudioStream(client));
queue->set_sample_rate(audiodevice_get_sample_rate());
{
Threading::MutexLocker const locker(m_pending_mutex);
m_pending_mixing.append(*queue);
}
// Signal the mixer thread to start back up, in case nobody was connected before.
m_mixing_necessary.signal();
return queue;
}
void Mixer::mix()
{
decltype(m_pending_mixing) active_mix_queues;
for (;;) {
{
Threading::MutexLocker const locker(m_pending_mutex);
// While we have nothing to mix, wait on the condition.
m_mixing_necessary.wait_while([this, &active_mix_queues]() { return m_pending_mixing.is_empty() && active_mix_queues.is_empty(); });
if (!m_pending_mixing.is_empty()) {
active_mix_queues.extend(move(m_pending_mixing));
m_pending_mixing.clear();
}
}
active_mix_queues.remove_all_matching([&](auto& entry) { return !entry->is_connected(); });
Array<Audio::Sample, HARDWARE_BUFFER_SIZE> mixed_buffer;
m_main_volume.advance_time();
// Mix the buffers together into the output
for (auto& queue : active_mix_queues) {
if (!queue->client()) {
queue->clear();
continue;
}
queue->volume().advance_time();
for (auto& mixed_sample : mixed_buffer) {
Audio::Sample sample;
if (!queue->get_next_sample(sample, audiodevice_get_sample_rate()))
break;
if (queue->is_muted())
continue;
sample.log_multiply(SAMPLE_HEADROOM);
sample.log_multiply(static_cast<float>(queue->volume()));
mixed_sample += sample;
}
}
// Even though it's not realistic, the user expects no sound at 0%.
if (m_muted || m_main_volume < 0.01) {
m_device->write_until_depleted(m_zero_filled_buffer).release_value_but_fixme_should_propagate_errors();
} else {
FixedMemoryStream stream { m_stream_buffer.span() };
for (auto& mixed_sample : mixed_buffer) {
mixed_sample.log_multiply(static_cast<float>(m_main_volume));
mixed_sample.clip();
LittleEndian<i16> out_sample;
out_sample = static_cast<i16>(mixed_sample.left * NumericLimits<i16>::max());
MUST(stream.write_value(out_sample));
out_sample = static_cast<i16>(mixed_sample.right * NumericLimits<i16>::max());
MUST(stream.write_value(out_sample));
}
auto buffered_bytes = MUST(stream.tell());
VERIFY(buffered_bytes == m_stream_buffer.size());
m_device->write_until_depleted({ m_stream_buffer.data(), buffered_bytes })
.release_value_but_fixme_should_propagate_errors();
}
}
}
void Mixer::set_main_volume(double volume)
{
if (volume < 0)
m_main_volume = 0;
else if (volume > 2)
m_main_volume = 2;
else
m_main_volume = volume;
m_config->write_num_entry("Master", "Volume", static_cast<int>(volume * 100));
request_setting_sync();
ConnectionFromManagerClient::for_each([&](auto& client) {
client.did_change_main_mix_volume({}, main_volume());
});
}
void Mixer::set_muted(bool muted)
{
if (m_muted == muted)
return;
m_muted = muted;
m_config->write_bool_entry("Master", "Mute", m_muted);
request_setting_sync();
ConnectionFromManagerClient::for_each([muted](auto& client) {
client.did_change_main_mix_muted_state({}, muted);
});
}
int Mixer::audiodevice_set_sample_rate(u32 sample_rate)
{
int code = ioctl(m_device->fd(), SOUNDCARD_IOCTL_SET_SAMPLE_RATE, sample_rate);
if (code != 0)
dbgln("Error while setting sample rate to {}: ioctl error: {}", sample_rate, strerror(errno));
// Note that the effective sample rate may be different depending on device restrictions.
// Therefore, we delete our cache, but for efficency don't immediately read the sample rate back.
m_cached_sample_rate = {};
return code;
}
u32 Mixer::audiodevice_get_sample_rate() const
{
if (m_cached_sample_rate.has_value())
return m_cached_sample_rate.value();
u32 sample_rate = 0;
int code = ioctl(m_device->fd(), SOUNDCARD_IOCTL_GET_SAMPLE_RATE, &sample_rate);
if (code != 0)
dbgln("Error while getting sample rate: ioctl error: {}", strerror(errno));
else
m_cached_sample_rate = sample_rate;
return sample_rate;
}
void Mixer::request_setting_sync()
{
if (m_config_write_timer.is_null() || !m_config_write_timer->is_active()) {
m_config_write_timer = Core::Timer::create_single_shot(
AUDIO_CONFIG_WRITE_INTERVAL,
[this] {
if (auto result = m_config->sync(); result.is_error())
dbgln("Failed to write audio mixer config: {}", result.error());
},
this)
.release_value_but_fixme_should_propagate_errors();
m_config_write_timer->start();
}
}
ClientAudioStream::ClientAudioStream(ConnectionFromClient& client)
: m_client(client)
{
}
}