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