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Rework audio output, connecting AudioOut into coretiming to fix desync during heavy loads.

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This commit is contained in:
Kelebek1 2022-08-01 02:58:13 +01:00
parent a83a5d2e4c
commit ea9ff71725
23 changed files with 550 additions and 841 deletions
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1
src/audio_core/CMakeLists.txt
View File

@ -194,6 +194,7 @@ add_library(audio_core STATIC
sink/sink.h
sink/sink_details.cpp
sink/sink_details.h
sink/sink_stream.cpp
sink/sink_stream.h
)

8
src/audio_core/audio_core.cpp
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@ -57,12 +57,4 @@ void AudioCore::PauseSinks(const bool pausing) const {
}
}
u32 AudioCore::GetStreamQueue() const {
return estimated_queue.load();
}
void AudioCore::SetStreamQueue(u32 size) {
estimated_queue.store(size);
}
} // namespace AudioCore

16
src/audio_core/audio_core.h
View File

@ -65,20 +65,6 @@ public:
*/
void PauseSinks(bool pausing) const;
/**
* Get the size of the current stream queue.
*
* @return Current stream queue size.
*/
u32 GetStreamQueue() const;
/**
* Get the size of the current stream queue.
*
* @param size - New stream size.
*/
void SetStreamQueue(u32 size);
private:
/**
* Create the sinks on startup.
@ -93,8 +79,6 @@ private:
std::unique_ptr<Sink::Sink> input_sink;
/// The ADSP in the sysmodule
std::unique_ptr<AudioRenderer::ADSP::ADSP> adsp;
/// Current size of the stream queue
std::atomic<u32> estimated_queue{0};
};
} // namespace AudioCore

4
src/audio_core/device/audio_buffer.h
View File

@ -8,6 +8,10 @@
namespace AudioCore {
struct AudioBuffer {
/// Timestamp this buffer started playing.
u64 start_timestamp;
/// Timestamp this buffer should finish playing.
u64 end_timestamp;
/// Timestamp this buffer completed playing.
s64 played_timestamp;
/// Game memory address for these samples.

13
src/audio_core/device/audio_buffers.h
View File

@ -58,6 +58,7 @@ public:
if (index < 0) {
index += N;
}
out_buffers.push_back(buffers[index]);
registered_count++;
registered_index = (registered_index + 1) % append_limit;
@ -100,7 +101,7 @@ public:
}
// Check with the backend if this buffer can be released yet.
if (!session.IsBufferConsumed(buffers[index].tag)) {
if (!session.IsBufferConsumed(buffers[index])) {
break;
}
@ -280,6 +281,16 @@ public:
return true;
}
u64 GetNextTimestamp() const {
// Iterate backwards through the buffer queue, and take the most recent buffer's end
std::scoped_lock l{lock};
auto index{appended_index - 1};
if (index < 0) {
index += append_limit;
}
return buffers[index].end_timestamp;
}
private:
/// Buffer lock
mutable std::recursive_mutex lock{};

52
src/audio_core/device/device_session.cpp
View File

@ -7,11 +7,20 @@
#include "audio_core/device/device_session.h"
#include "audio_core/sink/sink_stream.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/memory.h"
namespace AudioCore {
DeviceSession::DeviceSession(Core::System& system_) : system{system_} {}
using namespace std::literals;
constexpr auto INCREMENT_TIME{5ms};
DeviceSession::DeviceSession(Core::System& system_)
: system{system_}, thread_event{Core::Timing::CreateEvent(
"AudioOutSampleTick",
[this](std::uintptr_t, s64 time, std::chrono::nanoseconds) {
return ThreadFunc();
})} {}
DeviceSession::~DeviceSession() {
Finalize();
@ -50,20 +59,21 @@ void DeviceSession::Finalize() {
}
void DeviceSession::Start() {
stream->SetPlayedSampleCount(played_sample_count);
stream->Start();
if (stream) {
stream->Start();
system.CoreTiming().ScheduleLoopingEvent(std::chrono::nanoseconds::zero(), INCREMENT_TIME,
thread_event);
}
}
void DeviceSession::Stop() {
if (stream) {
played_sample_count = stream->GetPlayedSampleCount();
stream->Stop();
system.CoreTiming().UnscheduleEvent(thread_event, {});
}
}
void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
auto& memory{system.Memory()};
for (size_t i = 0; i < buffers.size(); i++) {
Sink::SinkBuffer new_buffer{
.frames = buffers[i].size / (channel_count * sizeof(s16)),
@ -77,7 +87,7 @@ void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
stream->AppendBuffer(new_buffer, samples);
} else {
std::vector<s16> samples(buffers[i].size / sizeof(s16));
memory.ReadBlockUnsafe(buffers[i].samples, samples.data(), buffers[i].size);
system.Memory().ReadBlockUnsafe(buffers[i].samples, samples.data(), buffers[i].size);
stream->AppendBuffer(new_buffer, samples);
}
}
@ -85,17 +95,13 @@ void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
void DeviceSession::ReleaseBuffer(AudioBuffer& buffer) const {
if (type == Sink::StreamType::In) {
auto& memory{system.Memory()};
auto samples{stream->ReleaseBuffer(buffer.size / sizeof(s16))};
memory.WriteBlockUnsafe(buffer.samples, samples.data(), buffer.size);
system.Memory().WriteBlockUnsafe(buffer.samples, samples.data(), buffer.size);
}
}
bool DeviceSession::IsBufferConsumed(u64 tag) const {
if (stream) {
return stream->IsBufferConsumed(tag);
}
return true;
bool DeviceSession::IsBufferConsumed(AudioBuffer& buffer) const {
return played_sample_count >= buffer.end_timestamp;
}
void DeviceSession::SetVolume(f32 volume) const {
@ -105,10 +111,22 @@ void DeviceSession::SetVolume(f32 volume) const {
}
u64 DeviceSession::GetPlayedSampleCount() const {
if (stream) {
return stream->GetPlayedSampleCount();
return played_sample_count;
}
std::optional<std::chrono::nanoseconds> DeviceSession::ThreadFunc() {
// Add 5ms of samples at a 48K sample rate.
played_sample_count += 48'000 * INCREMENT_TIME / 1s;
if (type == Sink::StreamType::Out) {
system.AudioCore().GetAudioManager().SetEvent(Event::Type::AudioOutManager, true);
} else {
system.AudioCore().GetAudioManager().SetEvent(Event::Type::AudioInManager, true);
}
return 0;
return std::nullopt;
}
void DeviceSession::SetRingSize(u32 ring_size) {
stream->SetRingSize(ring_size);
}
} // namespace AudioCore

27
src/audio_core/device/device_session.h
View File

@ -3,6 +3,9 @@
#pragma once
#include <chrono>
#include <memory>
#include <optional>
#include <span>
#include "audio_core/common/common.h"
@ -11,9 +14,13 @@
namespace Core {
class System;
}
namespace Timing {
struct EventType;
} // namespace Timing
} // namespace Core
namespace AudioCore {
namespace Sink {
class SinkStream;
struct SinkBuffer;
@ -70,7 +77,7 @@ public:
* @param tag - Unqiue tag of the buffer to check.
* @return true if the buffer has been consumed, otherwise false.
*/
bool IsBufferConsumed(u64 tag) const;
bool IsBufferConsumed(AudioBuffer& buffer) const;
/**
* Start this device session, starting the backend stream.
@ -96,6 +103,16 @@ public:
*/
u64 GetPlayedSampleCount() const;
/*
* CoreTiming callback to increment played_sample_count over time.
*/
std::optional<std::chrono::nanoseconds> ThreadFunc();
/*
* Set the size of the ring buffer.
*/
void SetRingSize(u32 ring_size);
private:
/// System
Core::System& system;
@ -118,9 +135,13 @@ private:
/// Applet resource user id of this device session
u64 applet_resource_user_id{};
/// Total number of samples played by this device session
u64 played_sample_count{};
std::atomic<u64> played_sample_count{};
/// Event increasing the played sample count every 5ms
std::shared_ptr<Core::Timing::EventType> thread_event;
/// Is this session initialised?
bool initialized{};
/// Buffer queue
std::vector<AudioBuffer> buffer_queue{};
};
} // namespace AudioCore

10
src/audio_core/in/audio_in_system.cpp
View File

@ -93,6 +93,7 @@ Result System::Start() {
std::vector<AudioBuffer> buffers_to_flush{};
buffers.RegisterBuffers(buffers_to_flush);
session->AppendBuffers(buffers_to_flush);
session->SetRingSize(static_cast<u32>(buffers_to_flush.size()));
return ResultSuccess;
}
@ -112,8 +113,13 @@ bool System::AppendBuffer(const AudioInBuffer& buffer, const u64 tag) {
return false;
}
AudioBuffer new_buffer{
.played_timestamp = 0, .samples = buffer.samples, .tag = tag, .size = buffer.size};
const auto timestamp{buffers.GetNextTimestamp()};
AudioBuffer new_buffer{.start_timestamp = timestamp,
.end_timestamp = timestamp + buffer.size / (channel_count * sizeof(s16)),
.played_timestamp = 0,
.samples = buffer.samples,
.tag = tag,
.size = buffer.size};
buffers.AppendBuffer(new_buffer);
RegisterBuffers();

10
src/audio_core/out/audio_out_system.cpp
View File

@ -92,6 +92,7 @@ Result System::Start() {
std::vector<AudioBuffer> buffers_to_flush{};
buffers.RegisterBuffers(buffers_to_flush);
session->AppendBuffers(buffers_to_flush);
session->SetRingSize(static_cast<u32>(buffers_to_flush.size()));
return ResultSuccess;
}
@ -111,8 +112,13 @@ bool System::AppendBuffer(const AudioOutBuffer& buffer, u64 tag) {
return false;
}
AudioBuffer new_buffer{
.played_timestamp = 0, .samples = buffer.samples, .tag = tag, .size = buffer.size};
const auto timestamp{buffers.GetNextTimestamp()};
AudioBuffer new_buffer{.start_timestamp = timestamp,
.end_timestamp = timestamp + buffer.size / (channel_count * sizeof(s16)),
.played_timestamp = 0,
.samples = buffer.samples,
.tag = tag,
.size = buffer.size};
buffers.AppendBuffer(new_buffer);
RegisterBuffers();

9
src/audio_core/renderer/adsp/audio_renderer.cpp
View File

@ -106,9 +106,6 @@ void AudioRenderer::Start(AudioRenderer_Mailbox* mailbox_) {
mailbox = mailbox_;
thread = std::thread(&AudioRenderer::ThreadFunc, this);
for (auto& stream : streams) {
stream->Start();
}
running = true;
}
@ -130,6 +127,7 @@ void AudioRenderer::CreateSinkStreams() {
std::string name{fmt::format("ADSP_RenderStream-{}", i)};
streams[i] =
sink.AcquireSinkStream(system, channels, name, ::AudioCore::Sink::StreamType::Render);
streams[i]->SetRingSize(4);
}
}
@ -198,11 +196,6 @@ void AudioRenderer::ThreadFunc() {
command_list_processor.Process(index) - start_time;
}
if (index == 0) {
auto stream{command_list_processor.GetOutputSinkStream()};
system.AudioCore().SetStreamQueue(stream->GetQueueSize());
}
const auto end_time{system.CoreTiming().GetClockTicks()};
command_buffer.remaining_command_count =

12
src/audio_core/renderer/behavior/behavior_info.cpp
View File

@ -43,13 +43,15 @@ void BehaviorInfo::AppendError(ErrorInfo& error) {
}
void BehaviorInfo::CopyErrorInfo(std::span<ErrorInfo> out_errors, u32& out_count) {
auto error_count_{std::min(error_count, MaxErrors)};
std::memset(out_errors.data(), 0, MaxErrors * sizeof(ErrorInfo));
out_count = std::min(error_count, MaxErrors);
for (size_t i = 0; i < error_count_; i++) {
out_errors[i] = errors[i];
for (size_t i = 0; i < MaxErrors; i++) {
if (i < out_count) {
out_errors[i] = errors[i];
} else {
out_errors[i] = {};
}
}
out_count = error_count_;
}
void BehaviorInfo::UpdateFlags(const Flags flags_) {

4
src/audio_core/renderer/command/sink/device.cpp
View File

@ -46,6 +46,10 @@ void DeviceSinkCommand::Process(const ADSP::CommandListProcessor& processor) {
out_buffer.tag = reinterpret_cast<u64>(samples.data());
stream->AppendBuffer(out_buffer, samples);
if (stream->IsPaused()) {
stream->Start();
}
}
bool DeviceSinkCommand::Verify(const ADSP::CommandListProcessor& processor) {

35
src/audio_core/renderer/system_manager.cpp
View File

@ -15,8 +15,7 @@ MICROPROFILE_DEFINE(Audio_RenderSystemManager, "Audio", "Render System Manager",
MP_RGB(60, 19, 97));
namespace AudioCore::AudioRenderer {
constexpr std::chrono::nanoseconds BaseRenderTime{5'000'000UL};
constexpr std::chrono::nanoseconds RenderTimeOffset{400'000UL};
constexpr std::chrono::nanoseconds RENDER_TIME{5'000'000UL};
SystemManager::SystemManager(Core::System& core_)
: core{core_}, adsp{core.AudioCore().GetADSP()}, mailbox{adsp.GetRenderMailbox()},
@ -36,8 +35,8 @@ bool SystemManager::InitializeUnsafe() {
if (adsp.Start()) {
active = true;
thread = std::jthread([this](std::stop_token stop_token) { ThreadFunc(); });
core.CoreTiming().ScheduleLoopingEvent(std::chrono::nanoseconds(0),
BaseRenderTime - RenderTimeOffset, thread_event);
core.CoreTiming().ScheduleLoopingEvent(std::chrono::nanoseconds(0), RENDER_TIME,
thread_event);
}
}
@ -121,35 +120,9 @@ void SystemManager::ThreadFunc() {
}
std::optional<std::chrono::nanoseconds> SystemManager::ThreadFunc2(s64 time) {
std::optional<std::chrono::nanoseconds> new_schedule_time{std::nullopt};
const auto queue_size{core.AudioCore().GetStreamQueue()};
switch (state) {
case StreamState::Filling:
if (queue_size >= 5) {
new_schedule_time = BaseRenderTime;
state = StreamState::Steady;
}
break;
case StreamState::Steady:
if (queue_size <= 2) {
new_schedule_time = BaseRenderTime - RenderTimeOffset;
state = StreamState::Filling;
} else if (queue_size > 5) {
new_schedule_time = BaseRenderTime + RenderTimeOffset;
state = StreamState::Draining;
}
break;
case StreamState::Draining:
if (queue_size <= 5) {
new_schedule_time = BaseRenderTime;
state = StreamState::Steady;
}
break;
}
update.store(true);
update.notify_all();
return new_schedule_time;
return std::nullopt;
}
void SystemManager::PauseCallback(bool paused) {

349
src/audio_core/sink/cubeb_sink.cpp
View File

@ -1,21 +1,13 @@
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include <atomic>
#include <span>
#include <vector>
#include "audio_core/audio_core.h"
#include "audio_core/audio_event.h"
#include "audio_core/audio_manager.h"
#include "audio_core/common/common.h"
#include "audio_core/sink/cubeb_sink.h"
#include "audio_core/sink/sink_stream.h"
#include "common/assert.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
#include "common/reader_writer_queue.h"
#include "common/ring_buffer.h"
#include "common/settings.h"
#include "core/core.h"
#ifdef _WIN32
@ -42,10 +34,10 @@ public:
* @param system_ - Core system.
* @param event - Event used only for audio renderer, signalled on buffer consume.
*/
CubebSinkStream(cubeb* ctx_, const u32 device_channels_, const u32 system_channels_,
CubebSinkStream(cubeb* ctx_, u32 device_channels_, u32 system_channels_,
cubeb_devid output_device, cubeb_devid input_device, const std::string& name_,
const StreamType type_, Core::System& system_)
: ctx{ctx_}, type{type_}, system{system_} {
StreamType type_, Core::System& system_)
: SinkStream(system_, type_), ctx{ctx_} {
#ifdef _WIN32
CoInitializeEx(nullptr, COINIT_MULTITHREADED);
#endif
@ -79,12 +71,10 @@ public:
minimum_latency = std::max(minimum_latency, 256u);
playing_buffer.consumed = true;
LOG_DEBUG(Service_Audio,
"Opening cubeb stream {} type {} with: rate {} channels {} (system channels {}) "
"latency {}",
name, type, params.rate, params.channels, system_channels, minimum_latency);
LOG_INFO(Service_Audio,
"Opening cubeb stream {} type {} with: rate {} channels {} (system channels {}) "
"latency {}",
name, type, params.rate, params.channels, system_channels, minimum_latency);
auto init_error{0};
if (type == StreamType::In) {
@ -111,6 +101,8 @@ public:
~CubebSinkStream() override {
LOG_DEBUG(Service_Audio, "Destructing cubeb stream {}", name);
Unstall();
if (!ctx) {
return;
}
@ -136,7 +128,7 @@ public:
* @param resume - Set to true if this is resuming the stream a previously-active stream.
* Default false.
*/
void Start(const bool resume = false) override {
void Start(bool resume = false) override {
if (!ctx) {
return;
}
@ -158,6 +150,7 @@ public:
* Stop the sink stream.
*/
void Stop() override {
Unstall();
if (!ctx) {
return;
}
@ -170,194 +163,7 @@ public:
paused = true;
}
/**
* Append a new buffer and its samples to a waiting queue to play.
*
* @param buffer - Audio buffer information to be queued.
* @param samples - The s16 samples to be queue for playback.
*/
void AppendBuffer(::AudioCore::Sink::SinkBuffer& buffer, std::vector<s16>& samples) override {
if (type == StreamType::In) {
queue.enqueue(buffer);
queued_buffers++;
} else {
constexpr s32 min{std::numeric_limits<s16>::min()};
constexpr s32 max{std::numeric_limits<s16>::max()};
auto yuzu_volume{Settings::Volume()};
if (yuzu_volume > 1.0f) {
yuzu_volume = 0.6f + 20 * std::log10(yuzu_volume);
}
auto volume{system_volume * device_volume * yuzu_volume};
if (system_channels == 6 && device_channels == 2) {
// We're given 6 channels, but our device only outputs 2, so downmix.
constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] *
down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] *
down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackLeft)] *
down_mix_coeff[3]) *
volume)
.to_int()};
const auto right_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] *
down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] *
down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackRight)] *
down_mix_coeff[3]) *
volume)
.to_int()};
samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
static_cast<s16>(std::clamp(left_sample, min, max));
samples[write_index + static_cast<u32>(Channels::FrontRight)] =
static_cast<s16>(std::clamp(right_sample, min, max));
}
samples.resize(samples.size() / system_channels * device_channels);
} else if (system_channels == 2 && device_channels == 6) {
// We need moar samples! Not all games will provide 6 channel audio.
// TODO: Implement some upmixing here. Currently just passthrough, with other
// channels left as silence.
std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
const auto right_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontRight)] =
right_sample;
}
samples = std::move(new_samples);
} else if (volume != 1.0f) {
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(std::clamp(
static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
}
samples_buffer.Push(samples);
queue.enqueue(buffer);
queued_buffers++;
}
}
/**
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
*
* @param num_samples - Maximum number of samples to receive.
* @return Vector of recorded samples. May have fewer than num_samples.
*/
std::vector<s16> ReleaseBuffer(const u64 num_samples) override {
static constexpr s32 min = std::numeric_limits<s16>::min();
static constexpr s32 max = std::numeric_limits<s16>::max();
auto samples{samples_buffer.Pop(num_samples)};
// TODO: Up-mix to 6 channels if the game expects it.
// For audio input this is unlikely to ever be the case though.
// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
// TODO: Play with this and find something that works better.
auto volume{system_volume * device_volume * 8};
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
if (samples.size() < num_samples) {
samples.resize(num_samples, 0);
}
return samples;
}
/**
* Check if a certain buffer has been consumed (fully played).
*
* @param tag - Unique tag of a buffer to check for.
* @return True if the buffer has been played, otherwise false.
*/
bool IsBufferConsumed(const u64 tag) override {
if (released_buffer.tag == 0) {
if (!released_buffers.try_dequeue(released_buffer)) {
return false;
}
}
if (released_buffer.tag == tag) {
released_buffer.tag = 0;
return true;
}
return false;
}
/**
* Empty out the buffer queue.
*/
void ClearQueue() override {
samples_buffer.Pop();
while (queue.pop()) {
}
while (released_buffers.pop()) {
}
queued_buffers = 0;
released_buffer = {};
playing_buffer = {};
playing_buffer.consumed = true;
}
private:
/**
* Signal events back to the audio system that a buffer was played/can be filled.
*
* @param buffer - Consumed audio buffer to be released.
*/
void SignalEvent(const ::AudioCore::Sink::SinkBuffer& buffer) {
auto& manager{system.AudioCore().GetAudioManager()};
switch (type) {
case StreamType::Out:
released_buffers.enqueue(buffer);
manager.SetEvent(Event::Type::AudioOutManager, true);
break;
case StreamType::In:
released_buffers.enqueue(buffer);
manager.SetEvent(Event::Type::AudioInManager, true);
break;
case StreamType::Render:
break;
}
}
/**
* Main callback from Cubeb. Either expects samples from us (audio render/audio out), or will
* provide samples to be copied (audio in).
@ -378,106 +184,15 @@ private:
const std::size_t num_channels = impl->GetDeviceChannels();
const std::size_t frame_size = num_channels;
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
const std::size_t num_frames{static_cast<size_t>(num_frames_)};
size_t frames_written{0};
[[maybe_unused]] bool underrun{false};
if (impl->type == StreamType::In) {
// INPUT
std::span<const s16> input_buffer{reinterpret_cast<const s16*>(in_buff),
num_frames * frame_size};
while (frames_written < num_frames) {
auto& playing_buffer{impl->playing_buffer};
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
// If no buffer was available we've underrun, just push the samples and
// continue.
underrun = true;
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
(num_frames - frames_written) * frame_size);
frames_written = num_frames;
continue;
} else {
// Successfully got a new buffer, mark the old one as consumed and signal.
impl->queued_buffers--;
impl->SignalEvent(impl->playing_buffer);
}
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{
std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
impl->playing_buffer.consumed = true;
}
}
std::memcpy(&impl->last_frame[0], &input_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
impl->ProcessAudioIn(input_buffer, num_frames);
} else {
// OUTPUT
std::span<s16> output_buffer{reinterpret_cast<s16*>(out_buff), num_frames * frame_size};
while (frames_written < num_frames) {
auto& playing_buffer{impl->playing_buffer};
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
// If no buffer was available we've underrun, fill the remaining buffer with
// the last written frame and continue.
underrun = true;
for (size_t i = frames_written; i < num_frames; i++) {
std::memcpy(&output_buffer[i * frame_size], &impl->last_frame[0],
frame_size_bytes);
}
frames_written = num_frames;
continue;
} else {
// Successfully got a new buffer, mark the old one as consumed and signal.
impl->queued_buffers--;
impl->SignalEvent(impl->playing_buffer);
}
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{
std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
impl->samples_buffer.Pop(&output_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
impl->playing_buffer.consumed = true;
}
}
std::memcpy(&impl->last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
impl->ProcessAudioOutAndRender(output_buffer, num_frames);
}
return num_frames_;
@ -490,32 +205,12 @@ private:
* @param user_data - Custom data pointer passed along, points to a CubebSinkStream.
* @param state - New state of the device.
*/
static void StateCallback([[maybe_unused]] cubeb_stream* stream,
[[maybe_unused]] void* user_data,
[[maybe_unused]] cubeb_state state) {}
static void StateCallback(cubeb_stream*, void*, cubeb_state) {}
/// Main Cubeb context
cubeb* ctx{};
/// Cubeb stream backend
cubeb_stream* stream_backend{};
/// Name of this stream
std::string name{};
/// Type of this stream
StreamType type;
/// Core system
Core::System& system;
/// Ring buffer of the samples waiting to be played or consumed
Common::RingBuffer<s16, 0x10000> samples_buffer;
/// Audio buffers queued and waiting to play
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> queue;
/// The currently-playing audio buffer
::AudioCore::Sink::SinkBuffer playing_buffer{};
/// Audio buffers which have been played and are in queue to be released by the audio system
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> released_buffers{};
/// Currently released buffer waiting to be taken by the audio system
::AudioCore::Sink::SinkBuffer released_buffer{};
/// The last played (or received) frame of audio, used when the callback underruns
std::array<s16, MaxChannels> last_frame{};
};
CubebSink::CubebSink(std::string_view target_device_name) {
@ -569,15 +264,15 @@ CubebSink::~CubebSink() {
#endif
}
SinkStream* CubebSink::AcquireSinkStream(Core::System& system, const u32 system_channels,
const std::string& name, const StreamType type) {
SinkStream* CubebSink::AcquireSinkStream(Core::System& system, u32 system_channels,
const std::string& name, StreamType type) {
SinkStreamPtr& stream = sink_streams.emplace_back(std::make_unique<CubebSinkStream>(
ctx, device_channels, system_channels, output_device, input_device, name, type, system));
return stream.get();
}
void CubebSink::CloseStream(const SinkStream* stream) {
void CubebSink::CloseStream(SinkStream* stream) {
for (size_t i = 0; i < sink_streams.size(); i++) {
if (sink_streams[i].get() == stream) {
sink_streams[i].reset();
@ -611,19 +306,19 @@ f32 CubebSink::GetDeviceVolume() const {
return sink_streams[0]->GetDeviceVolume();
}
void CubebSink::SetDeviceVolume(const f32 volume) {
void CubebSink::SetDeviceVolume(f32 volume) {
for (auto& stream : sink_streams) {
stream->SetDeviceVolume(volume);
}
}
void CubebSink::SetSystemVolume(const f32 volume) {
void CubebSink::SetSystemVolume(f32 volume) {
for (auto& stream : sink_streams) {
stream->SetSystemVolume(volume);
}
}
std::vector<std::string> ListCubebSinkDevices(const bool capture) {
std::vector<std::string> ListCubebSinkDevices(bool capture) {
std::vector<std::string> device_list;
cubeb* ctx;

2
src/audio_core/sink/cubeb_sink.h
View File

@ -46,7 +46,7 @@ public:
*
* @param stream - The stream to close.
*/
void CloseStream(const SinkStream* stream) override;
void CloseStream(SinkStream* stream) override;
/**
* Close all streams.

47
src/audio_core/sink/null_sink.h
View File

@ -3,10 +3,29 @@
#pragma once
#include <string>
#include <string_view>
#include <vector>
#include "audio_core/sink/sink.h"
#include "audio_core/sink/sink_stream.h"
namespace Core {
class System;
} // namespace Core
namespace AudioCore::Sink {
class NullSinkStreamImpl final : public SinkStream {
public:
explicit NullSinkStreamImpl(Core::System& system_, StreamType type_)
: SinkStream{system_, type_} {}
~NullSinkStreamImpl() override {}
void AppendBuffer(SinkBuffer&, std::vector<s16>&) override {}
std::vector<s16> ReleaseBuffer(u64) override {
return {};
}
};
/**
* A no-op sink for when no audio out is wanted.
*/
@ -15,14 +34,15 @@ public:
explicit NullSink(std::string_view) {}
~NullSink() override = default;
SinkStream* AcquireSinkStream([[maybe_unused]] Core::System& system,
[[maybe_unused]] u32 system_channels,
[[maybe_unused]] const std::string& name,
[[maybe_unused]] StreamType type) override {
return &null_sink_stream;
SinkStream* AcquireSinkStream(Core::System& system, u32, const std::string&,
StreamType type) override {
if (null_sink == nullptr) {
null_sink = std::make_unique<NullSinkStreamImpl>(system, type);
}
return null_sink.get();
}
void CloseStream([[maybe_unused]] const SinkStream* stream) override {}
void CloseStream(SinkStream*) override {}
void CloseStreams() override {}
void PauseStreams() override {}
void UnpauseStreams() override {}
@ -33,20 +53,7 @@ public:
void SetSystemVolume(f32 volume) override {}
private:
struct NullSinkStreamImpl final : SinkStream {
void Finalize() override {}
void Start(bool resume = false) override {}
void Stop() override {}
void AppendBuffer([[maybe_unused]] ::AudioCore::Sink::SinkBuffer& buffer,
[[maybe_unused]] std::vector<s16>& samples) override {}
std::vector<s16> ReleaseBuffer([[maybe_unused]] u64 num_samples) override {
return {};
}
bool IsBufferConsumed([[maybe_unused]] const u64 tag) {
return true;
}
void ClearQueue() override {}
} null_sink_stream;
SinkStreamPtr null_sink{};
};
} // namespace AudioCore::Sink

350
src/audio_core/sink/sdl2_sink.cpp
View File

@ -1,20 +1,13 @@
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include <atomic>
#include <span>
#include <vector>
#include "audio_core/audio_core.h"
#include "audio_core/audio_event.h"
#include "audio_core/audio_manager.h"
#include "audio_core/common/common.h"
#include "audio_core/sink/sdl2_sink.h"
#include "audio_core/sink/sink_stream.h"
#include "common/assert.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
#include "common/reader_writer_queue.h"
#include "common/ring_buffer.h"
#include "common/settings.h"
#include "core/core.h"
// Ignore -Wimplicit-fallthrough due to https://github.com/libsdl-org/SDL/issues/4307
@ -44,10 +37,9 @@ public:
* @param system_ - Core system.
* @param event - Event used only for audio renderer, signalled on buffer consume.
*/
SDLSinkStream(u32 device_channels_, const u32 system_channels_,
const std::string& output_device, const std::string& input_device,
const StreamType type_, Core::System& system_)
: type{type_}, system{system_} {
SDLSinkStream(u32 device_channels_, u32 system_channels_, const std::string& output_device,
const std::string& input_device, StreamType type_, Core::System& system_)
: SinkStream{system_, type_} {
system_channels = system_channels_;
device_channels = device_channels_;
@ -63,8 +55,6 @@ public:
spec.callback = &SDLSinkStream::DataCallback;
spec.userdata = this;
playing_buffer.consumed = true;
std::string device_name{output_device};
bool capture{false};
if (type == StreamType::In) {
@ -84,31 +74,30 @@ public:
return;
}
LOG_DEBUG(Service_Audio,
"Opening sdl stream {} with: rate {} channels {} (system channels {}) "
" samples {}",
device, obtained.freq, obtained.channels, system_channels, obtained.samples);
LOG_INFO(Service_Audio,
"Opening SDL stream {} with: rate {} channels {} (system channels {}) "
" samples {}",
device, obtained.freq, obtained.channels, system_channels, obtained.samples);
}
/**
* Destroy the sink stream.
*/
~SDLSinkStream() override {
if (device == 0) {
return;
}
SDL_CloseAudioDevice(device);
LOG_DEBUG(Service_Audio, "Destructing SDL stream {}", name);
Finalize();
}
/**
* Finalize the sink stream.
*/
void Finalize() override {
Unstall();
if (device == 0) {
return;
}
Stop();
SDL_CloseAudioDevice(device);
}
@ -118,7 +107,7 @@ public:
* @param resume - Set to true if this is resuming the stream a previously-active stream.
* Default false.
*/
void Start(const bool resume = false) override {
void Start(bool resume = false) override {
if (device == 0) {
return;
}
@ -135,7 +124,8 @@ public:
/**
* Stop the sink stream.
*/
void Stop() {
void Stop() override {
Unstall();
if (device == 0) {
return;
}
@ -143,191 +133,7 @@ public:
paused = true;
}
/**
* Append a new buffer and its samples to a waiting queue to play.
*
* @param buffer - Audio buffer information to be queued.
* @param samples - The s16 samples to be queue for playback.
*/
void AppendBuffer(::AudioCore::Sink::SinkBuffer& buffer, std::vector<s16>& samples) override {
if (type == StreamType::In) {
queue.enqueue(buffer);
queued_buffers++;
} else {
constexpr s32 min = std::numeric_limits<s16>::min();
constexpr s32 max = std::numeric_limits<s16>::max();
auto yuzu_volume{Settings::Volume()};
auto volume{system_volume * device_volume * yuzu_volume};
if (system_channels == 6 && device_channels == 2) {
// We're given 6 channels, but our device only outputs 2, so downmix.
constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] *
down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] *
down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackLeft)] *
down_mix_coeff[3]) *
volume)
.to_int()};
const auto right_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] *
down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] *
down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackRight)] *
down_mix_coeff[3]) *
volume)
.to_int()};
samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
static_cast<s16>(std::clamp(left_sample, min, max));
samples[write_index + static_cast<u32>(Channels::FrontRight)] =
static_cast<s16>(std::clamp(right_sample, min, max));
}
samples.resize(samples.size() / system_channels * device_channels);
} else if (system_channels == 2 && device_channels == 6) {
// We need moar samples! Not all games will provide 6 channel audio.
// TODO: Implement some upmixing here. Currently just passthrough, with other
// channels left as silence.
std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
const auto right_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontRight)] =
right_sample;
}
samples = std::move(new_samples);
} else if (volume != 1.0f) {
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(std::clamp(
static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
}
samples_buffer.Push(samples);
queue.enqueue(buffer);
queued_buffers++;
}
}
/**
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
*
* @param num_samples - Maximum number of samples to receive.
* @return Vector of recorded samples. May have fewer than num_samples.
*/
std::vector<s16> ReleaseBuffer(const u64 num_samples) override {
static constexpr s32 min = std::numeric_limits<s16>::min();
static constexpr s32 max = std::numeric_limits<s16>::max();
auto samples{samples_buffer.Pop(num_samples)};
// TODO: Up-mix to 6 channels if the game expects it.
// For audio input this is unlikely to ever be the case though.
// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
// TODO: Play with this and find something that works better.
auto volume{system_volume * device_volume * 8};
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
if (samples.size() < num_samples) {
samples.resize(num_samples, 0);
}
return samples;
}
/**
* Check if a certain buffer has been consumed (fully played).
*
* @param tag - Unique tag of a buffer to check for.
* @return True if the buffer has been played, otherwise false.
*/
bool IsBufferConsumed(const u64 tag) override {
if (released_buffer.tag == 0) {
if (!released_buffers.try_dequeue(released_buffer)) {
return false;
}
}
if (released_buffer.tag == tag) {
released_buffer.tag = 0;
return true;
}
return false;
}
/**
* Empty out the buffer queue.
*/
void ClearQueue() override {
samples_buffer.Pop();
while (queue.pop()) {
}
while (released_buffers.pop()) {
}
released_buffer = {};
playing_buffer = {};
playing_buffer.consumed = true;
queued_buffers = 0;
}
private:
/**
* Signal events back to the audio system that a buffer was played/can be filled.
*
* @param buffer - Consumed audio buffer to be released.
*/
void SignalEvent(const ::AudioCore::Sink::SinkBuffer& buffer) {
auto& manager{system.AudioCore().GetAudioManager()};
switch (type) {
case StreamType::Out:
released_buffers.enqueue(buffer);
manager.SetEvent(Event::Type::AudioOutManager, true);
break;
case StreamType::In:
released_buffers.enqueue(buffer);
manager.SetEvent(Event::Type::AudioInManager, true);
break;
case StreamType::Render:
break;
}
}
/**
* Main callback from SDL. Either expects samples from us (audio render/audio out), or will
* provide samples to be copied (audio in).
@ -345,122 +151,20 @@ private:
const std::size_t num_channels = impl->GetDeviceChannels();
const std::size_t frame_size = num_channels;
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
const std::size_t num_frames{len / num_channels / sizeof(s16)};
size_t frames_written{0};
[[maybe_unused]] bool underrun{false};
if (impl->type == StreamType::In) {
std::span<s16> input_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size};
while (frames_written < num_frames) {
auto& playing_buffer{impl->playing_buffer};
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
// If no buffer was available we've underrun, just push the samples and
// continue.
underrun = true;
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
(num_frames - frames_written) * frame_size);
frames_written = num_frames;
continue;
} else {
impl->queued_buffers--;
impl->SignalEvent(impl->playing_buffer);
}
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{
std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
impl->playing_buffer.consumed = true;
}
}
std::memcpy(&impl->last_frame[0], &input_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
std::span<const s16> input_buffer{reinterpret_cast<const s16*>(stream),
num_frames * frame_size};
impl->ProcessAudioIn(input_buffer, num_frames);
} else {
std::span<s16> output_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size};
while (frames_written < num_frames) {
auto& playing_buffer{impl->playing_buffer};
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
// If no buffer was available we've underrun, fill the remaining buffer with
// the last written frame and continue.
underrun = true;
for (size_t i = frames_written; i < num_frames; i++) {
std::memcpy(&output_buffer[i * frame_size], &impl->last_frame[0],
frame_size_bytes);
}
frames_written = num_frames;
continue;
} else {
impl->queued_buffers--;
impl->SignalEvent(impl->playing_buffer);
}
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{
std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
impl->samples_buffer.Pop(&output_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
impl->playing_buffer.consumed = true;
}
}
std::memcpy(&impl->last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
impl->ProcessAudioOutAndRender(output_buffer, num_frames);
}
}
/// SDL device id of the opened input/output device
SDL_AudioDeviceID device{};
/// Type of this stream
StreamType type;
/// Core system
Core::System& system;
/// Ring buffer of the samples waiting to be played or consumed
Common::RingBuffer<s16, 0x10000> samples_buffer;
/// Audio buffers queued and waiting to play
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> queue;
/// The currently-playing audio buffer
::AudioCore::Sink::SinkBuffer playing_buffer{};
/// Audio buffers which have been played and are in queue to be released by the audio system
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> released_buffers{};
/// Currently released buffer waiting to be taken by the audio system
::AudioCore::Sink::SinkBuffer released_buffer{};
/// The last played (or received) frame of audio, used when the callback underruns
std::array<s16, MaxChannels> last_frame{};
};
SDLSink::SDLSink(std::string_view target_device_name) {
@ -482,14 +186,14 @@ SDLSink::SDLSink(std::string_view target_device_name) {
SDLSink::~SDLSink() = default;
SinkStream* SDLSink::AcquireSinkStream(Core::System& system, const u32 system_channels,
const std::string&, const StreamType type) {
SinkStream* SDLSink::AcquireSinkStream(Core::System& system, u32 system_channels,
const std::string&, StreamType type) {
SinkStreamPtr& stream = sink_streams.emplace_back(std::make_unique<SDLSinkStream>(
device_channels, system_channels, output_device, input_device, type, system));
return stream.get();
}
void SDLSink::CloseStream(const SinkStream* stream) {
void SDLSink::CloseStream(SinkStream* stream) {
for (size_t i = 0; i < sink_streams.size(); i++) {
if (sink_streams[i].get() == stream) {
sink_streams[i].reset();
@ -523,19 +227,19 @@ f32 SDLSink::GetDeviceVolume() const {
return sink_streams[0]->GetDeviceVolume();
}
void SDLSink::SetDeviceVolume(const f32 volume) {
void SDLSink::SetDeviceVolume(f32 volume) {
for (auto& stream : sink_streams) {
stream->SetDeviceVolume(volume);
}
}
void SDLSink::SetSystemVolume(const f32 volume) {
void SDLSink::SetSystemVolume(f32 volume) {
for (auto& stream : sink_streams) {
stream->SetSystemVolume(volume);
}
}
std::vector<std::string> ListSDLSinkDevices(const bool capture) {
std::vector<std::string> ListSDLSinkDevices(bool capture) {
std::vector<std::string> device_list;
if (!SDL_WasInit(SDL_INIT_AUDIO)) {

2
src/audio_core/sink/sdl2_sink.h
View File

@ -44,7 +44,7 @@ public:
*
* @param stream - The stream to close.
*/
void CloseStream(const SinkStream* stream) override;
void CloseStream(SinkStream* stream) override;
/**
* Close all streams.

2
src/audio_core/sink/sink.h
View File

@ -32,7 +32,7 @@ public:
*
* @param stream - The stream to close.
*/
virtual void CloseStream(const SinkStream* stream) = 0;
virtual void CloseStream(SinkStream* stream) = 0;
/**
* Close all streams.

6
src/audio_core/sink/sink_details.cpp
View File

@ -5,7 +5,7 @@
#include <memory>
#include <string>
#include <vector>
#include "audio_core/sink/null_sink.h"
#include "audio_core/sink/sink_details.h"
#ifdef HAVE_CUBEB
#include "audio_core/sink/cubeb_sink.h"
@ -13,6 +13,7 @@
#ifdef HAVE_SDL2
#include "audio_core/sink/sdl2_sink.h"
#endif
#include "audio_core/sink/null_sink.h"
#include "common/logging/log.h"
namespace AudioCore::Sink {
@ -59,8 +60,7 @@ const SinkDetails& GetOutputSinkDetails(std::string_view sink_id) {
if (sink_id == "auto" || iter == std::end(sink_details)) {
if (sink_id != "auto") {
LOG_ERROR(Audio, "AudioCore::Sink::GetOutputSinkDetails given invalid sink_id {}",
sink_id);
LOG_ERROR(Audio, "Invalid sink_id {}", sink_id);
}
// Auto-select.
// sink_details is ordered in terms of desirability, with the best choice at the front.

259
src/audio_core/sink/sink_stream.cpp Normal file
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@ -0,0 +1,259 @@
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <atomic>
#include <memory>
#include <span>
#include <vector>
#include "audio_core/common/common.h"
#include "audio_core/sink/sink_stream.h"
#include "common/common_types.h"
#include "common/fixed_point.h"
#include "common/settings.h"
#include "core/core.h"
namespace AudioCore::Sink {
void SinkStream::AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) {
if (type == StreamType::In) {
queue.enqueue(buffer);
queued_buffers++;
return;
}
constexpr s32 min{std::numeric_limits<s16>::min()};
constexpr s32 max{std::numeric_limits<s16>::max()};
auto yuzu_volume{Settings::Volume()};
if (yuzu_volume > 1.0f) {
yuzu_volume = 0.6f + 20 * std::log10(yuzu_volume);
}
auto volume{system_volume * device_volume * yuzu_volume};
if (system_channels == 6 && device_channels == 2) {
// We're given 6 channels, but our device only outputs 2, so downmix.
constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] * down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] * down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackLeft)] * down_mix_coeff[3]) *
volume)
.to_int()};
const auto right_sample{
((Common::FixedPoint<49, 15>(
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
down_mix_coeff[0] +
samples[read_index + static_cast<u32>(Channels::Center)] * down_mix_coeff[1] +
samples[read_index + static_cast<u32>(Channels::LFE)] * down_mix_coeff[2] +
samples[read_index + static_cast<u32>(Channels::BackRight)] * down_mix_coeff[3]) *
volume)
.to_int()};
samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
static_cast<s16>(std::clamp(left_sample, min, max));
samples[write_index + static_cast<u32>(Channels::FrontRight)] =
static_cast<s16>(std::clamp(right_sample, min, max));
}
samples.resize(samples.size() / system_channels * device_channels);
} else if (system_channels == 2 && device_channels == 6) {
// We need moar samples! Not all games will provide 6 channel audio.
// TODO: Implement some upmixing here. Currently just passthrough, with other
// channels left as silence.
std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
read_index += system_channels, write_index += device_channels) {
const auto left_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
const auto right_sample{static_cast<s16>(std::clamp(
static_cast<s32>(
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
volume),
min, max))};
new_samples[write_index + static_cast<u32>(Channels::FrontRight)] = right_sample;
}
samples = std::move(new_samples);
} else if (volume != 1.0f) {
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
}
samples_buffer.Push(samples);
queue.enqueue(buffer);
queued_buffers++;
}
std::vector<s16> SinkStream::ReleaseBuffer(u64 num_samples) {
constexpr s32 min = std::numeric_limits<s16>::min();
constexpr s32 max = std::numeric_limits<s16>::max();
auto samples{samples_buffer.Pop(num_samples)};
// TODO: Up-mix to 6 channels if the game expects it.
// For audio input this is unlikely to ever be the case though.
// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
// TODO: Play with this and find something that works better.
auto volume{system_volume * device_volume * 8};
for (u32 i = 0; i < samples.size(); i++) {
samples[i] = static_cast<s16>(
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
}
if (samples.size() < num_samples) {
samples.resize(num_samples, 0);
}
return samples;
}
void SinkStream::ClearQueue() {
samples_buffer.Pop();
while (queue.pop()) {
}
queued_buffers = 0;
playing_buffer = {};
playing_buffer.consumed = true;
}
void SinkStream::ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames) {
const std::size_t num_channels = GetDeviceChannels();
const std::size_t frame_size = num_channels;
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
size_t frames_written{0};
if (queued_buffers > max_queue_size) {
Stall();
}
while (frames_written < num_frames) {
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!queue.try_dequeue(playing_buffer)) {
// If no buffer was available we've underrun, just push the samples and
// continue.
samples_buffer.Push(&input_buffer[frames_written * frame_size],
(num_frames - frames_written) * frame_size);
frames_written = num_frames;
continue;
}
// Successfully dequeued a new buffer.
queued_buffers--;
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
samples_buffer.Push(&input_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
playing_buffer.consumed = true;
}
}
std::memcpy(&last_frame[0], &input_buffer[(frames_written - 1) * frame_size], frame_size_bytes);
if (queued_buffers <= max_queue_size) {
Unstall();
}
}
void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames) {
const std::size_t num_channels = GetDeviceChannels();
const std::size_t frame_size = num_channels;
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
size_t frames_written{0};
if (queued_buffers > max_queue_size) {
Stall();
}
while (frames_written < num_frames) {
// If the playing buffer has been consumed or has no frames, we need a new one
if (playing_buffer.consumed || playing_buffer.frames == 0) {
if (!queue.try_dequeue(playing_buffer)) {
// If no buffer was available we've underrun, fill the remaining buffer with
// the last written frame and continue.
for (size_t i = frames_written; i < num_frames; i++) {
std::memcpy(&output_buffer[i * frame_size], &last_frame[0], frame_size_bytes);
}
frames_written = num_frames;
continue;
}
// Successfully dequeued a new buffer.
queued_buffers--;
}
// Get the minimum frames available between the currently playing buffer, and the
// amount we have left to fill
size_t frames_available{std::min(playing_buffer.frames - playing_buffer.frames_played,
num_frames - frames_written)};
samples_buffer.Pop(&output_buffer[frames_written * frame_size],
frames_available * frame_size);
frames_written += frames_available;
playing_buffer.frames_played += frames_available;
// If that's all the frames in the current buffer, add its samples and mark it as
// consumed
if (playing_buffer.frames_played >= playing_buffer.frames) {
playing_buffer.consumed = true;
}
}
std::memcpy(&last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
frame_size_bytes);
if (stalled && queued_buffers <= max_queue_size) {
Unstall();
}
}
void SinkStream::Stall() {
if (stalled) {
return;
}
stalled = true;
system.StallProcesses();
}
void SinkStream::Unstall() {
if (!stalled) {
return;
}
system.UnstallProcesses();
stalled = false;
}
} // namespace AudioCore::Sink

171
src/audio_core/sink/sink_stream.h
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@ -3,12 +3,20 @@
#pragma once
#include <array>
#include <atomic>
#include <memory>
#include <span>
#include <vector>
#include "audio_core/common/common.h"
#include "common/common_types.h"
#include "common/reader_writer_queue.h"
#include "common/ring_buffer.h"
namespace Core {
class System;
} // namespace Core
namespace AudioCore::Sink {
@ -34,20 +42,24 @@ struct SinkBuffer {
* You should regularly call IsBufferConsumed with the unique SinkBuffer tag to check if the buffer
* has been consumed.
*
* Since these are a FIFO queue, always check IsBufferConsumed in the same order you appended the
* buffers, skipping a buffer will result in all following buffers to never release.
* Since these are a FIFO queue, IsBufferConsumed must be checked in the same order buffers were
* appended, skipping a buffer will result in the queue getting stuck, and all following buffers to
* never release.
*
* If the buffers appear to be stuck, you can stop and re-open an IAudioIn/IAudioOut service (this
* is what games do), or call ClearQueue to flush all of the buffers without a full restart.
*/
class SinkStream {
public:
virtual ~SinkStream() = default;
explicit SinkStream(Core::System& system_, StreamType type_) : system{system_}, type{type_} {}
virtual ~SinkStream() {
Unstall();
}
/**
* Finalize the sink stream.
*/
virtual void Finalize() = 0;
virtual void Finalize() {}
/**
* Start the sink stream.
@ -55,48 +67,19 @@ public:
* @param resume - Set to true if this is resuming the stream a previously-active stream.
* Default false.
*/
virtual void Start(bool resume = false) = 0;
virtual void Start(bool resume = false) {}
/**
* Stop the sink stream.
*/
virtual void Stop() = 0;
/**
* Append a new buffer and its samples to a waiting queue to play.
*
* @param buffer - Audio buffer information to be queued.
* @param samples - The s16 samples to be queue for playback.
*/
virtual void AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) = 0;
/**
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
*
* @param num_samples - Maximum number of samples to receive.
* @return Vector of recorded samples. May have fewer than num_samples.
*/
virtual std::vector<s16> ReleaseBuffer(u64 num_samples) = 0;
/**
* Check if a certain buffer has been consumed (fully played).
*
* @param tag - Unique tag of a buffer to check for.
* @return True if the buffer has been played, otherwise false.
*/
virtual bool IsBufferConsumed(u64 tag) = 0;
/**
* Empty out the buffer queue.
*/
virtual void ClearQueue() = 0;
virtual void Stop() {}
/**
* Check if the stream is paused.
*
* @return True if paused, otherwise false.
*/
bool IsPaused() {
bool IsPaused() const {
return paused;
}
@ -127,34 +110,6 @@ public:
return device_channels;
}
/**
* Get the total number of samples played by this stream.
*
* @return Number of samples played.
*/
u64 GetPlayedSampleCount() const {
return played_sample_count;
}
/**
* Set the number of samples played.
* This is started and stopped on system start/stop.
*
* @param played_sample_count_ - Number of samples to set.
*/
void SetPlayedSampleCount(u64 played_sample_count_) {
played_sample_count = played_sample_count_;
}
/**
* Add to the played sample count.
*
* @param num_samples - Number of samples to add.
*/
void AddPlayedSampleCount(u64 num_samples) {
played_sample_count += num_samples;
}
/**
* Get the system volume.
*
@ -200,15 +155,65 @@ public:
return queued_buffers.load();
}
/**
* Set the maximum buffer queue size.
*/
void SetRingSize(u32 ring_size) {
max_queue_size = ring_size;
}
/**
* Append a new buffer and its samples to a waiting queue to play.
*
* @param buffer - Audio buffer information to be queued.
* @param samples - The s16 samples to be queue for playback.
*/
virtual void AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples);
/**
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
*
* @param num_samples - Maximum number of samples to receive.
* @return Vector of recorded samples. May have fewer than num_samples.
*/
virtual std::vector<s16> ReleaseBuffer(u64 num_samples);
/**
* Empty out the buffer queue.
*/
void ClearQueue();
/**
* Callback for AudioIn.
*
* @param input_buffer - Input buffer to be filled with samples.
* @param num_frames - Number of frames to be filled.
*/
void ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames);
/**
* Callback for AudioOut and AudioRenderer.
*
* @param output_buffer - Output buffer to be filled with samples.
* @param num_frames - Number of frames to be filled.
*/
void ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames);
/**
* Stall core processes if the audio thread falls too far behind.
*/
void Stall();
/**
* Unstall core processes.
*/
void Unstall();
protected:
/// Number of buffers waiting to be played
std::atomic<u32> queued_buffers{};
/// Total samples played by this stream
std::atomic<u64> played_sample_count{};
/// Set by the audio render/in/out system which uses this stream
f32 system_volume{1.0f};
/// Set via IAudioDevice service calls
f32 device_volume{1.0f};
/// Core system
Core::System& system;
/// Type of this stream
StreamType type;
/// Set by the audio render/in/out systen which uses this stream
u32 system_channels{2};
/// Channels supported by hardware
@ -217,6 +222,28 @@ protected:
std::atomic<bool> paused{true};
/// Was this stream previously playing?
std::atomic<bool> was_playing{false};
/// Name of this stream
std::string name{};
private:
/// Ring buffer of the samples waiting to be played or consumed
Common::RingBuffer<s16, 0x10000> samples_buffer;
/// Audio buffers queued and waiting to play
Common::ReaderWriterQueue<SinkBuffer> queue;
/// The currently-playing audio buffer
SinkBuffer playing_buffer{};
/// The last played (or received) frame of audio, used when the callback underruns
std::array<s16, MaxChannels> last_frame{};
/// Number of buffers waiting to be played
std::atomic<u32> queued_buffers{};
/// The ring size for audio out buffers (usually 4, rarely 2 or 8)
u32 max_queue_size{};
/// Set by the audio render/in/out system which uses this stream
f32 system_volume{1.0f};
/// Set via IAudioDevice service calls
f32 device_volume{1.0f};
/// True if coretiming has been stalled
bool stalled{false};
};
using SinkStreamPtr = std::unique_ptr<SinkStream>;

2
src/core/hle/result.h
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@ -117,6 +117,7 @@ union Result {
BitField<0, 9, ErrorModule> module;
BitField<9, 13, u32> description;
Result() = default;
constexpr explicit Result(u32 raw_) : raw(raw_) {}
constexpr Result(ErrorModule module_, u32 description_)
@ -130,6 +131,7 @@ union Result {
return !IsSuccess();
}
};
static_assert(std::is_trivial_v<Result>);
[[nodiscard]] constexpr bool operator==(const Result& a, const Result& b) {
return a.raw == b.raw;