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-rw-r--r--src/audio_core/CMakeLists.txt1
-rw-r--r--src/audio_core/audio_core.cpp18
-rw-r--r--src/audio_core/audio_core.h26
-rw-r--r--src/audio_core/audio_event.h4
-rw-r--r--src/audio_core/audio_in_manager.cpp2
-rw-r--r--src/audio_core/audio_in_manager.h5
-rw-r--r--src/audio_core/audio_manager.h2
-rw-r--r--src/audio_core/audio_out_manager.cpp2
-rw-r--r--src/audio_core/audio_render_manager.cpp6
-rw-r--r--src/audio_core/audio_render_manager.h12
-rw-r--r--src/audio_core/device/audio_buffer.h4
-rw-r--r--src/audio_core/device/audio_buffers.h21
-rw-r--r--src/audio_core/device/device_session.cpp64
-rw-r--r--src/audio_core/device/device_session.h34
-rw-r--r--src/audio_core/in/audio_in.cpp8
-rw-r--r--src/audio_core/in/audio_in.h8
-rw-r--r--src/audio_core/in/audio_in_system.cpp22
-rw-r--r--src/audio_core/in/audio_in_system.h12
-rw-r--r--src/audio_core/out/audio_out.cpp8
-rw-r--r--src/audio_core/out/audio_out.h8
-rw-r--r--src/audio_core/out/audio_out_system.cpp21
-rw-r--r--src/audio_core/out/audio_out_system.h10
-rw-r--r--src/audio_core/renderer/adsp/adsp.cpp2
-rw-r--r--src/audio_core/renderer/adsp/adsp.h4
-rw-r--r--src/audio_core/renderer/adsp/audio_renderer.cpp11
-rw-r--r--src/audio_core/renderer/adsp/audio_renderer.h8
-rw-r--r--src/audio_core/renderer/adsp/command_list_processor.h13
-rw-r--r--src/audio_core/renderer/audio_device.cpp34
-rw-r--r--src/audio_core/renderer/audio_device.h22
-rw-r--r--src/audio_core/renderer/behavior/behavior_info.cpp16
-rw-r--r--src/audio_core/renderer/behavior/behavior_info.h4
-rw-r--r--src/audio_core/renderer/behavior/info_updater.cpp2
-rw-r--r--src/audio_core/renderer/behavior/info_updater.h2
-rw-r--r--src/audio_core/renderer/command/command_buffer.h12
-rw-r--r--src/audio_core/renderer/command/command_generator.h46
-rw-r--r--src/audio_core/renderer/command/effect/compressor.cpp11
-rw-r--r--src/audio_core/renderer/command/mix/mix_ramp.cpp18
-rw-r--r--src/audio_core/renderer/command/mix/mix_ramp.h8
-rw-r--r--src/audio_core/renderer/command/mix/mix_ramp_grouped.h4
-rw-r--r--src/audio_core/renderer/command/sink/device.cpp4
-rw-r--r--src/audio_core/renderer/effect/effect_context.h14
-rw-r--r--src/audio_core/renderer/effect/effect_info_base.h4
-rw-r--r--src/audio_core/renderer/effect/i3dl2.h4
-rw-r--r--src/audio_core/renderer/effect/reverb.h4
-rw-r--r--src/audio_core/renderer/memory/address_info.h5
-rw-r--r--src/audio_core/renderer/nodes/node_states.h8
-rw-r--r--src/audio_core/renderer/performance/performance_manager.h8
-rw-r--r--src/audio_core/renderer/system_manager.cpp46
-rw-r--r--src/audio_core/renderer/system_manager.h9
-rw-r--r--src/audio_core/renderer/upsampler/upsampler_manager.h2
-rw-r--r--src/audio_core/renderer/voice/voice_info.h26
-rw-r--r--src/audio_core/sink/cubeb_sink.cpp383
-rw-r--r--src/audio_core/sink/cubeb_sink.h17
-rw-r--r--src/audio_core/sink/null_sink.h49
-rw-r--r--src/audio_core/sink/sdl2_sink.cpp377
-rw-r--r--src/audio_core/sink/sdl2_sink.h17
-rw-r--r--src/audio_core/sink/sink.h15
-rw-r--r--src/audio_core/sink/sink_details.cpp6
-rw-r--r--src/audio_core/sink/sink_stream.cpp279
-rw-r--r--src/audio_core/sink/sink_stream.h175
60 files changed, 810 insertions, 1157 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 5fe1d5fa5..144f1bab2 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -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
)
diff --git a/src/audio_core/audio_core.cpp b/src/audio_core/audio_core.cpp
index 78e615a10..c845330cd 100644
--- a/src/audio_core/audio_core.cpp
+++ b/src/audio_core/audio_core.cpp
@@ -47,22 +47,12 @@ AudioRenderer::ADSP::ADSP& AudioCore::GetADSP() {
return *adsp;
}
-void AudioCore::PauseSinks(const bool pausing) const {
- if (pausing) {
- output_sink->PauseStreams();
- input_sink->PauseStreams();
- } else {
- output_sink->UnpauseStreams();
- input_sink->UnpauseStreams();
- }
+void AudioCore::SetNVDECActive(bool active) {
+ nvdec_active = active;
}
-u32 AudioCore::GetStreamQueue() const {
- return estimated_queue.load();
-}
-
-void AudioCore::SetStreamQueue(u32 size) {
- estimated_queue.store(size);
+bool AudioCore::IsNVDECActive() const {
+ return nvdec_active;
}
} // namespace AudioCore
diff --git a/src/audio_core/audio_core.h b/src/audio_core/audio_core.h
index 0f7d61ee4..e33e00a3e 100644
--- a/src/audio_core/audio_core.h
+++ b/src/audio_core/audio_core.h
@@ -17,7 +17,7 @@ namespace AudioCore {
class AudioManager;
/**
- * Main audio class, sotred inside the core, and holding the audio manager, all sinks, and the ADSP.
+ * Main audio class, stored inside the core, and holding the audio manager, all sinks, and the ADSP.
*/
class AudioCore {
public:
@@ -58,26 +58,16 @@ public:
AudioRenderer::ADSP::ADSP& GetADSP();
/**
- * Pause the sink. Called from the core.
+ * Toggle NVDEC state, used to avoid stall in playback.
*
- * @param pausing - Is this pause due to an actual pause, or shutdown?
- * Unfortunately, shutdown also pauses streams, which can cause issues.
+ * @param active - Set true if nvdec is active, otherwise false.
*/
- void PauseSinks(bool pausing) const;
+ void SetNVDECActive(bool active);
/**
- * 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.
+ * Get NVDEC state.
*/
- void SetStreamQueue(u32 size);
+ bool IsNVDECActive() const;
private:
/**
@@ -93,8 +83,8 @@ 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};
+ /// Is NVDec currently active?
+ bool nvdec_active{false};
};
} // namespace AudioCore
diff --git a/src/audio_core/audio_event.h b/src/audio_core/audio_event.h
index 82dd32dca..012d2ed70 100644
--- a/src/audio_core/audio_event.h
+++ b/src/audio_core/audio_event.h
@@ -14,7 +14,7 @@ namespace AudioCore {
* Responsible for the input/output events, set by the stream backend when buffers are consumed, and
* waited on by the audio manager. These callbacks signal the game's events to keep the audio buffer
* recycling going.
- * In a real Switch this is not a seprate class, and exists entirely within the audio manager.
+ * In a real Switch this is not a separate class, and exists entirely within the audio manager.
* On the Switch it's implemented more simply through a MultiWaitEventHolder, where it can
* wait on multiple events at once, and the events are not needed by the backend.
*/
@@ -81,7 +81,7 @@ public:
void ClearEvents();
private:
- /// Lock, used bythe audio manager
+ /// Lock, used by the audio manager
std::mutex event_lock;
/// Array of events, one per system type (see Type), last event is used to terminate
std::array<std::atomic<bool>, 4> events_signalled;
diff --git a/src/audio_core/audio_in_manager.cpp b/src/audio_core/audio_in_manager.cpp
index 4aadb7fd6..f39fb4002 100644
--- a/src/audio_core/audio_in_manager.cpp
+++ b/src/audio_core/audio_in_manager.cpp
@@ -82,7 +82,7 @@ u32 Manager::GetDeviceNames(std::vector<AudioRenderer::AudioDevice::AudioDeviceN
auto input_devices{Sink::GetDeviceListForSink(Settings::values.sink_id.GetValue(), true)};
if (input_devices.size() > 1) {
- names.push_back(AudioRenderer::AudioDevice::AudioDeviceName("Uac"));
+ names.emplace_back("Uac");
return 1;
}
return 0;
diff --git a/src/audio_core/audio_in_manager.h b/src/audio_core/audio_in_manager.h
index 75b73a0b6..8a519df99 100644
--- a/src/audio_core/audio_in_manager.h
+++ b/src/audio_core/audio_in_manager.h
@@ -59,9 +59,10 @@ public:
/**
* Get a list of audio in device names.
*
- * @oaram names - Output container to write names to.
- * @param max_count - Maximum numebr of deivce names to write. Unused
+ * @param names - Output container to write names to.
+ * @param max_count - Maximum number of device names to write. Unused
* @param filter - Should the list be filtered? Unused.
+ *
* @return Number of names written.
*/
u32 GetDeviceNames(std::vector<AudioRenderer::AudioDevice::AudioDeviceName>& names,
diff --git a/src/audio_core/audio_manager.h b/src/audio_core/audio_manager.h
index 70316e9cb..8cbd95e22 100644
--- a/src/audio_core/audio_manager.h
+++ b/src/audio_core/audio_manager.h
@@ -76,7 +76,7 @@ public:
private:
/**
- * Main thread, waiting on a manager signal and calling the registered fucntion.
+ * Main thread, waiting on a manager signal and calling the registered function.
*/
void ThreadFunc();
diff --git a/src/audio_core/audio_out_manager.cpp b/src/audio_core/audio_out_manager.cpp
index 71d67de64..1766efde1 100644
--- a/src/audio_core/audio_out_manager.cpp
+++ b/src/audio_core/audio_out_manager.cpp
@@ -74,7 +74,7 @@ void Manager::BufferReleaseAndRegister() {
u32 Manager::GetAudioOutDeviceNames(
std::vector<AudioRenderer::AudioDevice::AudioDeviceName>& names) const {
- names.push_back({"DeviceOut"});
+ names.emplace_back("DeviceOut");
return 1;
}
diff --git a/src/audio_core/audio_render_manager.cpp b/src/audio_core/audio_render_manager.cpp
index 7a846835b..7aba2b423 100644
--- a/src/audio_core/audio_render_manager.cpp
+++ b/src/audio_core/audio_render_manager.cpp
@@ -25,8 +25,8 @@ SystemManager& Manager::GetSystemManager() {
return *system_manager;
}
-auto Manager::GetWorkBufferSize(const AudioRendererParameterInternal& params, u64& out_count)
- -> Result {
+Result Manager::GetWorkBufferSize(const AudioRendererParameterInternal& params,
+ u64& out_count) const {
if (!CheckValidRevision(params.revision)) {
return Service::Audio::ERR_INVALID_REVISION;
}
@@ -54,7 +54,7 @@ void Manager::ReleaseSessionId(const s32 session_id) {
session_ids[--session_count] = session_id;
}
-u32 Manager::GetSessionCount() {
+u32 Manager::GetSessionCount() const {
std::scoped_lock l{session_lock};
return session_count;
}
diff --git a/src/audio_core/audio_render_manager.h b/src/audio_core/audio_render_manager.h
index 6a508ec56..bf4837190 100644
--- a/src/audio_core/audio_render_manager.h
+++ b/src/audio_core/audio_render_manager.h
@@ -46,7 +46,7 @@ public:
* @param out_count - Output size of the required workbuffer.
* @return Result code.
*/
- Result GetWorkBufferSize(const AudioRendererParameterInternal& params, u64& out_count);
+ Result GetWorkBufferSize(const AudioRendererParameterInternal& params, u64& out_count) const;
/**
* Get a new session id.
@@ -60,14 +60,14 @@ public:
*
* @return The number of active sessions.
*/
- u32 GetSessionCount();
+ u32 GetSessionCount() const;
/**
* Add a renderer system to the manager.
- * The system will be reguarly called to generate commands for the AudioRenderer.
+ * The system will be regularly called to generate commands for the AudioRenderer.
*
* @param system - The system to add.
- * @return True if the system was sucessfully added, otherwise false.
+ * @return True if the system was successfully added, otherwise false.
*/
bool AddSystem(System& system);
@@ -75,7 +75,7 @@ public:
* Remove a renderer system from the manager.
*
* @param system - The system to remove.
- * @return True if the system was sucessfully removed, otherwise false.
+ * @return True if the system was successfully removed, otherwise false.
*/
bool RemoveSystem(System& system);
@@ -94,7 +94,7 @@ private:
/// Number of active renderers
u32 session_count{};
/// Lock for interacting with the sessions
- std::mutex session_lock{};
+ mutable std::mutex session_lock{};
/// Regularly generates commands from the registered systems for the AudioRenderer
std::unique_ptr<SystemManager> system_manager{};
};
diff --git a/src/audio_core/device/audio_buffer.h b/src/audio_core/device/audio_buffer.h
index cae7fa970..7128ef72a 100644
--- a/src/audio_core/device/audio_buffer.h
+++ b/src/audio_core/device/audio_buffer.h
@@ -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.
diff --git a/src/audio_core/device/audio_buffers.h b/src/audio_core/device/audio_buffers.h
index 5d1979ea0..3dae1a3b7 100644
--- a/src/audio_core/device/audio_buffers.h
+++ b/src/audio_core/device/audio_buffers.h
@@ -36,7 +36,7 @@ public:
*
* @param buffer - The new buffer.
*/
- void AppendBuffer(AudioBuffer& buffer) {
+ void AppendBuffer(const AudioBuffer& buffer) {
std::scoped_lock l{lock};
buffers[appended_index] = buffer;
appended_count++;
@@ -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;
@@ -87,10 +88,12 @@ public:
/**
* Release all registered buffers.
*
- * @param timestamp - The released timestamp for this buffer.
+ * @param core_timing - The CoreTiming instance
+ * @param session - The device session
+ *
* @return Is the buffer was released.
*/
- bool ReleaseBuffers(Core::Timing::CoreTiming& core_timing, DeviceSession& session) {
+ bool ReleaseBuffers(const Core::Timing::CoreTiming& core_timing, const DeviceSession& session) {
std::scoped_lock l{lock};
bool buffer_released{false};
while (registered_count > 0) {
@@ -100,7 +103,7 @@ 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 +283,16 @@ 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{};
diff --git a/src/audio_core/device/device_session.cpp b/src/audio_core/device/device_session.cpp
index 095fc96ce..995060414 100644
--- a/src/audio_core/device/device_session.cpp
+++ b/src/audio_core/device/device_session.cpp
@@ -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();
@@ -50,25 +59,26 @@ 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++) {
+void DeviceSession::AppendBuffers(std::span<const AudioBuffer> buffers) const {
+ for (const auto& buffer : buffers) {
Sink::SinkBuffer new_buffer{
- .frames = buffers[i].size / (channel_count * sizeof(s16)),
+ .frames = buffer.size / (channel_count * sizeof(s16)),
.frames_played = 0,
- .tag = buffers[i].tag,
+ .tag = buffer.tag,
.consumed = false,
};
@@ -76,26 +86,22 @@ void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
std::vector<s16> samples{};
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);
+ std::vector<s16> samples(buffer.size / sizeof(s16));
+ system.Memory().ReadBlockUnsafe(buffer.samples, samples.data(), buffer.size);
stream->AppendBuffer(new_buffer, samples);
}
}
}
-void DeviceSession::ReleaseBuffer(AudioBuffer& buffer) const {
+void DeviceSession::ReleaseBuffer(const 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(const 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
diff --git a/src/audio_core/device/device_session.h b/src/audio_core/device/device_session.h
index 4a031b765..74f4dc085 100644
--- a/src/audio_core/device/device_session.h
+++ b/src/audio_core/device/device_session.h
@@ -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;
@@ -55,22 +62,23 @@ public:
*
* @param buffers - The buffers to play.
*/
- void AppendBuffers(std::span<AudioBuffer> buffers) const;
+ void AppendBuffers(std::span<const AudioBuffer> buffers) const;
/**
* (Audio In only) Pop samples from the backend, and write them back to this buffer's address.
*
* @param buffer - The buffer to write to.
*/
- void ReleaseBuffer(AudioBuffer& buffer) const;
+ void ReleaseBuffer(const AudioBuffer& buffer) const;
/**
* Check if the buffer for the given tag has been consumed by the backend.
*
- * @param tag - Unqiue tag of the buffer to check.
+ * @param buffer - the buffer to check.
+ *
* @return true if the buffer has been consumed, otherwise false.
*/
- bool IsBufferConsumed(u64 tag) const;
+ bool IsBufferConsumed(const AudioBuffer& buffer) const;
/**
* Start this device session, starting the backend stream.
@@ -96,6 +104,16 @@ 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 +136,13 @@ 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
diff --git a/src/audio_core/in/audio_in.cpp b/src/audio_core/in/audio_in.cpp
index c946895d6..91ccd5ad7 100644
--- a/src/audio_core/in/audio_in.cpp
+++ b/src/audio_core/in/audio_in.cpp
@@ -72,7 +72,7 @@ Kernel::KReadableEvent& In::GetBufferEvent() {
return event->GetReadableEvent();
}
-f32 In::GetVolume() {
+f32 In::GetVolume() const {
std::scoped_lock l{parent_mutex};
return system.GetVolume();
}
@@ -82,17 +82,17 @@ void In::SetVolume(f32 volume) {
system.SetVolume(volume);
}
-bool In::ContainsAudioBuffer(u64 tag) {
+bool In::ContainsAudioBuffer(u64 tag) const {
std::scoped_lock l{parent_mutex};
return system.ContainsAudioBuffer(tag);
}
-u32 In::GetBufferCount() {
+u32 In::GetBufferCount() const {
std::scoped_lock l{parent_mutex};
return system.GetBufferCount();
}
-u64 In::GetPlayedSampleCount() {
+u64 In::GetPlayedSampleCount() const {
std::scoped_lock l{parent_mutex};
return system.GetPlayedSampleCount();
}
diff --git a/src/audio_core/in/audio_in.h b/src/audio_core/in/audio_in.h
index 6253891d5..092ab7236 100644
--- a/src/audio_core/in/audio_in.h
+++ b/src/audio_core/in/audio_in.h
@@ -102,7 +102,7 @@ public:
*
* @return The current volume.
*/
- f32 GetVolume();
+ f32 GetVolume() const;
/**
* Set the system volume.
@@ -117,21 +117,21 @@ public:
* @param tag - The tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
- bool ContainsAudioBuffer(u64 tag);
+ bool ContainsAudioBuffer(u64 tag) const;
/**
* Get the maximum number of buffers.
*
* @return The maximum number of buffers.
*/
- u32 GetBufferCount();
+ u32 GetBufferCount() const;
/**
* Get the total played sample count for this audio in.
*
* @return The played sample count.
*/
- u64 GetPlayedSampleCount();
+ u64 GetPlayedSampleCount() const;
private:
/// The AudioIn::Manager this audio in is registered with
diff --git a/src/audio_core/in/audio_in_system.cpp b/src/audio_core/in/audio_in_system.cpp
index ec5d37ed4..e7f918a47 100644
--- a/src/audio_core/in/audio_in_system.cpp
+++ b/src/audio_core/in/audio_in_system.cpp
@@ -34,16 +34,16 @@ size_t System::GetSessionId() const {
return session_id;
}
-std::string_view System::GetDefaultDeviceName() {
+std::string_view System::GetDefaultDeviceName() const {
return "BuiltInHeadset";
}
-std::string_view System::GetDefaultUacDeviceName() {
+std::string_view System::GetDefaultUacDeviceName() const {
return "Uac";
}
Result System::IsConfigValid(const std::string_view device_name,
- const AudioInParameter& in_params) {
+ const AudioInParameter& in_params) const {
if ((device_name.size() > 0) &&
(device_name != GetDefaultDeviceName() && device_name != GetDefaultUacDeviceName())) {
return Service::Audio::ERR_INVALID_DEVICE_NAME;
@@ -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 +113,15 @@ 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()};
+ const 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();
@@ -194,11 +202,11 @@ void System::SetVolume(const f32 volume_) {
session->SetVolume(volume_);
}
-bool System::ContainsAudioBuffer(const u64 tag) {
+bool System::ContainsAudioBuffer(const u64 tag) const {
return buffers.ContainsBuffer(tag);
}
-u32 System::GetBufferCount() {
+u32 System::GetBufferCount() const {
return buffers.GetAppendedRegisteredCount();
}
diff --git a/src/audio_core/in/audio_in_system.h b/src/audio_core/in/audio_in_system.h
index 165e35d83..b9dc0e60f 100644
--- a/src/audio_core/in/audio_in_system.h
+++ b/src/audio_core/in/audio_in_system.h
@@ -68,7 +68,7 @@ public:
*
* @return The default audio input device name.
*/
- std::string_view GetDefaultDeviceName();
+ std::string_view GetDefaultDeviceName() const;
/**
* Get the default USB audio input device name.
@@ -77,7 +77,7 @@ public:
*
* @return The default USB audio input device name.
*/
- std::string_view GetDefaultUacDeviceName();
+ std::string_view GetDefaultUacDeviceName() const;
/**
* Is the given initialize config valid?
@@ -86,7 +86,7 @@ public:
* @param in_params - Input parameters, see AudioInParameter.
* @return Result code.
*/
- Result IsConfigValid(std::string_view device_name, const AudioInParameter& in_params);
+ Result IsConfigValid(std::string_view device_name, const AudioInParameter& in_params) const;
/**
* Initialize this system.
@@ -208,7 +208,7 @@ public:
/**
* Set this system's current volume.
*
- * @param The new volume.
+ * @param volume The new volume.
*/
void SetVolume(f32 volume);
@@ -218,14 +218,14 @@ public:
* @param tag - Unique tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
- bool ContainsAudioBuffer(u64 tag);
+ bool ContainsAudioBuffer(u64 tag) const;
/**
* Get the maximum number of usable buffers (default 32).
*
* @return The number of buffers.
*/
- u32 GetBufferCount();
+ u32 GetBufferCount() const;
/**
* Get the total number of samples played by this system.
diff --git a/src/audio_core/out/audio_out.cpp b/src/audio_core/out/audio_out.cpp
index 9a8d8a742..d3ee4f0eb 100644
--- a/src/audio_core/out/audio_out.cpp
+++ b/src/audio_core/out/audio_out.cpp
@@ -72,7 +72,7 @@ Kernel::KReadableEvent& Out::GetBufferEvent() {
return event->GetReadableEvent();
}
-f32 Out::GetVolume() {
+f32 Out::GetVolume() const {
std::scoped_lock l{parent_mutex};
return system.GetVolume();
}
@@ -82,17 +82,17 @@ void Out::SetVolume(const f32 volume) {
system.SetVolume(volume);
}
-bool Out::ContainsAudioBuffer(const u64 tag) {
+bool Out::ContainsAudioBuffer(const u64 tag) const {
std::scoped_lock l{parent_mutex};
return system.ContainsAudioBuffer(tag);
}
-u32 Out::GetBufferCount() {
+u32 Out::GetBufferCount() const {
std::scoped_lock l{parent_mutex};
return system.GetBufferCount();
}
-u64 Out::GetPlayedSampleCount() {
+u64 Out::GetPlayedSampleCount() const {
std::scoped_lock l{parent_mutex};
return system.GetPlayedSampleCount();
}
diff --git a/src/audio_core/out/audio_out.h b/src/audio_core/out/audio_out.h
index f6b921645..946f345c6 100644
--- a/src/audio_core/out/audio_out.h
+++ b/src/audio_core/out/audio_out.h
@@ -102,7 +102,7 @@ public:
*
* @return The current volume.
*/
- f32 GetVolume();
+ f32 GetVolume() const;
/**
* Set the system volume.
@@ -117,21 +117,21 @@ public:
* @param tag - The tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
- bool ContainsAudioBuffer(u64 tag);
+ bool ContainsAudioBuffer(u64 tag) const;
/**
* Get the maximum number of buffers.
*
* @return The maximum number of buffers.
*/
- u32 GetBufferCount();
+ u32 GetBufferCount() const;
/**
* Get the total played sample count for this audio out.
*
* @return The played sample count.
*/
- u64 GetPlayedSampleCount();
+ u64 GetPlayedSampleCount() const;
private:
/// The AudioOut::Manager this audio out is registered with
diff --git a/src/audio_core/out/audio_out_system.cpp b/src/audio_core/out/audio_out_system.cpp
index 35afddf06..8b907590a 100644
--- a/src/audio_core/out/audio_out_system.cpp
+++ b/src/audio_core/out/audio_out_system.cpp
@@ -27,11 +27,12 @@ void System::Finalize() {
buffer_event->GetWritableEvent().Signal();
}
-std::string_view System::GetDefaultOutputDeviceName() {
+std::string_view System::GetDefaultOutputDeviceName() const {
return "DeviceOut";
}
-Result System::IsConfigValid(std::string_view device_name, const AudioOutParameter& in_params) {
+Result System::IsConfigValid(std::string_view device_name,
+ const AudioOutParameter& in_params) const {
if ((device_name.size() > 0) && (device_name != GetDefaultOutputDeviceName())) {
return Service::Audio::ERR_INVALID_DEVICE_NAME;
}
@@ -92,6 +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;
}
@@ -111,8 +113,15 @@ 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()};
+ const 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();
@@ -192,11 +201,11 @@ void System::SetVolume(const f32 volume_) {
session->SetVolume(volume_);
}
-bool System::ContainsAudioBuffer(const u64 tag) {
+bool System::ContainsAudioBuffer(const u64 tag) const {
return buffers.ContainsBuffer(tag);
}
-u32 System::GetBufferCount() {
+u32 System::GetBufferCount() const {
return buffers.GetAppendedRegisteredCount();
}
diff --git a/src/audio_core/out/audio_out_system.h b/src/audio_core/out/audio_out_system.h
index 4ca2f3417..0817b2f37 100644
--- a/src/audio_core/out/audio_out_system.h
+++ b/src/audio_core/out/audio_out_system.h
@@ -68,7 +68,7 @@ public:
*
* @return The default audio output device name.
*/
- std::string_view GetDefaultOutputDeviceName();
+ std::string_view GetDefaultOutputDeviceName() const;
/**
* Is the given initialize config valid?
@@ -77,7 +77,7 @@ public:
* @param in_params - Input parameters, see AudioOutParameter.
* @return Result code.
*/
- Result IsConfigValid(std::string_view device_name, const AudioOutParameter& in_params);
+ Result IsConfigValid(std::string_view device_name, const AudioOutParameter& in_params) const;
/**
* Initialize this system.
@@ -199,7 +199,7 @@ public:
/**
* Set this system's current volume.
*
- * @param The new volume.
+ * @param volume The new volume.
*/
void SetVolume(f32 volume);
@@ -209,14 +209,14 @@ public:
* @param tag - Unique tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
- bool ContainsAudioBuffer(u64 tag);
+ bool ContainsAudioBuffer(u64 tag) const;
/**
* Get the maximum number of usable buffers (default 32).
*
* @return The number of buffers.
*/
- u32 GetBufferCount();
+ u32 GetBufferCount() const;
/**
* Get the total number of samples played by this system.
diff --git a/src/audio_core/renderer/adsp/adsp.cpp b/src/audio_core/renderer/adsp/adsp.cpp
index e05a22d86..a28395663 100644
--- a/src/audio_core/renderer/adsp/adsp.cpp
+++ b/src/audio_core/renderer/adsp/adsp.cpp
@@ -50,7 +50,7 @@ u32 ADSP::GetRemainCommandCount(const u32 session_id) const {
return render_mailbox.GetRemainCommandCount(session_id);
}
-void ADSP::SendCommandBuffer(const u32 session_id, CommandBuffer& command_buffer) {
+void ADSP::SendCommandBuffer(const u32 session_id, const CommandBuffer& command_buffer) {
render_mailbox.SetCommandBuffer(session_id, command_buffer);
}
diff --git a/src/audio_core/renderer/adsp/adsp.h b/src/audio_core/renderer/adsp/adsp.h
index 4dfcef4a5..f7a2f25e4 100644
--- a/src/audio_core/renderer/adsp/adsp.h
+++ b/src/audio_core/renderer/adsp/adsp.h
@@ -63,8 +63,6 @@ public:
/**
* Stop the ADSP.
- *
- * @return True if started or already running, otherwise false.
*/
void Stop();
@@ -133,7 +131,7 @@ public:
* @param session_id - The session id to check (0 or 1).
* @param command_buffer - The command buffer to process.
*/
- void SendCommandBuffer(u32 session_id, CommandBuffer& command_buffer);
+ void SendCommandBuffer(u32 session_id, const CommandBuffer& command_buffer);
/**
* Clear the command buffers (does not clear the time taken or the remaining command count)
diff --git a/src/audio_core/renderer/adsp/audio_renderer.cpp b/src/audio_core/renderer/adsp/audio_renderer.cpp
index 3967ccfe6..bafe4822a 100644
--- a/src/audio_core/renderer/adsp/audio_renderer.cpp
+++ b/src/audio_core/renderer/adsp/audio_renderer.cpp
@@ -51,7 +51,7 @@ CommandBuffer& AudioRenderer_Mailbox::GetCommandBuffer(const s32 session_id) {
return command_buffers[session_id];
}
-void AudioRenderer_Mailbox::SetCommandBuffer(const u32 session_id, CommandBuffer& buffer) {
+void AudioRenderer_Mailbox::SetCommandBuffer(const u32 session_id, const CommandBuffer& buffer) {
command_buffers[session_id] = buffer;
}
@@ -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 =
diff --git a/src/audio_core/renderer/adsp/audio_renderer.h b/src/audio_core/renderer/adsp/audio_renderer.h
index b6ced9d2b..02e923c84 100644
--- a/src/audio_core/renderer/adsp/audio_renderer.h
+++ b/src/audio_core/renderer/adsp/audio_renderer.h
@@ -52,7 +52,7 @@ public:
/**
* Send a message from the host to the AudioRenderer.
*
- * @param message_ - The message to send to the AudioRenderer.
+ * @param message - The message to send to the AudioRenderer.
*/
void HostSendMessage(RenderMessage message);
@@ -66,7 +66,7 @@ public:
/**
* Send a message from the AudioRenderer to the host.
*
- * @param message_ - The message to send to the host.
+ * @param message - The message to send to the host.
*/
void ADSPSendMessage(RenderMessage message);
@@ -91,7 +91,7 @@ public:
* @param session_id - The session id to get (0 or 1).
* @param buffer - The command buffer to set.
*/
- void SetCommandBuffer(u32 session_id, CommandBuffer& buffer);
+ void SetCommandBuffer(u32 session_id, const CommandBuffer& buffer);
/**
* Get the total render time taken for the last command lists sent.
@@ -163,7 +163,7 @@ public:
/**
* Start the AudioRenderer.
*
- * @param The mailbox to use for this session.
+ * @param mailbox The mailbox to use for this session.
*/
void Start(AudioRenderer_Mailbox* mailbox);
diff --git a/src/audio_core/renderer/adsp/command_list_processor.h b/src/audio_core/renderer/adsp/command_list_processor.h
index 3f99173e3..d78269e1d 100644
--- a/src/audio_core/renderer/adsp/command_list_processor.h
+++ b/src/audio_core/renderer/adsp/command_list_processor.h
@@ -33,10 +33,10 @@ public:
/**
* Initialize the processor.
*
- * @param system_ - The core system.
- * @param buffer - The command buffer to process.
- * @param size - The size of the buffer.
- * @param stream_ - The stream to be used for sending the samples.
+ * @param system - The core system.
+ * @param buffer - The command buffer to process.
+ * @param size - The size of the buffer.
+ * @param stream - The stream to be used for sending the samples.
*/
void Initialize(Core::System& system, CpuAddr buffer, u64 size, Sink::SinkStream* stream);
@@ -72,7 +72,8 @@ public:
/**
* Process the command list.
*
- * @param index - Index of the current command list.
+ * @param session_id - Session ID for the commands being processed.
+ *
* @return The time taken to process.
*/
u64 Process(u32 session_id);
@@ -89,7 +90,7 @@ public:
u8* commands{};
/// The command buffer size
u64 commands_buffer_size{};
- /// The maximum processing time alloted
+ /// The maximum processing time allotted
u64 max_process_time{};
/// The number of commands in the buffer
u32 command_count{};
diff --git a/src/audio_core/renderer/audio_device.cpp b/src/audio_core/renderer/audio_device.cpp
index d5886e55e..0d9d8f6ce 100644
--- a/src/audio_core/renderer/audio_device.cpp
+++ b/src/audio_core/renderer/audio_device.cpp
@@ -1,6 +1,9 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
+#include <array>
+#include <span>
+
#include "audio_core/audio_core.h"
#include "audio_core/common/feature_support.h"
#include "audio_core/renderer/audio_device.h"
@@ -9,14 +12,33 @@
namespace AudioCore::AudioRenderer {
+constexpr std::array usb_device_names{
+ AudioDevice::AudioDeviceName{"AudioStereoJackOutput"},
+ AudioDevice::AudioDeviceName{"AudioBuiltInSpeakerOutput"},
+ AudioDevice::AudioDeviceName{"AudioTvOutput"},
+ AudioDevice::AudioDeviceName{"AudioUsbDeviceOutput"},
+};
+
+constexpr std::array device_names{
+ AudioDevice::AudioDeviceName{"AudioStereoJackOutput"},
+ AudioDevice::AudioDeviceName{"AudioBuiltInSpeakerOutput"},
+ AudioDevice::AudioDeviceName{"AudioTvOutput"},
+};
+
+constexpr std::array output_device_names{
+ AudioDevice::AudioDeviceName{"AudioBuiltInSpeakerOutput"},
+ AudioDevice::AudioDeviceName{"AudioTvOutput"},
+ AudioDevice::AudioDeviceName{"AudioExternalOutput"},
+};
+
AudioDevice::AudioDevice(Core::System& system, const u64 applet_resource_user_id_,
const u32 revision)
: output_sink{system.AudioCore().GetOutputSink()},
applet_resource_user_id{applet_resource_user_id_}, user_revision{revision} {}
u32 AudioDevice::ListAudioDeviceName(std::vector<AudioDeviceName>& out_buffer,
- const size_t max_count) {
- std::span<AudioDeviceName> names{};
+ const size_t max_count) const {
+ std::span<const AudioDeviceName> names{};
if (CheckFeatureSupported(SupportTags::AudioUsbDeviceOutput, user_revision)) {
names = usb_device_names;
@@ -24,7 +46,7 @@ u32 AudioDevice::ListAudioDeviceName(std::vector<AudioDeviceName>& out_buffer,
names = device_names;
}
- u32 out_count{static_cast<u32>(std::min(max_count, names.size()))};
+ const u32 out_count{static_cast<u32>(std::min(max_count, names.size()))};
for (u32 i = 0; i < out_count; i++) {
out_buffer.push_back(names[i]);
}
@@ -32,8 +54,8 @@ u32 AudioDevice::ListAudioDeviceName(std::vector<AudioDeviceName>& out_buffer,
}
u32 AudioDevice::ListAudioOutputDeviceName(std::vector<AudioDeviceName>& out_buffer,
- const size_t max_count) {
- u32 out_count{static_cast<u32>(std::min(max_count, output_device_names.size()))};
+ const size_t max_count) const {
+ const u32 out_count{static_cast<u32>(std::min(max_count, output_device_names.size()))};
for (u32 i = 0; i < out_count; i++) {
out_buffer.push_back(output_device_names[i]);
@@ -45,7 +67,7 @@ void AudioDevice::SetDeviceVolumes(const f32 volume) {
output_sink.SetDeviceVolume(volume);
}
-f32 AudioDevice::GetDeviceVolume([[maybe_unused]] std::string_view name) {
+f32 AudioDevice::GetDeviceVolume([[maybe_unused]] std::string_view name) const {
return output_sink.GetDeviceVolume();
}
diff --git a/src/audio_core/renderer/audio_device.h b/src/audio_core/renderer/audio_device.h
index 1f449f261..dd6be70ee 100644
--- a/src/audio_core/renderer/audio_device.h
+++ b/src/audio_core/renderer/audio_device.h
@@ -3,7 +3,7 @@
#pragma once
-#include <span>
+#include <string_view>
#include "audio_core/audio_render_manager.h"
@@ -23,21 +23,13 @@ namespace AudioRenderer {
class AudioDevice {
public:
struct AudioDeviceName {
- std::array<char, 0x100> name;
+ std::array<char, 0x100> name{};
- AudioDeviceName(const char* name_) {
- std::strncpy(name.data(), name_, name.size());
+ constexpr AudioDeviceName(std::string_view name_) {
+ name_.copy(name.data(), name.size() - 1);
}
};
- std::array<AudioDeviceName, 4> usb_device_names{"AudioStereoJackOutput",
- "AudioBuiltInSpeakerOutput", "AudioTvOutput",
- "AudioUsbDeviceOutput"};
- std::array<AudioDeviceName, 3> device_names{"AudioStereoJackOutput",
- "AudioBuiltInSpeakerOutput", "AudioTvOutput"};
- std::array<AudioDeviceName, 3> output_device_names{"AudioBuiltInSpeakerOutput", "AudioTvOutput",
- "AudioExternalOutput"};
-
explicit AudioDevice(Core::System& system, u64 applet_resource_user_id, u32 revision);
/**
@@ -47,7 +39,7 @@ public:
* @param max_count - Maximum number of devices to write (count of out_buffer).
* @return Number of device names written.
*/
- u32 ListAudioDeviceName(std::vector<AudioDeviceName>& out_buffer, size_t max_count);
+ u32 ListAudioDeviceName(std::vector<AudioDeviceName>& out_buffer, size_t max_count) const;
/**
* Get a list of the available output devices.
@@ -57,7 +49,7 @@ public:
* @param max_count - Maximum number of devices to write (count of out_buffer).
* @return Number of device names written.
*/
- u32 ListAudioOutputDeviceName(std::vector<AudioDeviceName>& out_buffer, size_t max_count);
+ u32 ListAudioOutputDeviceName(std::vector<AudioDeviceName>& out_buffer, size_t max_count) const;
/**
* Set the volume of all streams in the backend sink.
@@ -73,7 +65,7 @@ public:
* @param name - Name of the device to check. Unused.
* @return Volume of the device.
*/
- f32 GetDeviceVolume(std::string_view name);
+ f32 GetDeviceVolume(std::string_view name) const;
private:
/// Backend output sink for the device
diff --git a/src/audio_core/renderer/behavior/behavior_info.cpp b/src/audio_core/renderer/behavior/behavior_info.cpp
index c5d4d66d8..3d2a91312 100644
--- a/src/audio_core/renderer/behavior/behavior_info.cpp
+++ b/src/audio_core/renderer/behavior/behavior_info.cpp
@@ -34,7 +34,7 @@ void BehaviorInfo::ClearError() {
error_count = 0;
}
-void BehaviorInfo::AppendError(ErrorInfo& error) {
+void BehaviorInfo::AppendError(const ErrorInfo& error) {
LOG_ERROR(Service_Audio, "Error during RequestUpdate, reporting code {:04X} address {:08X}",
error.error_code.raw, error.address);
if (error_count < MaxErrors) {
@@ -42,14 +42,16 @@ 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));
+void BehaviorInfo::CopyErrorInfo(std::span<ErrorInfo> out_errors, u32& out_count) const {
+ 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_) {
diff --git a/src/audio_core/renderer/behavior/behavior_info.h b/src/audio_core/renderer/behavior/behavior_info.h
index 7333c297f..15c948344 100644
--- a/src/audio_core/renderer/behavior/behavior_info.h
+++ b/src/audio_core/renderer/behavior/behavior_info.h
@@ -94,7 +94,7 @@ public:
*
* @param error - The new error.
*/
- void AppendError(ErrorInfo& error);
+ void AppendError(const ErrorInfo& error);
/**
* Copy errors to the given output container.
@@ -102,7 +102,7 @@ public:
* @param out_errors - Output container to receive the errors.
* @param out_count - The number of errors written.
*/
- void CopyErrorInfo(std::span<ErrorInfo> out_errors, u32& out_count);
+ void CopyErrorInfo(std::span<ErrorInfo> out_errors, u32& out_count) const;
/**
* Update the behaviour flags.
diff --git a/src/audio_core/renderer/behavior/info_updater.cpp b/src/audio_core/renderer/behavior/info_updater.cpp
index 06a37e1a6..c0a307b89 100644
--- a/src/audio_core/renderer/behavior/info_updater.cpp
+++ b/src/audio_core/renderer/behavior/info_updater.cpp
@@ -485,7 +485,7 @@ Result InfoUpdater::UpdateBehaviorInfo(BehaviorInfo& behaviour_) {
return ResultSuccess;
}
-Result InfoUpdater::UpdateErrorInfo(BehaviorInfo& behaviour_) {
+Result InfoUpdater::UpdateErrorInfo(const BehaviorInfo& behaviour_) {
auto out_params{reinterpret_cast<BehaviorInfo::OutStatus*>(output)};
behaviour_.CopyErrorInfo(out_params->errors, out_params->error_count);
diff --git a/src/audio_core/renderer/behavior/info_updater.h b/src/audio_core/renderer/behavior/info_updater.h
index f0b445d9c..c817d8d8d 100644
--- a/src/audio_core/renderer/behavior/info_updater.h
+++ b/src/audio_core/renderer/behavior/info_updater.h
@@ -130,7 +130,7 @@ public:
* @param behaviour - Behaviour to update.
* @return Result code.
*/
- Result UpdateErrorInfo(BehaviorInfo& behaviour);
+ Result UpdateErrorInfo(const BehaviorInfo& behaviour);
/**
* Update splitter.
diff --git a/src/audio_core/renderer/command/command_buffer.h b/src/audio_core/renderer/command/command_buffer.h
index 496b0e50a..162170846 100644
--- a/src/audio_core/renderer/command/command_buffer.h
+++ b/src/audio_core/renderer/command/command_buffer.h
@@ -191,6 +191,7 @@ public:
* @param volume - Current mix volume used for calculating the ramp.
* @param prev_volume - Previous mix volume, used for calculating the ramp,
* also applied to the input.
+ * @param prev_samples - Previous sample buffer. Used for depopping.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixRampCommand(s32 node_id, s16 buffer_count, s16 input_index, s16 output_index,
@@ -208,6 +209,7 @@ public:
* @param volumes - Current mix volumes used for calculating the ramp.
* @param prev_volumes - Previous mix volumes, used for calculating the ramp,
* also applied to the input.
+ * @param prev_samples - Previous sample buffer. Used for depopping.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixRampGroupedCommand(s32 node_id, s16 buffer_count, s16 input_index,
@@ -297,11 +299,11 @@ public:
/**
* Generate a device sink command, adding it to the command list.
*
- * @param node_id - Node id of the voice this command is generated for.
- * @param buffer_offset - Base mix buffer offset to use.
- * @param sink_info - The sink_info to generate this command from.
- * @session_id - System session id this command is generated from.
- * @samples_buffer - The buffer to be sent to the sink if upsampling is not used.
+ * @param node_id - Node id of the voice this command is generated for.
+ * @param buffer_offset - Base mix buffer offset to use.
+ * @param sink_info - The sink_info to generate this command from.
+ * @param session_id - System session id this command is generated from.
+ * @param samples_buffer - The buffer to be sent to the sink if upsampling is not used.
*/
void GenerateDeviceSinkCommand(s32 node_id, s16 buffer_offset, SinkInfoBase& sink_info,
u32 session_id, std::span<s32> samples_buffer);
diff --git a/src/audio_core/renderer/command/command_generator.h b/src/audio_core/renderer/command/command_generator.h
index d80d9b0d8..b3cd7b408 100644
--- a/src/audio_core/renderer/command/command_generator.h
+++ b/src/audio_core/renderer/command/command_generator.h
@@ -197,9 +197,9 @@ public:
/**
* Generate an I3DL2 reverb effect command.
*
- * @param buffer_offset - Base mix buffer offset to use.
- * @param effect_info_base - I3DL2Reverb effect info.
- * @param node_id - Node id of the mix this command is generated for.
+ * @param buffer_offset - Base mix buffer offset to use.
+ * @param effect_info - I3DL2Reverb effect info.
+ * @param node_id - Node id of the mix this command is generated for.
*/
void GenerateI3dl2ReverbEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id);
@@ -207,18 +207,18 @@ public:
/**
* Generate an aux effect command.
*
- * @param buffer_offset - Base mix buffer offset to use.
- * @param effect_info_base - Aux effect info.
- * @param node_id - Node id of the mix this command is generated for.
+ * @param buffer_offset - Base mix buffer offset to use.
+ * @param effect_info - Aux effect info.
+ * @param node_id - Node id of the mix this command is generated for.
*/
void GenerateAuxCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate a biquad filter effect command.
*
- * @param buffer_offset - Base mix buffer offset to use.
- * @param effect_info_base - Aux effect info.
- * @param node_id - Node id of the mix this command is generated for.
+ * @param buffer_offset - Base mix buffer offset to use.
+ * @param effect_info - Aux effect info.
+ * @param node_id - Node id of the mix this command is generated for.
*/
void GenerateBiquadFilterEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id);
@@ -226,10 +226,10 @@ public:
/**
* Generate a light limiter effect command.
*
- * @param buffer_offset - Base mix buffer offset to use.
- * @param effect_info_base - Limiter effect info.
- * @param node_id - Node id of the mix this command is generated for.
- * @param effect_index - Index for the statistics state.
+ * @param buffer_offset - Base mix buffer offset to use.
+ * @param effect_info - Limiter effect info.
+ * @param node_id - Node id of the mix this command is generated for.
+ * @param effect_index - Index for the statistics state.
*/
void GenerateLightLimiterEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id, u32 effect_index);
@@ -238,21 +238,20 @@ public:
* Generate a capture effect command.
* Writes a mix buffer back to game memory.
*
- * @param buffer_offset - Base mix buffer offset to use.
- * @param effect_info_base - Capture effect info.
- * @param node_id - Node id of the mix this command is generated for.
+ * @param buffer_offset - Base mix buffer offset to use.
+ * @param effect_info - Capture effect info.
+ * @param node_id - Node id of the mix this command is generated for.
*/
void GenerateCaptureCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate a compressor effect command.
*
- * @param buffer_offset - Base mix buffer offset to use.
- * @param effect_info_base - Compressor effect info.
- * @param node_id - Node id of the mix this command is generated for.
+ * @param buffer_offset - Base mix buffer offset to use.
+ * @param effect_info - Compressor effect info.
+ * @param node_id - Node id of the mix this command is generated for.
*/
- void GenerateCompressorCommand(const s16 buffer_offset, EffectInfoBase& effect_info,
- const s32 node_id);
+ void GenerateCompressorCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate all effect commands for a mix.
@@ -318,8 +317,9 @@ public:
* Generate a performance command.
* Used to report performance metrics of the AudioRenderer back to the game.
*
- * @param buffer_offset - Base mix buffer offset to use.
- * @param sink_info - Sink info to generate the commands from.
+ * @param node_id - Node ID of the mix this command is generated for
+ * @param state - Output state of the generated performance command
+ * @param entry_addresses - Addresses to be written
*/
void GeneratePerformanceCommand(s32 node_id, PerformanceState state,
const PerformanceEntryAddresses& entry_addresses);
diff --git a/src/audio_core/renderer/command/effect/compressor.cpp b/src/audio_core/renderer/command/effect/compressor.cpp
index 2ebc140f1..7229618e8 100644
--- a/src/audio_core/renderer/command/effect/compressor.cpp
+++ b/src/audio_core/renderer/command/effect/compressor.cpp
@@ -11,7 +11,7 @@
namespace AudioCore::AudioRenderer {
-static void SetCompressorEffectParameter(CompressorInfo::ParameterVersion2& params,
+static void SetCompressorEffectParameter(const CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state) {
const auto ratio{1.0f / params.compressor_ratio};
auto makeup_gain{0.0f};
@@ -31,9 +31,9 @@ static void SetCompressorEffectParameter(CompressorInfo::ParameterVersion2& para
state.unk_20 = c;
}
-static void InitializeCompressorEffect(CompressorInfo::ParameterVersion2& params,
+static void InitializeCompressorEffect(const CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state) {
- std::memset(&state, 0, sizeof(CompressorInfo::State));
+ state = {};
state.unk_00 = 0;
state.unk_04 = 1.0f;
@@ -42,7 +42,7 @@ static void InitializeCompressorEffect(CompressorInfo::ParameterVersion2& params
SetCompressorEffectParameter(params, state);
}
-static void ApplyCompressorEffect(CompressorInfo::ParameterVersion2& params,
+static void ApplyCompressorEffect(const CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state, bool enabled,
std::vector<std::span<const s32>> input_buffers,
std::vector<std::span<s32>> output_buffers, u32 sample_count) {
@@ -103,8 +103,7 @@ static void ApplyCompressorEffect(CompressorInfo::ParameterVersion2& params,
} else {
for (s16 channel = 0; channel < params.channel_count; channel++) {
if (params.inputs[channel] != params.outputs[channel]) {
- std::memcpy((char*)output_buffers[channel].data(),
- (char*)input_buffers[channel].data(),
+ std::memcpy(output_buffers[channel].data(), input_buffers[channel].data(),
output_buffers[channel].size_bytes());
}
}
diff --git a/src/audio_core/renderer/command/mix/mix_ramp.cpp b/src/audio_core/renderer/command/mix/mix_ramp.cpp
index ffdafa1c8..d67123cd8 100644
--- a/src/audio_core/renderer/command/mix/mix_ramp.cpp
+++ b/src/audio_core/renderer/command/mix/mix_ramp.cpp
@@ -7,17 +7,7 @@
#include "common/logging/log.h"
namespace AudioCore::AudioRenderer {
-/**
- * Mix input mix buffer into output mix buffer, with volume applied to the input.
- *
- * @tparam Q - Number of bits for fixed point operations.
- * @param output - Output mix buffer.
- * @param input - Input mix buffer.
- * @param volume - Volume applied to the input.
- * @param ramp - Ramp applied to volume every sample.
- * @param sample_count - Number of samples to process.
- * @return The final gained input sample, used for depopping.
- */
+
template <size_t Q>
s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, const f32 volume_,
const f32 ramp_, const u32 sample_count) {
@@ -40,10 +30,8 @@ s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, const f32 vo
return sample.to_int();
}
-template s32 ApplyMixRamp<15>(std::span<s32>, std::span<const s32>, const f32, const f32,
- const u32);
-template s32 ApplyMixRamp<23>(std::span<s32>, std::span<const s32>, const f32, const f32,
- const u32);
+template s32 ApplyMixRamp<15>(std::span<s32>, std::span<const s32>, f32, f32, u32);
+template s32 ApplyMixRamp<23>(std::span<s32>, std::span<const s32>, f32, f32, u32);
void MixRampCommand::Dump(const ADSP::CommandListProcessor& processor, std::string& string) {
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
diff --git a/src/audio_core/renderer/command/mix/mix_ramp.h b/src/audio_core/renderer/command/mix/mix_ramp.h
index 770f57e80..52f74a273 100644
--- a/src/audio_core/renderer/command/mix/mix_ramp.h
+++ b/src/audio_core/renderer/command/mix/mix_ramp.h
@@ -61,13 +61,13 @@ struct MixRampCommand : ICommand {
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffer.
* @param input - Input mix buffer.
- * @param volume - Volume applied to the input.
- * @param ramp - Ramp applied to volume every sample.
+ * @param volume_ - Volume applied to the input.
+ * @param ramp_ - Ramp applied to volume every sample.
* @param sample_count - Number of samples to process.
* @return The final gained input sample, used for depopping.
*/
template <size_t Q>
-s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, const f32 volume_,
- const f32 ramp_, const u32 sample_count);
+s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, f32 volume_, f32 ramp_,
+ u32 sample_count);
} // namespace AudioCore::AudioRenderer
diff --git a/src/audio_core/renderer/command/mix/mix_ramp_grouped.h b/src/audio_core/renderer/command/mix/mix_ramp_grouped.h
index 027276e5a..3b0ce67ef 100644
--- a/src/audio_core/renderer/command/mix/mix_ramp_grouped.h
+++ b/src/audio_core/renderer/command/mix/mix_ramp_grouped.h
@@ -50,9 +50,9 @@ struct MixRampGroupedCommand : ICommand {
std::array<s16, MaxMixBuffers> inputs;
/// Output mix buffer indexes for each mix buffer
std::array<s16, MaxMixBuffers> outputs;
- /// Previous mix vloumes for each mix buffer
+ /// Previous mix volumes for each mix buffer
std::array<f32, MaxMixBuffers> prev_volumes;
- /// Current mix vloumes for each mix buffer
+ /// Current mix volumes for each mix buffer
std::array<f32, MaxMixBuffers> volumes;
/// Pointer to the previous sample buffer, used for depop
CpuAddr previous_samples;
diff --git a/src/audio_core/renderer/command/sink/device.cpp b/src/audio_core/renderer/command/sink/device.cpp
index 47e0c6722..e88372a75 100644
--- a/src/audio_core/renderer/command/sink/device.cpp
+++ b/src/audio_core/renderer/command/sink/device.cpp
@@ -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) {
diff --git a/src/audio_core/renderer/effect/effect_context.h b/src/audio_core/renderer/effect/effect_context.h
index 85955bd9c..8f6d6e7d8 100644
--- a/src/audio_core/renderer/effect/effect_context.h
+++ b/src/audio_core/renderer/effect/effect_context.h
@@ -15,15 +15,15 @@ class EffectContext {
public:
/**
* Initialize the effect context
- * @param effect_infos List of effect infos for this context
- * @param effect_count The number of effects in the list
- * @param result_states_cpu The workbuffer of result states for the CPU for this context
- * @param result_states_dsp The workbuffer of result states for the DSP for this context
- * @param state_count The number of result states
+ * @param effect_infos_ - List of effect infos for this context
+ * @param effect_count_ - The number of effects in the list
+ * @param result_states_cpu_ - The workbuffer of result states for the CPU for this context
+ * @param result_states_dsp_ - The workbuffer of result states for the DSP for this context
+ * @param dsp_state_count - The number of result states
*/
- void Initialize(std::span<EffectInfoBase> effect_infos_, const u32 effect_count_,
+ void Initialize(std::span<EffectInfoBase> effect_infos_, u32 effect_count_,
std::span<EffectResultState> result_states_cpu_,
- std::span<EffectResultState> result_states_dsp_, const size_t dsp_state_count);
+ std::span<EffectResultState> result_states_dsp_, size_t dsp_state_count);
/**
* Get the EffectInfo for a given index
diff --git a/src/audio_core/renderer/effect/effect_info_base.h b/src/audio_core/renderer/effect/effect_info_base.h
index 8c9583878..8525fde05 100644
--- a/src/audio_core/renderer/effect/effect_info_base.h
+++ b/src/audio_core/renderer/effect/effect_info_base.h
@@ -291,7 +291,7 @@ public:
* Update the info with new parameters, version 1.
*
* @param error_info - Used to write call result code.
- * @param in_params - New parameters to update the info with.
+ * @param params - New parameters to update the info with.
* @param pool_mapper - Pool for mapping buffers.
*/
virtual void Update(BehaviorInfo::ErrorInfo& error_info,
@@ -305,7 +305,7 @@ public:
* Update the info with new parameters, version 2.
*
* @param error_info - Used to write call result code.
- * @param in_params - New parameters to update the info with.
+ * @param params - New parameters to update the info with.
* @param pool_mapper - Pool for mapping buffers.
*/
virtual void Update(BehaviorInfo::ErrorInfo& error_info,
diff --git a/src/audio_core/renderer/effect/i3dl2.h b/src/audio_core/renderer/effect/i3dl2.h
index 7a088a627..1ebbc5c4c 100644
--- a/src/audio_core/renderer/effect/i3dl2.h
+++ b/src/audio_core/renderer/effect/i3dl2.h
@@ -99,7 +99,7 @@ public:
return out_sample;
}
- Common::FixedPoint<50, 14> Read() {
+ Common::FixedPoint<50, 14> Read() const {
return *output;
}
@@ -110,7 +110,7 @@ public:
}
}
- Common::FixedPoint<50, 14> TapOut(const s32 index) {
+ Common::FixedPoint<50, 14> TapOut(const s32 index) const {
auto out{input - (index + 1)};
if (out < buffer.data()) {
out += max_delay + 1;
diff --git a/src/audio_core/renderer/effect/reverb.h b/src/audio_core/renderer/effect/reverb.h
index b4df9f6ef..a72475c3c 100644
--- a/src/audio_core/renderer/effect/reverb.h
+++ b/src/audio_core/renderer/effect/reverb.h
@@ -95,7 +95,7 @@ public:
return out_sample;
}
- Common::FixedPoint<50, 14> Read() {
+ Common::FixedPoint<50, 14> Read() const {
return *output;
}
@@ -106,7 +106,7 @@ public:
}
}
- Common::FixedPoint<50, 14> TapOut(const s32 index) {
+ Common::FixedPoint<50, 14> TapOut(const s32 index) const {
auto out{input - (index + 1)};
if (out < buffer.data()) {
out += sample_count;
diff --git a/src/audio_core/renderer/memory/address_info.h b/src/audio_core/renderer/memory/address_info.h
index 4cfefea8e..bb5c930e1 100644
--- a/src/audio_core/renderer/memory/address_info.h
+++ b/src/audio_core/renderer/memory/address_info.h
@@ -19,8 +19,8 @@ public:
/**
* Setup a new AddressInfo.
*
- * @param cpu_address - The CPU address of this region.
- * @param size - The size of this region.
+ * @param cpu_address_ - The CPU address of this region.
+ * @param size_ - The size of this region.
*/
void Setup(CpuAddr cpu_address_, u64 size_) {
cpu_address = cpu_address_;
@@ -42,7 +42,6 @@ public:
* Assign this region to a memory pool.
*
* @param memory_pool_ - Memory pool to assign.
- * @return The CpuAddr address of this region.
*/
void SetPool(MemoryPoolInfo* memory_pool_) {
memory_pool = memory_pool_;
diff --git a/src/audio_core/renderer/nodes/node_states.h b/src/audio_core/renderer/nodes/node_states.h
index a1e0958a2..94b1d1254 100644
--- a/src/audio_core/renderer/nodes/node_states.h
+++ b/src/audio_core/renderer/nodes/node_states.h
@@ -56,7 +56,7 @@ class NodeStates {
*
* @return The current stack position.
*/
- u32 Count() {
+ u32 Count() const {
return pos;
}
@@ -83,7 +83,7 @@ class NodeStates {
*
* @return The node on the top of the stack.
*/
- u32 top() {
+ u32 top() const {
return stack[pos - 1];
}
@@ -112,11 +112,11 @@ public:
/**
* Initialize the node states.
*
- * @param buffer - The workbuffer to use. Unused.
+ * @param buffer_ - The workbuffer to use. Unused.
* @param node_buffer_size - The size of the workbuffer. Unused.
* @param count - The number of nodes in the graph.
*/
- void Initialize(std::span<u8> nodes, u64 node_buffer_size, u32 count);
+ void Initialize(std::span<u8> buffer_, u64 node_buffer_size, u32 count);
/**
* Sort the graph. Only calls DepthFirstSearch.
diff --git a/src/audio_core/renderer/performance/performance_manager.h b/src/audio_core/renderer/performance/performance_manager.h
index b82176bef..b65caa9b6 100644
--- a/src/audio_core/renderer/performance/performance_manager.h
+++ b/src/audio_core/renderer/performance/performance_manager.h
@@ -73,7 +73,8 @@ public:
* Calculate the required size for the performance workbuffer.
*
* @param behavior - Check which version is supported.
- * @param params - Input parameters.
+ * @param params - Input parameters.
+ *
* @return Required workbuffer size.
*/
static u64 GetRequiredBufferSizeForPerformanceMetricsPerFrame(
@@ -104,7 +105,7 @@ public:
* @param workbuffer - Workbuffer to use for performance frames.
* @param workbuffer_size - Size of the workbuffer.
* @param params - Input parameters.
- * @param behavior - Behaviour to check version and data format.
+ * @param behavior - Behaviour to check version and data format.
* @param memory_pool - Used to translate the workbuffer address for the DSP.
*/
virtual void Initialize(std::span<u8> workbuffer, u64 workbuffer_size,
@@ -160,7 +161,8 @@ public:
* workbuffer, to be written by the AudioRenderer.
*
* @param addresses - Filled with pointers to the new detail, which should be passed
- * to the AudioRenderer with Performance commands to be written.
+ * to the AudioRenderer with Performance commands to be written.
+ * @param detail_type - Performance detail type.
* @param entry_type - The type of this detail. See PerformanceEntryType
* @param node_id - Node id for this detail.
* @return True if a new detail was created and the offsets are valid, otherwise false.
diff --git a/src/audio_core/renderer/system_manager.cpp b/src/audio_core/renderer/system_manager.cpp
index b326819ed..9c1331e19 100644
--- a/src/audio_core/renderer/system_manager.cpp
+++ b/src/audio_core/renderer/system_manager.cpp
@@ -15,17 +15,14 @@ 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()},
thread_event{Core::Timing::CreateEvent(
"AudioRendererSystemManager", [this](std::uintptr_t, s64 time, std::chrono::nanoseconds) {
return ThreadFunc2(time);
- })} {
- core.CoreTiming().RegisterPauseCallback([this](bool paused) { PauseCallback(paused); });
-}
+ })} {}
SystemManager::~SystemManager() {
Stop();
@@ -36,8 +33,8 @@ 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);
}
}
@@ -121,42 +118,9 @@ 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;
-}
-
-void SystemManager::PauseCallback(bool paused) {
- if (paused && core.IsPoweredOn() && core.IsShuttingDown()) {
- update.store(true);
- update.notify_all();
- }
+ return std::nullopt;
}
} // namespace AudioCore::AudioRenderer
diff --git a/src/audio_core/renderer/system_manager.h b/src/audio_core/renderer/system_manager.h
index 1291e9e0e..81457a3a1 100644
--- a/src/audio_core/renderer/system_manager.h
+++ b/src/audio_core/renderer/system_manager.h
@@ -73,13 +73,6 @@ private:
*/
std::optional<std::chrono::nanoseconds> ThreadFunc2(s64 time);
- /**
- * Callback from core timing when pausing, used to detect shutdowns and stop ThreadFunc.
- *
- * @param paused - Are we pausing or resuming?
- */
- void PauseCallback(bool paused);
-
enum class StreamState {
Filling,
Steady,
@@ -106,8 +99,6 @@ private:
std::shared_ptr<Core::Timing::EventType> thread_event;
/// Atomic for main thread to wait on
std::atomic<bool> update{};
- /// Current state of the streams
- StreamState state{StreamState::Filling};
};
} // namespace AudioCore::AudioRenderer
diff --git a/src/audio_core/renderer/upsampler/upsampler_manager.h b/src/audio_core/renderer/upsampler/upsampler_manager.h
index 70cd42b08..83c697c0c 100644
--- a/src/audio_core/renderer/upsampler/upsampler_manager.h
+++ b/src/audio_core/renderer/upsampler/upsampler_manager.h
@@ -27,7 +27,7 @@ public:
/**
* Free the given upsampler.
*
- * @param The upsampler to be freed.
+ * @param info The upsampler to be freed.
*/
void Free(UpsamplerInfo* info);
diff --git a/src/audio_core/renderer/voice/voice_info.h b/src/audio_core/renderer/voice/voice_info.h
index 896723e0c..930180895 100644
--- a/src/audio_core/renderer/voice/voice_info.h
+++ b/src/audio_core/renderer/voice/voice_info.h
@@ -185,7 +185,8 @@ public:
/**
* Does this voice ned an update?
*
- * @param params - Input parametetrs to check matching.
+ * @param params - Input parameters to check matching.
+ *
* @return True if this voice needs an update, otherwise false.
*/
bool ShouldUpdateParameters(const InParameter& params) const;
@@ -194,9 +195,9 @@ public:
* Update the parameters of this voice.
*
* @param error_info - Output error code.
- * @param params - Input parametters to udpate from.
+ * @param params - Input parameters to update from.
* @param pool_mapper - Used to map buffers.
- * @param behavior - behavior to check supported features.
+ * @param behavior - behavior to check supported features.
*/
void UpdateParameters(BehaviorInfo::ErrorInfo& error_info, const InParameter& params,
const PoolMapper& pool_mapper, const BehaviorInfo& behavior);
@@ -218,12 +219,12 @@ public:
/**
* Update all wavebuffers.
*
- * @param error_infos - Output 2D array of errors, 2 per wavebuffer.
- * @param error_count - Number of errors provided. Unused.
- * @param params - Input parametters to be used for the update.
+ * @param error_infos - Output 2D array of errors, 2 per wavebuffer.
+ * @param error_count - Number of errors provided. Unused.
+ * @param params - Input parameters to be used for the update.
* @param voice_states - The voice states for each channel in this voice to be updated.
- * @param pool_mapper - Used to map the wavebuffers.
- * @param behavior - Used to check for supported features.
+ * @param pool_mapper - Used to map the wavebuffers.
+ * @param behavior - Used to check for supported features.
*/
void UpdateWaveBuffers(std::span<std::array<BehaviorInfo::ErrorInfo, 2>> error_infos,
u32 error_count, const InParameter& params,
@@ -233,13 +234,13 @@ public:
/**
* Update a wavebuffer.
*
- * @param error_infos - Output array of errors.
+ * @param error_info - Output array of errors.
* @param wave_buffer - The wavebuffer to be updated.
* @param wave_buffer_internal - Input parametters to be used for the update.
* @param sample_format - Sample format of the wavebuffer.
* @param valid - Is this wavebuffer valid?
* @param pool_mapper - Used to map the wavebuffers.
- * @param behavior - Used to check for supported features.
+ * @param behavior - Used to check for supported features.
*/
void UpdateWaveBuffer(std::span<BehaviorInfo::ErrorInfo> error_info, WaveBuffer& wave_buffer,
const WaveBufferInternal& wave_buffer_internal,
@@ -276,7 +277,7 @@ public:
/**
* Check if this voice has any mixing connections.
*
- * @return True if this voice participes in mixing, otherwise false.
+ * @return True if this voice participates in mixing, otherwise false.
*/
bool HasAnyConnection() const;
@@ -301,7 +302,8 @@ public:
/**
* Update this voice on command generation.
*
- * @param voice_states - Voice states for these wavebuffers.
+ * @param voice_context - Voice context for these wavebuffers.
+ *
* @return True if this voice should be generated, otherwise false.
*/
bool UpdateForCommandGeneration(VoiceContext& voice_context);
diff --git a/src/audio_core/sink/cubeb_sink.cpp b/src/audio_core/sink/cubeb_sink.cpp
index 90d049e8e..36b115ad6 100644
--- a/src/audio_core/sink/cubeb_sink.cpp
+++ b/src/audio_core/sink/cubeb_sink.cpp
@@ -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;
}
@@ -136,21 +128,14 @@ 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 {
- if (!ctx) {
+ void Start(bool resume = false) override {
+ if (!ctx || !paused) {
return;
}
- if (resume && was_playing) {
- if (cubeb_stream_start(stream_backend) != CUBEB_OK) {
- LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream");
- }
- paused = false;
- } else if (!resume) {
- if (cubeb_stream_start(stream_backend) != CUBEB_OK) {
- LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream");
- }
- paused = false;
+ paused = false;
+ if (cubeb_stream_start(stream_backend) != CUBEB_OK) {
+ LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream");
}
}
@@ -158,207 +143,20 @@ public:
* Stop the sink stream.
*/
void Stop() override {
- if (!ctx) {
+ Unstall();
+
+ if (!ctx || paused) {
return;
}
+ paused = true;
if (cubeb_stream_stop(stream_backend) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error stopping cubeb stream");
}
-
- was_playing.store(!paused);
- 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 +176,15 @@ 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 +197,12 @@ 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 +256,15 @@ 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();
@@ -591,18 +278,6 @@ void CubebSink::CloseStreams() {
sink_streams.clear();
}
-void CubebSink::PauseStreams() {
- for (auto& stream : sink_streams) {
- stream->Stop();
- }
-}
-
-void CubebSink::UnpauseStreams() {
- for (auto& stream : sink_streams) {
- stream->Start(true);
- }
-}
-
f32 CubebSink::GetDeviceVolume() const {
if (sink_streams.empty()) {
return 1.0f;
@@ -611,19 +286,19 @@ 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;
diff --git a/src/audio_core/sink/cubeb_sink.h b/src/audio_core/sink/cubeb_sink.h
index f0f43dfa1..4b0cb160d 100644
--- a/src/audio_core/sink/cubeb_sink.h
+++ b/src/audio_core/sink/cubeb_sink.h
@@ -34,8 +34,7 @@ public:
* May differ from the device's channel count.
* @param name - Name of this stream.
* @param type - Type of this stream, render/in/out.
- * @param event - Audio render only, a signal used to prevent the renderer running too
- * fast.
+ *
* @return A pointer to the created SinkStream
*/
SinkStream* AcquireSinkStream(Core::System& system, u32 system_channels,
@@ -46,7 +45,7 @@ public:
*
* @param stream - The stream to close.
*/
- void CloseStream(const SinkStream* stream) override;
+ void CloseStream(SinkStream* stream) override;
/**
* Close all streams.
@@ -54,16 +53,6 @@ public:
void CloseStreams() override;
/**
- * Pause all streams.
- */
- void PauseStreams() override;
-
- /**
- * Unpause all streams.
- */
- void UnpauseStreams() override;
-
- /**
* Get the device volume. Set from calls to the IAudioDevice service.
*
* @return Volume of the device.
@@ -101,7 +90,7 @@ private:
};
/**
- * Get a list of conencted devices from Cubeb.
+ * Get a list of connected devices from Cubeb.
*
* @param capture - Return input (capture) devices if true, otherwise output devices.
*/
diff --git a/src/audio_core/sink/null_sink.h b/src/audio_core/sink/null_sink.h
index 47a342171..1215d3cd2 100644
--- a/src/audio_core/sink/null_sink.h
+++ b/src/audio_core/sink/null_sink.h
@@ -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.
*/
@@ -15,17 +34,16 @@ 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 {}
f32 GetDeviceVolume() const override {
return 1.0f;
}
@@ -33,20 +51,7 @@ 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
diff --git a/src/audio_core/sink/sdl2_sink.cpp b/src/audio_core/sink/sdl2_sink.cpp
index d6c9ec90d..1bd001b94 100644
--- a/src/audio_core/sink/sdl2_sink.cpp
+++ b/src/audio_core/sink/sdl2_sink.cpp
@@ -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);
}
@@ -118,217 +107,29 @@ 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 {
- if (device == 0) {
+ void Start(bool resume = false) override {
+ if (device == 0 || !paused) {
return;
}
- if (resume && was_playing) {
- SDL_PauseAudioDevice(device, 0);
- paused = false;
- } else if (!resume) {
- SDL_PauseAudioDevice(device, 0);
- paused = false;
- }
+ paused = false;
+ SDL_PauseAudioDevice(device, 0);
}
/**
* Stop the sink stream.
*/
- void Stop() {
- if (device == 0) {
+ void Stop() override {
+ Unstall();
+ if (device == 0 || paused) {
return;
}
- SDL_PauseAudioDevice(device, 1);
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;
+ SDL_PauseAudioDevice(device, 1);
}
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 +146,20 @@ 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 +181,14 @@ 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();
@@ -503,18 +202,6 @@ void SDLSink::CloseStreams() {
sink_streams.clear();
}
-void SDLSink::PauseStreams() {
- for (auto& stream : sink_streams) {
- stream->Stop();
- }
-}
-
-void SDLSink::UnpauseStreams() {
- for (auto& stream : sink_streams) {
- stream->Start();
- }
-}
-
f32 SDLSink::GetDeviceVolume() const {
if (sink_streams.empty()) {
return 1.0f;
@@ -523,19 +210,19 @@ 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)) {
diff --git a/src/audio_core/sink/sdl2_sink.h b/src/audio_core/sink/sdl2_sink.h
index 186bc2fa3..f01eddc1b 100644
--- a/src/audio_core/sink/sdl2_sink.h
+++ b/src/audio_core/sink/sdl2_sink.h
@@ -32,8 +32,7 @@ public:
* May differ from the device's channel count.
* @param name - Name of this stream.
* @param type - Type of this stream, render/in/out.
- * @param event - Audio render only, a signal used to prevent the renderer running too
- * fast.
+ *
* @return A pointer to the created SinkStream
*/
SinkStream* AcquireSinkStream(Core::System& system, u32 system_channels,
@@ -44,7 +43,7 @@ public:
*
* @param stream - The stream to close.
*/
- void CloseStream(const SinkStream* stream) override;
+ void CloseStream(SinkStream* stream) override;
/**
* Close all streams.
@@ -52,16 +51,6 @@ public:
void CloseStreams() override;
/**
- * Pause all streams.
- */
- void PauseStreams() override;
-
- /**
- * Unpause all streams.
- */
- void UnpauseStreams() override;
-
- /**
* Get the device volume. Set from calls to the IAudioDevice service.
*
* @return Volume of the device.
@@ -92,7 +81,7 @@ private:
};
/**
- * Get a list of conencted devices from Cubeb.
+ * Get a list of connected devices from SDL.
*
* @param capture - Return input (capture) devices if true, otherwise output devices.
*/
diff --git a/src/audio_core/sink/sink.h b/src/audio_core/sink/sink.h
index 91fe455e4..f28c6d126 100644
--- a/src/audio_core/sink/sink.h
+++ b/src/audio_core/sink/sink.h
@@ -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.
@@ -40,16 +40,6 @@ public:
virtual void CloseStreams() = 0;
/**
- * Pause all streams.
- */
- virtual void PauseStreams() = 0;
-
- /**
- * Unpause all streams.
- */
- virtual void UnpauseStreams() = 0;
-
- /**
* Create a new sink stream, kept within this sink, with a pointer returned for use.
* Do not free the returned pointer. When done with the stream, call CloseStream on the sink.
*
@@ -58,8 +48,7 @@ public:
* May differ from the device's channel count.
* @param name - Name of this stream.
* @param type - Type of this stream, render/in/out.
- * @param event - Audio render only, a signal used to prevent the renderer running too
- * fast.
+ *
* @return A pointer to the created SinkStream
*/
virtual SinkStream* AcquireSinkStream(Core::System& system, u32 system_channels,
diff --git a/src/audio_core/sink/sink_details.cpp b/src/audio_core/sink/sink_details.cpp
index 253c0fd1e..67bdab779 100644
--- a/src/audio_core/sink/sink_details.cpp
+++ b/src/audio_core/sink/sink_details.cpp
@@ -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.
diff --git a/src/audio_core/sink/sink_stream.cpp b/src/audio_core/sink/sink_stream.cpp
new file mode 100644
index 000000000..37fe725e4
--- /dev/null
+++ b/src/audio_core/sink/sink_stream.cpp
@@ -0,0 +1,279 @@
+// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
+// SPDX-License-Identifier: GPL-2.0-or-later
+
+#include <array>
+#include <atomic>
+#include <memory>
+#include <span>
+#include <vector>
+
+#include "audio_core/audio_core.h"
+#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 we're paused or going to shut down, we don't want to consume buffers as coretiming is
+ // paused and we'll desync, so just return.
+ if (system.IsPaused() || system.IsShuttingDown()) {
+ return;
+ }
+
+ 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 we're paused or going to shut down, we don't want to consume buffers as coretiming is
+ // paused and we'll desync, so just play silence.
+ if (system.IsPaused() || system.IsShuttingDown()) {
+ constexpr std::array<s16, 6> silence{};
+ for (size_t i = frames_written; i < num_frames; i++) {
+ std::memcpy(&output_buffer[i * frame_size], &silence[0], frame_size_bytes);
+ }
+ return;
+ }
+
+ // Due to many frames being queued up with nvdec (5 frames or so?), a lot of buffers also get
+ // queued up (30+) but not all at once, which causes constant stalling here, so just let the
+ // video play out without attempting to stall.
+ // Can hopefully remove this later with a more complete NVDEC implementation.
+ const auto nvdec_active{system.AudioCore().IsNVDECActive()};
+ if (!nvdec_active && 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
diff --git a/src/audio_core/sink/sink_stream.h b/src/audio_core/sink/sink_stream.h
index 17ed6593f..38a4b2f51 100644
--- a/src/audio_core/sink/sink_stream.h
+++ b/src/audio_core/sink/sink_stream.h
@@ -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;
}
@@ -128,34 +111,6 @@ public:
}
/**
- * 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.
*
* @return The current system volume.
@@ -196,27 +151,97 @@ public:
*
* @return The number of queued buffers.
*/
- u32 GetQueueSize() {
+ u32 GetQueueSize() const {
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{};
+ /// Core system
+ Core::System& system;
+ /// Type of this stream
+ StreamType type;
/// 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};
- /// Set by the audio render/in/out systen which uses this stream
u32 system_channels{2};
/// Channels supported by hardware
u32 device_channels{2};
/// Is this stream currently paused?
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>;