Rework audio output, connecting AudioOut into coretiming to fix desync during heavy loads.
This commit is contained in:
parent
a83a5d2e4c
commit
ea9ff71725
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@ -194,6 +194,7 @@ add_library(audio_core STATIC
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sink/sink.h
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sink/sink_details.cpp
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sink/sink_details.h
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sink/sink_stream.cpp
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sink/sink_stream.h
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)
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@ -57,12 +57,4 @@ void AudioCore::PauseSinks(const bool pausing) const {
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}
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}
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u32 AudioCore::GetStreamQueue() const {
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return estimated_queue.load();
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}
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void AudioCore::SetStreamQueue(u32 size) {
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estimated_queue.store(size);
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}
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} // namespace AudioCore
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@ -65,20 +65,6 @@ public:
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*/
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void PauseSinks(bool pausing) const;
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/**
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* Get the size of the current stream queue.
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*
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* @return Current stream queue size.
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*/
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u32 GetStreamQueue() const;
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/**
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* Get the size of the current stream queue.
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*
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* @param size - New stream size.
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*/
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void SetStreamQueue(u32 size);
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private:
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/**
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* Create the sinks on startup.
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@ -93,8 +79,6 @@ private:
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std::unique_ptr<Sink::Sink> input_sink;
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/// The ADSP in the sysmodule
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std::unique_ptr<AudioRenderer::ADSP::ADSP> adsp;
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/// Current size of the stream queue
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std::atomic<u32> estimated_queue{0};
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};
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} // namespace AudioCore
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@ -8,6 +8,10 @@
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namespace AudioCore {
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struct AudioBuffer {
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/// Timestamp this buffer started playing.
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u64 start_timestamp;
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/// Timestamp this buffer should finish playing.
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u64 end_timestamp;
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/// Timestamp this buffer completed playing.
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s64 played_timestamp;
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/// Game memory address for these samples.
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@ -58,6 +58,7 @@ public:
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if (index < 0) {
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index += N;
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}
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out_buffers.push_back(buffers[index]);
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registered_count++;
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registered_index = (registered_index + 1) % append_limit;
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@ -100,7 +101,7 @@ public:
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}
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// Check with the backend if this buffer can be released yet.
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if (!session.IsBufferConsumed(buffers[index].tag)) {
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if (!session.IsBufferConsumed(buffers[index])) {
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break;
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}
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@ -280,6 +281,16 @@ public:
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return true;
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}
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u64 GetNextTimestamp() const {
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// Iterate backwards through the buffer queue, and take the most recent buffer's end
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std::scoped_lock l{lock};
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auto index{appended_index - 1};
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if (index < 0) {
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index += append_limit;
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}
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return buffers[index].end_timestamp;
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}
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private:
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/// Buffer lock
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mutable std::recursive_mutex lock{};
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@ -7,11 +7,20 @@
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#include "audio_core/device/device_session.h"
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#include "audio_core/sink/sink_stream.h"
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#include "core/core.h"
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#include "core/core_timing.h"
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#include "core/memory.h"
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namespace AudioCore {
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DeviceSession::DeviceSession(Core::System& system_) : system{system_} {}
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using namespace std::literals;
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constexpr auto INCREMENT_TIME{5ms};
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DeviceSession::DeviceSession(Core::System& system_)
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: system{system_}, thread_event{Core::Timing::CreateEvent(
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"AudioOutSampleTick",
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[this](std::uintptr_t, s64 time, std::chrono::nanoseconds) {
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return ThreadFunc();
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})} {}
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DeviceSession::~DeviceSession() {
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Finalize();
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@ -50,20 +59,21 @@ void DeviceSession::Finalize() {
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}
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void DeviceSession::Start() {
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stream->SetPlayedSampleCount(played_sample_count);
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stream->Start();
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if (stream) {
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stream->Start();
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system.CoreTiming().ScheduleLoopingEvent(std::chrono::nanoseconds::zero(), INCREMENT_TIME,
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thread_event);
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}
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}
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void DeviceSession::Stop() {
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if (stream) {
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played_sample_count = stream->GetPlayedSampleCount();
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stream->Stop();
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system.CoreTiming().UnscheduleEvent(thread_event, {});
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}
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}
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void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
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auto& memory{system.Memory()};
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for (size_t i = 0; i < buffers.size(); i++) {
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Sink::SinkBuffer new_buffer{
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.frames = buffers[i].size / (channel_count * sizeof(s16)),
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@ -77,7 +87,7 @@ void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
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stream->AppendBuffer(new_buffer, samples);
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} else {
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std::vector<s16> samples(buffers[i].size / sizeof(s16));
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memory.ReadBlockUnsafe(buffers[i].samples, samples.data(), buffers[i].size);
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system.Memory().ReadBlockUnsafe(buffers[i].samples, samples.data(), buffers[i].size);
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stream->AppendBuffer(new_buffer, samples);
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}
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}
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@ -85,17 +95,13 @@ void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
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void DeviceSession::ReleaseBuffer(AudioBuffer& buffer) const {
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if (type == Sink::StreamType::In) {
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auto& memory{system.Memory()};
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auto samples{stream->ReleaseBuffer(buffer.size / sizeof(s16))};
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memory.WriteBlockUnsafe(buffer.samples, samples.data(), buffer.size);
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system.Memory().WriteBlockUnsafe(buffer.samples, samples.data(), buffer.size);
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}
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}
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bool DeviceSession::IsBufferConsumed(u64 tag) const {
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if (stream) {
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return stream->IsBufferConsumed(tag);
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}
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return true;
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bool DeviceSession::IsBufferConsumed(AudioBuffer& buffer) const {
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return played_sample_count >= buffer.end_timestamp;
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}
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void DeviceSession::SetVolume(f32 volume) const {
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@ -105,10 +111,22 @@ void DeviceSession::SetVolume(f32 volume) const {
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}
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u64 DeviceSession::GetPlayedSampleCount() const {
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if (stream) {
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return stream->GetPlayedSampleCount();
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return played_sample_count;
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}
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std::optional<std::chrono::nanoseconds> DeviceSession::ThreadFunc() {
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// Add 5ms of samples at a 48K sample rate.
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played_sample_count += 48'000 * INCREMENT_TIME / 1s;
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if (type == Sink::StreamType::Out) {
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system.AudioCore().GetAudioManager().SetEvent(Event::Type::AudioOutManager, true);
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} else {
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system.AudioCore().GetAudioManager().SetEvent(Event::Type::AudioInManager, true);
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}
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return 0;
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return std::nullopt;
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}
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void DeviceSession::SetRingSize(u32 ring_size) {
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stream->SetRingSize(ring_size);
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}
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} // namespace AudioCore
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@ -3,6 +3,9 @@
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#pragma once
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#include <chrono>
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#include <memory>
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#include <optional>
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#include <span>
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#include "audio_core/common/common.h"
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@ -11,9 +14,13 @@
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namespace Core {
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class System;
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}
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namespace Timing {
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struct EventType;
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} // namespace Timing
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} // namespace Core
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namespace AudioCore {
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namespace Sink {
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class SinkStream;
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struct SinkBuffer;
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@ -70,7 +77,7 @@ public:
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* @param tag - Unqiue tag of the buffer to check.
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* @return true if the buffer has been consumed, otherwise false.
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*/
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bool IsBufferConsumed(u64 tag) const;
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bool IsBufferConsumed(AudioBuffer& buffer) const;
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/**
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* Start this device session, starting the backend stream.
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@ -96,6 +103,16 @@ public:
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*/
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u64 GetPlayedSampleCount() const;
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/*
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* CoreTiming callback to increment played_sample_count over time.
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*/
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std::optional<std::chrono::nanoseconds> ThreadFunc();
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/*
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* Set the size of the ring buffer.
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*/
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void SetRingSize(u32 ring_size);
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private:
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/// System
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Core::System& system;
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@ -118,9 +135,13 @@ private:
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/// Applet resource user id of this device session
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u64 applet_resource_user_id{};
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/// Total number of samples played by this device session
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u64 played_sample_count{};
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std::atomic<u64> played_sample_count{};
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/// Event increasing the played sample count every 5ms
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std::shared_ptr<Core::Timing::EventType> thread_event;
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/// Is this session initialised?
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bool initialized{};
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/// Buffer queue
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std::vector<AudioBuffer> buffer_queue{};
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};
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} // namespace AudioCore
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@ -93,6 +93,7 @@ Result System::Start() {
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std::vector<AudioBuffer> buffers_to_flush{};
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buffers.RegisterBuffers(buffers_to_flush);
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session->AppendBuffers(buffers_to_flush);
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session->SetRingSize(static_cast<u32>(buffers_to_flush.size()));
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return ResultSuccess;
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}
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@ -112,8 +113,13 @@ bool System::AppendBuffer(const AudioInBuffer& buffer, const u64 tag) {
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return false;
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}
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AudioBuffer new_buffer{
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.played_timestamp = 0, .samples = buffer.samples, .tag = tag, .size = buffer.size};
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const auto timestamp{buffers.GetNextTimestamp()};
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AudioBuffer new_buffer{.start_timestamp = timestamp,
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.end_timestamp = timestamp + buffer.size / (channel_count * sizeof(s16)),
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.played_timestamp = 0,
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.samples = buffer.samples,
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.tag = tag,
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.size = buffer.size};
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buffers.AppendBuffer(new_buffer);
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RegisterBuffers();
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@ -92,6 +92,7 @@ Result System::Start() {
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std::vector<AudioBuffer> buffers_to_flush{};
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buffers.RegisterBuffers(buffers_to_flush);
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session->AppendBuffers(buffers_to_flush);
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session->SetRingSize(static_cast<u32>(buffers_to_flush.size()));
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return ResultSuccess;
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}
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@ -111,8 +112,13 @@ bool System::AppendBuffer(const AudioOutBuffer& buffer, u64 tag) {
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return false;
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}
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AudioBuffer new_buffer{
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.played_timestamp = 0, .samples = buffer.samples, .tag = tag, .size = buffer.size};
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const auto timestamp{buffers.GetNextTimestamp()};
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AudioBuffer new_buffer{.start_timestamp = timestamp,
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.end_timestamp = timestamp + buffer.size / (channel_count * sizeof(s16)),
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.played_timestamp = 0,
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.samples = buffer.samples,
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.tag = tag,
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.size = buffer.size};
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buffers.AppendBuffer(new_buffer);
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RegisterBuffers();
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@ -106,9 +106,6 @@ void AudioRenderer::Start(AudioRenderer_Mailbox* mailbox_) {
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mailbox = mailbox_;
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thread = std::thread(&AudioRenderer::ThreadFunc, this);
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for (auto& stream : streams) {
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stream->Start();
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}
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running = true;
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}
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@ -130,6 +127,7 @@ void AudioRenderer::CreateSinkStreams() {
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std::string name{fmt::format("ADSP_RenderStream-{}", i)};
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streams[i] =
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sink.AcquireSinkStream(system, channels, name, ::AudioCore::Sink::StreamType::Render);
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streams[i]->SetRingSize(4);
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}
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}
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@ -198,11 +196,6 @@ void AudioRenderer::ThreadFunc() {
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command_list_processor.Process(index) - start_time;
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}
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if (index == 0) {
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auto stream{command_list_processor.GetOutputSinkStream()};
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system.AudioCore().SetStreamQueue(stream->GetQueueSize());
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}
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const auto end_time{system.CoreTiming().GetClockTicks()};
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command_buffer.remaining_command_count =
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@ -43,13 +43,15 @@ void BehaviorInfo::AppendError(ErrorInfo& error) {
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}
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void BehaviorInfo::CopyErrorInfo(std::span<ErrorInfo> out_errors, u32& out_count) {
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auto error_count_{std::min(error_count, MaxErrors)};
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std::memset(out_errors.data(), 0, MaxErrors * sizeof(ErrorInfo));
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out_count = std::min(error_count, MaxErrors);
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for (size_t i = 0; i < error_count_; i++) {
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out_errors[i] = errors[i];
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for (size_t i = 0; i < MaxErrors; i++) {
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if (i < out_count) {
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out_errors[i] = errors[i];
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} else {
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out_errors[i] = {};
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}
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}
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out_count = error_count_;
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}
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void BehaviorInfo::UpdateFlags(const Flags flags_) {
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@ -46,6 +46,10 @@ void DeviceSinkCommand::Process(const ADSP::CommandListProcessor& processor) {
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out_buffer.tag = reinterpret_cast<u64>(samples.data());
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stream->AppendBuffer(out_buffer, samples);
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if (stream->IsPaused()) {
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stream->Start();
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}
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}
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bool DeviceSinkCommand::Verify(const ADSP::CommandListProcessor& processor) {
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@ -15,8 +15,7 @@ MICROPROFILE_DEFINE(Audio_RenderSystemManager, "Audio", "Render System Manager",
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MP_RGB(60, 19, 97));
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namespace AudioCore::AudioRenderer {
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constexpr std::chrono::nanoseconds BaseRenderTime{5'000'000UL};
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constexpr std::chrono::nanoseconds RenderTimeOffset{400'000UL};
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constexpr std::chrono::nanoseconds RENDER_TIME{5'000'000UL};
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SystemManager::SystemManager(Core::System& core_)
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: core{core_}, adsp{core.AudioCore().GetADSP()}, mailbox{adsp.GetRenderMailbox()},
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@ -36,8 +35,8 @@ bool SystemManager::InitializeUnsafe() {
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if (adsp.Start()) {
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active = true;
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thread = std::jthread([this](std::stop_token stop_token) { ThreadFunc(); });
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core.CoreTiming().ScheduleLoopingEvent(std::chrono::nanoseconds(0),
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BaseRenderTime - RenderTimeOffset, thread_event);
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core.CoreTiming().ScheduleLoopingEvent(std::chrono::nanoseconds(0), RENDER_TIME,
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thread_event);
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}
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}
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@ -121,35 +120,9 @@ void SystemManager::ThreadFunc() {
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}
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std::optional<std::chrono::nanoseconds> SystemManager::ThreadFunc2(s64 time) {
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std::optional<std::chrono::nanoseconds> new_schedule_time{std::nullopt};
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const auto queue_size{core.AudioCore().GetStreamQueue()};
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switch (state) {
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case StreamState::Filling:
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if (queue_size >= 5) {
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new_schedule_time = BaseRenderTime;
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state = StreamState::Steady;
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}
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break;
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case StreamState::Steady:
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if (queue_size <= 2) {
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new_schedule_time = BaseRenderTime - RenderTimeOffset;
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state = StreamState::Filling;
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} else if (queue_size > 5) {
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new_schedule_time = BaseRenderTime + RenderTimeOffset;
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state = StreamState::Draining;
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}
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break;
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case StreamState::Draining:
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if (queue_size <= 5) {
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new_schedule_time = BaseRenderTime;
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state = StreamState::Steady;
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}
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break;
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}
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update.store(true);
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update.notify_all();
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return new_schedule_time;
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return std::nullopt;
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}
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void SystemManager::PauseCallback(bool paused) {
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@ -1,21 +1,13 @@
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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
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// SPDX-License-Identifier: GPL-2.0-or-later
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#include <algorithm>
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#include <atomic>
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#include <span>
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#include <vector>
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#include "audio_core/audio_core.h"
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#include "audio_core/audio_event.h"
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#include "audio_core/audio_manager.h"
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#include "audio_core/common/common.h"
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#include "audio_core/sink/cubeb_sink.h"
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#include "audio_core/sink/sink_stream.h"
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#include "common/assert.h"
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#include "common/fixed_point.h"
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#include "common/logging/log.h"
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#include "common/reader_writer_queue.h"
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#include "common/ring_buffer.h"
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#include "common/settings.h"
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#include "core/core.h"
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#ifdef _WIN32
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@ -42,10 +34,10 @@ public:
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* @param system_ - Core system.
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* @param event - Event used only for audio renderer, signalled on buffer consume.
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*/
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CubebSinkStream(cubeb* ctx_, const u32 device_channels_, const u32 system_channels_,
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CubebSinkStream(cubeb* ctx_, u32 device_channels_, u32 system_channels_,
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cubeb_devid output_device, cubeb_devid input_device, const std::string& name_,
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const StreamType type_, Core::System& system_)
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: ctx{ctx_}, type{type_}, system{system_} {
|
||||
StreamType type_, Core::System& system_)
|
||||
: SinkStream(system_, type_), ctx{ctx_} {
|
||||
#ifdef _WIN32
|
||||
CoInitializeEx(nullptr, COINIT_MULTITHREADED);
|
||||
#endif
|
||||
|
@ -79,12 +71,10 @@ public:
|
|||
|
||||
minimum_latency = std::max(minimum_latency, 256u);
|
||||
|
||||
playing_buffer.consumed = true;
|
||||
|
||||
LOG_DEBUG(Service_Audio,
|
||||
"Opening cubeb stream {} type {} with: rate {} channels {} (system channels {}) "
|
||||
"latency {}",
|
||||
name, type, params.rate, params.channels, system_channels, minimum_latency);
|
||||
LOG_INFO(Service_Audio,
|
||||
"Opening cubeb stream {} type {} with: rate {} channels {} (system channels {}) "
|
||||
"latency {}",
|
||||
name, type, params.rate, params.channels, system_channels, minimum_latency);
|
||||
|
||||
auto init_error{0};
|
||||
if (type == StreamType::In) {
|
||||
|
@ -111,6 +101,8 @@ public:
|
|||
~CubebSinkStream() override {
|
||||
LOG_DEBUG(Service_Audio, "Destructing cubeb stream {}", name);
|
||||
|
||||
Unstall();
|
||||
|
||||
if (!ctx) {
|
||||
return;
|
||||
}
|
||||
|
@ -136,7 +128,7 @@ public:
|
|||
* @param resume - Set to true if this is resuming the stream a previously-active stream.
|
||||
* Default false.
|
||||
*/
|
||||
void Start(const bool resume = false) override {
|
||||
void Start(bool resume = false) override {
|
||||
if (!ctx) {
|
||||
return;
|
||||
}
|
||||
|
@ -158,6 +150,7 @@ public:
|
|||
* Stop the sink stream.
|
||||
*/
|
||||
void Stop() override {
|
||||
Unstall();
|
||||
if (!ctx) {
|
||||
return;
|
||||
}
|
||||
|
@ -170,194 +163,7 @@ public:
|
|||
paused = true;
|
||||
}
|
||||
|
||||
/**
|
||||
* Append a new buffer and its samples to a waiting queue to play.
|
||||
*
|
||||
* @param buffer - Audio buffer information to be queued.
|
||||
* @param samples - The s16 samples to be queue for playback.
|
||||
*/
|
||||
void AppendBuffer(::AudioCore::Sink::SinkBuffer& buffer, std::vector<s16>& samples) override {
|
||||
if (type == StreamType::In) {
|
||||
queue.enqueue(buffer);
|
||||
queued_buffers++;
|
||||
} else {
|
||||
constexpr s32 min{std::numeric_limits<s16>::min()};
|
||||
constexpr s32 max{std::numeric_limits<s16>::max()};
|
||||
|
||||
auto yuzu_volume{Settings::Volume()};
|
||||
if (yuzu_volume > 1.0f) {
|
||||
yuzu_volume = 0.6f + 20 * std::log10(yuzu_volume);
|
||||
}
|
||||
auto volume{system_volume * device_volume * yuzu_volume};
|
||||
|
||||
if (system_channels == 6 && device_channels == 2) {
|
||||
// We're given 6 channels, but our device only outputs 2, so downmix.
|
||||
constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
|
||||
|
||||
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
|
||||
read_index += system_channels, write_index += device_channels) {
|
||||
const auto left_sample{
|
||||
((Common::FixedPoint<49, 15>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
|
||||
down_mix_coeff[0] +
|
||||
samples[read_index + static_cast<u32>(Channels::Center)] *
|
||||
down_mix_coeff[1] +
|
||||
samples[read_index + static_cast<u32>(Channels::LFE)] *
|
||||
down_mix_coeff[2] +
|
||||
samples[read_index + static_cast<u32>(Channels::BackLeft)] *
|
||||
down_mix_coeff[3]) *
|
||||
volume)
|
||||
.to_int()};
|
||||
|
||||
const auto right_sample{
|
||||
((Common::FixedPoint<49, 15>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
|
||||
down_mix_coeff[0] +
|
||||
samples[read_index + static_cast<u32>(Channels::Center)] *
|
||||
down_mix_coeff[1] +
|
||||
samples[read_index + static_cast<u32>(Channels::LFE)] *
|
||||
down_mix_coeff[2] +
|
||||
samples[read_index + static_cast<u32>(Channels::BackRight)] *
|
||||
down_mix_coeff[3]) *
|
||||
volume)
|
||||
.to_int()};
|
||||
|
||||
samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
|
||||
static_cast<s16>(std::clamp(left_sample, min, max));
|
||||
samples[write_index + static_cast<u32>(Channels::FrontRight)] =
|
||||
static_cast<s16>(std::clamp(right_sample, min, max));
|
||||
}
|
||||
|
||||
samples.resize(samples.size() / system_channels * device_channels);
|
||||
|
||||
} else if (system_channels == 2 && device_channels == 6) {
|
||||
// We need moar samples! Not all games will provide 6 channel audio.
|
||||
// TODO: Implement some upmixing here. Currently just passthrough, with other
|
||||
// channels left as silence.
|
||||
std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
|
||||
|
||||
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
|
||||
read_index += system_channels, write_index += device_channels) {
|
||||
const auto left_sample{static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(
|
||||
static_cast<f32>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
|
||||
volume),
|
||||
min, max))};
|
||||
|
||||
new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
|
||||
|
||||
const auto right_sample{static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(
|
||||
static_cast<f32>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
|
||||
volume),
|
||||
min, max))};
|
||||
|
||||
new_samples[write_index + static_cast<u32>(Channels::FrontRight)] =
|
||||
right_sample;
|
||||
}
|
||||
samples = std::move(new_samples);
|
||||
|
||||
} else if (volume != 1.0f) {
|
||||
for (u32 i = 0; i < samples.size(); i++) {
|
||||
samples[i] = static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
|
||||
}
|
||||
}
|
||||
|
||||
samples_buffer.Push(samples);
|
||||
queue.enqueue(buffer);
|
||||
queued_buffers++;
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
|
||||
*
|
||||
* @param num_samples - Maximum number of samples to receive.
|
||||
* @return Vector of recorded samples. May have fewer than num_samples.
|
||||
*/
|
||||
std::vector<s16> ReleaseBuffer(const u64 num_samples) override {
|
||||
static constexpr s32 min = std::numeric_limits<s16>::min();
|
||||
static constexpr s32 max = std::numeric_limits<s16>::max();
|
||||
|
||||
auto samples{samples_buffer.Pop(num_samples)};
|
||||
|
||||
// TODO: Up-mix to 6 channels if the game expects it.
|
||||
// For audio input this is unlikely to ever be the case though.
|
||||
|
||||
// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
|
||||
// TODO: Play with this and find something that works better.
|
||||
auto volume{system_volume * device_volume * 8};
|
||||
for (u32 i = 0; i < samples.size(); i++) {
|
||||
samples[i] = static_cast<s16>(
|
||||
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
|
||||
}
|
||||
|
||||
if (samples.size() < num_samples) {
|
||||
samples.resize(num_samples, 0);
|
||||
}
|
||||
return samples;
|
||||
}
|
||||
|
||||
/**
|
||||
* Check if a certain buffer has been consumed (fully played).
|
||||
*
|
||||
* @param tag - Unique tag of a buffer to check for.
|
||||
* @return True if the buffer has been played, otherwise false.
|
||||
*/
|
||||
bool IsBufferConsumed(const u64 tag) override {
|
||||
if (released_buffer.tag == 0) {
|
||||
if (!released_buffers.try_dequeue(released_buffer)) {
|
||||
return false;
|
||||
}
|
||||
}
|
||||
|
||||
if (released_buffer.tag == tag) {
|
||||
released_buffer.tag = 0;
|
||||
return true;
|
||||
}
|
||||
return false;
|
||||
}
|
||||
|
||||
/**
|
||||
* Empty out the buffer queue.
|
||||
*/
|
||||
void ClearQueue() override {
|
||||
samples_buffer.Pop();
|
||||
while (queue.pop()) {
|
||||
}
|
||||
while (released_buffers.pop()) {
|
||||
}
|
||||
queued_buffers = 0;
|
||||
released_buffer = {};
|
||||
playing_buffer = {};
|
||||
playing_buffer.consumed = true;
|
||||
}
|
||||
|
||||
private:
|
||||
/**
|
||||
* Signal events back to the audio system that a buffer was played/can be filled.
|
||||
*
|
||||
* @param buffer - Consumed audio buffer to be released.
|
||||
*/
|
||||
void SignalEvent(const ::AudioCore::Sink::SinkBuffer& buffer) {
|
||||
auto& manager{system.AudioCore().GetAudioManager()};
|
||||
switch (type) {
|
||||
case StreamType::Out:
|
||||
released_buffers.enqueue(buffer);
|
||||
manager.SetEvent(Event::Type::AudioOutManager, true);
|
||||
break;
|
||||
case StreamType::In:
|
||||
released_buffers.enqueue(buffer);
|
||||
manager.SetEvent(Event::Type::AudioInManager, true);
|
||||
break;
|
||||
case StreamType::Render:
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Main callback from Cubeb. Either expects samples from us (audio render/audio out), or will
|
||||
* provide samples to be copied (audio in).
|
||||
|
@ -378,106 +184,15 @@ private:
|
|||
|
||||
const std::size_t num_channels = impl->GetDeviceChannels();
|
||||
const std::size_t frame_size = num_channels;
|
||||
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
|
||||
const std::size_t num_frames{static_cast<size_t>(num_frames_)};
|
||||
size_t frames_written{0};
|
||||
[[maybe_unused]] bool underrun{false};
|
||||
|
||||
if (impl->type == StreamType::In) {
|
||||
// INPUT
|
||||
std::span<const s16> input_buffer{reinterpret_cast<const s16*>(in_buff),
|
||||
num_frames * frame_size};
|
||||
|
||||
while (frames_written < num_frames) {
|
||||
auto& playing_buffer{impl->playing_buffer};
|
||||
|
||||
// If the playing buffer has been consumed or has no frames, we need a new one
|
||||
if (playing_buffer.consumed || playing_buffer.frames == 0) {
|
||||
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
|
||||
// If no buffer was available we've underrun, just push the samples and
|
||||
// continue.
|
||||
underrun = true;
|
||||
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
|
||||
(num_frames - frames_written) * frame_size);
|
||||
frames_written = num_frames;
|
||||
continue;
|
||||
} else {
|
||||
// Successfully got a new buffer, mark the old one as consumed and signal.
|
||||
impl->queued_buffers--;
|
||||
impl->SignalEvent(impl->playing_buffer);
|
||||
}
|
||||
}
|
||||
|
||||
// Get the minimum frames available between the currently playing buffer, and the
|
||||
// amount we have left to fill
|
||||
size_t frames_available{
|
||||
std::min(playing_buffer.frames - playing_buffer.frames_played,
|
||||
num_frames - frames_written)};
|
||||
|
||||
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
|
||||
frames_available * frame_size);
|
||||
|
||||
frames_written += frames_available;
|
||||
playing_buffer.frames_played += frames_available;
|
||||
|
||||
// If that's all the frames in the current buffer, add its samples and mark it as
|
||||
// consumed
|
||||
if (playing_buffer.frames_played >= playing_buffer.frames) {
|
||||
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
|
||||
impl->playing_buffer.consumed = true;
|
||||
}
|
||||
}
|
||||
|
||||
std::memcpy(&impl->last_frame[0], &input_buffer[(frames_written - 1) * frame_size],
|
||||
frame_size_bytes);
|
||||
impl->ProcessAudioIn(input_buffer, num_frames);
|
||||
} else {
|
||||
// OUTPUT
|
||||
std::span<s16> output_buffer{reinterpret_cast<s16*>(out_buff), num_frames * frame_size};
|
||||
|
||||
while (frames_written < num_frames) {
|
||||
auto& playing_buffer{impl->playing_buffer};
|
||||
|
||||
// If the playing buffer has been consumed or has no frames, we need a new one
|
||||
if (playing_buffer.consumed || playing_buffer.frames == 0) {
|
||||
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
|
||||
// If no buffer was available we've underrun, fill the remaining buffer with
|
||||
// the last written frame and continue.
|
||||
underrun = true;
|
||||
for (size_t i = frames_written; i < num_frames; i++) {
|
||||
std::memcpy(&output_buffer[i * frame_size], &impl->last_frame[0],
|
||||
frame_size_bytes);
|
||||
}
|
||||
frames_written = num_frames;
|
||||
continue;
|
||||
} else {
|
||||
// Successfully got a new buffer, mark the old one as consumed and signal.
|
||||
impl->queued_buffers--;
|
||||
impl->SignalEvent(impl->playing_buffer);
|
||||
}
|
||||
}
|
||||
|
||||
// Get the minimum frames available between the currently playing buffer, and the
|
||||
// amount we have left to fill
|
||||
size_t frames_available{
|
||||
std::min(playing_buffer.frames - playing_buffer.frames_played,
|
||||
num_frames - frames_written)};
|
||||
|
||||
impl->samples_buffer.Pop(&output_buffer[frames_written * frame_size],
|
||||
frames_available * frame_size);
|
||||
|
||||
frames_written += frames_available;
|
||||
playing_buffer.frames_played += frames_available;
|
||||
|
||||
// If that's all the frames in the current buffer, add its samples and mark it as
|
||||
// consumed
|
||||
if (playing_buffer.frames_played >= playing_buffer.frames) {
|
||||
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
|
||||
impl->playing_buffer.consumed = true;
|
||||
}
|
||||
}
|
||||
|
||||
std::memcpy(&impl->last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
|
||||
frame_size_bytes);
|
||||
impl->ProcessAudioOutAndRender(output_buffer, num_frames);
|
||||
}
|
||||
|
||||
return num_frames_;
|
||||
|
@ -490,32 +205,12 @@ private:
|
|||
* @param user_data - Custom data pointer passed along, points to a CubebSinkStream.
|
||||
* @param state - New state of the device.
|
||||
*/
|
||||
static void StateCallback([[maybe_unused]] cubeb_stream* stream,
|
||||
[[maybe_unused]] void* user_data,
|
||||
[[maybe_unused]] cubeb_state state) {}
|
||||
static void StateCallback(cubeb_stream*, void*, cubeb_state) {}
|
||||
|
||||
/// Main Cubeb context
|
||||
cubeb* ctx{};
|
||||
/// Cubeb stream backend
|
||||
cubeb_stream* stream_backend{};
|
||||
/// Name of this stream
|
||||
std::string name{};
|
||||
/// Type of this stream
|
||||
StreamType type;
|
||||
/// Core system
|
||||
Core::System& system;
|
||||
/// Ring buffer of the samples waiting to be played or consumed
|
||||
Common::RingBuffer<s16, 0x10000> samples_buffer;
|
||||
/// Audio buffers queued and waiting to play
|
||||
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> queue;
|
||||
/// The currently-playing audio buffer
|
||||
::AudioCore::Sink::SinkBuffer playing_buffer{};
|
||||
/// Audio buffers which have been played and are in queue to be released by the audio system
|
||||
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> released_buffers{};
|
||||
/// Currently released buffer waiting to be taken by the audio system
|
||||
::AudioCore::Sink::SinkBuffer released_buffer{};
|
||||
/// The last played (or received) frame of audio, used when the callback underruns
|
||||
std::array<s16, MaxChannels> last_frame{};
|
||||
};
|
||||
|
||||
CubebSink::CubebSink(std::string_view target_device_name) {
|
||||
|
@ -569,15 +264,15 @@ CubebSink::~CubebSink() {
|
|||
#endif
|
||||
}
|
||||
|
||||
SinkStream* CubebSink::AcquireSinkStream(Core::System& system, const u32 system_channels,
|
||||
const std::string& name, const StreamType type) {
|
||||
SinkStream* CubebSink::AcquireSinkStream(Core::System& system, u32 system_channels,
|
||||
const std::string& name, StreamType type) {
|
||||
SinkStreamPtr& stream = sink_streams.emplace_back(std::make_unique<CubebSinkStream>(
|
||||
ctx, device_channels, system_channels, output_device, input_device, name, type, system));
|
||||
|
||||
return stream.get();
|
||||
}
|
||||
|
||||
void CubebSink::CloseStream(const SinkStream* stream) {
|
||||
void CubebSink::CloseStream(SinkStream* stream) {
|
||||
for (size_t i = 0; i < sink_streams.size(); i++) {
|
||||
if (sink_streams[i].get() == stream) {
|
||||
sink_streams[i].reset();
|
||||
|
@ -611,19 +306,19 @@ f32 CubebSink::GetDeviceVolume() const {
|
|||
return sink_streams[0]->GetDeviceVolume();
|
||||
}
|
||||
|
||||
void CubebSink::SetDeviceVolume(const f32 volume) {
|
||||
void CubebSink::SetDeviceVolume(f32 volume) {
|
||||
for (auto& stream : sink_streams) {
|
||||
stream->SetDeviceVolume(volume);
|
||||
}
|
||||
}
|
||||
|
||||
void CubebSink::SetSystemVolume(const f32 volume) {
|
||||
void CubebSink::SetSystemVolume(f32 volume) {
|
||||
for (auto& stream : sink_streams) {
|
||||
stream->SetSystemVolume(volume);
|
||||
}
|
||||
}
|
||||
|
||||
std::vector<std::string> ListCubebSinkDevices(const bool capture) {
|
||||
std::vector<std::string> ListCubebSinkDevices(bool capture) {
|
||||
std::vector<std::string> device_list;
|
||||
cubeb* ctx;
|
||||
|
||||
|
|
|
@ -46,7 +46,7 @@ public:
|
|||
*
|
||||
* @param stream - The stream to close.
|
||||
*/
|
||||
void CloseStream(const SinkStream* stream) override;
|
||||
void CloseStream(SinkStream* stream) override;
|
||||
|
||||
/**
|
||||
* Close all streams.
|
||||
|
|
|
@ -3,10 +3,29 @@
|
|||
|
||||
#pragma once
|
||||
|
||||
#include <string>
|
||||
#include <string_view>
|
||||
#include <vector>
|
||||
|
||||
#include "audio_core/sink/sink.h"
|
||||
#include "audio_core/sink/sink_stream.h"
|
||||
|
||||
namespace Core {
|
||||
class System;
|
||||
} // namespace Core
|
||||
|
||||
namespace AudioCore::Sink {
|
||||
class NullSinkStreamImpl final : public SinkStream {
|
||||
public:
|
||||
explicit NullSinkStreamImpl(Core::System& system_, StreamType type_)
|
||||
: SinkStream{system_, type_} {}
|
||||
~NullSinkStreamImpl() override {}
|
||||
void AppendBuffer(SinkBuffer&, std::vector<s16>&) override {}
|
||||
std::vector<s16> ReleaseBuffer(u64) override {
|
||||
return {};
|
||||
}
|
||||
};
|
||||
|
||||
/**
|
||||
* A no-op sink for when no audio out is wanted.
|
||||
*/
|
||||
|
@ -15,14 +34,15 @@ public:
|
|||
explicit NullSink(std::string_view) {}
|
||||
~NullSink() override = default;
|
||||
|
||||
SinkStream* AcquireSinkStream([[maybe_unused]] Core::System& system,
|
||||
[[maybe_unused]] u32 system_channels,
|
||||
[[maybe_unused]] const std::string& name,
|
||||
[[maybe_unused]] StreamType type) override {
|
||||
return &null_sink_stream;
|
||||
SinkStream* AcquireSinkStream(Core::System& system, u32, const std::string&,
|
||||
StreamType type) override {
|
||||
if (null_sink == nullptr) {
|
||||
null_sink = std::make_unique<NullSinkStreamImpl>(system, type);
|
||||
}
|
||||
return null_sink.get();
|
||||
}
|
||||
|
||||
void CloseStream([[maybe_unused]] const SinkStream* stream) override {}
|
||||
void CloseStream(SinkStream*) override {}
|
||||
void CloseStreams() override {}
|
||||
void PauseStreams() override {}
|
||||
void UnpauseStreams() override {}
|
||||
|
@ -33,20 +53,7 @@ public:
|
|||
void SetSystemVolume(f32 volume) override {}
|
||||
|
||||
private:
|
||||
struct NullSinkStreamImpl final : SinkStream {
|
||||
void Finalize() override {}
|
||||
void Start(bool resume = false) override {}
|
||||
void Stop() override {}
|
||||
void AppendBuffer([[maybe_unused]] ::AudioCore::Sink::SinkBuffer& buffer,
|
||||
[[maybe_unused]] std::vector<s16>& samples) override {}
|
||||
std::vector<s16> ReleaseBuffer([[maybe_unused]] u64 num_samples) override {
|
||||
return {};
|
||||
}
|
||||
bool IsBufferConsumed([[maybe_unused]] const u64 tag) {
|
||||
return true;
|
||||
}
|
||||
void ClearQueue() override {}
|
||||
} null_sink_stream;
|
||||
SinkStreamPtr null_sink{};
|
||||
};
|
||||
|
||||
} // namespace AudioCore::Sink
|
||||
|
|
|
@ -1,20 +1,13 @@
|
|||
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
|
||||
// SPDX-License-Identifier: GPL-2.0-or-later
|
||||
|
||||
#include <algorithm>
|
||||
#include <atomic>
|
||||
#include <span>
|
||||
#include <vector>
|
||||
|
||||
#include "audio_core/audio_core.h"
|
||||
#include "audio_core/audio_event.h"
|
||||
#include "audio_core/audio_manager.h"
|
||||
#include "audio_core/common/common.h"
|
||||
#include "audio_core/sink/sdl2_sink.h"
|
||||
#include "audio_core/sink/sink_stream.h"
|
||||
#include "common/assert.h"
|
||||
#include "common/fixed_point.h"
|
||||
#include "common/logging/log.h"
|
||||
#include "common/reader_writer_queue.h"
|
||||
#include "common/ring_buffer.h"
|
||||
#include "common/settings.h"
|
||||
#include "core/core.h"
|
||||
|
||||
// Ignore -Wimplicit-fallthrough due to https://github.com/libsdl-org/SDL/issues/4307
|
||||
|
@ -44,10 +37,9 @@ public:
|
|||
* @param system_ - Core system.
|
||||
* @param event - Event used only for audio renderer, signalled on buffer consume.
|
||||
*/
|
||||
SDLSinkStream(u32 device_channels_, const u32 system_channels_,
|
||||
const std::string& output_device, const std::string& input_device,
|
||||
const StreamType type_, Core::System& system_)
|
||||
: type{type_}, system{system_} {
|
||||
SDLSinkStream(u32 device_channels_, u32 system_channels_, const std::string& output_device,
|
||||
const std::string& input_device, StreamType type_, Core::System& system_)
|
||||
: SinkStream{system_, type_} {
|
||||
system_channels = system_channels_;
|
||||
device_channels = device_channels_;
|
||||
|
||||
|
@ -63,8 +55,6 @@ public:
|
|||
spec.callback = &SDLSinkStream::DataCallback;
|
||||
spec.userdata = this;
|
||||
|
||||
playing_buffer.consumed = true;
|
||||
|
||||
std::string device_name{output_device};
|
||||
bool capture{false};
|
||||
if (type == StreamType::In) {
|
||||
|
@ -84,31 +74,30 @@ public:
|
|||
return;
|
||||
}
|
||||
|
||||
LOG_DEBUG(Service_Audio,
|
||||
"Opening sdl stream {} with: rate {} channels {} (system channels {}) "
|
||||
" samples {}",
|
||||
device, obtained.freq, obtained.channels, system_channels, obtained.samples);
|
||||
LOG_INFO(Service_Audio,
|
||||
"Opening SDL stream {} with: rate {} channels {} (system channels {}) "
|
||||
" samples {}",
|
||||
device, obtained.freq, obtained.channels, system_channels, obtained.samples);
|
||||
}
|
||||
|
||||
/**
|
||||
* Destroy the sink stream.
|
||||
*/
|
||||
~SDLSinkStream() override {
|
||||
if (device == 0) {
|
||||
return;
|
||||
}
|
||||
|
||||
SDL_CloseAudioDevice(device);
|
||||
LOG_DEBUG(Service_Audio, "Destructing SDL stream {}", name);
|
||||
Finalize();
|
||||
}
|
||||
|
||||
/**
|
||||
* Finalize the sink stream.
|
||||
*/
|
||||
void Finalize() override {
|
||||
Unstall();
|
||||
if (device == 0) {
|
||||
return;
|
||||
}
|
||||
|
||||
Stop();
|
||||
SDL_CloseAudioDevice(device);
|
||||
}
|
||||
|
||||
|
@ -118,7 +107,7 @@ public:
|
|||
* @param resume - Set to true if this is resuming the stream a previously-active stream.
|
||||
* Default false.
|
||||
*/
|
||||
void Start(const bool resume = false) override {
|
||||
void Start(bool resume = false) override {
|
||||
if (device == 0) {
|
||||
return;
|
||||
}
|
||||
|
@ -135,7 +124,8 @@ public:
|
|||
/**
|
||||
* Stop the sink stream.
|
||||
*/
|
||||
void Stop() {
|
||||
void Stop() override {
|
||||
Unstall();
|
||||
if (device == 0) {
|
||||
return;
|
||||
}
|
||||
|
@ -143,191 +133,7 @@ public:
|
|||
paused = true;
|
||||
}
|
||||
|
||||
/**
|
||||
* Append a new buffer and its samples to a waiting queue to play.
|
||||
*
|
||||
* @param buffer - Audio buffer information to be queued.
|
||||
* @param samples - The s16 samples to be queue for playback.
|
||||
*/
|
||||
void AppendBuffer(::AudioCore::Sink::SinkBuffer& buffer, std::vector<s16>& samples) override {
|
||||
if (type == StreamType::In) {
|
||||
queue.enqueue(buffer);
|
||||
queued_buffers++;
|
||||
} else {
|
||||
constexpr s32 min = std::numeric_limits<s16>::min();
|
||||
constexpr s32 max = std::numeric_limits<s16>::max();
|
||||
|
||||
auto yuzu_volume{Settings::Volume()};
|
||||
auto volume{system_volume * device_volume * yuzu_volume};
|
||||
|
||||
if (system_channels == 6 && device_channels == 2) {
|
||||
// We're given 6 channels, but our device only outputs 2, so downmix.
|
||||
constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
|
||||
|
||||
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
|
||||
read_index += system_channels, write_index += device_channels) {
|
||||
const auto left_sample{
|
||||
((Common::FixedPoint<49, 15>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
|
||||
down_mix_coeff[0] +
|
||||
samples[read_index + static_cast<u32>(Channels::Center)] *
|
||||
down_mix_coeff[1] +
|
||||
samples[read_index + static_cast<u32>(Channels::LFE)] *
|
||||
down_mix_coeff[2] +
|
||||
samples[read_index + static_cast<u32>(Channels::BackLeft)] *
|
||||
down_mix_coeff[3]) *
|
||||
volume)
|
||||
.to_int()};
|
||||
|
||||
const auto right_sample{
|
||||
((Common::FixedPoint<49, 15>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
|
||||
down_mix_coeff[0] +
|
||||
samples[read_index + static_cast<u32>(Channels::Center)] *
|
||||
down_mix_coeff[1] +
|
||||
samples[read_index + static_cast<u32>(Channels::LFE)] *
|
||||
down_mix_coeff[2] +
|
||||
samples[read_index + static_cast<u32>(Channels::BackRight)] *
|
||||
down_mix_coeff[3]) *
|
||||
volume)
|
||||
.to_int()};
|
||||
|
||||
samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
|
||||
static_cast<s16>(std::clamp(left_sample, min, max));
|
||||
samples[write_index + static_cast<u32>(Channels::FrontRight)] =
|
||||
static_cast<s16>(std::clamp(right_sample, min, max));
|
||||
}
|
||||
|
||||
samples.resize(samples.size() / system_channels * device_channels);
|
||||
|
||||
} else if (system_channels == 2 && device_channels == 6) {
|
||||
// We need moar samples! Not all games will provide 6 channel audio.
|
||||
// TODO: Implement some upmixing here. Currently just passthrough, with other
|
||||
// channels left as silence.
|
||||
std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
|
||||
|
||||
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
|
||||
read_index += system_channels, write_index += device_channels) {
|
||||
const auto left_sample{static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(
|
||||
static_cast<f32>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
|
||||
volume),
|
||||
min, max))};
|
||||
|
||||
new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
|
||||
|
||||
const auto right_sample{static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(
|
||||
static_cast<f32>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
|
||||
volume),
|
||||
min, max))};
|
||||
|
||||
new_samples[write_index + static_cast<u32>(Channels::FrontRight)] =
|
||||
right_sample;
|
||||
}
|
||||
samples = std::move(new_samples);
|
||||
|
||||
} else if (volume != 1.0f) {
|
||||
for (u32 i = 0; i < samples.size(); i++) {
|
||||
samples[i] = static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
|
||||
}
|
||||
}
|
||||
|
||||
samples_buffer.Push(samples);
|
||||
queue.enqueue(buffer);
|
||||
queued_buffers++;
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
|
||||
*
|
||||
* @param num_samples - Maximum number of samples to receive.
|
||||
* @return Vector of recorded samples. May have fewer than num_samples.
|
||||
*/
|
||||
std::vector<s16> ReleaseBuffer(const u64 num_samples) override {
|
||||
static constexpr s32 min = std::numeric_limits<s16>::min();
|
||||
static constexpr s32 max = std::numeric_limits<s16>::max();
|
||||
|
||||
auto samples{samples_buffer.Pop(num_samples)};
|
||||
|
||||
// TODO: Up-mix to 6 channels if the game expects it.
|
||||
// For audio input this is unlikely to ever be the case though.
|
||||
|
||||
// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
|
||||
// TODO: Play with this and find something that works better.
|
||||
auto volume{system_volume * device_volume * 8};
|
||||
for (u32 i = 0; i < samples.size(); i++) {
|
||||
samples[i] = static_cast<s16>(
|
||||
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
|
||||
}
|
||||
|
||||
if (samples.size() < num_samples) {
|
||||
samples.resize(num_samples, 0);
|
||||
}
|
||||
return samples;
|
||||
}
|
||||
|
||||
/**
|
||||
* Check if a certain buffer has been consumed (fully played).
|
||||
*
|
||||
* @param tag - Unique tag of a buffer to check for.
|
||||
* @return True if the buffer has been played, otherwise false.
|
||||
*/
|
||||
bool IsBufferConsumed(const u64 tag) override {
|
||||
if (released_buffer.tag == 0) {
|
||||
if (!released_buffers.try_dequeue(released_buffer)) {
|
||||
return false;
|
||||
}
|
||||
}
|
||||
|
||||
if (released_buffer.tag == tag) {
|
||||
released_buffer.tag = 0;
|
||||
return true;
|
||||
}
|
||||
return false;
|
||||
}
|
||||
|
||||
/**
|
||||
* Empty out the buffer queue.
|
||||
*/
|
||||
void ClearQueue() override {
|
||||
samples_buffer.Pop();
|
||||
while (queue.pop()) {
|
||||
}
|
||||
while (released_buffers.pop()) {
|
||||
}
|
||||
released_buffer = {};
|
||||
playing_buffer = {};
|
||||
playing_buffer.consumed = true;
|
||||
queued_buffers = 0;
|
||||
}
|
||||
|
||||
private:
|
||||
/**
|
||||
* Signal events back to the audio system that a buffer was played/can be filled.
|
||||
*
|
||||
* @param buffer - Consumed audio buffer to be released.
|
||||
*/
|
||||
void SignalEvent(const ::AudioCore::Sink::SinkBuffer& buffer) {
|
||||
auto& manager{system.AudioCore().GetAudioManager()};
|
||||
switch (type) {
|
||||
case StreamType::Out:
|
||||
released_buffers.enqueue(buffer);
|
||||
manager.SetEvent(Event::Type::AudioOutManager, true);
|
||||
break;
|
||||
case StreamType::In:
|
||||
released_buffers.enqueue(buffer);
|
||||
manager.SetEvent(Event::Type::AudioInManager, true);
|
||||
break;
|
||||
case StreamType::Render:
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Main callback from SDL. Either expects samples from us (audio render/audio out), or will
|
||||
* provide samples to be copied (audio in).
|
||||
|
@ -345,122 +151,20 @@ private:
|
|||
|
||||
const std::size_t num_channels = impl->GetDeviceChannels();
|
||||
const std::size_t frame_size = num_channels;
|
||||
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
|
||||
const std::size_t num_frames{len / num_channels / sizeof(s16)};
|
||||
size_t frames_written{0};
|
||||
[[maybe_unused]] bool underrun{false};
|
||||
|
||||
if (impl->type == StreamType::In) {
|
||||
std::span<s16> input_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size};
|
||||
|
||||
while (frames_written < num_frames) {
|
||||
auto& playing_buffer{impl->playing_buffer};
|
||||
|
||||
// If the playing buffer has been consumed or has no frames, we need a new one
|
||||
if (playing_buffer.consumed || playing_buffer.frames == 0) {
|
||||
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
|
||||
// If no buffer was available we've underrun, just push the samples and
|
||||
// continue.
|
||||
underrun = true;
|
||||
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
|
||||
(num_frames - frames_written) * frame_size);
|
||||
frames_written = num_frames;
|
||||
continue;
|
||||
} else {
|
||||
impl->queued_buffers--;
|
||||
impl->SignalEvent(impl->playing_buffer);
|
||||
}
|
||||
}
|
||||
|
||||
// Get the minimum frames available between the currently playing buffer, and the
|
||||
// amount we have left to fill
|
||||
size_t frames_available{
|
||||
std::min(playing_buffer.frames - playing_buffer.frames_played,
|
||||
num_frames - frames_written)};
|
||||
|
||||
impl->samples_buffer.Push(&input_buffer[frames_written * frame_size],
|
||||
frames_available * frame_size);
|
||||
|
||||
frames_written += frames_available;
|
||||
playing_buffer.frames_played += frames_available;
|
||||
|
||||
// If that's all the frames in the current buffer, add its samples and mark it as
|
||||
// consumed
|
||||
if (playing_buffer.frames_played >= playing_buffer.frames) {
|
||||
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
|
||||
impl->playing_buffer.consumed = true;
|
||||
}
|
||||
}
|
||||
|
||||
std::memcpy(&impl->last_frame[0], &input_buffer[(frames_written - 1) * frame_size],
|
||||
frame_size_bytes);
|
||||
std::span<const s16> input_buffer{reinterpret_cast<const s16*>(stream),
|
||||
num_frames * frame_size};
|
||||
impl->ProcessAudioIn(input_buffer, num_frames);
|
||||
} else {
|
||||
std::span<s16> output_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size};
|
||||
|
||||
while (frames_written < num_frames) {
|
||||
auto& playing_buffer{impl->playing_buffer};
|
||||
|
||||
// If the playing buffer has been consumed or has no frames, we need a new one
|
||||
if (playing_buffer.consumed || playing_buffer.frames == 0) {
|
||||
if (!impl->queue.try_dequeue(impl->playing_buffer)) {
|
||||
// If no buffer was available we've underrun, fill the remaining buffer with
|
||||
// the last written frame and continue.
|
||||
underrun = true;
|
||||
for (size_t i = frames_written; i < num_frames; i++) {
|
||||
std::memcpy(&output_buffer[i * frame_size], &impl->last_frame[0],
|
||||
frame_size_bytes);
|
||||
}
|
||||
frames_written = num_frames;
|
||||
continue;
|
||||
} else {
|
||||
impl->queued_buffers--;
|
||||
impl->SignalEvent(impl->playing_buffer);
|
||||
}
|
||||
}
|
||||
|
||||
// Get the minimum frames available between the currently playing buffer, and the
|
||||
// amount we have left to fill
|
||||
size_t frames_available{
|
||||
std::min(playing_buffer.frames - playing_buffer.frames_played,
|
||||
num_frames - frames_written)};
|
||||
|
||||
impl->samples_buffer.Pop(&output_buffer[frames_written * frame_size],
|
||||
frames_available * frame_size);
|
||||
|
||||
frames_written += frames_available;
|
||||
playing_buffer.frames_played += frames_available;
|
||||
|
||||
// If that's all the frames in the current buffer, add its samples and mark it as
|
||||
// consumed
|
||||
if (playing_buffer.frames_played >= playing_buffer.frames) {
|
||||
impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels);
|
||||
impl->playing_buffer.consumed = true;
|
||||
}
|
||||
}
|
||||
|
||||
std::memcpy(&impl->last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
|
||||
frame_size_bytes);
|
||||
impl->ProcessAudioOutAndRender(output_buffer, num_frames);
|
||||
}
|
||||
}
|
||||
|
||||
/// SDL device id of the opened input/output device
|
||||
SDL_AudioDeviceID device{};
|
||||
/// Type of this stream
|
||||
StreamType type;
|
||||
/// Core system
|
||||
Core::System& system;
|
||||
/// Ring buffer of the samples waiting to be played or consumed
|
||||
Common::RingBuffer<s16, 0x10000> samples_buffer;
|
||||
/// Audio buffers queued and waiting to play
|
||||
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> queue;
|
||||
/// The currently-playing audio buffer
|
||||
::AudioCore::Sink::SinkBuffer playing_buffer{};
|
||||
/// Audio buffers which have been played and are in queue to be released by the audio system
|
||||
Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> released_buffers{};
|
||||
/// Currently released buffer waiting to be taken by the audio system
|
||||
::AudioCore::Sink::SinkBuffer released_buffer{};
|
||||
/// The last played (or received) frame of audio, used when the callback underruns
|
||||
std::array<s16, MaxChannels> last_frame{};
|
||||
};
|
||||
|
||||
SDLSink::SDLSink(std::string_view target_device_name) {
|
||||
|
@ -482,14 +186,14 @@ SDLSink::SDLSink(std::string_view target_device_name) {
|
|||
|
||||
SDLSink::~SDLSink() = default;
|
||||
|
||||
SinkStream* SDLSink::AcquireSinkStream(Core::System& system, const u32 system_channels,
|
||||
const std::string&, const StreamType type) {
|
||||
SinkStream* SDLSink::AcquireSinkStream(Core::System& system, u32 system_channels,
|
||||
const std::string&, StreamType type) {
|
||||
SinkStreamPtr& stream = sink_streams.emplace_back(std::make_unique<SDLSinkStream>(
|
||||
device_channels, system_channels, output_device, input_device, type, system));
|
||||
return stream.get();
|
||||
}
|
||||
|
||||
void SDLSink::CloseStream(const SinkStream* stream) {
|
||||
void SDLSink::CloseStream(SinkStream* stream) {
|
||||
for (size_t i = 0; i < sink_streams.size(); i++) {
|
||||
if (sink_streams[i].get() == stream) {
|
||||
sink_streams[i].reset();
|
||||
|
@ -523,19 +227,19 @@ f32 SDLSink::GetDeviceVolume() const {
|
|||
return sink_streams[0]->GetDeviceVolume();
|
||||
}
|
||||
|
||||
void SDLSink::SetDeviceVolume(const f32 volume) {
|
||||
void SDLSink::SetDeviceVolume(f32 volume) {
|
||||
for (auto& stream : sink_streams) {
|
||||
stream->SetDeviceVolume(volume);
|
||||
}
|
||||
}
|
||||
|
||||
void SDLSink::SetSystemVolume(const f32 volume) {
|
||||
void SDLSink::SetSystemVolume(f32 volume) {
|
||||
for (auto& stream : sink_streams) {
|
||||
stream->SetSystemVolume(volume);
|
||||
}
|
||||
}
|
||||
|
||||
std::vector<std::string> ListSDLSinkDevices(const bool capture) {
|
||||
std::vector<std::string> ListSDLSinkDevices(bool capture) {
|
||||
std::vector<std::string> device_list;
|
||||
|
||||
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
|
||||
|
|
|
@ -44,7 +44,7 @@ public:
|
|||
*
|
||||
* @param stream - The stream to close.
|
||||
*/
|
||||
void CloseStream(const SinkStream* stream) override;
|
||||
void CloseStream(SinkStream* stream) override;
|
||||
|
||||
/**
|
||||
* Close all streams.
|
||||
|
|
|
@ -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.
|
||||
|
|
|
@ -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.
|
||||
|
|
|
@ -0,0 +1,259 @@
|
|||
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
|
||||
// SPDX-License-Identifier: GPL-2.0-or-later
|
||||
|
||||
#pragma once
|
||||
|
||||
#include <array>
|
||||
#include <atomic>
|
||||
#include <memory>
|
||||
#include <span>
|
||||
#include <vector>
|
||||
|
||||
#include "audio_core/common/common.h"
|
||||
#include "audio_core/sink/sink_stream.h"
|
||||
#include "common/common_types.h"
|
||||
#include "common/fixed_point.h"
|
||||
#include "common/settings.h"
|
||||
#include "core/core.h"
|
||||
|
||||
namespace AudioCore::Sink {
|
||||
|
||||
void SinkStream::AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) {
|
||||
if (type == StreamType::In) {
|
||||
queue.enqueue(buffer);
|
||||
queued_buffers++;
|
||||
return;
|
||||
}
|
||||
|
||||
constexpr s32 min{std::numeric_limits<s16>::min()};
|
||||
constexpr s32 max{std::numeric_limits<s16>::max()};
|
||||
|
||||
auto yuzu_volume{Settings::Volume()};
|
||||
if (yuzu_volume > 1.0f) {
|
||||
yuzu_volume = 0.6f + 20 * std::log10(yuzu_volume);
|
||||
}
|
||||
auto volume{system_volume * device_volume * yuzu_volume};
|
||||
|
||||
if (system_channels == 6 && device_channels == 2) {
|
||||
// We're given 6 channels, but our device only outputs 2, so downmix.
|
||||
constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f};
|
||||
|
||||
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
|
||||
read_index += system_channels, write_index += device_channels) {
|
||||
const auto left_sample{
|
||||
((Common::FixedPoint<49, 15>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
|
||||
down_mix_coeff[0] +
|
||||
samples[read_index + static_cast<u32>(Channels::Center)] * down_mix_coeff[1] +
|
||||
samples[read_index + static_cast<u32>(Channels::LFE)] * down_mix_coeff[2] +
|
||||
samples[read_index + static_cast<u32>(Channels::BackLeft)] * down_mix_coeff[3]) *
|
||||
volume)
|
||||
.to_int()};
|
||||
|
||||
const auto right_sample{
|
||||
((Common::FixedPoint<49, 15>(
|
||||
samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
|
||||
down_mix_coeff[0] +
|
||||
samples[read_index + static_cast<u32>(Channels::Center)] * down_mix_coeff[1] +
|
||||
samples[read_index + static_cast<u32>(Channels::LFE)] * down_mix_coeff[2] +
|
||||
samples[read_index + static_cast<u32>(Channels::BackRight)] * down_mix_coeff[3]) *
|
||||
volume)
|
||||
.to_int()};
|
||||
|
||||
samples[write_index + static_cast<u32>(Channels::FrontLeft)] =
|
||||
static_cast<s16>(std::clamp(left_sample, min, max));
|
||||
samples[write_index + static_cast<u32>(Channels::FrontRight)] =
|
||||
static_cast<s16>(std::clamp(right_sample, min, max));
|
||||
}
|
||||
|
||||
samples.resize(samples.size() / system_channels * device_channels);
|
||||
|
||||
} else if (system_channels == 2 && device_channels == 6) {
|
||||
// We need moar samples! Not all games will provide 6 channel audio.
|
||||
// TODO: Implement some upmixing here. Currently just passthrough, with other
|
||||
// channels left as silence.
|
||||
std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0);
|
||||
|
||||
for (u32 read_index = 0, write_index = 0; read_index < samples.size();
|
||||
read_index += system_channels, write_index += device_channels) {
|
||||
const auto left_sample{static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(
|
||||
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontLeft)]) *
|
||||
volume),
|
||||
min, max))};
|
||||
|
||||
new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample;
|
||||
|
||||
const auto right_sample{static_cast<s16>(std::clamp(
|
||||
static_cast<s32>(
|
||||
static_cast<f32>(samples[read_index + static_cast<u32>(Channels::FrontRight)]) *
|
||||
volume),
|
||||
min, max))};
|
||||
|
||||
new_samples[write_index + static_cast<u32>(Channels::FrontRight)] = right_sample;
|
||||
}
|
||||
samples = std::move(new_samples);
|
||||
|
||||
} else if (volume != 1.0f) {
|
||||
for (u32 i = 0; i < samples.size(); i++) {
|
||||
samples[i] = static_cast<s16>(
|
||||
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
|
||||
}
|
||||
}
|
||||
|
||||
samples_buffer.Push(samples);
|
||||
queue.enqueue(buffer);
|
||||
queued_buffers++;
|
||||
}
|
||||
|
||||
std::vector<s16> SinkStream::ReleaseBuffer(u64 num_samples) {
|
||||
constexpr s32 min = std::numeric_limits<s16>::min();
|
||||
constexpr s32 max = std::numeric_limits<s16>::max();
|
||||
|
||||
auto samples{samples_buffer.Pop(num_samples)};
|
||||
|
||||
// TODO: Up-mix to 6 channels if the game expects it.
|
||||
// For audio input this is unlikely to ever be the case though.
|
||||
|
||||
// Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here.
|
||||
// TODO: Play with this and find something that works better.
|
||||
auto volume{system_volume * device_volume * 8};
|
||||
for (u32 i = 0; i < samples.size(); i++) {
|
||||
samples[i] = static_cast<s16>(
|
||||
std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max));
|
||||
}
|
||||
|
||||
if (samples.size() < num_samples) {
|
||||
samples.resize(num_samples, 0);
|
||||
}
|
||||
return samples;
|
||||
}
|
||||
|
||||
void SinkStream::ClearQueue() {
|
||||
samples_buffer.Pop();
|
||||
while (queue.pop()) {
|
||||
}
|
||||
queued_buffers = 0;
|
||||
playing_buffer = {};
|
||||
playing_buffer.consumed = true;
|
||||
}
|
||||
|
||||
void SinkStream::ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames) {
|
||||
const std::size_t num_channels = GetDeviceChannels();
|
||||
const std::size_t frame_size = num_channels;
|
||||
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
|
||||
size_t frames_written{0};
|
||||
|
||||
if (queued_buffers > max_queue_size) {
|
||||
Stall();
|
||||
}
|
||||
|
||||
while (frames_written < num_frames) {
|
||||
// If the playing buffer has been consumed or has no frames, we need a new one
|
||||
if (playing_buffer.consumed || playing_buffer.frames == 0) {
|
||||
if (!queue.try_dequeue(playing_buffer)) {
|
||||
// If no buffer was available we've underrun, just push the samples and
|
||||
// continue.
|
||||
samples_buffer.Push(&input_buffer[frames_written * frame_size],
|
||||
(num_frames - frames_written) * frame_size);
|
||||
frames_written = num_frames;
|
||||
continue;
|
||||
}
|
||||
// Successfully dequeued a new buffer.
|
||||
queued_buffers--;
|
||||
}
|
||||
|
||||
// Get the minimum frames available between the currently playing buffer, and the
|
||||
// amount we have left to fill
|
||||
size_t frames_available{std::min(playing_buffer.frames - playing_buffer.frames_played,
|
||||
num_frames - frames_written)};
|
||||
|
||||
samples_buffer.Push(&input_buffer[frames_written * frame_size],
|
||||
frames_available * frame_size);
|
||||
|
||||
frames_written += frames_available;
|
||||
playing_buffer.frames_played += frames_available;
|
||||
|
||||
// If that's all the frames in the current buffer, add its samples and mark it as
|
||||
// consumed
|
||||
if (playing_buffer.frames_played >= playing_buffer.frames) {
|
||||
playing_buffer.consumed = true;
|
||||
}
|
||||
}
|
||||
|
||||
std::memcpy(&last_frame[0], &input_buffer[(frames_written - 1) * frame_size], frame_size_bytes);
|
||||
|
||||
if (queued_buffers <= max_queue_size) {
|
||||
Unstall();
|
||||
}
|
||||
}
|
||||
|
||||
void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames) {
|
||||
const std::size_t num_channels = GetDeviceChannels();
|
||||
const std::size_t frame_size = num_channels;
|
||||
const std::size_t frame_size_bytes = frame_size * sizeof(s16);
|
||||
size_t frames_written{0};
|
||||
|
||||
if (queued_buffers > max_queue_size) {
|
||||
Stall();
|
||||
}
|
||||
|
||||
while (frames_written < num_frames) {
|
||||
// If the playing buffer has been consumed or has no frames, we need a new one
|
||||
if (playing_buffer.consumed || playing_buffer.frames == 0) {
|
||||
if (!queue.try_dequeue(playing_buffer)) {
|
||||
// If no buffer was available we've underrun, fill the remaining buffer with
|
||||
// the last written frame and continue.
|
||||
for (size_t i = frames_written; i < num_frames; i++) {
|
||||
std::memcpy(&output_buffer[i * frame_size], &last_frame[0], frame_size_bytes);
|
||||
}
|
||||
frames_written = num_frames;
|
||||
continue;
|
||||
}
|
||||
// Successfully dequeued a new buffer.
|
||||
queued_buffers--;
|
||||
}
|
||||
|
||||
// Get the minimum frames available between the currently playing buffer, and the
|
||||
// amount we have left to fill
|
||||
size_t frames_available{std::min(playing_buffer.frames - playing_buffer.frames_played,
|
||||
num_frames - frames_written)};
|
||||
|
||||
samples_buffer.Pop(&output_buffer[frames_written * frame_size],
|
||||
frames_available * frame_size);
|
||||
|
||||
frames_written += frames_available;
|
||||
playing_buffer.frames_played += frames_available;
|
||||
|
||||
// If that's all the frames in the current buffer, add its samples and mark it as
|
||||
// consumed
|
||||
if (playing_buffer.frames_played >= playing_buffer.frames) {
|
||||
playing_buffer.consumed = true;
|
||||
}
|
||||
}
|
||||
|
||||
std::memcpy(&last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
|
||||
frame_size_bytes);
|
||||
|
||||
if (stalled && queued_buffers <= max_queue_size) {
|
||||
Unstall();
|
||||
}
|
||||
}
|
||||
|
||||
void SinkStream::Stall() {
|
||||
if (stalled) {
|
||||
return;
|
||||
}
|
||||
stalled = true;
|
||||
system.StallProcesses();
|
||||
}
|
||||
|
||||
void SinkStream::Unstall() {
|
||||
if (!stalled) {
|
||||
return;
|
||||
}
|
||||
system.UnstallProcesses();
|
||||
stalled = false;
|
||||
}
|
||||
|
||||
} // namespace AudioCore::Sink
|
|
@ -3,12 +3,20 @@
|
|||
|
||||
#pragma once
|
||||
|
||||
#include <array>
|
||||
#include <atomic>
|
||||
#include <memory>
|
||||
#include <span>
|
||||
#include <vector>
|
||||
|
||||
#include "audio_core/common/common.h"
|
||||
#include "common/common_types.h"
|
||||
#include "common/reader_writer_queue.h"
|
||||
#include "common/ring_buffer.h"
|
||||
|
||||
namespace Core {
|
||||
class System;
|
||||
} // namespace Core
|
||||
|
||||
namespace AudioCore::Sink {
|
||||
|
||||
|
@ -34,20 +42,24 @@ struct SinkBuffer {
|
|||
* You should regularly call IsBufferConsumed with the unique SinkBuffer tag to check if the buffer
|
||||
* has been consumed.
|
||||
*
|
||||
* Since these are a FIFO queue, always check IsBufferConsumed in the same order you appended the
|
||||
* buffers, skipping a buffer will result in all following buffers to never release.
|
||||
* Since these are a FIFO queue, IsBufferConsumed must be checked in the same order buffers were
|
||||
* appended, skipping a buffer will result in the queue getting stuck, and all following buffers to
|
||||
* never release.
|
||||
*
|
||||
* If the buffers appear to be stuck, you can stop and re-open an IAudioIn/IAudioOut service (this
|
||||
* is what games do), or call ClearQueue to flush all of the buffers without a full restart.
|
||||
*/
|
||||
class SinkStream {
|
||||
public:
|
||||
virtual ~SinkStream() = default;
|
||||
explicit SinkStream(Core::System& system_, StreamType type_) : system{system_}, type{type_} {}
|
||||
virtual ~SinkStream() {
|
||||
Unstall();
|
||||
}
|
||||
|
||||
/**
|
||||
* Finalize the sink stream.
|
||||
*/
|
||||
virtual void Finalize() = 0;
|
||||
virtual void Finalize() {}
|
||||
|
||||
/**
|
||||
* Start the sink stream.
|
||||
|
@ -55,48 +67,19 @@ public:
|
|||
* @param resume - Set to true if this is resuming the stream a previously-active stream.
|
||||
* Default false.
|
||||
*/
|
||||
virtual void Start(bool resume = false) = 0;
|
||||
virtual void Start(bool resume = false) {}
|
||||
|
||||
/**
|
||||
* Stop the sink stream.
|
||||
*/
|
||||
virtual void Stop() = 0;
|
||||
|
||||
/**
|
||||
* Append a new buffer and its samples to a waiting queue to play.
|
||||
*
|
||||
* @param buffer - Audio buffer information to be queued.
|
||||
* @param samples - The s16 samples to be queue for playback.
|
||||
*/
|
||||
virtual void AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) = 0;
|
||||
|
||||
/**
|
||||
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
|
||||
*
|
||||
* @param num_samples - Maximum number of samples to receive.
|
||||
* @return Vector of recorded samples. May have fewer than num_samples.
|
||||
*/
|
||||
virtual std::vector<s16> ReleaseBuffer(u64 num_samples) = 0;
|
||||
|
||||
/**
|
||||
* Check if a certain buffer has been consumed (fully played).
|
||||
*
|
||||
* @param tag - Unique tag of a buffer to check for.
|
||||
* @return True if the buffer has been played, otherwise false.
|
||||
*/
|
||||
virtual bool IsBufferConsumed(u64 tag) = 0;
|
||||
|
||||
/**
|
||||
* Empty out the buffer queue.
|
||||
*/
|
||||
virtual void ClearQueue() = 0;
|
||||
virtual void Stop() {}
|
||||
|
||||
/**
|
||||
* Check if the stream is paused.
|
||||
*
|
||||
* @return True if paused, otherwise false.
|
||||
*/
|
||||
bool IsPaused() {
|
||||
bool IsPaused() const {
|
||||
return paused;
|
||||
}
|
||||
|
||||
|
@ -127,34 +110,6 @@ public:
|
|||
return device_channels;
|
||||
}
|
||||
|
||||
/**
|
||||
* Get the total number of samples played by this stream.
|
||||
*
|
||||
* @return Number of samples played.
|
||||
*/
|
||||
u64 GetPlayedSampleCount() const {
|
||||
return played_sample_count;
|
||||
}
|
||||
|
||||
/**
|
||||
* Set the number of samples played.
|
||||
* This is started and stopped on system start/stop.
|
||||
*
|
||||
* @param played_sample_count_ - Number of samples to set.
|
||||
*/
|
||||
void SetPlayedSampleCount(u64 played_sample_count_) {
|
||||
played_sample_count = played_sample_count_;
|
||||
}
|
||||
|
||||
/**
|
||||
* Add to the played sample count.
|
||||
*
|
||||
* @param num_samples - Number of samples to add.
|
||||
*/
|
||||
void AddPlayedSampleCount(u64 num_samples) {
|
||||
played_sample_count += num_samples;
|
||||
}
|
||||
|
||||
/**
|
||||
* Get the system volume.
|
||||
*
|
||||
|
@ -200,15 +155,65 @@ public:
|
|||
return queued_buffers.load();
|
||||
}
|
||||
|
||||
/**
|
||||
* Set the maximum buffer queue size.
|
||||
*/
|
||||
void SetRingSize(u32 ring_size) {
|
||||
max_queue_size = ring_size;
|
||||
}
|
||||
|
||||
/**
|
||||
* Append a new buffer and its samples to a waiting queue to play.
|
||||
*
|
||||
* @param buffer - Audio buffer information to be queued.
|
||||
* @param samples - The s16 samples to be queue for playback.
|
||||
*/
|
||||
virtual void AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples);
|
||||
|
||||
/**
|
||||
* Release a buffer. Audio In only, will fill a buffer with recorded samples.
|
||||
*
|
||||
* @param num_samples - Maximum number of samples to receive.
|
||||
* @return Vector of recorded samples. May have fewer than num_samples.
|
||||
*/
|
||||
virtual std::vector<s16> ReleaseBuffer(u64 num_samples);
|
||||
|
||||
/**
|
||||
* Empty out the buffer queue.
|
||||
*/
|
||||
void ClearQueue();
|
||||
|
||||
/**
|
||||
* Callback for AudioIn.
|
||||
*
|
||||
* @param input_buffer - Input buffer to be filled with samples.
|
||||
* @param num_frames - Number of frames to be filled.
|
||||
*/
|
||||
void ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames);
|
||||
|
||||
/**
|
||||
* Callback for AudioOut and AudioRenderer.
|
||||
*
|
||||
* @param output_buffer - Output buffer to be filled with samples.
|
||||
* @param num_frames - Number of frames to be filled.
|
||||
*/
|
||||
void ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames);
|
||||
|
||||
/**
|
||||
* Stall core processes if the audio thread falls too far behind.
|
||||
*/
|
||||
void Stall();
|
||||
|
||||
/**
|
||||
* Unstall core processes.
|
||||
*/
|
||||
void Unstall();
|
||||
|
||||
protected:
|
||||
/// Number of buffers waiting to be played
|
||||
std::atomic<u32> queued_buffers{};
|
||||
/// Total samples played by this stream
|
||||
std::atomic<u64> played_sample_count{};
|
||||
/// Set by the audio render/in/out system which uses this stream
|
||||
f32 system_volume{1.0f};
|
||||
/// Set via IAudioDevice service calls
|
||||
f32 device_volume{1.0f};
|
||||
/// Core system
|
||||
Core::System& system;
|
||||
/// Type of this stream
|
||||
StreamType type;
|
||||
/// Set by the audio render/in/out systen which uses this stream
|
||||
u32 system_channels{2};
|
||||
/// Channels supported by hardware
|
||||
|
@ -217,6 +222,28 @@ protected:
|
|||
std::atomic<bool> paused{true};
|
||||
/// Was this stream previously playing?
|
||||
std::atomic<bool> was_playing{false};
|
||||
/// Name of this stream
|
||||
std::string name{};
|
||||
|
||||
private:
|
||||
/// Ring buffer of the samples waiting to be played or consumed
|
||||
Common::RingBuffer<s16, 0x10000> samples_buffer;
|
||||
/// Audio buffers queued and waiting to play
|
||||
Common::ReaderWriterQueue<SinkBuffer> queue;
|
||||
/// The currently-playing audio buffer
|
||||
SinkBuffer playing_buffer{};
|
||||
/// The last played (or received) frame of audio, used when the callback underruns
|
||||
std::array<s16, MaxChannels> last_frame{};
|
||||
/// Number of buffers waiting to be played
|
||||
std::atomic<u32> queued_buffers{};
|
||||
/// The ring size for audio out buffers (usually 4, rarely 2 or 8)
|
||||
u32 max_queue_size{};
|
||||
/// Set by the audio render/in/out system which uses this stream
|
||||
f32 system_volume{1.0f};
|
||||
/// Set via IAudioDevice service calls
|
||||
f32 device_volume{1.0f};
|
||||
/// True if coretiming has been stalled
|
||||
bool stalled{false};
|
||||
};
|
||||
|
||||
using SinkStreamPtr = std::unique_ptr<SinkStream>;
|
||||
|
|
|
@ -117,6 +117,7 @@ union Result {
|
|||
BitField<0, 9, ErrorModule> module;
|
||||
BitField<9, 13, u32> description;
|
||||
|
||||
Result() = default;
|
||||
constexpr explicit Result(u32 raw_) : raw(raw_) {}
|
||||
|
||||
constexpr Result(ErrorModule module_, u32 description_)
|
||||
|
@ -130,6 +131,7 @@ union Result {
|
|||
return !IsSuccess();
|
||||
}
|
||||
};
|
||||
static_assert(std::is_trivial_v<Result>);
|
||||
|
||||
[[nodiscard]] constexpr bool operator==(const Result& a, const Result& b) {
|
||||
return a.raw == b.raw;
|
||||
|
|
Reference in New Issue