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https://github.com/starr-dusT/yuzu-mainline
synced 2024-03-05 21:12:25 -08:00
audio: Interpolate system manager sample count using host sink sample info
This avoids the need to stall if the host sink sporadically misses the deadline, in such a case the previous implementation would report them samples as being played on-time, causing the guest to send more samples and leading to a gradual buildup.
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@ -121,8 +121,7 @@ u64 DeviceSession::GetPlayedSampleCount() const {
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}
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}
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std::optional<std::chrono::nanoseconds> DeviceSession::ThreadFunc() {
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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 = stream->GetExpectedPlayedSampleCount();
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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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if (type == Sink::StreamType::Out) {
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system.AudioCore().GetAudioManager().SetEvent(Event::Type::AudioOutManager, true);
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system.AudioCore().GetAudioManager().SetEvent(Event::Type::AudioOutManager, true);
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} else {
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} else {
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@ -15,7 +15,6 @@ MICROPROFILE_DEFINE(Audio_RenderSystemManager, "Audio", "Render System Manager",
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MP_RGB(60, 19, 97));
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MP_RGB(60, 19, 97));
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namespace AudioCore::AudioRenderer {
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namespace AudioCore::AudioRenderer {
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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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SystemManager::SystemManager(Core::System& core_)
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: core{core_}, adsp{core.AudioCore().GetADSP()}, mailbox{adsp.GetRenderMailbox()},
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: core{core_}, adsp{core.AudioCore().GetADSP()}, mailbox{adsp.GetRenderMailbox()},
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@ -14,6 +14,8 @@
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#include "common/fixed_point.h"
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#include "common/fixed_point.h"
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#include "common/settings.h"
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#include "common/settings.h"
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#include "core/core.h"
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#include "core/core.h"
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#include "core/core_timing.h"
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#include "core/core_timing_util.h"
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namespace AudioCore::Sink {
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namespace AudioCore::Sink {
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@ -198,6 +200,7 @@ void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::siz
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const std::size_t frame_size = num_channels;
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const std::size_t frame_size = num_channels;
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const std::size_t frame_size_bytes = frame_size * sizeof(s16);
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const std::size_t frame_size_bytes = frame_size * sizeof(s16);
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size_t frames_written{0};
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size_t frames_written{0};
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size_t actual_frames_written{0};
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// If we're paused or going to shut down, we don't want to consume buffers as coretiming is
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// If we're paused or going to shut down, we don't want to consume buffers as coretiming is
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// paused and we'll desync, so just play silence.
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// paused and we'll desync, so just play silence.
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@ -248,6 +251,7 @@ void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::siz
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frames_available * frame_size);
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frames_available * frame_size);
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frames_written += frames_available;
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frames_written += frames_available;
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actual_frames_written += frames_available;
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playing_buffer.frames_played += frames_available;
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playing_buffer.frames_played += frames_available;
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// If that's all the frames in the current buffer, add its samples and mark it as
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// If that's all the frames in the current buffer, add its samples and mark it as
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@ -260,6 +264,13 @@ void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::siz
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std::memcpy(&last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
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std::memcpy(&last_frame[0], &output_buffer[(frames_written - 1) * frame_size],
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frame_size_bytes);
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frame_size_bytes);
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{
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std::scoped_lock lk{sample_count_lock};
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last_sample_count_update_time = Core::Timing::CyclesToUs(system.CoreTiming().GetClockTicks());
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min_played_sample_count = max_played_sample_count;
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max_played_sample_count += actual_frames_written;
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}
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if (system.IsMulticore() && queued_buffers <= max_queue_size) {
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if (system.IsMulticore() && queued_buffers <= max_queue_size) {
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Unstall();
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Unstall();
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}
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}
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@ -282,4 +293,14 @@ void SinkStream::Unstall() {
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stalled_lock.unlock();
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stalled_lock.unlock();
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}
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}
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u64 SinkStream::GetExpectedPlayedSampleCount() {
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std::scoped_lock lk{sample_count_lock};
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auto cur_time{Core::Timing::CyclesToUs(system.CoreTiming().GetClockTicks())};
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auto time_delta{cur_time - last_sample_count_update_time};
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auto exp_played_sample_count{min_played_sample_count +
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(TargetSampleRate * time_delta) / std::chrono::seconds{1}};
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return std::min<u64>(exp_played_sample_count, max_played_sample_count);
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}
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} // namespace AudioCore::Sink
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} // namespace AudioCore::Sink
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@ -5,6 +5,7 @@
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#include <array>
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#include <array>
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#include <atomic>
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#include <atomic>
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#include <chrono>
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#include <memory>
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#include <memory>
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#include <mutex>
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#include <mutex>
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#include <span>
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#include <span>
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@ -14,6 +15,7 @@
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#include "common/common_types.h"
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#include "common/common_types.h"
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#include "common/reader_writer_queue.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/ring_buffer.h"
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#include "common/thread.h"
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namespace Core {
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namespace Core {
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class System;
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class System;
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@ -210,6 +212,13 @@ public:
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*/
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*/
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void Unstall();
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void Unstall();
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/**
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* Get the total number of samples expected to have been played by this stream.
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*
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* @return The number of samples.
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*/
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u64 GetExpectedPlayedSampleCount();
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protected:
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protected:
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/// Core system
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/// Core system
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Core::System& system;
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Core::System& system;
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@ -237,6 +246,14 @@ private:
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std::atomic<u32> queued_buffers{};
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std::atomic<u32> queued_buffers{};
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/// The ring size for audio out buffers (usually 4, rarely 2 or 8)
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/// The ring size for audio out buffers (usually 4, rarely 2 or 8)
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u32 max_queue_size{};
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u32 max_queue_size{};
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/// Locks access to sample count tracking info
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std::mutex sample_count_lock;
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/// Minimum number of total samples that have been played since the last callback
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u64 min_played_sample_count{};
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/// Maximum number of total samples that can be played since the last callback
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u64 max_played_sample_count{};
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/// The time the two above tracking variables were last written to
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std::chrono::microseconds last_sample_count_update_time{};
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/// Set by the audio render/in/out system which uses this stream
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/// Set by the audio render/in/out system which uses this stream
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f32 system_volume{1.0f};
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f32 system_volume{1.0f};
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/// Set via IAudioDevice service calls
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/// Set via IAudioDevice service calls
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