// SPDX-FileCopyrightText: Copyright 2025 Eden Emulator Project // SPDX-License-Identifier: GPL-3.0-or-later // SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project // SPDX-License-Identifier: GPL-2.0-or-later #include #include #include #include #include #include "audio_core/audio_core.h" #include "audio_core/common/common.h" #include "audio_core/sink/sink_stream.h" #include "common/common_types.h" #include "common/fixed_point.h" #include "common/scope_exit.h" #include "common/settings.h" #include "core/core.h" #include "core/core_timing.h" namespace AudioCore::Sink { void SinkStream::AppendBuffer(SinkBuffer& buffer, std::span samples) { if (type == StreamType::In) return; constexpr s32 min = (std::numeric_limits::min)(); constexpr s32 max = (std::numeric_limits::max)(); auto yuzu_volume = Settings::Volume(); if (yuzu_volume > 1.0f) yuzu_volume = 0.6f + 20.0f * std::log10(yuzu_volume); yuzu_volume = std::max(yuzu_volume, 0.001f); auto const volume = system_volume * device_volume * yuzu_volume; if (system_channels > device_channels) { static constexpr std::array tcoeff{1.0f, 0.596f, 0.354f, 0.707f}; for (u32 r_offs = 0, w_offs = 0; r_offs < samples.size(); r_offs += system_channels, w_offs += device_channels) { std::array ccoeff{0.f}; for (u32 i = 0; i < system_channels; ++i) ccoeff[i] = f32(samples[r_offs + i]); std::array rcoeff{ ccoeff[u32(Channels::FrontLeft)], ccoeff[u32(Channels::BackLeft)], ccoeff[u32(Channels::Center)], ccoeff[u32(Channels::LFE)], ccoeff[u32(Channels::BackRight)], ccoeff[u32(Channels::FrontRight)], }; const f32 left = rcoeff[0] * tcoeff[0] + rcoeff[2] * tcoeff[1] + rcoeff[3] * tcoeff[2] + rcoeff[1] * tcoeff[3]; const f32 right = rcoeff[5] * tcoeff[0] + rcoeff[2] * tcoeff[1] + rcoeff[3] * tcoeff[2] + rcoeff[4] * tcoeff[3]; samples[w_offs + 0] = s16(std::clamp(s32(left * volume), min, max)); samples[w_offs + 1] = s16(std::clamp(s32(right * volume), min, max)); } queue.EmplaceWait(buffer); samples_buffer.Push(samples.subspan(0, samples.size() / system_channels * device_channels)); } else if (system_channels < device_channels) { std::vector new_samples(samples.size() / system_channels * device_channels); for (u32 r_offs = 0, w_offs = 0; r_offs < samples.size(); r_offs += system_channels, w_offs += device_channels) for (u32 channel = 0; channel < system_channels; ++channel) new_samples[w_offs + channel] = s16(std::clamp(s32(f32(samples[r_offs + channel]) * volume), min, max)); queue.EmplaceWait(buffer); samples_buffer.Push(new_samples); } else { if (volume != 1.0f) { for (u32 i = 0; i < samples.size(); ++i) samples[i] = s16(std::clamp(s32(f32(samples[i]) * volume), min, max)); } queue.EmplaceWait(buffer); samples_buffer.Push(samples); } ++queued_buffers; } std::vector SinkStream::ReleaseBuffer(u64 num_samples) { 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. constexpr s32 min = (std::numeric_limits::min)(); constexpr s32 max = (std::numeric_limits::max)(); auto volume{system_volume * device_volume * 8}; for (u32 i = 0; i < samples.size(); i++) samples[i] = s16(std::clamp(s32(f32(samples[i]) * volume), min, max)); if (samples.size() < num_samples) samples.resize(num_samples, 0); return samples; } void SinkStream::ClearQueue() { std::scoped_lock lk{release_mutex}; samples_buffer.Pop(); SinkBuffer tmp; while (queue.TryPop(tmp)); queued_buffers = 0; playing_buffer = {}; playing_buffer.consumed = true; } void SinkStream::ProcessAudioIn(std::span input_buffer, std::size_t num_frames) { const std::size_t num_channels = GetDeviceChannels(); const std::size_t frame_size = num_channels; const std::size_t frame_size_bytes = frame_size * sizeof(s16); size_t frames_written{0}; // If we're paused or going to shut down, we don't want to consume buffers as coretiming is // paused and we'll desync, so just return. if (system.IsPaused() || system.IsShuttingDown()) return; 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.TryPop(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); } void SinkStream::ProcessAudioOutAndRender(std::span 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}; size_t actual_frames_written{0}; if (system.IsPaused() || system.IsShuttingDown()) { if (system.IsShuttingDown()) { std::scoped_lock lk{release_mutex}; queued_buffers.store(0); release_cv.notify_one(); } static constexpr std::array silence{}; for (size_t i = 0; i < num_frames; i++) std::memcpy(&output_buffer[i * frame_size], silence.data(), frame_size_bytes); return; } while (frames_written < num_frames) { if (playing_buffer.consumed || playing_buffer.frames == 0) { std::unique_lock lk{release_mutex}; if (!queue.TryPop(playing_buffer)) { lk.unlock(); for (size_t i = frames_written; i < num_frames; i++) std::memcpy(&output_buffer[i * frame_size], last_frame.data(), frame_size_bytes); frames_written = num_frames; continue; } --queued_buffers; lk.unlock(); release_cv.notify_one(); } const 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; actual_frames_written += frames_available; playing_buffer.frames_played += frames_available; if (playing_buffer.frames_played >= playing_buffer.frames) playing_buffer.consumed = true; } std::memcpy(last_frame.data(), &output_buffer[(frames_written - 1) * frame_size], frame_size_bytes); { std::scoped_lock lk{sample_count_lock}; last_sample_count_update_time = system.CoreTiming().GetGlobalTimeNs(); min_played_sample_count = max_played_sample_count; max_played_sample_count += actual_frames_written; } } u64 SinkStream::GetExpectedPlayedSampleCount() { std::scoped_lock lk{sample_count_lock}; auto cur_time{system.CoreTiming().GetGlobalTimeNs()}; auto time_delta{cur_time - last_sample_count_update_time}; auto exp_played_sample_count{min_played_sample_count + (TargetSampleRate * time_delta) / std::chrono::seconds{1}}; // Add 15ms of latency in sample reporting to allow for some leeway in scheduler timings return std::min(exp_played_sample_count, max_played_sample_count) + TargetSampleCount * 3; } void SinkStream::WaitFreeSpace(std::stop_token stop_token) { std::unique_lock lk{release_mutex}; auto can_continue = [this]() { return paused || queued_buffers < max_queue_size; }; release_cv.wait_for(lk, std::chrono::milliseconds(10), can_continue); if (queued_buffers > max_queue_size + 10) { release_cv.wait(lk, stop_token, can_continue); } } void SinkStream::SignalPause() { { std::scoped_lock lk{release_mutex}; paused = true; } release_cv.notify_one(); } } // namespace AudioCore::Sink