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-rw-r--r--src/audio_core/CMakeLists.txt31
-rw-r--r--src/audio_core/algorithm/filter.cpp79
-rw-r--r--src/audio_core/algorithm/filter.h62
-rw-r--r--src/audio_core/algorithm/interpolate.cpp71
-rw-r--r--src/audio_core/algorithm/interpolate.h43
-rw-r--r--src/audio_core/audio_out.cpp58
-rw-r--r--src/audio_core/audio_out.h43
-rw-r--r--src/audio_core/audio_renderer.cpp249
-rw-r--r--src/audio_core/audio_renderer.h211
-rw-r--r--src/audio_core/buffer.h45
-rw-r--r--src/audio_core/codec.cpp77
-rw-r--r--src/audio_core/codec.h44
-rw-r--r--src/audio_core/cubeb_sink.cpp206
-rw-r--r--src/audio_core/cubeb_sink.h32
-rw-r--r--src/audio_core/null_sink.h27
-rw-r--r--src/audio_core/sink.h31
-rw-r--r--src/audio_core/sink_details.cpp44
-rw-r--r--src/audio_core/sink_details.h35
-rw-r--r--src/audio_core/sink_stream.h32
-rw-r--r--src/audio_core/stream.cpp127
-rw-r--r--src/audio_core/stream.h102
21 files changed, 1649 insertions, 0 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
new file mode 100644
index 000000000..82e4850f7
--- /dev/null
+++ b/src/audio_core/CMakeLists.txt
@@ -0,0 +1,31 @@
+add_library(audio_core STATIC
+ algorithm/filter.cpp
+ algorithm/filter.h
+ algorithm/interpolate.cpp
+ algorithm/interpolate.h
+ audio_out.cpp
+ audio_out.h
+ audio_renderer.cpp
+ audio_renderer.h
+ buffer.h
+ codec.cpp
+ codec.h
+ null_sink.h
+ sink.h
+ sink_details.cpp
+ sink_details.h
+ sink_stream.h
+ stream.cpp
+ stream.h
+
+ $<$<BOOL:${ENABLE_CUBEB}>:cubeb_sink.cpp cubeb_sink.h>
+)
+
+create_target_directory_groups(audio_core)
+
+target_link_libraries(audio_core PUBLIC common core)
+
+if(ENABLE_CUBEB)
+ target_link_libraries(audio_core PRIVATE cubeb)
+ target_compile_definitions(audio_core PRIVATE -DHAVE_CUBEB=1)
+endif()
diff --git a/src/audio_core/algorithm/filter.cpp b/src/audio_core/algorithm/filter.cpp
new file mode 100644
index 000000000..403b8503f
--- /dev/null
+++ b/src/audio_core/algorithm/filter.cpp
@@ -0,0 +1,79 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#define _USE_MATH_DEFINES
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <vector>
+#include "audio_core/algorithm/filter.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+Filter Filter::LowPass(double cutoff, double Q) {
+ const double w0 = 2.0 * M_PI * cutoff;
+ const double sin_w0 = std::sin(w0);
+ const double cos_w0 = std::cos(w0);
+ const double alpha = sin_w0 / (2 * Q);
+
+ const double a0 = 1 + alpha;
+ const double a1 = -2.0 * cos_w0;
+ const double a2 = 1 - alpha;
+ const double b0 = 0.5 * (1 - cos_w0);
+ const double b1 = 1.0 * (1 - cos_w0);
+ const double b2 = 0.5 * (1 - cos_w0);
+
+ return {a0, a1, a2, b0, b1, b2};
+}
+
+Filter::Filter() : Filter(1.0, 0.0, 0.0, 1.0, 0.0, 0.0) {}
+
+Filter::Filter(double a0, double a1, double a2, double b0, double b1, double b2)
+ : a1(a1 / a0), a2(a2 / a0), b0(b0 / a0), b1(b1 / a0), b2(b2 / a0) {}
+
+void Filter::Process(std::vector<s16>& signal) {
+ const size_t num_frames = signal.size() / 2;
+ for (size_t i = 0; i < num_frames; i++) {
+ std::rotate(in.begin(), in.end() - 1, in.end());
+ std::rotate(out.begin(), out.end() - 1, out.end());
+
+ for (size_t ch = 0; ch < channel_count; ch++) {
+ in[0][ch] = signal[i * channel_count + ch];
+
+ out[0][ch] = b0 * in[0][ch] + b1 * in[1][ch] + b2 * in[2][ch] - a1 * out[1][ch] -
+ a2 * out[2][ch];
+
+ signal[i * 2 + ch] = std::clamp(out[0][ch], -32768.0, 32767.0);
+ }
+ }
+}
+
+/// Calculates the appropriate Q for each biquad in a cascading filter.
+/// @param total_count The total number of biquads to be cascaded.
+/// @param index 0-index of the biquad to calculate the Q value for.
+static double CascadingBiquadQ(size_t total_count, size_t index) {
+ const double pole = M_PI * (2 * index + 1) / (4.0 * total_count);
+ return 1.0 / (2.0 * std::cos(pole));
+}
+
+CascadingFilter CascadingFilter::LowPass(double cutoff, size_t cascade_size) {
+ std::vector<Filter> cascade(cascade_size);
+ for (size_t i = 0; i < cascade_size; i++) {
+ cascade[i] = Filter::LowPass(cutoff, CascadingBiquadQ(cascade_size, i));
+ }
+ return CascadingFilter{std::move(cascade)};
+}
+
+CascadingFilter::CascadingFilter() = default;
+CascadingFilter::CascadingFilter(std::vector<Filter> filters) : filters(std::move(filters)) {}
+
+void CascadingFilter::Process(std::vector<s16>& signal) {
+ for (auto& filter : filters) {
+ filter.Process(signal);
+ }
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/algorithm/filter.h b/src/audio_core/algorithm/filter.h
new file mode 100644
index 000000000..a41beef98
--- /dev/null
+++ b/src/audio_core/algorithm/filter.h
@@ -0,0 +1,62 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <vector>
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/// Digital biquad filter:
+///
+/// b0 + b1 z^-1 + b2 z^-2
+/// H(z) = ------------------------
+/// a0 + a1 z^-1 + b2 z^-2
+class Filter {
+public:
+ /// Creates a low-pass filter.
+ /// @param cutoff Determines the cutoff frequency. A value from 0.0 to 1.0.
+ /// @param Q Determines the quality factor of this filter.
+ static Filter LowPass(double cutoff, double Q = 0.7071);
+
+ /// Passthrough filter.
+ Filter();
+
+ Filter(double a0, double a1, double a2, double b0, double b1, double b2);
+
+ void Process(std::vector<s16>& signal);
+
+private:
+ static constexpr size_t channel_count = 2;
+
+ /// Coefficients are in normalized form (a0 = 1.0).
+ double a1, a2, b0, b1, b2;
+ /// Input History
+ std::array<std::array<double, channel_count>, 3> in;
+ /// Output History
+ std::array<std::array<double, channel_count>, 3> out;
+};
+
+/// Cascade filters to build up higher-order filters from lower-order ones.
+class CascadingFilter {
+public:
+ /// Creates a cascading low-pass filter.
+ /// @param cutoff Determines the cutoff frequency. A value from 0.0 to 1.0.
+ /// @param cascade_size Number of biquads in cascade.
+ static CascadingFilter LowPass(double cutoff, size_t cascade_size);
+
+ /// Passthrough.
+ CascadingFilter();
+
+ explicit CascadingFilter(std::vector<Filter> filters);
+
+ void Process(std::vector<s16>& signal);
+
+private:
+ std::vector<Filter> filters;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/algorithm/interpolate.cpp b/src/audio_core/algorithm/interpolate.cpp
new file mode 100644
index 000000000..11459821f
--- /dev/null
+++ b/src/audio_core/algorithm/interpolate.cpp
@@ -0,0 +1,71 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#define _USE_MATH_DEFINES
+
+#include <algorithm>
+#include <cmath>
+#include <vector>
+#include "audio_core/algorithm/interpolate.h"
+#include "common/common_types.h"
+#include "common/logging/log.h"
+
+namespace AudioCore {
+
+/// The Lanczos kernel
+static double Lanczos(size_t a, double x) {
+ if (x == 0.0)
+ return 1.0;
+ const double px = M_PI * x;
+ return a * std::sin(px) * std::sin(px / a) / (px * px);
+}
+
+std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input, double ratio) {
+ if (input.size() < 2)
+ return {};
+
+ if (ratio <= 0) {
+ LOG_CRITICAL(Audio, "Nonsensical interpolation ratio {}", ratio);
+ ratio = 1.0;
+ }
+
+ if (ratio != state.current_ratio) {
+ const double cutoff_frequency = std::min(0.5 / ratio, 0.5 * ratio);
+ state.nyquist = CascadingFilter::LowPass(std::clamp(cutoff_frequency, 0.0, 0.4), 3);
+ state.current_ratio = ratio;
+ }
+ state.nyquist.Process(input);
+
+ constexpr size_t taps = InterpolationState::lanczos_taps;
+ const size_t num_frames = input.size() / 2;
+
+ std::vector<s16> output;
+ output.reserve(static_cast<size_t>(input.size() / ratio + 4));
+
+ double& pos = state.position;
+ auto& h = state.history;
+ for (size_t i = 0; i < num_frames; ++i) {
+ std::rotate(h.begin(), h.end() - 1, h.end());
+ h[0][0] = input[i * 2 + 0];
+ h[0][1] = input[i * 2 + 1];
+
+ while (pos <= 1.0) {
+ double l = 0.0;
+ double r = 0.0;
+ for (size_t j = 0; j < h.size(); j++) {
+ l += Lanczos(taps, pos + j - taps + 1) * h[j][0];
+ r += Lanczos(taps, pos + j - taps + 1) * h[j][1];
+ }
+ output.emplace_back(static_cast<s16>(std::clamp(l, -32768.0, 32767.0)));
+ output.emplace_back(static_cast<s16>(std::clamp(r, -32768.0, 32767.0)));
+
+ pos += ratio;
+ }
+ pos -= 1.0;
+ }
+
+ return output;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/algorithm/interpolate.h b/src/audio_core/algorithm/interpolate.h
new file mode 100644
index 000000000..c79c2eef4
--- /dev/null
+++ b/src/audio_core/algorithm/interpolate.h
@@ -0,0 +1,43 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <vector>
+#include "audio_core/algorithm/filter.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+struct InterpolationState {
+ static constexpr size_t lanczos_taps = 4;
+ static constexpr size_t history_size = lanczos_taps * 2 - 1;
+
+ double current_ratio = 0.0;
+ CascadingFilter nyquist;
+ std::array<std::array<s16, 2>, history_size> history = {};
+ double position = 0;
+};
+
+/// Interpolates input signal to produce output signal.
+/// @param input The signal to interpolate.
+/// @param ratio Interpolation ratio.
+/// ratio > 1.0 results in fewer output samples.
+/// ratio < 1.0 results in more output samples.
+/// @returns Output signal.
+std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input, double ratio);
+
+/// Interpolates input signal to produce output signal.
+/// @param input The signal to interpolate.
+/// @param input_rate The sample rate of input.
+/// @param output_rate The desired sample rate of the output.
+/// @returns Output signal.
+inline std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
+ u32 input_rate, u32 output_rate) {
+ const double ratio = static_cast<double>(input_rate) / static_cast<double>(output_rate);
+ return Interpolate(state, std::move(input), ratio);
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_out.cpp b/src/audio_core/audio_out.cpp
new file mode 100644
index 000000000..12632a95c
--- /dev/null
+++ b/src/audio_core/audio_out.cpp
@@ -0,0 +1,58 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio_core/audio_out.h"
+#include "audio_core/sink.h"
+#include "audio_core/sink_details.h"
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "core/settings.h"
+
+namespace AudioCore {
+
+/// Returns the stream format from the specified number of channels
+static Stream::Format ChannelsToStreamFormat(u32 num_channels) {
+ switch (num_channels) {
+ case 1:
+ return Stream::Format::Mono16;
+ case 2:
+ return Stream::Format::Stereo16;
+ case 6:
+ return Stream::Format::Multi51Channel16;
+ }
+
+ LOG_CRITICAL(Audio, "Unimplemented num_channels={}", num_channels);
+ UNREACHABLE();
+ return {};
+}
+
+StreamPtr AudioOut::OpenStream(u32 sample_rate, u32 num_channels, std::string&& name,
+ Stream::ReleaseCallback&& release_callback) {
+ if (!sink) {
+ const SinkDetails& sink_details = GetSinkDetails(Settings::values.sink_id);
+ sink = sink_details.factory(Settings::values.audio_device_id);
+ }
+
+ return std::make_shared<Stream>(
+ sample_rate, ChannelsToStreamFormat(num_channels), std::move(release_callback),
+ sink->AcquireSinkStream(sample_rate, num_channels, name), std::move(name));
+}
+
+std::vector<Buffer::Tag> AudioOut::GetTagsAndReleaseBuffers(StreamPtr stream, size_t max_count) {
+ return stream->GetTagsAndReleaseBuffers(max_count);
+}
+
+void AudioOut::StartStream(StreamPtr stream) {
+ stream->Play();
+}
+
+void AudioOut::StopStream(StreamPtr stream) {
+ stream->Stop();
+}
+
+bool AudioOut::QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data) {
+ return stream->QueueBuffer(std::make_shared<Buffer>(tag, std::move(data)));
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_out.h b/src/audio_core/audio_out.h
new file mode 100644
index 000000000..39b7e656b
--- /dev/null
+++ b/src/audio_core/audio_out.h
@@ -0,0 +1,43 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "audio_core/buffer.h"
+#include "audio_core/sink.h"
+#include "audio_core/stream.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/**
+ * Represents an audio playback interface, used to open and play audio streams
+ */
+class AudioOut {
+public:
+ /// Opens a new audio stream
+ StreamPtr OpenStream(u32 sample_rate, u32 num_channels, std::string&& name,
+ Stream::ReleaseCallback&& release_callback);
+
+ /// Returns a vector of recently released buffers specified by tag for the specified stream
+ std::vector<Buffer::Tag> GetTagsAndReleaseBuffers(StreamPtr stream, size_t max_count);
+
+ /// Starts an audio stream for playback
+ void StartStream(StreamPtr stream);
+
+ /// Stops an audio stream that is currently playing
+ void StopStream(StreamPtr stream);
+
+ /// Queues a buffer into the specified audio stream, returns true on success
+ bool QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data);
+
+private:
+ SinkPtr sink;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_renderer.cpp b/src/audio_core/audio_renderer.cpp
new file mode 100644
index 000000000..397b107f5
--- /dev/null
+++ b/src/audio_core/audio_renderer.cpp
@@ -0,0 +1,249 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include "audio_core/algorithm/interpolate.h"
+#include "audio_core/audio_renderer.h"
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "core/memory.h"
+
+namespace AudioCore {
+
+constexpr u32 STREAM_SAMPLE_RATE{48000};
+constexpr u32 STREAM_NUM_CHANNELS{2};
+
+AudioRenderer::AudioRenderer(AudioRendererParameter params,
+ Kernel::SharedPtr<Kernel::Event> buffer_event)
+ : worker_params{params}, buffer_event{buffer_event}, voices(params.voice_count) {
+
+ audio_core = std::make_unique<AudioCore::AudioOut>();
+ stream = audio_core->OpenStream(STREAM_SAMPLE_RATE, STREAM_NUM_CHANNELS, "AudioRenderer",
+ [=]() { buffer_event->Signal(); });
+ audio_core->StartStream(stream);
+
+ QueueMixedBuffer(0);
+ QueueMixedBuffer(1);
+ QueueMixedBuffer(2);
+}
+
+u32 AudioRenderer::GetSampleRate() const {
+ return worker_params.sample_rate;
+}
+
+u32 AudioRenderer::GetSampleCount() const {
+ return worker_params.sample_count;
+}
+
+u32 AudioRenderer::GetMixBufferCount() const {
+ return worker_params.mix_buffer_count;
+}
+
+std::vector<u8> AudioRenderer::UpdateAudioRenderer(const std::vector<u8>& input_params) {
+ // Copy UpdateDataHeader struct
+ UpdateDataHeader config{};
+ std::memcpy(&config, input_params.data(), sizeof(UpdateDataHeader));
+ u32 memory_pool_count = worker_params.effect_count + (worker_params.voice_count * 4);
+
+ // Copy MemoryPoolInfo structs
+ std::vector<MemoryPoolInfo> mem_pool_info(memory_pool_count);
+ std::memcpy(mem_pool_info.data(),
+ input_params.data() + sizeof(UpdateDataHeader) + config.behavior_size,
+ memory_pool_count * sizeof(MemoryPoolInfo));
+
+ // Copy VoiceInfo structs
+ size_t offset{sizeof(UpdateDataHeader) + config.behavior_size + config.memory_pools_size +
+ config.voice_resource_size};
+ for (auto& voice : voices) {
+ std::memcpy(&voice.Info(), input_params.data() + offset, sizeof(VoiceInfo));
+ offset += sizeof(VoiceInfo);
+ }
+
+ // Update voices
+ for (auto& voice : voices) {
+ voice.UpdateState();
+ if (!voice.GetInfo().is_in_use) {
+ continue;
+ }
+ if (voice.GetInfo().is_new) {
+ voice.SetWaveIndex(voice.GetInfo().wave_buffer_head);
+ }
+ }
+
+ // Update memory pool state
+ std::vector<MemoryPoolEntry> memory_pool(memory_pool_count);
+ for (size_t index = 0; index < memory_pool.size(); ++index) {
+ if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestAttach) {
+ memory_pool[index].state = MemoryPoolStates::Attached;
+ } else if (mem_pool_info[index].pool_state == MemoryPoolStates::RequestDetach) {
+ memory_pool[index].state = MemoryPoolStates::Detached;
+ }
+ }
+
+ // Release previous buffers and queue next ones for playback
+ ReleaseAndQueueBuffers();
+
+ // Copy output header
+ UpdateDataHeader response_data{worker_params};
+ std::vector<u8> output_params(response_data.total_size);
+ std::memcpy(output_params.data(), &response_data, sizeof(UpdateDataHeader));
+
+ // Copy output memory pool entries
+ std::memcpy(output_params.data() + sizeof(UpdateDataHeader), memory_pool.data(),
+ response_data.memory_pools_size);
+
+ // Copy output voice status
+ size_t voice_out_status_offset{sizeof(UpdateDataHeader) + response_data.memory_pools_size};
+ for (const auto& voice : voices) {
+ std::memcpy(output_params.data() + voice_out_status_offset, &voice.GetOutStatus(),
+ sizeof(VoiceOutStatus));
+ voice_out_status_offset += sizeof(VoiceOutStatus);
+ }
+
+ return output_params;
+}
+
+void AudioRenderer::VoiceState::SetWaveIndex(size_t index) {
+ wave_index = index & 3;
+ is_refresh_pending = true;
+}
+
+std::vector<s16> AudioRenderer::VoiceState::DequeueSamples(size_t sample_count) {
+ if (!IsPlaying()) {
+ return {};
+ }
+
+ if (is_refresh_pending) {
+ RefreshBuffer();
+ }
+
+ const size_t max_size{samples.size() - offset};
+ const size_t dequeue_offset{offset};
+ size_t size{sample_count * STREAM_NUM_CHANNELS};
+ if (size > max_size) {
+ size = max_size;
+ }
+
+ out_status.played_sample_count += size / STREAM_NUM_CHANNELS;
+ offset += size;
+
+ const auto& wave_buffer{info.wave_buffer[wave_index]};
+ if (offset == samples.size()) {
+ offset = 0;
+
+ if (!wave_buffer.is_looping) {
+ SetWaveIndex(wave_index + 1);
+ }
+
+ out_status.wave_buffer_consumed++;
+
+ if (wave_buffer.end_of_stream) {
+ info.play_state = PlayState::Paused;
+ }
+ }
+
+ return {samples.begin() + dequeue_offset, samples.begin() + dequeue_offset + size};
+}
+
+void AudioRenderer::VoiceState::UpdateState() {
+ if (is_in_use && !info.is_in_use) {
+ // No longer in use, reset state
+ is_refresh_pending = true;
+ wave_index = 0;
+ offset = 0;
+ out_status = {};
+ }
+ is_in_use = info.is_in_use;
+}
+
+void AudioRenderer::VoiceState::RefreshBuffer() {
+ std::vector<s16> new_samples(info.wave_buffer[wave_index].buffer_sz / sizeof(s16));
+ Memory::ReadBlock(info.wave_buffer[wave_index].buffer_addr, new_samples.data(),
+ info.wave_buffer[wave_index].buffer_sz);
+
+ switch (static_cast<Codec::PcmFormat>(info.sample_format)) {
+ case Codec::PcmFormat::Int16: {
+ // PCM16 is played as-is
+ break;
+ }
+ case Codec::PcmFormat::Adpcm: {
+ // Decode ADPCM to PCM16
+ Codec::ADPCM_Coeff coeffs;
+ Memory::ReadBlock(info.additional_params_addr, coeffs.data(), sizeof(Codec::ADPCM_Coeff));
+ new_samples = Codec::DecodeADPCM(reinterpret_cast<u8*>(new_samples.data()),
+ new_samples.size() * sizeof(s16), coeffs, adpcm_state);
+ break;
+ }
+ default:
+ LOG_CRITICAL(Audio, "Unimplemented sample_format={}", info.sample_format);
+ UNREACHABLE();
+ break;
+ }
+
+ switch (info.channel_count) {
+ case 1:
+ // 1 channel is upsampled to 2 channel
+ samples.resize(new_samples.size() * 2);
+ for (size_t index = 0; index < new_samples.size(); ++index) {
+ samples[index * 2] = new_samples[index];
+ samples[index * 2 + 1] = new_samples[index];
+ }
+ break;
+ case 2: {
+ // 2 channel is played as is
+ samples = std::move(new_samples);
+ break;
+ }
+ default:
+ LOG_CRITICAL(Audio, "Unimplemented channel_count={}", info.channel_count);
+ UNREACHABLE();
+ break;
+ }
+
+ samples = Interpolate(interp_state, std::move(samples), Info().sample_rate, STREAM_SAMPLE_RATE);
+
+ is_refresh_pending = false;
+}
+
+static constexpr s16 ClampToS16(s32 value) {
+ return static_cast<s16>(std::clamp(value, -32768, 32767));
+}
+
+void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
+ constexpr size_t BUFFER_SIZE{512};
+ std::vector<s16> buffer(BUFFER_SIZE * stream->GetNumChannels());
+
+ for (auto& voice : voices) {
+ if (!voice.IsPlaying()) {
+ continue;
+ }
+
+ size_t offset{};
+ s64 samples_remaining{BUFFER_SIZE};
+ while (samples_remaining > 0) {
+ const std::vector<s16> samples{voice.DequeueSamples(samples_remaining)};
+
+ if (samples.empty()) {
+ break;
+ }
+
+ samples_remaining -= samples.size() / stream->GetNumChannels();
+
+ for (const auto& sample : samples) {
+ const s32 buffer_sample{buffer[offset]};
+ buffer[offset++] =
+ ClampToS16(buffer_sample + static_cast<s32>(sample * voice.GetInfo().volume));
+ }
+ }
+ }
+ audio_core->QueueBuffer(stream, tag, std::move(buffer));
+}
+
+void AudioRenderer::ReleaseAndQueueBuffers() {
+ const auto released_buffers{audio_core->GetTagsAndReleaseBuffers(stream, 2)};
+ for (const auto& tag : released_buffers) {
+ QueueMixedBuffer(tag);
+ }
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/audio_renderer.h b/src/audio_core/audio_renderer.h
new file mode 100644
index 000000000..eba67f28e
--- /dev/null
+++ b/src/audio_core/audio_renderer.h
@@ -0,0 +1,211 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "audio_core/algorithm/interpolate.h"
+#include "audio_core/audio_out.h"
+#include "audio_core/codec.h"
+#include "audio_core/stream.h"
+#include "common/common_types.h"
+#include "common/swap.h"
+#include "core/hle/kernel/event.h"
+
+namespace AudioCore {
+
+enum class PlayState : u8 {
+ Started = 0,
+ Stopped = 1,
+ Paused = 2,
+};
+
+struct AudioRendererParameter {
+ u32_le sample_rate;
+ u32_le sample_count;
+ u32_le mix_buffer_count;
+ u32_le unknown_c;
+ u32_le voice_count;
+ u32_le sink_count;
+ u32_le effect_count;
+ u32_le unknown_1c;
+ u8 unknown_20;
+ INSERT_PADDING_BYTES(3);
+ u32_le splitter_count;
+ u32_le unknown_2c;
+ INSERT_PADDING_WORDS(1);
+ u32_le revision;
+};
+static_assert(sizeof(AudioRendererParameter) == 52, "AudioRendererParameter is an invalid size");
+
+enum class MemoryPoolStates : u32 { // Should be LE
+ Invalid = 0x0,
+ Unknown = 0x1,
+ RequestDetach = 0x2,
+ Detached = 0x3,
+ RequestAttach = 0x4,
+ Attached = 0x5,
+ Released = 0x6,
+};
+
+struct MemoryPoolEntry {
+ MemoryPoolStates state;
+ u32_le unknown_4;
+ u32_le unknown_8;
+ u32_le unknown_c;
+};
+static_assert(sizeof(MemoryPoolEntry) == 0x10, "MemoryPoolEntry has wrong size");
+
+struct MemoryPoolInfo {
+ u64_le pool_address;
+ u64_le pool_size;
+ MemoryPoolStates pool_state;
+ INSERT_PADDING_WORDS(3); // Unknown
+};
+static_assert(sizeof(MemoryPoolInfo) == 0x20, "MemoryPoolInfo has wrong size");
+struct BiquadFilter {
+ u8 enable;
+ INSERT_PADDING_BYTES(1);
+ std::array<s16_le, 3> numerator;
+ std::array<s16_le, 2> denominator;
+};
+static_assert(sizeof(BiquadFilter) == 0xc, "BiquadFilter has wrong size");
+
+struct WaveBuffer {
+ u64_le buffer_addr;
+ u64_le buffer_sz;
+ s32_le start_sample_offset;
+ s32_le end_sample_offset;
+ u8 is_looping;
+ u8 end_of_stream;
+ u8 sent_to_server;
+ INSERT_PADDING_BYTES(5);
+ u64 context_addr;
+ u64 context_sz;
+ INSERT_PADDING_BYTES(8);
+};
+static_assert(sizeof(WaveBuffer) == 0x38, "WaveBuffer has wrong size");
+
+struct VoiceInfo {
+ u32_le id;
+ u32_le node_id;
+ u8 is_new;
+ u8 is_in_use;
+ PlayState play_state;
+ u8 sample_format;
+ u32_le sample_rate;
+ u32_le priority;
+ u32_le sorting_order;
+ u32_le channel_count;
+ float_le pitch;
+ float_le volume;
+ std::array<BiquadFilter, 2> biquad_filter;
+ u32_le wave_buffer_count;
+ u32_le wave_buffer_head;
+ INSERT_PADDING_WORDS(1);
+ u64_le additional_params_addr;
+ u64_le additional_params_sz;
+ u32_le mix_id;
+ u32_le splitter_info_id;
+ std::array<WaveBuffer, 4> wave_buffer;
+ std::array<u32_le, 6> voice_channel_resource_ids;
+ INSERT_PADDING_BYTES(24);
+};
+static_assert(sizeof(VoiceInfo) == 0x170, "VoiceInfo is wrong size");
+
+struct VoiceOutStatus {
+ u64_le played_sample_count;
+ u32_le wave_buffer_consumed;
+ u32_le voice_drops_count;
+};
+static_assert(sizeof(VoiceOutStatus) == 0x10, "VoiceOutStatus has wrong size");
+
+struct UpdateDataHeader {
+ UpdateDataHeader() {}
+
+ explicit UpdateDataHeader(const AudioRendererParameter& config) {
+ revision = Common::MakeMagic('R', 'E', 'V', '4'); // 5.1.0 Revision
+ behavior_size = 0xb0;
+ memory_pools_size = (config.effect_count + (config.voice_count * 4)) * 0x10;
+ voices_size = config.voice_count * 0x10;
+ voice_resource_size = 0x0;
+ effects_size = config.effect_count * 0x10;
+ mixes_size = 0x0;
+ sinks_size = config.sink_count * 0x20;
+ performance_manager_size = 0x10;
+ total_size = sizeof(UpdateDataHeader) + behavior_size + memory_pools_size + voices_size +
+ effects_size + sinks_size + performance_manager_size;
+ }
+
+ u32_le revision;
+ u32_le behavior_size;
+ u32_le memory_pools_size;
+ u32_le voices_size;
+ u32_le voice_resource_size;
+ u32_le effects_size;
+ u32_le mixes_size;
+ u32_le sinks_size;
+ u32_le performance_manager_size;
+ INSERT_PADDING_WORDS(6);
+ u32_le total_size;
+};
+static_assert(sizeof(UpdateDataHeader) == 0x40, "UpdateDataHeader has wrong size");
+
+class AudioRenderer {
+public:
+ AudioRenderer(AudioRendererParameter params, Kernel::SharedPtr<Kernel::Event> buffer_event);
+ std::vector<u8> UpdateAudioRenderer(const std::vector<u8>& input_params);
+ void QueueMixedBuffer(Buffer::Tag tag);
+ void ReleaseAndQueueBuffers();
+ u32 GetSampleRate() const;
+ u32 GetSampleCount() const;
+ u32 GetMixBufferCount() const;
+
+private:
+ class VoiceState {
+ public:
+ bool IsPlaying() const {
+ return is_in_use && info.play_state == PlayState::Started;
+ }
+
+ const VoiceOutStatus& GetOutStatus() const {
+ return out_status;
+ }
+
+ const VoiceInfo& GetInfo() const {
+ return info;
+ }
+
+ VoiceInfo& Info() {
+ return info;
+ }
+
+ void SetWaveIndex(size_t index);
+ std::vector<s16> DequeueSamples(size_t sample_count);
+ void UpdateState();
+ void RefreshBuffer();
+
+ private:
+ bool is_in_use{};
+ bool is_refresh_pending{};
+ size_t wave_index{};
+ size_t offset{};
+ Codec::ADPCMState adpcm_state{};
+ InterpolationState interp_state{};
+ std::vector<s16> samples;
+ VoiceOutStatus out_status{};
+ VoiceInfo info{};
+ };
+
+ AudioRendererParameter worker_params;
+ Kernel::SharedPtr<Kernel::Event> buffer_event;
+ std::vector<VoiceState> voices;
+ std::unique_ptr<AudioCore::AudioOut> audio_core;
+ AudioCore::StreamPtr stream;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/buffer.h b/src/audio_core/buffer.h
new file mode 100644
index 000000000..a323b23ec
--- /dev/null
+++ b/src/audio_core/buffer.h
@@ -0,0 +1,45 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/**
+ * Represents a buffer of audio samples to be played in an audio stream
+ */
+class Buffer {
+public:
+ using Tag = u64;
+
+ Buffer(Tag tag, std::vector<s16>&& samples) : tag{tag}, samples{std::move(samples)} {}
+
+ /// Returns the raw audio data for the buffer
+ std::vector<s16>& Samples() {
+ return samples;
+ }
+
+ /// Returns the raw audio data for the buffer
+ const std::vector<s16>& GetSamples() const {
+ return samples;
+ }
+
+ /// Returns the buffer tag, this is provided by the game to the audout service
+ Tag GetTag() const {
+ return tag;
+ }
+
+private:
+ Tag tag;
+ std::vector<s16> samples;
+};
+
+using BufferPtr = std::shared_ptr<Buffer>;
+
+} // namespace AudioCore
diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
new file mode 100644
index 000000000..c3021403f
--- /dev/null
+++ b/src/audio_core/codec.cpp
@@ -0,0 +1,77 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+
+#include "audio_core/codec.h"
+
+namespace AudioCore::Codec {
+
+std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
+ ADPCMState& state) {
+ // GC-ADPCM with scale factor and variable coefficients.
+ // Frames are 8 bytes long containing 14 samples each.
+ // Samples are 4 bits (one nibble) long.
+
+ constexpr size_t FRAME_LEN = 8;
+ constexpr size_t SAMPLES_PER_FRAME = 14;
+ constexpr std::array<int, 16> SIGNED_NIBBLES = {
+ {0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}};
+
+ const size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
+ const size_t ret_size =
+ sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
+ std::vector<s16> ret(ret_size);
+
+ int yn1 = state.yn1, yn2 = state.yn2;
+
+ const size_t NUM_FRAMES =
+ (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
+ for (size_t framei = 0; framei < NUM_FRAMES; framei++) {
+ const int frame_header = data[framei * FRAME_LEN];
+ const int scale = 1 << (frame_header & 0xF);
+ const int idx = (frame_header >> 4) & 0x7;
+
+ // Coefficients are fixed point with 11 bits fractional part.
+ const int coef1 = coeff[idx * 2 + 0];
+ const int coef2 = coeff[idx * 2 + 1];
+
+ // Decodes an audio sample. One nibble produces one sample.
+ const auto decode_sample = [&](const int nibble) -> s16 {
+ const int xn = nibble * scale;
+ // We first transform everything into 11 bit fixed point, perform the second order
+ // digital filter, then transform back.
+ // 0x400 == 0.5 in 11 bit fixed point.
+ // Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
+ int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
+ // Clamp to output range.
+ val = std::clamp<s32>(val, -32768, 32767);
+ // Advance output feedback.
+ yn2 = yn1;
+ yn1 = val;
+ return static_cast<s16>(val);
+ };
+
+ size_t outputi = framei * SAMPLES_PER_FRAME;
+ size_t datai = framei * FRAME_LEN + 1;
+ for (size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
+ const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
+ ret[outputi] = sample1;
+ outputi++;
+
+ const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]);
+ ret[outputi] = sample2;
+ outputi++;
+
+ datai++;
+ }
+ }
+
+ state.yn1 = yn1;
+ state.yn2 = yn2;
+
+ return ret;
+}
+
+} // namespace AudioCore::Codec
diff --git a/src/audio_core/codec.h b/src/audio_core/codec.h
new file mode 100644
index 000000000..3f845c42c
--- /dev/null
+++ b/src/audio_core/codec.h
@@ -0,0 +1,44 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioCore::Codec {
+
+enum class PcmFormat : u32 {
+ Invalid = 0,
+ Int8 = 1,
+ Int16 = 2,
+ Int24 = 3,
+ Int32 = 4,
+ PcmFloat = 5,
+ Adpcm = 6,
+};
+
+/// See: Codec::DecodeADPCM
+struct ADPCMState {
+ // Two historical samples from previous processed buffer,
+ // required for ADPCM decoding
+ s16 yn1; ///< y[n-1]
+ s16 yn2; ///< y[n-2]
+};
+
+using ADPCM_Coeff = std::array<s16, 16>;
+
+/**
+ * @param data Pointer to buffer that contains ADPCM data to decode
+ * @param size Size of buffer in bytes
+ * @param coeff ADPCM coefficients
+ * @param state ADPCM state, this is updated with new state
+ * @return Decoded stereo signed PCM16 data, sample_count in length
+ */
+std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff,
+ ADPCMState& state);
+
+}; // namespace AudioCore::Codec
diff --git a/src/audio_core/cubeb_sink.cpp b/src/audio_core/cubeb_sink.cpp
new file mode 100644
index 000000000..5a1177d0c
--- /dev/null
+++ b/src/audio_core/cubeb_sink.cpp
@@ -0,0 +1,206 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <cstring>
+#include <mutex>
+
+#include "audio_core/cubeb_sink.h"
+#include "audio_core/stream.h"
+#include "common/logging/log.h"
+
+namespace AudioCore {
+
+class SinkStreamImpl final : public SinkStream {
+public:
+ SinkStreamImpl(cubeb* ctx, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
+ const std::string& name)
+ : ctx{ctx}, num_channels{num_channels_} {
+
+ if (num_channels == 6) {
+ // 6-channel audio does not seem to work with cubeb + SDL, so we downsample this to 2
+ // channel for now
+ is_6_channel = true;
+ num_channels = 2;
+ }
+
+ cubeb_stream_params params{};
+ params.rate = sample_rate;
+ params.channels = num_channels;
+ params.format = CUBEB_SAMPLE_S16NE;
+ params.layout = num_channels == 1 ? CUBEB_LAYOUT_MONO : CUBEB_LAYOUT_STEREO;
+
+ u32 minimum_latency{};
+ if (cubeb_get_min_latency(ctx, &params, &minimum_latency) != CUBEB_OK) {
+ LOG_CRITICAL(Audio_Sink, "Error getting minimum latency");
+ }
+
+ if (cubeb_stream_init(ctx, &stream_backend, name.c_str(), nullptr, nullptr, output_device,
+ &params, std::max(512u, minimum_latency),
+ &SinkStreamImpl::DataCallback, &SinkStreamImpl::StateCallback,
+ this) != CUBEB_OK) {
+ LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream");
+ return;
+ }
+
+ if (cubeb_stream_start(stream_backend) != CUBEB_OK) {
+ LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream");
+ return;
+ }
+ }
+
+ ~SinkStreamImpl() {
+ if (!ctx) {
+ return;
+ }
+
+ if (cubeb_stream_stop(stream_backend) != CUBEB_OK) {
+ LOG_CRITICAL(Audio_Sink, "Error stopping cubeb stream");
+ }
+
+ cubeb_stream_destroy(stream_backend);
+ }
+
+ void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) override {
+ if (!ctx) {
+ return;
+ }
+
+ std::lock_guard lock{queue_mutex};
+
+ queue.reserve(queue.size() + samples.size() * GetNumChannels());
+
+ if (is_6_channel) {
+ // Downsample 6 channels to 2
+ const size_t sample_count_copy_size = samples.size() * 2;
+ queue.reserve(sample_count_copy_size);
+ for (size_t i = 0; i < samples.size(); i += num_channels) {
+ queue.push_back(samples[i]);
+ queue.push_back(samples[i + 1]);
+ }
+ } else {
+ // Copy as-is
+ std::copy(samples.begin(), samples.end(), std::back_inserter(queue));
+ }
+ }
+
+ u32 GetNumChannels() const {
+ return num_channels;
+ }
+
+private:
+ std::vector<std::string> device_list;
+
+ cubeb* ctx{};
+ cubeb_stream* stream_backend{};
+ u32 num_channels{};
+ bool is_6_channel{};
+
+ std::mutex queue_mutex;
+ std::vector<s16> queue;
+
+ static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
+ void* output_buffer, long num_frames);
+ static void StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state);
+};
+
+CubebSink::CubebSink(std::string target_device_name) {
+ if (cubeb_init(&ctx, "yuzu", nullptr) != CUBEB_OK) {
+ LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
+ return;
+ }
+
+ if (target_device_name != auto_device_name && !target_device_name.empty()) {
+ cubeb_device_collection collection;
+ if (cubeb_enumerate_devices(ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) {
+ LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported");
+ } else {
+ const auto collection_end{collection.device + collection.count};
+ const auto device{std::find_if(collection.device, collection_end,
+ [&](const cubeb_device_info& device) {
+ return target_device_name == device.friendly_name;
+ })};
+ if (device != collection_end) {
+ output_device = device->devid;
+ }
+ cubeb_device_collection_destroy(ctx, &collection);
+ }
+ }
+}
+
+CubebSink::~CubebSink() {
+ if (!ctx) {
+ return;
+ }
+
+ for (auto& sink_stream : sink_streams) {
+ sink_stream.reset();
+ }
+
+ cubeb_destroy(ctx);
+}
+
+SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels,
+ const std::string& name) {
+ sink_streams.push_back(
+ std::make_unique<SinkStreamImpl>(ctx, sample_rate, num_channels, output_device, name));
+ return *sink_streams.back();
+}
+
+long SinkStreamImpl::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
+ void* output_buffer, long num_frames) {
+ SinkStreamImpl* impl = static_cast<SinkStreamImpl*>(user_data);
+ u8* buffer = reinterpret_cast<u8*>(output_buffer);
+
+ if (!impl) {
+ return {};
+ }
+
+ std::lock_guard lock{impl->queue_mutex};
+
+ const size_t frames_to_write{
+ std::min(impl->queue.size() / impl->GetNumChannels(), static_cast<size_t>(num_frames))};
+
+ memcpy(buffer, impl->queue.data(), frames_to_write * sizeof(s16) * impl->GetNumChannels());
+ impl->queue.erase(impl->queue.begin(),
+ impl->queue.begin() + frames_to_write * impl->GetNumChannels());
+
+ if (frames_to_write < num_frames) {
+ // Fill the rest of the frames with silence
+ memset(buffer + frames_to_write * sizeof(s16) * impl->GetNumChannels(), 0,
+ (num_frames - frames_to_write) * sizeof(s16) * impl->GetNumChannels());
+ }
+
+ return num_frames;
+}
+
+void SinkStreamImpl::StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state) {}
+
+std::vector<std::string> ListCubebSinkDevices() {
+ std::vector<std::string> device_list;
+ cubeb* ctx;
+
+ if (cubeb_init(&ctx, "Citra Device Enumerator", nullptr) != CUBEB_OK) {
+ LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
+ return {};
+ }
+
+ cubeb_device_collection collection;
+ if (cubeb_enumerate_devices(ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) {
+ LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported");
+ } else {
+ for (size_t i = 0; i < collection.count; i++) {
+ const cubeb_device_info& device = collection.device[i];
+ if (device.friendly_name) {
+ device_list.emplace_back(device.friendly_name);
+ }
+ }
+ cubeb_device_collection_destroy(ctx, &collection);
+ }
+
+ cubeb_destroy(ctx);
+ return device_list;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/cubeb_sink.h b/src/audio_core/cubeb_sink.h
new file mode 100644
index 000000000..59cbf05e9
--- /dev/null
+++ b/src/audio_core/cubeb_sink.h
@@ -0,0 +1,32 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <string>
+#include <vector>
+
+#include <cubeb/cubeb.h>
+
+#include "audio_core/sink.h"
+
+namespace AudioCore {
+
+class CubebSink final : public Sink {
+public:
+ explicit CubebSink(std::string device_id);
+ ~CubebSink() override;
+
+ SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels,
+ const std::string& name) override;
+
+private:
+ cubeb* ctx{};
+ cubeb_devid output_device{};
+ std::vector<SinkStreamPtr> sink_streams;
+};
+
+std::vector<std::string> ListCubebSinkDevices();
+
+} // namespace AudioCore
diff --git a/src/audio_core/null_sink.h b/src/audio_core/null_sink.h
new file mode 100644
index 000000000..f235d93e5
--- /dev/null
+++ b/src/audio_core/null_sink.h
@@ -0,0 +1,27 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include "audio_core/sink.h"
+
+namespace AudioCore {
+
+class NullSink final : public Sink {
+public:
+ explicit NullSink(std::string){};
+ ~NullSink() override = default;
+
+ SinkStream& AcquireSinkStream(u32 /*sample_rate*/, u32 /*num_channels*/,
+ const std::string& /*name*/) override {
+ return null_sink_stream;
+ }
+
+private:
+ struct NullSinkStreamImpl final : SinkStream {
+ void EnqueueSamples(u32 /*num_channels*/, const std::vector<s16>& /*samples*/) override {}
+ } null_sink_stream;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink.h b/src/audio_core/sink.h
new file mode 100644
index 000000000..95c7b2b6e
--- /dev/null
+++ b/src/audio_core/sink.h
@@ -0,0 +1,31 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <string>
+
+#include "audio_core/sink_stream.h"
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+constexpr char auto_device_name[] = "auto";
+
+/**
+ * This class is an interface for an audio sink. An audio sink accepts samples in stereo signed
+ * PCM16 format to be output. Sinks *do not* handle resampling and expect the correct sample rate.
+ * They are dumb outputs.
+ */
+class Sink {
+public:
+ virtual ~Sink() = default;
+ virtual SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels,
+ const std::string& name) = 0;
+};
+
+using SinkPtr = std::unique_ptr<Sink>;
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink_details.cpp b/src/audio_core/sink_details.cpp
new file mode 100644
index 000000000..955ba20fb
--- /dev/null
+++ b/src/audio_core/sink_details.cpp
@@ -0,0 +1,44 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+#include "audio_core/null_sink.h"
+#include "audio_core/sink_details.h"
+#ifdef HAVE_CUBEB
+#include "audio_core/cubeb_sink.h"
+#endif
+#include "common/logging/log.h"
+
+namespace AudioCore {
+
+// g_sink_details is ordered in terms of desirability, with the best choice at the top.
+const std::vector<SinkDetails> g_sink_details = {
+#ifdef HAVE_CUBEB
+ SinkDetails{"cubeb", &std::make_unique<CubebSink, std::string>, &ListCubebSinkDevices},
+#endif
+ SinkDetails{"null", &std::make_unique<NullSink, std::string>,
+ [] { return std::vector<std::string>{"null"}; }},
+};
+
+const SinkDetails& GetSinkDetails(std::string sink_id) {
+ auto iter =
+ std::find_if(g_sink_details.begin(), g_sink_details.end(),
+ [sink_id](const auto& sink_detail) { return sink_detail.id == sink_id; });
+
+ if (sink_id == "auto" || iter == g_sink_details.end()) {
+ if (sink_id != "auto") {
+ LOG_ERROR(Audio, "AudioCore::SelectSink given invalid sink_id {}", sink_id);
+ }
+ // Auto-select.
+ // g_sink_details is ordered in terms of desirability, with the best choice at the front.
+ iter = g_sink_details.begin();
+ }
+
+ return *iter;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink_details.h b/src/audio_core/sink_details.h
new file mode 100644
index 000000000..ea666c554
--- /dev/null
+++ b/src/audio_core/sink_details.h
@@ -0,0 +1,35 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <utility>
+#include <vector>
+
+namespace AudioCore {
+
+class Sink;
+
+struct SinkDetails {
+ using FactoryFn = std::function<std::unique_ptr<Sink>(std::string)>;
+ using ListDevicesFn = std::function<std::vector<std::string>()>;
+
+ SinkDetails(const char* id_, FactoryFn factory_, ListDevicesFn list_devices_)
+ : id(id_), factory(std::move(factory_)), list_devices(std::move(list_devices_)) {}
+
+ /// Name for this sink.
+ const char* id;
+ /// A method to call to construct an instance of this type of sink.
+ FactoryFn factory;
+ /// A method to call to list available devices.
+ ListDevicesFn list_devices;
+};
+
+extern const std::vector<SinkDetails> g_sink_details;
+
+const SinkDetails& GetSinkDetails(std::string sink_id);
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink_stream.h b/src/audio_core/sink_stream.h
new file mode 100644
index 000000000..41b6736d8
--- /dev/null
+++ b/src/audio_core/sink_stream.h
@@ -0,0 +1,32 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <memory>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+/**
+ * Accepts samples in stereo signed PCM16 format to be output. Sinks *do not* handle resampling and
+ * expect the correct sample rate. They are dumb outputs.
+ */
+class SinkStream {
+public:
+ virtual ~SinkStream() = default;
+
+ /**
+ * Feed stereo samples to sink.
+ * @param num_channels Number of channels used.
+ * @param samples Samples in interleaved stereo PCM16 format.
+ */
+ virtual void EnqueueSamples(u32 num_channels, const std::vector<s16>& samples) = 0;
+};
+
+using SinkStreamPtr = std::unique_ptr<SinkStream>;
+
+} // namespace AudioCore
diff --git a/src/audio_core/stream.cpp b/src/audio_core/stream.cpp
new file mode 100644
index 000000000..ad9e2915c
--- /dev/null
+++ b/src/audio_core/stream.cpp
@@ -0,0 +1,127 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <cmath>
+
+#include "audio_core/sink.h"
+#include "audio_core/sink_details.h"
+#include "audio_core/stream.h"
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "core/core_timing.h"
+#include "core/core_timing_util.h"
+#include "core/settings.h"
+
+namespace AudioCore {
+
+constexpr size_t MaxAudioBufferCount{32};
+
+u32 Stream::GetNumChannels() const {
+ switch (format) {
+ case Format::Mono16:
+ return 1;
+ case Format::Stereo16:
+ return 2;
+ case Format::Multi51Channel16:
+ return 6;
+ }
+ LOG_CRITICAL(Audio, "Unimplemented format={}", static_cast<u32>(format));
+ UNREACHABLE();
+ return {};
+}
+
+Stream::Stream(u32 sample_rate, Format format, ReleaseCallback&& release_callback,
+ SinkStream& sink_stream, std::string&& name_)
+ : sample_rate{sample_rate}, format{format}, release_callback{std::move(release_callback)},
+ sink_stream{sink_stream}, name{std::move(name_)} {
+
+ release_event = CoreTiming::RegisterEvent(
+ name, [this](u64 userdata, int cycles_late) { ReleaseActiveBuffer(); });
+}
+
+void Stream::Play() {
+ state = State::Playing;
+ PlayNextBuffer();
+}
+
+void Stream::Stop() {
+ ASSERT_MSG(false, "Unimplemented");
+}
+
+s64 Stream::GetBufferReleaseCycles(const Buffer& buffer) const {
+ const size_t num_samples{buffer.GetSamples().size() / GetNumChannels()};
+ return CoreTiming::usToCycles((static_cast<u64>(num_samples) * 1000000) / sample_rate);
+}
+
+static void VolumeAdjustSamples(std::vector<s16>& samples) {
+ const float volume{std::clamp(Settings::values.volume, 0.0f, 1.0f)};
+
+ if (volume == 1.0f) {
+ return;
+ }
+
+ // Implementation of a volume slider with a dynamic range of 60 dB
+ const float volume_scale_factor{std::exp(6.90775f * volume) * 0.001f};
+ for (auto& sample : samples) {
+ sample = static_cast<s16>(sample * volume_scale_factor);
+ }
+}
+
+void Stream::PlayNextBuffer() {
+ if (!IsPlaying()) {
+ // Ensure we are in playing state before playing the next buffer
+ return;
+ }
+
+ if (active_buffer) {
+ // Do not queue a new buffer if we are already playing a buffer
+ return;
+ }
+
+ if (queued_buffers.empty()) {
+ // No queued buffers - we are effectively paused
+ return;
+ }
+
+ active_buffer = queued_buffers.front();
+ queued_buffers.pop();
+
+ VolumeAdjustSamples(active_buffer->Samples());
+ sink_stream.EnqueueSamples(GetNumChannels(), active_buffer->GetSamples());
+
+ CoreTiming::ScheduleEventThreadsafe(GetBufferReleaseCycles(*active_buffer), release_event, {});
+}
+
+void Stream::ReleaseActiveBuffer() {
+ ASSERT(active_buffer);
+ released_buffers.push(std::move(active_buffer));
+ release_callback();
+ PlayNextBuffer();
+}
+
+bool Stream::QueueBuffer(BufferPtr&& buffer) {
+ if (queued_buffers.size() < MaxAudioBufferCount) {
+ queued_buffers.push(std::move(buffer));
+ PlayNextBuffer();
+ return true;
+ }
+ return false;
+}
+
+bool Stream::ContainsBuffer(Buffer::Tag tag) const {
+ ASSERT_MSG(false, "Unimplemented");
+ return {};
+}
+
+std::vector<Buffer::Tag> Stream::GetTagsAndReleaseBuffers(size_t max_count) {
+ std::vector<Buffer::Tag> tags;
+ for (size_t count = 0; count < max_count && !released_buffers.empty(); ++count) {
+ tags.push_back(released_buffers.front()->GetTag());
+ released_buffers.pop();
+ }
+ return tags;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/stream.h b/src/audio_core/stream.h
new file mode 100644
index 000000000..049b92ca9
--- /dev/null
+++ b/src/audio_core/stream.h
@@ -0,0 +1,102 @@
+// Copyright 2018 yuzu Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+#include <queue>
+
+#include "audio_core/buffer.h"
+#include "audio_core/sink_stream.h"
+#include "common/assert.h"
+#include "common/common_types.h"
+#include "core/core_timing.h"
+
+namespace AudioCore {
+
+/**
+ * Represents an audio stream, which is a sequence of queued buffers, to be outputed by AudioOut
+ */
+class Stream {
+public:
+ /// Audio format of the stream
+ enum class Format {
+ Mono16,
+ Stereo16,
+ Multi51Channel16,
+ };
+
+ /// Callback function type, used to change guest state on a buffer being released
+ using ReleaseCallback = std::function<void()>;
+
+ Stream(u32 sample_rate, Format format, ReleaseCallback&& release_callback,
+ SinkStream& sink_stream, std::string&& name_);
+
+ /// Plays the audio stream
+ void Play();
+
+ /// Stops the audio stream
+ void Stop();
+
+ /// Queues a buffer into the audio stream, returns true on success
+ bool QueueBuffer(BufferPtr&& buffer);
+
+ /// Returns true if the audio stream contains a buffer with the specified tag
+ bool ContainsBuffer(Buffer::Tag tag) const;
+
+ /// Returns a vector of recently released buffers specified by tag
+ std::vector<Buffer::Tag> GetTagsAndReleaseBuffers(size_t max_count);
+
+ /// Returns true if the stream is currently playing
+ bool IsPlaying() const {
+ return state == State::Playing;
+ }
+
+ /// Returns the number of queued buffers
+ size_t GetQueueSize() const {
+ return queued_buffers.size();
+ }
+
+ /// Gets the sample rate
+ u32 GetSampleRate() const {
+ return sample_rate;
+ }
+
+ /// Gets the number of channels
+ u32 GetNumChannels() const;
+
+private:
+ /// Current state of the stream
+ enum class State {
+ Stopped,
+ Playing,
+ };
+
+ /// Plays the next queued buffer in the audio stream, starting playback if necessary
+ void PlayNextBuffer();
+
+ /// Releases the actively playing buffer, signalling that it has been completed
+ void ReleaseActiveBuffer();
+
+ /// Gets the number of core cycles when the specified buffer will be released
+ s64 GetBufferReleaseCycles(const Buffer& buffer) const;
+
+ u32 sample_rate; ///< Sample rate of the stream
+ Format format; ///< Format of the stream
+ ReleaseCallback release_callback; ///< Buffer release callback for the stream
+ State state{State::Stopped}; ///< Playback state of the stream
+ CoreTiming::EventType* release_event{}; ///< Core timing release event for the stream
+ BufferPtr active_buffer; ///< Actively playing buffer in the stream
+ std::queue<BufferPtr> queued_buffers; ///< Buffers queued to be played in the stream
+ std::queue<BufferPtr> released_buffers; ///< Buffers recently released from the stream
+ SinkStream& sink_stream; ///< Output sink for the stream
+ std::string name; ///< Name of the stream, must be unique
+};
+
+using StreamPtr = std::shared_ptr<Stream>;
+
+} // namespace AudioCore
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