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-rw-r--r--src/audio_core/CMakeLists.txt17
-rw-r--r--src/audio_core/hle/common.h2
-rw-r--r--src/audio_core/hle/dsp.cpp81
-rw-r--r--src/audio_core/hle/dsp.h16
-rw-r--r--src/audio_core/hle/filter.h1
-rw-r--r--src/audio_core/hle/mixers.cpp201
-rw-r--r--src/audio_core/hle/mixers.h63
-rw-r--r--src/audio_core/hle/pipe.cpp9
-rw-r--r--src/audio_core/hle/pipe.h12
-rw-r--r--src/audio_core/hle/source.cpp320
-rw-r--r--src/audio_core/hle/source.h144
-rw-r--r--src/audio_core/sdl2_sink.cpp126
-rw-r--r--src/audio_core/sdl2_sink.h30
-rw-r--r--src/audio_core/sink.h2
-rw-r--r--src/audio_core/sink_details.cpp7
-rw-r--r--src/audio_core/time_stretch.cpp144
-rw-r--r--src/audio_core/time_stretch.h57
17 files changed, 1215 insertions, 17 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 5a2747e78..a72a907ef 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/src/audio_core/CMakeLists.txt
@@ -3,9 +3,12 @@ set(SRCS
codec.cpp
hle/dsp.cpp
hle/filter.cpp
+ hle/mixers.cpp
hle/pipe.cpp
+ hle/source.cpp
interpolate.cpp
sink_details.cpp
+ time_stretch.cpp
)
set(HEADERS
@@ -14,16 +17,30 @@ set(HEADERS
hle/common.h
hle/dsp.h
hle/filter.h
+ hle/mixers.h
hle/pipe.h
+ hle/source.h
interpolate.h
null_sink.h
sink.h
sink_details.h
+ time_stretch.h
)
include_directories(../../externals/soundtouch/include)
+if(SDL2_FOUND)
+ set(SRCS ${SRCS} sdl2_sink.cpp)
+ set(HEADERS ${HEADERS} sdl2_sink.h)
+ include_directories(${SDL2_INCLUDE_DIR})
+endif()
+
create_directory_groups(${SRCS} ${HEADERS})
add_library(audio_core STATIC ${SRCS} ${HEADERS})
target_link_libraries(audio_core SoundTouch)
+
+if(SDL2_FOUND)
+ target_link_libraries(audio_core ${SDL2_LIBRARY})
+ set_property(TARGET audio_core APPEND PROPERTY COMPILE_DEFINITIONS HAVE_SDL2)
+endif()
diff --git a/src/audio_core/hle/common.h b/src/audio_core/hle/common.h
index 7910f42ae..596b67eaf 100644
--- a/src/audio_core/hle/common.h
+++ b/src/audio_core/hle/common.h
@@ -27,7 +27,7 @@ using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
*/
template<typename FrameT, typename FilterT>
void FilterFrame(FrameT& frame, FilterT& filter) {
- std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const typename FrameT::value_type& sample) {
+ std::transform(frame.begin(), frame.end(), frame.begin(), [&filter](const auto& sample) {
return filter.ProcessSample(sample);
});
}
diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp
index 4d44bd2d9..0640e1eff 100644
--- a/src/audio_core/hle/dsp.cpp
+++ b/src/audio_core/hle/dsp.cpp
@@ -2,15 +2,21 @@
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
+#include <array>
#include <memory>
#include "audio_core/hle/dsp.h"
+#include "audio_core/hle/mixers.h"
#include "audio_core/hle/pipe.h"
+#include "audio_core/hle/source.h"
#include "audio_core/sink.h"
+#include "audio_core/time_stretch.h"
namespace DSP {
namespace HLE {
+// Region management
+
std::array<SharedMemory, 2> g_regions;
static size_t CurrentRegionIndex() {
@@ -38,21 +44,96 @@ static SharedMemory& WriteRegion() {
return g_regions[1 - CurrentRegionIndex()];
}
+// Audio processing and mixing
+
+static std::array<Source, num_sources> sources = {
+ Source(0), Source(1), Source(2), Source(3), Source(4), Source(5),
+ Source(6), Source(7), Source(8), Source(9), Source(10), Source(11),
+ Source(12), Source(13), Source(14), Source(15), Source(16), Source(17),
+ Source(18), Source(19), Source(20), Source(21), Source(22), Source(23)
+};
+static Mixers mixers;
+
+static StereoFrame16 GenerateCurrentFrame() {
+ SharedMemory& read = ReadRegion();
+ SharedMemory& write = WriteRegion();
+
+ std::array<QuadFrame32, 3> intermediate_mixes = {};
+
+ // Generate intermediate mixes
+ for (size_t i = 0; i < num_sources; i++) {
+ write.source_statuses.status[i] = sources[i].Tick(read.source_configurations.config[i], read.adpcm_coefficients.coeff[i]);
+ for (size_t mix = 0; mix < 3; mix++) {
+ sources[i].MixInto(intermediate_mixes[mix], mix);
+ }
+ }
+
+ // Generate final mix
+ write.dsp_status = mixers.Tick(read.dsp_configuration, read.intermediate_mix_samples, write.intermediate_mix_samples, intermediate_mixes);
+
+ StereoFrame16 output_frame = mixers.GetOutput();
+
+ // Write current output frame to the shared memory region
+ for (size_t samplei = 0; samplei < output_frame.size(); samplei++) {
+ for (size_t channeli = 0; channeli < output_frame[0].size(); channeli++) {
+ write.final_samples.pcm16[samplei][channeli] = s16_le(output_frame[samplei][channeli]);
+ }
+ }
+
+ return output_frame;
+}
+
+// Audio output
+
static std::unique_ptr<AudioCore::Sink> sink;
+static AudioCore::TimeStretcher time_stretcher;
+
+static void OutputCurrentFrame(const StereoFrame16& frame) {
+ time_stretcher.AddSamples(&frame[0][0], frame.size());
+ sink->EnqueueSamples(time_stretcher.Process(sink->SamplesInQueue()));
+}
+
+// Public Interface
void Init() {
DSP::HLE::ResetPipes();
+
+ for (auto& source : sources) {
+ source.Reset();
+ }
+
+ mixers.Reset();
+
+ time_stretcher.Reset();
+ if (sink) {
+ time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
+ }
}
void Shutdown() {
+ time_stretcher.Flush();
+ while (true) {
+ std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
+ if (residual_audio.empty())
+ break;
+ sink->EnqueueSamples(residual_audio);
+ }
}
bool Tick() {
+ StereoFrame16 current_frame = {};
+
+ // TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to shared memory region)
+ current_frame = GenerateCurrentFrame();
+
+ OutputCurrentFrame(current_frame);
+
return true;
}
void SetSink(std::unique_ptr<AudioCore::Sink> sink_) {
sink = std::move(sink_);
+ time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
}
} // namespace HLE
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h
index 4f2410c27..9275cd7de 100644
--- a/src/audio_core/hle/dsp.h
+++ b/src/audio_core/hle/dsp.h
@@ -33,13 +33,9 @@ namespace HLE {
// double-buffer. The frame counter is located as the very last u16 of each region and is incremented
// each audio tick.
-struct SharedMemory;
-
constexpr VAddr region0_base = 0x1FF50000;
constexpr VAddr region1_base = 0x1FF70000;
-extern std::array<SharedMemory, 2> g_regions;
-
/**
* The DSP is native 16-bit. The DSP also appears to be big-endian. When reading 32-bit numbers from
* its memory regions, the higher and lower 16-bit halves are swapped compared to the little-endian
@@ -169,9 +165,9 @@ struct SourceConfiguration {
float_le rate_multiplier;
enum class InterpolationMode : u8 {
- None = 0,
+ Polyphase = 0,
Linear = 1,
- Polyphase = 2
+ None = 2
};
InterpolationMode interpolation_mode;
@@ -318,10 +314,10 @@ ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
struct SourceStatus {
struct Status {
u8 is_enabled; ///< Is this channel enabled? (Doesn't have to be playing anything.)
- u8 previous_buffer_id_dirty; ///< Non-zero when previous_buffer_id changes
+ u8 current_buffer_id_dirty; ///< Non-zero when current_buffer_id changes
u16_le sync; ///< Is set by the DSP to the value of SourceConfiguration::sync
u32_dsp buffer_position; ///< Number of samples into the current buffer
- u16_le previous_buffer_id; ///< Updated when a buffer finishes playing
+ u16_le current_buffer_id; ///< Updated when a buffer finishes playing
INSERT_PADDING_DSPWORDS(1);
};
@@ -432,7 +428,7 @@ ASSERT_DSP_STRUCT(DspStatus, 32);
/// Final mixed output in PCM16 stereo format, what you hear out of the speakers.
/// When the application writes to this region it has no effect.
struct FinalMixSamples {
- s16_le pcm16[2 * samples_per_frame];
+ s16_le pcm16[samples_per_frame][2];
};
ASSERT_DSP_STRUCT(FinalMixSamples, 640);
@@ -507,6 +503,8 @@ struct SharedMemory {
};
ASSERT_DSP_STRUCT(SharedMemory, 0x8000);
+extern std::array<SharedMemory, 2> g_regions;
+
// Structures must have an offset that is a multiple of two.
static_assert(offsetof(SharedMemory, frame_counter) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0, "Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
diff --git a/src/audio_core/hle/filter.h b/src/audio_core/hle/filter.h
index 75738f600..43d2035cd 100644
--- a/src/audio_core/hle/filter.h
+++ b/src/audio_core/hle/filter.h
@@ -16,6 +16,7 @@ namespace HLE {
/// Preprocessing filters. There is an independent set of filters for each Source.
class SourceFilters final {
+public:
SourceFilters() { Reset(); }
/// Reset internal state.
diff --git a/src/audio_core/hle/mixers.cpp b/src/audio_core/hle/mixers.cpp
new file mode 100644
index 000000000..18335f7f0
--- /dev/null
+++ b/src/audio_core/hle/mixers.cpp
@@ -0,0 +1,201 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <cstddef>
+
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/dsp.h"
+#include "audio_core/hle/mixers.h"
+
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include "common/math_util.h"
+
+namespace DSP {
+namespace HLE {
+
+void Mixers::Reset() {
+ current_frame.fill({});
+ state = {};
+}
+
+DspStatus Mixers::Tick(DspConfiguration& config,
+ const IntermediateMixSamples& read_samples,
+ IntermediateMixSamples& write_samples,
+ const std::array<QuadFrame32, 3>& input)
+{
+ ParseConfig(config);
+
+ AuxReturn(read_samples);
+ AuxSend(write_samples, input);
+
+ MixCurrentFrame();
+
+ return GetCurrentStatus();
+}
+
+void Mixers::ParseConfig(DspConfiguration& config) {
+ if (!config.dirty_raw) {
+ return;
+ }
+
+ if (config.mixer1_enabled_dirty) {
+ config.mixer1_enabled_dirty.Assign(0);
+ state.mixer1_enabled = config.mixer1_enabled != 0;
+ LOG_TRACE(Audio_DSP, "mixers mixer1_enabled = %hu", config.mixer1_enabled);
+ }
+
+ if (config.mixer2_enabled_dirty) {
+ config.mixer2_enabled_dirty.Assign(0);
+ state.mixer2_enabled = config.mixer2_enabled != 0;
+ LOG_TRACE(Audio_DSP, "mixers mixer2_enabled = %hu", config.mixer2_enabled);
+ }
+
+ if (config.volume_0_dirty) {
+ config.volume_0_dirty.Assign(0);
+ state.intermediate_mixer_volume[0] = config.volume[0];
+ LOG_TRACE(Audio_DSP, "mixers volume[0] = %f", config.volume[0]);
+ }
+
+ if (config.volume_1_dirty) {
+ config.volume_1_dirty.Assign(0);
+ state.intermediate_mixer_volume[1] = config.volume[1];
+ LOG_TRACE(Audio_DSP, "mixers volume[1] = %f", config.volume[1]);
+ }
+
+ if (config.volume_2_dirty) {
+ config.volume_2_dirty.Assign(0);
+ state.intermediate_mixer_volume[2] = config.volume[2];
+ LOG_TRACE(Audio_DSP, "mixers volume[2] = %f", config.volume[2]);
+ }
+
+ if (config.output_format_dirty) {
+ config.output_format_dirty.Assign(0);
+ state.output_format = config.output_format;
+ LOG_TRACE(Audio_DSP, "mixers output_format = %zu", static_cast<size_t>(config.output_format));
+ }
+
+ if (config.headphones_connected_dirty) {
+ config.headphones_connected_dirty.Assign(0);
+ // Do nothing.
+ // (Note: Whether headphones are connected does affect coefficients used for surround sound.)
+ LOG_TRACE(Audio_DSP, "mixers headphones_connected=%hu", config.headphones_connected);
+ }
+
+ if (config.dirty_raw) {
+ LOG_DEBUG(Audio_DSP, "mixers remaining_dirty=%x", config.dirty_raw);
+ }
+
+ config.dirty_raw = 0;
+}
+
+static s16 ClampToS16(s32 value) {
+ return static_cast<s16>(MathUtil::Clamp(value, -32768, 32767));
+}
+
+static std::array<s16, 2> AddAndClampToS16(const std::array<s16, 2>& a, const std::array<s16, 2>& b) {
+ return {
+ ClampToS16(static_cast<s32>(a[0]) + static_cast<s32>(b[0])),
+ ClampToS16(static_cast<s32>(a[1]) + static_cast<s32>(b[1]))
+ };
+}
+
+void Mixers::DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples) {
+ // TODO(merry): Limiter. (Currently we're performing final mixing assuming a disabled limiter.)
+
+ switch (state.output_format) {
+ case OutputFormat::Mono:
+ std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
+ [gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> {
+ // Downmix to mono
+ s16 mono = ClampToS16(static_cast<s32>((gain * sample[0] + gain * sample[1] + gain * sample[2] + gain * sample[3]) / 2));
+ // Mix into current frame
+ return AddAndClampToS16(accumulator, { mono, mono });
+ });
+ return;
+
+ case OutputFormat::Surround:
+ // TODO(merry): Implement surround sound.
+ // fallthrough
+
+ case OutputFormat::Stereo:
+ std::transform(current_frame.begin(), current_frame.end(), samples.begin(), current_frame.begin(),
+ [gain](const std::array<s16, 2>& accumulator, const std::array<s32, 4>& sample) -> std::array<s16, 2> {
+ // Downmix to stereo
+ s16 left = ClampToS16(static_cast<s32>(gain * sample[0] + gain * sample[2]));
+ s16 right = ClampToS16(static_cast<s32>(gain * sample[1] + gain * sample[3]));
+ // Mix into current frame
+ return AddAndClampToS16(accumulator, { left, right });
+ });
+ return;
+ }
+
+ UNREACHABLE_MSG("Invalid output_format %zu", static_cast<size_t>(state.output_format));
+}
+
+void Mixers::AuxReturn(const IntermediateMixSamples& read_samples) {
+ // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32.
+
+ if (state.mixer1_enabled) {
+ for (size_t sample = 0; sample < samples_per_frame; sample++) {
+ for (size_t channel = 0; channel < 4; channel++) {
+ state.intermediate_mix_buffer[1][sample][channel] = read_samples.mix1.pcm32[channel][sample];
+ }
+ }
+ }
+
+ if (state.mixer2_enabled) {
+ for (size_t sample = 0; sample < samples_per_frame; sample++) {
+ for (size_t channel = 0; channel < 4; channel++) {
+ state.intermediate_mix_buffer[2][sample][channel] = read_samples.mix2.pcm32[channel][sample];
+ }
+ }
+ }
+}
+
+void Mixers::AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input) {
+ // NOTE: read_samples.mix{1,2}.pcm32 annoyingly have their dimensions in reverse order to QuadFrame32.
+
+ state.intermediate_mix_buffer[0] = input[0];
+
+ if (state.mixer1_enabled) {
+ for (size_t sample = 0; sample < samples_per_frame; sample++) {
+ for (size_t channel = 0; channel < 4; channel++) {
+ write_samples.mix1.pcm32[channel][sample] = input[1][sample][channel];
+ }
+ }
+ } else {
+ state.intermediate_mix_buffer[1] = input[1];
+ }
+
+ if (state.mixer2_enabled) {
+ for (size_t sample = 0; sample < samples_per_frame; sample++) {
+ for (size_t channel = 0; channel < 4; channel++) {
+ write_samples.mix2.pcm32[channel][sample] = input[2][sample][channel];
+ }
+ }
+ } else {
+ state.intermediate_mix_buffer[2] = input[2];
+ }
+}
+
+void Mixers::MixCurrentFrame() {
+ current_frame.fill({});
+
+ for (size_t mix = 0; mix < 3; mix++) {
+ DownmixAndMixIntoCurrentFrame(state.intermediate_mixer_volume[mix], state.intermediate_mix_buffer[mix]);
+ }
+
+ // TODO(merry): Compressor. (We currently assume a disabled compressor.)
+}
+
+DspStatus Mixers::GetCurrentStatus() const {
+ DspStatus status;
+ status.unknown = 0;
+ status.dropped_frames = 0;
+ return status;
+}
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/hle/mixers.h b/src/audio_core/hle/mixers.h
new file mode 100644
index 000000000..b52952eb5
--- /dev/null
+++ b/src/audio_core/hle/mixers.h
@@ -0,0 +1,63 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/dsp.h"
+
+namespace DSP {
+namespace HLE {
+
+class Mixers final {
+public:
+ Mixers() {
+ Reset();
+ }
+
+ void Reset();
+
+ DspStatus Tick(DspConfiguration& config,
+ const IntermediateMixSamples& read_samples,
+ IntermediateMixSamples& write_samples,
+ const std::array<QuadFrame32, 3>& input);
+
+ StereoFrame16 GetOutput() const {
+ return current_frame;
+ }
+
+private:
+ StereoFrame16 current_frame = {};
+
+ using OutputFormat = DspConfiguration::OutputFormat;
+
+ struct {
+ std::array<float, 3> intermediate_mixer_volume = {};
+
+ bool mixer1_enabled = false;
+ bool mixer2_enabled = false;
+ std::array<QuadFrame32, 3> intermediate_mix_buffer = {};
+
+ OutputFormat output_format = OutputFormat::Stereo;
+
+ } state;
+
+ /// INTERNAL: Update our internal state based on the current config.
+ void ParseConfig(DspConfiguration& config);
+ /// INTERNAL: Read samples from shared memory that have been modified by the ARM11.
+ void AuxReturn(const IntermediateMixSamples& read_samples);
+ /// INTERNAL: Write samples to shared memory for the ARM11 to modify.
+ void AuxSend(IntermediateMixSamples& write_samples, const std::array<QuadFrame32, 3>& input);
+ /// INTERNAL: Mix current_frame.
+ void MixCurrentFrame();
+ /// INTERNAL: Downmix from quadraphonic to stereo based on status.output_format and accumulate into current_frame.
+ void DownmixAndMixIntoCurrentFrame(float gain, const QuadFrame32& samples);
+ /// INTERNAL: Generate DspStatus based on internal state.
+ DspStatus GetCurrentStatus() const;
+};
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/hle/pipe.cpp b/src/audio_core/hle/pipe.cpp
index 03280780f..44dff1345 100644
--- a/src/audio_core/hle/pipe.cpp
+++ b/src/audio_core/hle/pipe.cpp
@@ -36,12 +36,17 @@ std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
return {};
}
+ if (length > UINT16_MAX) { // Can only read at most UINT16_MAX from the pipe
+ LOG_ERROR(Audio_DSP, "length of %u greater than max of %u", length, UINT16_MAX);
+ return {};
+ }
+
std::vector<u8>& data = pipe_data[pipe_index];
if (length > data.size()) {
LOG_WARNING(Audio_DSP, "pipe_number = %zu is out of data, application requested read of %u but %zu remain",
pipe_index, length, data.size());
- length = data.size();
+ length = static_cast<u32>(data.size());
}
if (length == 0)
@@ -94,7 +99,7 @@ static void AudioPipeWriteStructAddresses() {
};
// Begin with a u16 denoting the number of structs.
- WriteU16(DspPipe::Audio, struct_addresses.size());
+ WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size()));
// Then write the struct addresses.
for (u16 addr : struct_addresses) {
WriteU16(DspPipe::Audio, addr);
diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h
index 64d97f8ba..b714c0496 100644
--- a/src/audio_core/hle/pipe.h
+++ b/src/audio_core/hle/pipe.h
@@ -24,10 +24,14 @@ enum class DspPipe {
constexpr size_t NUM_DSP_PIPE = 8;
/**
- * Read a DSP pipe.
- * @param pipe_number The Pipe ID
- * @param length How much data to request.
- * @return The data read from the pipe. The size of this vector can be less than the length requested.
+ * Reads `length` bytes from the DSP pipe identified with `pipe_number`.
+ * @note Can read up to the maximum value of a u16 in bytes (65,535).
+ * @note IF an error is encoutered with either an invalid `pipe_number` or `length` value, an empty vector will be returned.
+ * @note IF `length` is set to 0, an empty vector will be returned.
+ * @note IF `length` is greater than the amount of data available, this function will only read the available amount.
+ * @param pipe_number a `DspPipe`
+ * @param length the number of bytes to read. The max is 65,535 (max of u16).
+ * @returns a vector of bytes from the specified pipe. On error, will be empty.
*/
std::vector<u8> PipeRead(DspPipe pipe_number, u32 length);
diff --git a/src/audio_core/hle/source.cpp b/src/audio_core/hle/source.cpp
new file mode 100644
index 000000000..30552fe26
--- /dev/null
+++ b/src/audio_core/hle/source.cpp
@@ -0,0 +1,320 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <array>
+
+#include "audio_core/codec.h"
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/source.h"
+#include "audio_core/interpolate.h"
+
+#include "common/assert.h"
+#include "common/logging/log.h"
+
+#include "core/memory.h"
+
+namespace DSP {
+namespace HLE {
+
+SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
+ ParseConfig(config, adpcm_coeffs);
+
+ if (state.enabled) {
+ GenerateFrame();
+ }
+
+ return GetCurrentStatus();
+}
+
+void Source::MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const {
+ if (!state.enabled)
+ return;
+
+ const std::array<float, 4>& gains = state.gain.at(intermediate_mix_id);
+ for (size_t samplei = 0; samplei < samples_per_frame; samplei++) {
+ // Conversion from stereo (current_frame) to quadraphonic (dest) occurs here.
+ dest[samplei][0] += static_cast<s32>(gains[0] * current_frame[samplei][0]);
+ dest[samplei][1] += static_cast<s32>(gains[1] * current_frame[samplei][1]);
+ dest[samplei][2] += static_cast<s32>(gains[2] * current_frame[samplei][0]);
+ dest[samplei][3] += static_cast<s32>(gains[3] * current_frame[samplei][1]);
+ }
+}
+
+void Source::Reset() {
+ current_frame.fill({});
+ state = {};
+}
+
+void Source::ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]) {
+ if (!config.dirty_raw) {
+ return;
+ }
+
+ if (config.reset_flag) {
+ config.reset_flag.Assign(0);
+ Reset();
+ LOG_TRACE(Audio_DSP, "source_id=%zu reset", source_id);
+ }
+
+ if (config.partial_reset_flag) {
+ config.partial_reset_flag.Assign(0);
+ state.input_queue = std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder>{};
+ LOG_TRACE(Audio_DSP, "source_id=%zu partial_reset", source_id);
+ }
+
+ if (config.enable_dirty) {
+ config.enable_dirty.Assign(0);
+ state.enabled = config.enable != 0;
+ LOG_TRACE(Audio_DSP, "source_id=%zu enable=%d", source_id, state.enabled);
+ }
+
+ if (config.sync_dirty) {
+ config.sync_dirty.Assign(0);
+ state.sync = config.sync;
+ LOG_TRACE(Audio_DSP, "source_id=%zu sync=%u", source_id, state.sync);
+ }
+
+ if (config.rate_multiplier_dirty) {
+ config.rate_multiplier_dirty.Assign(0);
+ state.rate_multiplier = config.rate_multiplier;
+ LOG_TRACE(Audio_DSP, "source_id=%zu rate=%f", source_id, state.rate_multiplier);
+
+ if (state.rate_multiplier <= 0) {
+ LOG_ERROR(Audio_DSP, "Was given an invalid rate multiplier: source_id=%zu rate=%f", source_id, state.rate_multiplier);
+ state.rate_multiplier = 1.0f;
+ // Note: Actual firmware starts producing garbage if this occurs.
+ }
+ }
+
+ if (config.adpcm_coefficients_dirty) {
+ config.adpcm_coefficients_dirty.Assign(0);
+ std::transform(adpcm_coeffs, adpcm_coeffs + state.adpcm_coeffs.size(), state.adpcm_coeffs.begin(),
+ [](const auto& coeff) { return static_cast<s16>(coeff); });
+ LOG_TRACE(Audio_DSP, "source_id=%zu adpcm update", source_id);
+ }
+
+ if (config.gain_0_dirty) {
+ config.gain_0_dirty.Assign(0);
+ std::transform(config.gain[0], config.gain[0] + state.gain[0].size(), state.gain[0].begin(),
+ [](const auto& coeff) { return static_cast<float>(coeff); });
+ LOG_TRACE(Audio_DSP, "source_id=%zu gain 0 update", source_id);
+ }
+
+ if (config.gain_1_dirty) {
+ config.gain_1_dirty.Assign(0);
+ std::transform(config.gain[1], config.gain[1] + state.gain[1].size(), state.gain[1].begin(),
+ [](const auto& coeff) { return static_cast<float>(coeff); });
+ LOG_TRACE(Audio_DSP, "source_id=%zu gain 1 update", source_id);
+ }
+
+ if (config.gain_2_dirty) {
+ config.gain_2_dirty.Assign(0);
+ std::transform(config.gain[2], config.gain[2] + state.gain[2].size(), state.gain[2].begin(),
+ [](const auto& coeff) { return static_cast<float>(coeff); });
+ LOG_TRACE(Audio_DSP, "source_id=%zu gain 2 update", source_id);
+ }
+
+ if (config.filters_enabled_dirty) {
+ config.filters_enabled_dirty.Assign(0);
+ state.filters.Enable(config.simple_filter_enabled.ToBool(), config.biquad_filter_enabled.ToBool());
+ LOG_TRACE(Audio_DSP, "source_id=%zu enable_simple=%hu enable_biquad=%hu",
+ source_id, config.simple_filter_enabled.Value(), config.biquad_filter_enabled.Value());
+ }
+
+ if (config.simple_filter_dirty) {
+ config.simple_filter_dirty.Assign(0);
+ state.filters.Configure(config.simple_filter);
+ LOG_TRACE(Audio_DSP, "source_id=%zu simple filter update", source_id);
+ }
+
+ if (config.biquad_filter_dirty) {
+ config.biquad_filter_dirty.Assign(0);
+ state.filters.Configure(config.biquad_filter);
+ LOG_TRACE(Audio_DSP, "source_id=%zu biquad filter update", source_id);
+ }
+
+ if (config.interpolation_dirty) {
+ config.interpolation_dirty.Assign(0);
+ state.interpolation_mode = config.interpolation_mode;
+ LOG_TRACE(Audio_DSP, "source_id=%zu interpolation_mode=%zu", source_id, static_cast<size_t>(state.interpolation_mode));
+ }
+
+ if (config.format_dirty || config.embedded_buffer_dirty) {
+ config.format_dirty.Assign(0);
+ state.format = config.format;
+ LOG_TRACE(Audio_DSP, "source_id=%zu format=%zu", source_id, static_cast<size_t>(state.format));
+ }
+
+ if (config.mono_or_stereo_dirty || config.embedded_buffer_dirty) {
+ config.mono_or_stereo_dirty.Assign(0);
+ state.mono_or_stereo = config.mono_or_stereo;
+ LOG_TRACE(Audio_DSP, "source_id=%zu mono_or_stereo=%zu", source_id, static_cast<size_t>(state.mono_or_stereo));
+ }
+
+ if (config.embedded_buffer_dirty) {
+ config.embedded_buffer_dirty.Assign(0);
+ state.input_queue.emplace(Buffer{
+ config.physical_address,
+ config.length,
+ static_cast<u8>(config.adpcm_ps),
+ { config.adpcm_yn[0], config.adpcm_yn[1] },
+ config.adpcm_dirty.ToBool(),
+ config.is_looping.ToBool(),
+ config.buffer_id,
+ state.mono_or_stereo,
+ state.format,
+ false
+ });
+ LOG_TRACE(Audio_DSP, "enqueuing embedded addr=0x%08x len=%u id=%hu", config.physical_address, config.length, config.buffer_id);
+ }
+
+ if (config.buffer_queue_dirty) {
+ config.buffer_queue_dirty.Assign(0);
+ for (size_t i = 0; i < 4; i++) {
+ if (config.buffers_dirty & (1 << i)) {
+ const auto& b = config.buffers[i];
+ state.input_queue.emplace(Buffer{
+ b.physical_address,
+ b.length,
+ static_cast<u8>(b.adpcm_ps),
+ { b.adpcm_yn[0], b.adpcm_yn[1] },
+ b.adpcm_dirty != 0,
+ b.is_looping != 0,
+ b.buffer_id,
+ state.mono_or_stereo,
+ state.format,
+ true
+ });
+ LOG_TRACE(Audio_DSP, "enqueuing queued %zu addr=0x%08x len=%u id=%hu", i, b.physical_address, b.length, b.buffer_id);
+ }
+ }
+ config.buffers_dirty = 0;
+ }
+
+ if (config.dirty_raw) {
+ LOG_DEBUG(Audio_DSP, "source_id=%zu remaining_dirty=%x", source_id, config.dirty_raw);
+ }
+
+ config.dirty_raw = 0;
+}
+
+void Source::GenerateFrame() {
+ current_frame.fill({});
+
+ if (state.current_buffer.empty() && !DequeueBuffer()) {
+ state.enabled = false;
+ state.buffer_update = true;
+ state.current_buffer_id = 0;
+ return;
+ }
+
+ size_t frame_position = 0;
+
+ state.current_sample_number = state.next_sample_number;
+ while (frame_position < current_frame.size()) {
+ if (state.current_buffer.empty() && !DequeueBuffer()) {
+ break;
+ }
+
+ const size_t size_to_copy = std::min(state.current_buffer.size(), current_frame.size() - frame_position);
+
+ std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy, current_frame.begin() + frame_position);
+ state.current_buffer.erase(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy);
+
+ frame_position += size_to_copy;
+ state.next_sample_number += static_cast<u32>(size_to_copy);
+ }
+
+ state.filters.ProcessFrame(current_frame);
+}
+
+
+bool Source::DequeueBuffer() {
+ ASSERT_MSG(state.current_buffer.empty(), "Shouldn't dequeue; we still have data in current_buffer");
+
+ if (state.input_queue.empty())
+ return false;
+
+ const Buffer buf = state.input_queue.top();
+ state.input_queue.pop();
+
+ if (buf.adpcm_dirty) {
+ state.adpcm_state.yn1 = buf.adpcm_yn[0];
+ state.adpcm_state.yn2 = buf.adpcm_yn[1];
+ }
+
+ if (buf.is_looping) {
+ LOG_ERROR(Audio_DSP, "Looped buffers are unimplemented at the moment");
+ }
+
+ const u8* const memory = Memory::GetPhysicalPointer(buf.physical_address);
+ if (memory) {
+ const unsigned num_channels = buf.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1;
+ switch (buf.format) {
+ case Format::PCM8:
+ state.current_buffer = Codec::DecodePCM8(num_channels, memory, buf.length);
+ break;
+ case Format::PCM16:
+ state.current_buffer = Codec::DecodePCM16(num_channels, memory, buf.length);
+ break;
+ case Format::ADPCM:
+ DEBUG_ASSERT(num_channels == 1);
+ state.current_buffer = Codec::DecodeADPCM(memory, buf.length, state.adpcm_coeffs, state.adpcm_state);
+ break;
+ default:
+ UNIMPLEMENTED();
+ break;
+ }
+ } else {
+ LOG_WARNING(Audio_DSP, "source_id=%zu buffer_id=%hu length=%u: Invalid physical address 0x%08X",
+ source_id, buf.buffer_id, buf.length, buf.physical_address);
+ state.current_buffer.clear();
+ return true;
+ }
+
+ switch (state.interpolation_mode) {
+ case InterpolationMode::None:
+ state.current_buffer = AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
+ break;
+ case InterpolationMode::Linear:
+ state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
+ break;
+ case InterpolationMode::Polyphase:
+ // TODO(merry): Implement polyphase interpolation
+ state.current_buffer = AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
+ break;
+ default:
+ UNIMPLEMENTED();
+ break;
+ }
+
+ state.current_sample_number = 0;
+ state.next_sample_number = 0;
+ state.current_buffer_id = buf.buffer_id;
+ state.buffer_update = buf.from_queue;
+
+ LOG_TRACE(Audio_DSP, "source_id=%zu buffer_id=%hu from_queue=%s current_buffer.size()=%zu",
+ source_id, buf.buffer_id, buf.from_queue ? "true" : "false", state.current_buffer.size());
+ return true;
+}
+
+SourceStatus::Status Source::GetCurrentStatus() {
+ SourceStatus::Status ret;
+
+ // Applications depend on the correct emulation of
+ // current_buffer_id_dirty and current_buffer_id to synchronise
+ // audio with video.
+ ret.is_enabled = state.enabled;
+ ret.current_buffer_id_dirty = state.buffer_update ? 1 : 0;
+ state.buffer_update = false;
+ ret.current_buffer_id = state.current_buffer_id;
+ ret.buffer_position = state.current_sample_number;
+ ret.sync = state.sync;
+
+ return ret;
+}
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/hle/source.h b/src/audio_core/hle/source.h
new file mode 100644
index 000000000..7ee08d424
--- /dev/null
+++ b/src/audio_core/hle/source.h
@@ -0,0 +1,144 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <array>
+#include <queue>
+#include <vector>
+
+#include "audio_core/codec.h"
+#include "audio_core/hle/common.h"
+#include "audio_core/hle/dsp.h"
+#include "audio_core/hle/filter.h"
+#include "audio_core/interpolate.h"
+
+#include "common/common_types.h"
+
+namespace DSP {
+namespace HLE {
+
+/**
+ * This module performs:
+ * - Buffer management
+ * - Decoding of buffers
+ * - Buffer resampling and interpolation
+ * - Per-source filtering (SimpleFilter, BiquadFilter)
+ * - Per-source gain
+ * - Other per-source processing
+ */
+class Source final {
+public:
+ explicit Source(size_t source_id_) : source_id(source_id_) {
+ Reset();
+ }
+
+ /// Resets internal state.
+ void Reset();
+
+ /**
+ * This is called once every audio frame. This performs per-source processing every frame.
+ * @param config The new configuration we've got for this Source from the application.
+ * @param adpcm_coeffs ADPCM coefficients to use if config tells us to use them (may contain invalid values otherwise).
+ * @return The current status of this Source. This is given back to the emulated application via SharedMemory.
+ */
+ SourceStatus::Status Tick(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
+
+ /**
+ * Mix this source's output into dest, using the gains for the `intermediate_mix_id`-th intermediate mixer.
+ * @param dest The QuadFrame32 to mix into.
+ * @param intermediate_mix_id The id of the intermediate mix whose gains we are using.
+ */
+ void MixInto(QuadFrame32& dest, size_t intermediate_mix_id) const;
+
+private:
+ const size_t source_id;
+ StereoFrame16 current_frame;
+
+ using Format = SourceConfiguration::Configuration::Format;
+ using InterpolationMode = SourceConfiguration::Configuration::InterpolationMode;
+ using MonoOrStereo = SourceConfiguration::Configuration::MonoOrStereo;
+
+ /// Internal representation of a buffer for our buffer queue
+ struct Buffer {
+ PAddr physical_address;
+ u32 length;
+ u8 adpcm_ps;
+ std::array<u16, 2> adpcm_yn;
+ bool adpcm_dirty;
+ bool is_looping;
+ u16 buffer_id;
+
+ MonoOrStereo mono_or_stereo;
+ Format format;
+
+ bool from_queue;
+ };
+
+ struct BufferOrder {
+ bool operator() (const Buffer& a, const Buffer& b) const {
+ // Lower buffer_id comes first.
+ return a.buffer_id > b.buffer_id;
+ }
+ };
+
+ struct {
+
+ // State variables
+
+ bool enabled = false;
+ u16 sync = 0;
+
+ // Mixing
+
+ std::array<std::array<float, 4>, 3> gain = {};
+
+ // Buffer queue
+
+ std::priority_queue<Buffer, std::vector<Buffer>, BufferOrder> input_queue;
+ MonoOrStereo mono_or_stereo = MonoOrStereo::Mono;
+ Format format = Format::ADPCM;
+
+ // Current buffer
+
+ u32 current_sample_number = 0;
+ u32 next_sample_number = 0;
+ std::vector<std::array<s16, 2>> current_buffer;
+
+ // buffer_id state
+
+ bool buffer_update = false;
+ u32 current_buffer_id = 0;
+
+ // Decoding state
+
+ std::array<s16, 16> adpcm_coeffs = {};
+ Codec::ADPCMState adpcm_state = {};
+
+ // Resampling state
+
+ float rate_multiplier = 1.0;
+ InterpolationMode interpolation_mode = InterpolationMode::Polyphase;
+ AudioInterp::State interp_state = {};
+
+ // Filter state
+
+ SourceFilters filters;
+
+ } state;
+
+ // Internal functions
+
+ /// INTERNAL: Update our internal state based on the current config.
+ void ParseConfig(SourceConfiguration::Configuration& config, const s16_le (&adpcm_coeffs)[16]);
+ /// INTERNAL: Generate the current audio output for this frame based on our internal state.
+ void GenerateFrame();
+ /// INTERNAL: Dequeues a buffer and does preprocessing on it (decoding, resampling). Puts it into current_buffer.
+ bool DequeueBuffer();
+ /// INTERNAL: Generates a SourceStatus::Status based on our internal state.
+ SourceStatus::Status GetCurrentStatus();
+};
+
+} // namespace HLE
+} // namespace DSP
diff --git a/src/audio_core/sdl2_sink.cpp b/src/audio_core/sdl2_sink.cpp
new file mode 100644
index 000000000..dc75c04ee
--- /dev/null
+++ b/src/audio_core/sdl2_sink.cpp
@@ -0,0 +1,126 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <list>
+#include <vector>
+
+#include <SDL.h>
+
+#include "audio_core/audio_core.h"
+#include "audio_core/sdl2_sink.h"
+
+#include "common/assert.h"
+#include "common/logging/log.h"
+#include <numeric>
+
+namespace AudioCore {
+
+struct SDL2Sink::Impl {
+ unsigned int sample_rate = 0;
+
+ SDL_AudioDeviceID audio_device_id = 0;
+
+ std::list<std::vector<s16>> queue;
+
+ static void Callback(void* impl_, u8* buffer, int buffer_size_in_bytes);
+};
+
+SDL2Sink::SDL2Sink() : impl(std::make_unique<Impl>()) {
+ if (SDL_Init(SDL_INIT_AUDIO) < 0) {
+ LOG_CRITICAL(Audio_Sink, "SDL_Init(SDL_INIT_AUDIO) failed");
+ impl->audio_device_id = 0;
+ return;
+ }
+
+ SDL_AudioSpec desired_audiospec;
+ SDL_zero(desired_audiospec);
+ desired_audiospec.format = AUDIO_S16;
+ desired_audiospec.channels = 2;
+ desired_audiospec.freq = native_sample_rate;
+ desired_audiospec.samples = 1024;
+ desired_audiospec.userdata = impl.get();
+ desired_audiospec.callback = &Impl::Callback;
+
+ SDL_AudioSpec obtained_audiospec;
+ SDL_zero(obtained_audiospec);
+
+ impl->audio_device_id = SDL_OpenAudioDevice(nullptr, false, &desired_audiospec, &obtained_audiospec, 0);
+ if (impl->audio_device_id <= 0) {
+ LOG_CRITICAL(Audio_Sink, "SDL_OpenAudioDevice failed");
+ return;
+ }
+
+ impl->sample_rate = obtained_audiospec.freq;
+
+ // SDL2 audio devices start out paused, unpause it:
+ SDL_PauseAudioDevice(impl->audio_device_id, 0);
+}
+
+SDL2Sink::~SDL2Sink() {
+ if (impl->audio_device_id <= 0)
+ return;
+
+ SDL_CloseAudioDevice(impl->audio_device_id);
+}
+
+unsigned int SDL2Sink::GetNativeSampleRate() const {
+ if (impl->audio_device_id <= 0)
+ return native_sample_rate;
+
+ return impl->sample_rate;
+}
+
+void SDL2Sink::EnqueueSamples(const std::vector<s16>& samples) {
+ if (impl->audio_device_id <= 0)
+ return;
+
+ ASSERT_MSG(samples.size() % 2 == 0, "Samples must be in interleaved stereo PCM16 format (size must be a multiple of two)");
+
+ SDL_LockAudioDevice(impl->audio_device_id);
+ impl->queue.emplace_back(samples);
+ SDL_UnlockAudioDevice(impl->audio_device_id);
+}
+
+size_t SDL2Sink::SamplesInQueue() const {
+ if (impl->audio_device_id <= 0)
+ return 0;
+
+ SDL_LockAudioDevice(impl->audio_device_id);
+
+ size_t total_size = std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<size_t>(0),
+ [](size_t sum, const auto& buffer) {
+ // Division by two because each stereo sample is made of two s16.
+ return sum + buffer.size() / 2;
+ });
+
+ SDL_UnlockAudioDevice(impl->audio_device_id);
+
+ return total_size;
+}
+
+void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) {
+ Impl* impl = reinterpret_cast<Impl*>(impl_);
+
+ size_t remaining_size = static_cast<size_t>(buffer_size_in_bytes) / sizeof(s16); // Keep track of size in 16-bit increments.
+
+ while (remaining_size > 0 && !impl->queue.empty()) {
+ if (impl->queue.front().size() <= remaining_size) {
+ memcpy(buffer, impl->queue.front().data(), impl->queue.front().size() * sizeof(s16));
+ buffer += impl->queue.front().size() * sizeof(s16);
+ remaining_size -= impl->queue.front().size();
+ impl->queue.pop_front();
+ } else {
+ memcpy(buffer, impl->queue.front().data(), remaining_size * sizeof(s16));
+ buffer += remaining_size * sizeof(s16);
+ impl->queue.front().erase(impl->queue.front().begin(), impl->queue.front().begin() + remaining_size);
+ remaining_size = 0;
+ }
+ }
+
+ if (remaining_size > 0) {
+ memset(buffer, 0, remaining_size * sizeof(s16));
+ }
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/sdl2_sink.h b/src/audio_core/sdl2_sink.h
new file mode 100644
index 000000000..0f296b673
--- /dev/null
+++ b/src/audio_core/sdl2_sink.h
@@ -0,0 +1,30 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#pragma once
+
+#include <cstddef>
+#include <memory>
+
+#include "audio_core/sink.h"
+
+namespace AudioCore {
+
+class SDL2Sink final : public Sink {
+public:
+ SDL2Sink();
+ ~SDL2Sink() override;
+
+ unsigned int GetNativeSampleRate() const override;
+
+ void EnqueueSamples(const std::vector<s16>& samples) override;
+
+ size_t SamplesInQueue() const override;
+
+private:
+ struct Impl;
+ std::unique_ptr<Impl> impl;
+};
+
+} // namespace AudioCore
diff --git a/src/audio_core/sink.h b/src/audio_core/sink.h
index cad21a85e..1c881c3d2 100644
--- a/src/audio_core/sink.h
+++ b/src/audio_core/sink.h
@@ -19,7 +19,7 @@ public:
virtual ~Sink() = default;
/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units: samples/sec)
- virtual unsigned GetNativeSampleRate() const = 0;
+ virtual unsigned int GetNativeSampleRate() const = 0;
/**
* Feed stereo samples to sink.
diff --git a/src/audio_core/sink_details.cpp b/src/audio_core/sink_details.cpp
index d2cc74103..ba5e83d17 100644
--- a/src/audio_core/sink_details.cpp
+++ b/src/audio_core/sink_details.cpp
@@ -8,10 +8,17 @@
#include "audio_core/null_sink.h"
#include "audio_core/sink_details.h"
+#ifdef HAVE_SDL2
+#include "audio_core/sdl2_sink.h"
+#endif
+
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_SDL2
+ { "sdl2", []() { return std::make_unique<SDL2Sink>(); } },
+#endif
{ "null", []() { return std::make_unique<NullSink>(); } },
};
diff --git a/src/audio_core/time_stretch.cpp b/src/audio_core/time_stretch.cpp
new file mode 100644
index 000000000..ea38f40d0
--- /dev/null
+++ b/src/audio_core/time_stretch.cpp
@@ -0,0 +1,144 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <chrono>
+#include <cmath>
+#include <vector>
+
+#include <SoundTouch.h>
+
+#include "audio_core/audio_core.h"
+#include "audio_core/time_stretch.h"
+
+#include "common/common_types.h"
+#include "common/logging/log.h"
+#include "common/math_util.h"
+
+using steady_clock = std::chrono::steady_clock;
+
+namespace AudioCore {
+
+constexpr double MIN_RATIO = 0.1;
+constexpr double MAX_RATIO = 100.0;
+
+static double ClampRatio(double ratio) {
+ return MathUtil::Clamp(ratio, MIN_RATIO, MAX_RATIO);
+}
+
+constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
+constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
+constexpr size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples
+
+constexpr double SMOOTHING_FACTOR = 0.007;
+
+struct TimeStretcher::Impl {
+ soundtouch::SoundTouch soundtouch;
+
+ steady_clock::time_point frame_timer = steady_clock::now();
+ size_t samples_queued = 0;
+
+ double smoothed_ratio = 1.0;
+
+ double sample_rate = static_cast<double>(native_sample_rate);
+};
+
+std::vector<s16> TimeStretcher::Process(size_t samples_in_queue) {
+ // This is a very simple algorithm without any fancy control theory. It works and is stable.
+
+ double ratio = CalculateCurrentRatio();
+ ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue);
+ impl->smoothed_ratio = (1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
+ impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio);
+
+ // SoundTouch's tempo definition the inverse of our ratio definition.
+ impl->soundtouch.setTempo(1.0 / impl->smoothed_ratio);
+
+ std::vector<s16> samples = GetSamples();
+ if (samples_in_queue >= DROP_FRAMES_SAMPLE_DELAY) {
+ samples.clear();
+ LOG_DEBUG(Audio, "Dropping frames!");
+ }
+ return samples;
+}
+
+TimeStretcher::TimeStretcher() : impl(std::make_unique<Impl>()) {
+ impl->soundtouch.setPitch(1.0);
+ impl->soundtouch.setChannels(2);
+ impl->soundtouch.setSampleRate(native_sample_rate);
+ Reset();
+}
+
+TimeStretcher::~TimeStretcher() {
+ impl->soundtouch.clear();
+}
+
+void TimeStretcher::SetOutputSampleRate(unsigned int sample_rate) {
+ impl->sample_rate = static_cast<double>(sample_rate);
+ impl->soundtouch.setRate(static_cast<double>(native_sample_rate) / impl->sample_rate);
+}
+
+void TimeStretcher::AddSamples(const s16* buffer, size_t num_samples) {
+ impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples));
+ impl->samples_queued += num_samples;
+}
+
+void TimeStretcher::Flush() {
+ impl->soundtouch.flush();
+}
+
+void TimeStretcher::Reset() {
+ impl->soundtouch.setTempo(1.0);
+ impl->soundtouch.clear();
+ impl->smoothed_ratio = 1.0;
+ impl->frame_timer = steady_clock::now();
+ impl->samples_queued = 0;
+ SetOutputSampleRate(native_sample_rate);
+}
+
+double TimeStretcher::CalculateCurrentRatio() {
+ const steady_clock::time_point now = steady_clock::now();
+ const std::chrono::duration<double> duration = now - impl->frame_timer;
+
+ const double expected_time = static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
+ const double actual_time = duration.count();
+
+ double ratio;
+ if (expected_time != 0) {
+ ratio = ClampRatio(actual_time / expected_time);
+ } else {
+ ratio = impl->smoothed_ratio;
+ }
+
+ impl->frame_timer = now;
+ impl->samples_queued = 0;
+
+ return ratio;
+}
+
+double TimeStretcher::CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const {
+ const size_t min_sample_delay = static_cast<size_t>(MIN_DELAY_TIME * impl->sample_rate);
+ const size_t max_sample_delay = static_cast<size_t>(MAX_DELAY_TIME * impl->sample_rate);
+
+ if (sample_delay < min_sample_delay) {
+ // Make the ratio bigger.
+ ratio = ratio > 1.0 ? ratio * ratio : sqrt(ratio);
+ } else if (sample_delay > max_sample_delay) {
+ // Make the ratio smaller.
+ ratio = ratio > 1.0 ? sqrt(ratio) : ratio * ratio;
+ }
+
+ return ClampRatio(ratio);
+}
+
+std::vector<s16> TimeStretcher::GetSamples() {
+ uint available = impl->soundtouch.numSamples();
+
+ std::vector<s16> output(static_cast<size_t>(available) * 2);
+
+ impl->soundtouch.receiveSamples(output.data(), available);
+
+ return output;
+}
+
+} // namespace AudioCore
diff --git a/src/audio_core/time_stretch.h b/src/audio_core/time_stretch.h
new file mode 100644
index 000000000..1fde3f72a
--- /dev/null
+++ b/src/audio_core/time_stretch.h
@@ -0,0 +1,57 @@
+// Copyright 2016 Citra Emulator Project
+// Licensed under GPLv2 or any later version
+// Refer to the license.txt file included.
+
+#include <cstddef>
+#include <memory>
+#include <vector>
+
+#include "common/common_types.h"
+
+namespace AudioCore {
+
+class TimeStretcher final {
+public:
+ TimeStretcher();
+ ~TimeStretcher();
+
+ /**
+ * Set sample rate for the samples that Process returns.
+ * @param sample_rate The sample rate.
+ */
+ void SetOutputSampleRate(unsigned int sample_rate);
+
+ /**
+ * Add samples to be processed.
+ * @param sample_buffer Buffer of samples in interleaved stereo PCM16 format.
+ * @param num_sample Number of samples.
+ */
+ void AddSamples(const s16* sample_buffer, size_t num_samples);
+
+ /// Flush audio remaining in internal buffers.
+ void Flush();
+
+ /// Resets internal state and clears buffers.
+ void Reset();
+
+ /**
+ * Does audio stretching and produces the time-stretched samples.
+ * Timer calculations use sample_delay to determine how much of a margin we have.
+ * @param sample_delay How many samples are buffered downstream of this module and haven't been played yet.
+ * @return Samples to play in interleaved stereo PCM16 format.
+ */
+ std::vector<s16> Process(size_t sample_delay);
+
+private:
+ struct Impl;
+ std::unique_ptr<Impl> impl;
+
+ /// INTERNAL: ratio = wallclock time / emulated time
+ double CalculateCurrentRatio();
+ /// INTERNAL: If we have too many or too few samples downstream, nudge ratio in the appropriate direction.
+ double CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const;
+ /// INTERNAL: Gets the time-stretched samples from SoundTouch.
+ std::vector<s16> GetSamples();
+};
+
+} // namespace AudioCore
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-09 14:54:06 +0100