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audio_core: Remove global state

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This commit is contained in:
MerryMage 2017-12-20 18:44:32 +00:00
parent dca5fd291f
commit ab3d53131a
34 changed files with 711 additions and 650 deletions
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12
src/audio_core/CMakeLists.txt
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@ -1,17 +1,17 @@
add_library(audio_core STATIC
audio_core.cpp
audio_core.h
audio_types.h
codec.cpp
codec.h
dsp_interface.cpp
dsp_interface.h
hle/common.h
hle/dsp.cpp
hle/dsp.h
hle/filter.cpp
hle/filter.h
hle/hle.cpp
hle/hle.h
hle/mixers.cpp
hle/mixers.h
hle/pipe.cpp
hle/pipe.h
hle/shared_memory.h
hle/source.cpp
hle/source.h
interpolate.cpp

61
src/audio_core/audio_core.cpp
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@ -1,61 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <array>
#include <memory>
#include <string>
#include "audio_core/audio_core.h"
#include "audio_core/hle/dsp.h"
#include "audio_core/hle/pipe.h"
#include "audio_core/null_sink.h"
#include "audio_core/sink.h"
#include "audio_core/sink_details.h"
#include "common/common_types.h"
#include "core/core_timing.h"
#include "core/hle/service/dsp_dsp.h"
namespace AudioCore {
// Audio Ticks occur about every 5 miliseconds.
static CoreTiming::EventType* tick_event; ///< CoreTiming event
static constexpr u64 audio_frame_ticks = 1310252ull; ///< Units: ARM11 cycles
static void AudioTickCallback(u64 /*userdata*/, int cycles_late) {
if (DSP::HLE::Tick()) {
// TODO(merry): Signal all the other interrupts as appropriate.
Service::DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Audio);
// HACK(merry): Added to prevent regressions. Will remove soon.
Service::DSP_DSP::SignalPipeInterrupt(DSP::HLE::DspPipe::Binary);
}
// Reschedule recurrent event
CoreTiming::ScheduleEvent(audio_frame_ticks - cycles_late, tick_event);
}
void Init() {
DSP::HLE::Init();
tick_event = CoreTiming::RegisterEvent("AudioCore::tick_event", AudioTickCallback);
CoreTiming::ScheduleEvent(audio_frame_ticks, tick_event);
}
std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory() {
return DSP::HLE::g_dsp_memory.raw_memory;
}
void SelectSink(std::string sink_id) {
const SinkDetails& sink_details = GetSinkDetails(sink_id);
DSP::HLE::SetSink(sink_details.factory());
}
void EnableStretching(bool enable) {
DSP::HLE::EnableStretching(enable);
}
void Shutdown() {
CoreTiming::UnscheduleEvent(tick_event, 0);
DSP::HLE::Shutdown();
}
} // namespace AudioCore

31
src/audio_core/audio_core.h
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@ -1,31 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <array>
#include <string>
#include "common/common_types.h"
#include "core/memory.h"
namespace AudioCore {
constexpr int native_sample_rate = 32728; ///< 32kHz
/// Initialise Audio Core
void Init();
/// Returns a reference to the array backing DSP memory
std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory();
/// Select the sink to use based on sink id.
void SelectSink(std::string sink_id);
/// Enable/Disable stretching.
void EnableStretching(bool enable);
/// Shutdown Audio Core
void Shutdown();
} // namespace AudioCore

43
src/audio_core/audio_types.h Normal file
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@ -0,0 +1,43 @@
// Copyright 2017 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <array>
#include <cstddef>
#include <deque>
#include "common/common_types.h"
namespace AudioCore {
/// Samples per second which the 3DS's audio hardware natively outputs at
constexpr int native_sample_rate = 32728; // Hz
/// Samples per audio frame at native sample rate
constexpr int samples_per_frame = 160;
/// The final output to the speakers is stereo. Preprocessing output in Source is also stereo.
using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>;
/// The DSP is quadraphonic internally.
using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
/// A variable length buffer of signed PCM16 stereo samples.
using StereoBuffer16 = std::deque<std::array<s16, 2>>;
constexpr size_t num_dsp_pipe = 8;
enum class DspPipe {
Debug = 0,
Dma = 1,
Audio = 2,
Binary = 3,
};
enum class DspState {
Off,
On,
Sleeping,
};
} // namespace AudioCore

6
src/audio_core/codec.cpp
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@ -5,12 +5,13 @@
#include <array>
#include <cstddef>
#include <cstring>
#include <vector>
#include "audio_core/audio_types.h"
#include "audio_core/codec.h"
#include "common/assert.h"
#include "common/common_types.h"
#include "common/math_util.h"
namespace AudioCore {
namespace Codec {
StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count,
@ -124,4 +125,5 @@ StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
return ret;
}
};
} // namespace Codec
} // namespace AudioCore

9
src/audio_core/codec.h
View file

@ -5,14 +5,12 @@
#pragma once
#include <array>
#include <deque>
#include "audio_core/audio_types.h"
#include "common/common_types.h"
namespace AudioCore {
namespace Codec {
/// A variable length buffer of signed PCM16 stereo samples.
using StereoBuffer16 = std::deque<std::array<s16, 2>>;
/// See: Codec::DecodeADPCM
struct ADPCMState {
// Two historical samples from previous processed buffer,
@ -48,4 +46,5 @@ StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data,
*/
StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
const size_t sample_count);
};
} // namespace Codec
} // namespace AudioCore

75
src/audio_core/dsp_interface.cpp Normal file
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@ -0,0 +1,75 @@
// Copyright 2017 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <cstddef>
#include "audio_core/dsp_interface.h"
#include "audio_core/sink.h"
#include "audio_core/sink_details.h"
#include "common/assert.h"
namespace AudioCore {
DspInterface::DspInterface() = default;
DspInterface::~DspInterface() {
if (perform_time_stretching) {
FlushResidualStretcherAudio();
}
}
void DspInterface::SetSink(const std::string& sink_id) {
const SinkDetails& sink_details = GetSinkDetails(sink_id);
sink = sink_details.factory();
time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
}
Sink& DspInterface::GetSink() {
ASSERT(sink);
return *sink.get();
}
void DspInterface::EnableStretching(bool enable) {
if (perform_time_stretching == enable)
return;
if (!enable) {
FlushResidualStretcherAudio();
}
perform_time_stretching = enable;
}
void DspInterface::OutputFrame(const StereoFrame16& frame) {
if (!sink)
return;
if (perform_time_stretching) {
time_stretcher.AddSamples(&frame[0][0], frame.size());
std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue());
sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2);
} else {
constexpr size_t maximum_sample_latency = 2048; // about 64 miliseconds
if (sink->SamplesInQueue() > maximum_sample_latency) {
// This can occur if we're running too fast and samples are starting to back up.
// Just drop the samples.
return;
}
sink->EnqueueSamples(&frame[0][0], frame.size());
}
}
void DspInterface::FlushResidualStretcherAudio() {
if (!sink)
return;
time_stretcher.Flush();
while (true) {
std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
if (residual_audio.empty())
break;
sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
}
}
} // namespace AudioCore

81
src/audio_core/dsp_interface.h Normal file
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@ -0,0 +1,81 @@
// Copyright 2017 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <memory>
#include <vector>
#include "audio_core/audio_types.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "core/memory.h"
namespace AudioCore {
class Sink;
class DspInterface {
public:
DspInterface();
virtual ~DspInterface();
DspInterface(const DspInterface&) = delete;
DspInterface(DspInterface&&) = delete;
DspInterface& operator=(const DspInterface&) = delete;
DspInterface& operator=(DspInterface&&) = delete;
/// Get the state of the DSP
virtual DspState GetDspState() const = 0;
/**
* 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.
*/
virtual std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) = 0;
/**
* How much data is left in pipe
* @param pipe_number The Pipe ID
* @return The amount of data remaning in the pipe. This is the maximum length PipeRead will
* return.
*/
virtual size_t GetPipeReadableSize(DspPipe pipe_number) const = 0;
/**
* Write to a DSP pipe.
* @param pipe_number The Pipe ID
* @param buffer The data to write to the pipe.
*/
virtual void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) = 0;
/// Returns a reference to the array backing DSP memory
virtual std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory() = 0;
/// Select the sink to use based on sink id.
void SetSink(const std::string& sink_id);
/// Get the current sink
Sink& GetSink();
/// Enable/Disable audio stretching.
void EnableStretching(bool enable);
protected:
void OutputFrame(const StereoFrame16& frame);
private:
void FlushResidualStretcherAudio();
std::unique_ptr<Sink> sink;
bool perform_time_stretching = false;
TimeStretcher time_stretcher;
};
} // namespace AudioCore

16
src/audio_core/hle/common.h
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@ -5,20 +5,12 @@
#pragma once
#include <algorithm>
#include <array>
#include "common/common_types.h"
#include <cstddef>
namespace DSP {
namespace AudioCore {
namespace HLE {
constexpr int num_sources = 24;
constexpr int samples_per_frame = 160; ///< Samples per audio frame at native sample rate
/// The final output to the speakers is stereo. Preprocessing output in Source is also stereo.
using StereoFrame16 = std::array<std::array<s16, 2>, samples_per_frame>;
/// The DSP is quadraphonic internally.
using QuadFrame32 = std::array<std::array<s32, 4>, samples_per_frame>;
constexpr size_t num_sources = 24;
/**
* This performs the filter operation defined by FilterT::ProcessSample on the frame in-place.
@ -31,4 +23,4 @@ void FilterFrame(FrameT& frame, FilterT& filter) {
}
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

172
src/audio_core/hle/dsp.cpp
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@ -1,172 +0,0 @@
// Copyright 2016 Citra Emulator Project
// 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
DspMemory g_dsp_memory;
static size_t CurrentRegionIndex() {
// The region with the higher frame counter is chosen unless there is wraparound.
// This function only returns a 0 or 1.
u16 frame_counter_0 = g_dsp_memory.region_0.frame_counter;
u16 frame_counter_1 = g_dsp_memory.region_1.frame_counter;
if (frame_counter_0 == 0xFFFFu && frame_counter_1 != 0xFFFEu) {
// Wraparound has occurred.
return 1;
}
if (frame_counter_1 == 0xFFFFu && frame_counter_0 != 0xFFFEu) {
// Wraparound has occurred.
return 0;
}
return (frame_counter_0 > frame_counter_1) ? 0 : 1;
}
static SharedMemory& ReadRegion() {
return CurrentRegionIndex() == 0 ? g_dsp_memory.region_0 : g_dsp_memory.region_1;
}
static SharedMemory& WriteRegion() {
return CurrentRegionIndex() != 0 ? g_dsp_memory.region_0 : g_dsp_memory.region_1;
}
// 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 bool perform_time_stretching = true;
static std::unique_ptr<AudioCore::Sink> sink;
static AudioCore::TimeStretcher time_stretcher;
static void FlushResidualStretcherAudio() {
time_stretcher.Flush();
while (true) {
std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
if (residual_audio.empty())
break;
sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
}
}
static void OutputCurrentFrame(const StereoFrame16& frame) {
if (perform_time_stretching) {
time_stretcher.AddSamples(&frame[0][0], frame.size());
std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue());
sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2);
} else {
constexpr size_t maximum_sample_latency = 2048; // about 64 miliseconds
if (sink->SamplesInQueue() > maximum_sample_latency) {
// This can occur if we're running too fast and samples are starting to back up.
// Just drop the samples.
return;
}
sink->EnqueueSamples(&frame[0][0], frame.size());
}
}
void EnableStretching(bool enable) {
if (perform_time_stretching == enable)
return;
if (!enable) {
FlushResidualStretcherAudio();
}
perform_time_stretching = enable;
}
// 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() {
if (perform_time_stretching) {
FlushResidualStretcherAudio();
}
}
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
} // namespace DSP

6
src/audio_core/hle/filter.cpp
View file

@ -5,12 +5,12 @@
#include <array>
#include <cstddef>
#include "audio_core/hle/common.h"
#include "audio_core/hle/dsp.h"
#include "audio_core/hle/filter.h"
#include "audio_core/hle/shared_memory.h"
#include "common/common_types.h"
#include "common/math_util.h"
namespace DSP {
namespace AudioCore {
namespace HLE {
void SourceFilters::Reset() {
@ -114,4 +114,4 @@ std::array<s16, 2> SourceFilters::BiquadFilter::ProcessSample(const std::array<s
}
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

8
src/audio_core/hle/filter.h
View file

@ -5,11 +5,11 @@
#pragma once
#include <array>
#include "audio_core/hle/common.h"
#include "audio_core/hle/dsp.h"
#include "audio_core/audio_types.h"
#include "audio_core/hle/shared_memory.h"
#include "common/common_types.h"
namespace DSP {
namespace AudioCore {
namespace HLE {
/// Preprocessing filters. There is an independent set of filters for each Source.
@ -114,4 +114,4 @@ private:
};
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

341
src/audio_core/hle/hle.cpp Normal file
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@ -0,0 +1,341 @@
// Copyright 2017 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include "audio_core/audio_types.h"
#include "audio_core/hle/common.h"
#include "audio_core/hle/hle.h"
#include "audio_core/hle/mixers.h"
#include "audio_core/hle/shared_memory.h"
#include "audio_core/hle/source.h"
#include "audio_core/sink.h"
#include "common/assert.h"
#include "common/common_types.h"
#include "common/logging/log.h"
#include "core/core_timing.h"
#include "core/hle/service/dsp_dsp.h"
namespace AudioCore {
static constexpr u64 audio_frame_ticks = 1310252ull; ///< Units: ARM11 cycles
struct DspHle::Impl final {
public:
explicit Impl(DspHle& parent);
~Impl();
DspState GetDspState() const;
std::vector<u8> PipeRead(DspPipe pipe_number, u32 length);
size_t GetPipeReadableSize(DspPipe pipe_number) const;
void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer);
std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory();
private:
void ResetPipes();
void WriteU16(DspPipe pipe_number, u16 value);
void AudioPipeWriteStructAddresses();
size_t CurrentRegionIndex() const;
HLE::SharedMemory& ReadRegion();
HLE::SharedMemory& WriteRegion();
StereoFrame16 GenerateCurrentFrame();
bool Tick();
void AudioTickCallback(int cycles_late);
DspState dsp_state = DspState::Off;
std::array<std::vector<u8>, num_dsp_pipe> pipe_data;
HLE::DspMemory dsp_memory;
std::array<HLE::Source, HLE::num_sources> sources{{
HLE::Source(0), HLE::Source(1), HLE::Source(2), HLE::Source(3), HLE::Source(4),
HLE::Source(5), HLE::Source(6), HLE::Source(7), HLE::Source(8), HLE::Source(9),
HLE::Source(10), HLE::Source(11), HLE::Source(12), HLE::Source(13), HLE::Source(14),
HLE::Source(15), HLE::Source(16), HLE::Source(17), HLE::Source(18), HLE::Source(19),
HLE::Source(20), HLE::Source(21), HLE::Source(22), HLE::Source(23),
}};
HLE::Mixers mixers;
DspHle& parent;
CoreTiming::EventType* tick_event;
};
DspHle::Impl::Impl(DspHle& parent_) : parent(parent_) {
tick_event =
CoreTiming::RegisterEvent("AudioCore::DspHle::tick_event", [this](u64, int cycles_late) {
this->AudioTickCallback(cycles_late);
});
CoreTiming::ScheduleEvent(audio_frame_ticks, tick_event);
}
DspHle::Impl::~Impl() {
CoreTiming::UnscheduleEvent(tick_event, 0);
}
DspState DspHle::Impl::GetDspState() const {
return dsp_state;
}
std::vector<u8> DspHle::Impl::PipeRead(DspPipe pipe_number, u32 length) {
const size_t pipe_index = static_cast<size_t>(pipe_number);
if (pipe_index >= num_dsp_pipe) {
LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
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 = static_cast<u32>(data.size());
}
if (length == 0)
return {};
std::vector<u8> ret(data.begin(), data.begin() + length);
data.erase(data.begin(), data.begin() + length);
return ret;
}
size_t DspHle::Impl::GetPipeReadableSize(DspPipe pipe_number) const {
const size_t pipe_index = static_cast<size_t>(pipe_number);
if (pipe_index >= num_dsp_pipe) {
LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
return 0;
}
return pipe_data[pipe_index].size();
}
void DspHle::Impl::PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
switch (pipe_number) {
case DspPipe::Audio: {
if (buffer.size() != 4) {
LOG_ERROR(Audio_DSP, "DspPipe::Audio: Unexpected buffer length %zu was written",
buffer.size());
return;
}
enum class StateChange {
Initialize = 0,
Shutdown = 1,
Wakeup = 2,
Sleep = 3,
};
// The difference between Initialize and Wakeup is that Input state is maintained
// when sleeping but isn't when turning it off and on again. (TODO: Implement this.)
// Waking up from sleep garbles some of the structs in the memory region. (TODO:
// Implement this.) Applications store away the state of these structs before
// sleeping and reset it back after wakeup on behalf of the DSP.
switch (static_cast<StateChange>(buffer[0])) {
case StateChange::Initialize:
LOG_INFO(Audio_DSP, "Application has requested initialization of DSP hardware");
ResetPipes();
AudioPipeWriteStructAddresses();
dsp_state = DspState::On;
break;
case StateChange::Shutdown:
LOG_INFO(Audio_DSP, "Application has requested shutdown of DSP hardware");
dsp_state = DspState::Off;
break;
case StateChange::Wakeup:
LOG_INFO(Audio_DSP, "Application has requested wakeup of DSP hardware");
ResetPipes();
AudioPipeWriteStructAddresses();
dsp_state = DspState::On;
break;
case StateChange::Sleep:
LOG_INFO(Audio_DSP, "Application has requested sleep of DSP hardware");
UNIMPLEMENTED();
dsp_state = DspState::Sleeping;
break;
default:
LOG_ERROR(Audio_DSP,
"Application has requested unknown state transition of DSP hardware %hhu",
buffer[0]);
dsp_state = DspState::Off;
break;
}
return;
}
default:
LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented",
static_cast<size_t>(pipe_number));
UNIMPLEMENTED();
return;
}
}
std::array<u8, Memory::DSP_RAM_SIZE>& DspHle::Impl::GetDspMemory() {
return dsp_memory.raw_memory;
}
void DspHle::Impl::ResetPipes() {
for (auto& data : pipe_data) {
data.clear();
}
dsp_state = DspState::Off;
}
void DspHle::Impl::WriteU16(DspPipe pipe_number, u16 value) {
const size_t pipe_index = static_cast<size_t>(pipe_number);
std::vector<u8>& data = pipe_data.at(pipe_index);
// Little endian
data.emplace_back(value & 0xFF);
data.emplace_back(value >> 8);
}
void DspHle::Impl::AudioPipeWriteStructAddresses() {
// These struct addresses are DSP dram addresses.
// See also: DSP_DSP::ConvertProcessAddressFromDspDram
static const std::array<u16, 15> struct_addresses = {
0x8000 + offsetof(HLE::SharedMemory, frame_counter) / 2,
0x8000 + offsetof(HLE::SharedMemory, source_configurations) / 2,
0x8000 + offsetof(HLE::SharedMemory, source_statuses) / 2,
0x8000 + offsetof(HLE::SharedMemory, adpcm_coefficients) / 2,
0x8000 + offsetof(HLE::SharedMemory, dsp_configuration) / 2,
0x8000 + offsetof(HLE::SharedMemory, dsp_status) / 2,
0x8000 + offsetof(HLE::SharedMemory, final_samples) / 2,
0x8000 + offsetof(HLE::SharedMemory, intermediate_mix_samples) / 2,
0x8000 + offsetof(HLE::SharedMemory, compressor) / 2,
0x8000 + offsetof(HLE::SharedMemory, dsp_debug) / 2,
0x8000 + offsetof(HLE::SharedMemory, unknown10) / 2,
0x8000 + offsetof(HLE::SharedMemory, unknown11) / 2,
0x8000 + offsetof(HLE::SharedMemory, unknown12) / 2,
0x8000 + offsetof(HLE::SharedMemory, unknown13) / 2,
0x8000 + offsetof(HLE::SharedMemory, unknown14) / 2,
};
// Begin with a u16 denoting the number of structs.
WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size()));
// Then write the struct addresses.
for (u16 addr : struct_addresses) {
WriteU16(DspPipe::Audio, addr);
}
// Signal that we have data on this pipe.
Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
}
size_t DspHle::Impl::CurrentRegionIndex() const {
// The region with the higher frame counter is chosen unless there is wraparound.
// This function only returns a 0 or 1.
const u16 frame_counter_0 = dsp_memory.region_0.frame_counter;
const u16 frame_counter_1 = dsp_memory.region_1.frame_counter;
if (frame_counter_0 == 0xFFFFu && frame_counter_1 != 0xFFFEu) {
// Wraparound has occurred.
return 1;
}
if (frame_counter_1 == 0xFFFFu && frame_counter_0 != 0xFFFEu) {
// Wraparound has occurred.
return 0;
}
return (frame_counter_0 > frame_counter_1) ? 0 : 1;
}
HLE::SharedMemory& DspHle::Impl::ReadRegion() {
return CurrentRegionIndex() == 0 ? dsp_memory.region_0 : dsp_memory.region_1;
}
HLE::SharedMemory& DspHle::Impl::WriteRegion() {
return CurrentRegionIndex() != 0 ? dsp_memory.region_0 : dsp_memory.region_1;
}
StereoFrame16 DspHle::Impl::GenerateCurrentFrame() {
HLE::SharedMemory& read = ReadRegion();
HLE::SharedMemory& write = WriteRegion();
std::array<QuadFrame32, 3> intermediate_mixes = {};
// Generate intermediate mixes
for (size_t i = 0; i < HLE::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;
}
bool DspHle::Impl::Tick() {
StereoFrame16 current_frame = {};
// TODO: Check dsp::DSP semaphore (which indicates emulated application has finished writing to
// shared memory region)
current_frame = GenerateCurrentFrame();
parent.OutputFrame(current_frame);
return true;
}
void DspHle::Impl::AudioTickCallback(int cycles_late) {
if (Tick()) {
// TODO(merry): Signal all the other interrupts as appropriate.
Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
// HACK(merry): Added to prevent regressions. Will remove soon.
Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Binary);
}
// Reschedule recurrent event
CoreTiming::ScheduleEvent(audio_frame_ticks - cycles_late, tick_event);
}
DspHle::DspHle() : impl(std::make_unique<Impl>(*this)) {}
DspHle::~DspHle() = default;
DspState DspHle::GetDspState() const {
return impl->GetDspState();
}
std::vector<u8> DspHle::PipeRead(DspPipe pipe_number, u32 length) {
return impl->PipeRead(pipe_number, length);
}
size_t DspHle::GetPipeReadableSize(DspPipe pipe_number) const {
return impl->GetPipeReadableSize(pipe_number);
}
void DspHle::PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
impl->PipeWrite(pipe_number, buffer);
}
std::array<u8, Memory::DSP_RAM_SIZE>& DspHle::GetDspMemory() {
return impl->GetDspMemory();
}
} // namespace AudioCore

36
src/audio_core/hle/hle.h Normal file
View file

@ -0,0 +1,36 @@
// Copyright 2017 Citra 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/audio_types.h"
#include "audio_core/dsp_interface.h"
#include "common/common_types.h"
#include "core/memory.h"
namespace AudioCore {
class DspHle final : public DspInterface {
public:
DspHle();
~DspHle();
DspState GetDspState() const override;
std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) override;
size_t GetPipeReadableSize(DspPipe pipe_number) const override;
void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) override;
std::array<u8, Memory::DSP_RAM_SIZE>& GetDspMemory() override;
private:
struct Impl;
friend struct Impl;
std::unique_ptr<Impl> impl;
};
} // namespace AudioCore

6
src/audio_core/hle/mixers.cpp
View file

@ -4,14 +4,12 @@
#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 AudioCore {
namespace HLE {
void Mixers::Reset() {
@ -207,4 +205,4 @@ DspStatus Mixers::GetCurrentStatus() const {
}
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

8
src/audio_core/hle/mixers.h
View file

@ -5,10 +5,10 @@
#pragma once
#include <array>
#include "audio_core/hle/common.h"
#include "audio_core/hle/dsp.h"
#include "audio_core/audio_types.h"
#include "audio_core/hle/shared_memory.h"
namespace DSP {
namespace AudioCore {
namespace HLE {
class Mixers final {
@ -58,4 +58,4 @@ private:
};
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

177
src/audio_core/hle/pipe.cpp
View file

@ -1,177 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <array>
#include <vector>
#include "audio_core/hle/dsp.h"
#include "audio_core/hle/pipe.h"
#include "common/assert.h"
#include "common/common_types.h"
#include "common/logging/log.h"
#include "core/hle/service/dsp_dsp.h"
namespace DSP {
namespace HLE {
static DspState dsp_state = DspState::Off;
static std::array<std::vector<u8>, NUM_DSP_PIPE> pipe_data;
void ResetPipes() {
for (auto& data : pipe_data) {
data.clear();
}
dsp_state = DspState::Off;
}
std::vector<u8> PipeRead(DspPipe pipe_number, u32 length) {
const size_t pipe_index = static_cast<size_t>(pipe_number);
if (pipe_index >= NUM_DSP_PIPE) {
LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
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 = static_cast<u32>(data.size());
}
if (length == 0)
return {};
std::vector<u8> ret(data.begin(), data.begin() + length);
data.erase(data.begin(), data.begin() + length);
return ret;
}
size_t GetPipeReadableSize(DspPipe pipe_number) {
const size_t pipe_index = static_cast<size_t>(pipe_number);
if (pipe_index >= NUM_DSP_PIPE) {
LOG_ERROR(Audio_DSP, "pipe_number = %zu invalid", pipe_index);
return 0;
}
return pipe_data[pipe_index].size();
}
static void WriteU16(DspPipe pipe_number, u16 value) {
const size_t pipe_index = static_cast<size_t>(pipe_number);
std::vector<u8>& data = pipe_data.at(pipe_index);
// Little endian
data.emplace_back(value & 0xFF);
data.emplace_back(value >> 8);
}
static void AudioPipeWriteStructAddresses() {
// These struct addresses are DSP dram addresses.
// See also: DSP_DSP::ConvertProcessAddressFromDspDram
static const std::array<u16, 15> struct_addresses = {
0x8000 + offsetof(SharedMemory, frame_counter) / 2,
0x8000 + offsetof(SharedMemory, source_configurations) / 2,
0x8000 + offsetof(SharedMemory, source_statuses) / 2,
0x8000 + offsetof(SharedMemory, adpcm_coefficients) / 2,
0x8000 + offsetof(SharedMemory, dsp_configuration) / 2,
0x8000 + offsetof(SharedMemory, dsp_status) / 2,
0x8000 + offsetof(SharedMemory, final_samples) / 2,
0x8000 + offsetof(SharedMemory, intermediate_mix_samples) / 2,
0x8000 + offsetof(SharedMemory, compressor) / 2,
0x8000 + offsetof(SharedMemory, dsp_debug) / 2,
0x8000 + offsetof(SharedMemory, unknown10) / 2,
0x8000 + offsetof(SharedMemory, unknown11) / 2,
0x8000 + offsetof(SharedMemory, unknown12) / 2,
0x8000 + offsetof(SharedMemory, unknown13) / 2,
0x8000 + offsetof(SharedMemory, unknown14) / 2,
};
// Begin with a u16 denoting the number of structs.
WriteU16(DspPipe::Audio, static_cast<u16>(struct_addresses.size()));
// Then write the struct addresses.
for (u16 addr : struct_addresses) {
WriteU16(DspPipe::Audio, addr);
}
// Signal that we have data on this pipe.
Service::DSP_DSP::SignalPipeInterrupt(DspPipe::Audio);
}
void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer) {
switch (pipe_number) {
case DspPipe::Audio: {
if (buffer.size() != 4) {
LOG_ERROR(Audio_DSP, "DspPipe::Audio: Unexpected buffer length %zu was written",
buffer.size());
return;
}
enum class StateChange {
Initialize = 0,
Shutdown = 1,
Wakeup = 2,
Sleep = 3,
};
// The difference between Initialize and Wakeup is that Input state is maintained
// when sleeping but isn't when turning it off and on again. (TODO: Implement this.)
// Waking up from sleep garbles some of the structs in the memory region. (TODO:
// Implement this.) Applications store away the state of these structs before
// sleeping and reset it back after wakeup on behalf of the DSP.
switch (static_cast<StateChange>(buffer[0])) {
case StateChange::Initialize:
LOG_INFO(Audio_DSP, "Application has requested initialization of DSP hardware");
ResetPipes();
AudioPipeWriteStructAddresses();
dsp_state = DspState::On;
break;
case StateChange::Shutdown:
LOG_INFO(Audio_DSP, "Application has requested shutdown of DSP hardware");
dsp_state = DspState::Off;
break;
case StateChange::Wakeup:
LOG_INFO(Audio_DSP, "Application has requested wakeup of DSP hardware");
ResetPipes();
AudioPipeWriteStructAddresses();
dsp_state = DspState::On;
break;
case StateChange::Sleep:
LOG_INFO(Audio_DSP, "Application has requested sleep of DSP hardware");
UNIMPLEMENTED();
dsp_state = DspState::Sleeping;
break;
default:
LOG_ERROR(Audio_DSP,
"Application has requested unknown state transition of DSP hardware %hhu",
buffer[0]);
dsp_state = DspState::Off;
break;
}
return;
}
default:
LOG_CRITICAL(Audio_DSP, "pipe_number = %zu unimplemented",
static_cast<size_t>(pipe_number));
UNIMPLEMENTED();
return;
}
}
DspState GetDspState() {
return dsp_state;
}
} // namespace HLE
} // namespace DSP

63
src/audio_core/hle/pipe.h
View file

@ -1,63 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <cstddef>
#include <vector>
#include "common/common_types.h"
namespace DSP {
namespace HLE {
/// Reset the pipes by setting pipe positions back to the beginning.
void ResetPipes();
enum class DspPipe {
Debug = 0,
Dma = 1,
Audio = 2,
Binary = 3,
};
constexpr size_t NUM_DSP_PIPE = 8;
/**
* 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);
/**
* How much data is left in pipe
* @param pipe_number The Pipe ID
* @return The amount of data remaning in the pipe. This is the maximum length PipeRead will return.
*/
size_t GetPipeReadableSize(DspPipe pipe_number);
/**
* Write to a DSP pipe.
* @param pipe_number The Pipe ID
* @param buffer The data to write to the pipe.
*/
void PipeWrite(DspPipe pipe_number, const std::vector<u8>& buffer);
enum class DspState {
Off,
On,
Sleeping,
};
/// Get the state of the DSP
DspState GetDspState();
} // namespace HLE
} // namespace DSP

68
src/audio_core/hle/dsp.h → src/audio_core/hle/shared_memory.h
View file

@ -8,6 +8,7 @@
#include <cstddef>
#include <memory>
#include <type_traits>
#include "audio_core/audio_types.h"
#include "audio_core/hle/common.h"
#include "common/bit_field.h"
#include "common/common_funcs.h"
@ -15,10 +16,6 @@
#include "common/swap.h"
namespace AudioCore {
class Sink;
}
namespace DSP {
namespace HLE {
// The application-accessible region of DSP memory consists of two parts. Both are marked as IO and
@ -86,7 +83,7 @@ static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivial
// 0 0xBFFF Frame Counter Application
//
// #: This refers to the order in which they appear in the DspPipe::Audio DSP pipe.
// See also: DSP::HLE::PipeRead.
// See also: HLE::PipeRead.
//
// Note that the above addresses do vary slightly between audio firmwares observed; the addresses
// are not fixed in stone. The addresses above are only an examplar; they're what this
@ -527,69 +524,40 @@ static_assert(offsetof(DspMemory, region_0) == region0_offset,
static_assert(offsetof(DspMemory, region_1) == region1_offset,
"DSP region 1 is at the wrong offset");
extern DspMemory g_dsp_memory;
// 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");
"Structures in 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");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_statuses) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, adpcm_coefficients) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_configuration) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_status) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, final_samples) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, intermediate_mix_samples) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, compressor) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_debug) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown10) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown11) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown12) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown13) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown14) % 2 == 0,
"Structures in DSP::HLE::SharedMemory must be 2-byte aligned");
"Structures in HLE::SharedMemory must be 2-byte aligned");
#undef INSERT_PADDING_DSPWORDS
#undef ASSERT_DSP_STRUCT
/// Initialize DSP hardware
void Init();
/// Shutdown DSP hardware
void Shutdown();
/**
* Perform processing and updates state of current shared memory buffer.
* This function is called every audio tick before triggering the audio interrupt.
* @return Whether an audio interrupt should be triggered this frame.
*/
bool Tick();
/**
* Set the output sink. This must be called before calling Tick().
* @param sink The sink to which audio will be output to.
*/
void SetSink(std::unique_ptr<AudioCore::Sink> sink);
/**
* Enables/Disables audio-stretching.
* Audio stretching is an enhancement that stretches audio to match emulation
* speed to prevent stuttering at the cost of some audio latency.
* @param enable true to enable, false to disable.
*/
void EnableStretching(bool enable);
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

4
src/audio_core/hle/source.cpp
View file

@ -12,7 +12,7 @@
#include "common/logging/log.h"
#include "core/memory.h"
namespace DSP {
namespace AudioCore {
namespace HLE {
SourceStatus::Status Source::Tick(SourceConfiguration::Configuration& config,
@ -345,4 +345,4 @@ SourceStatus::Status Source::GetCurrentStatus() {
}
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

6
src/audio_core/hle/source.h
View file

@ -7,14 +7,14 @@
#include <array>
#include <queue>
#include <vector>
#include "audio_core/audio_types.h"
#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 AudioCore {
namespace HLE {
/**
@ -146,4 +146,4 @@ private:
};
} // namespace HLE
} // namespace DSP
} // namespace AudioCore

11
src/audio_core/interpolate.cpp
View file

@ -6,6 +6,7 @@
#include "common/assert.h"
#include "common/math_util.h"
namespace AudioCore {
namespace AudioInterp {
// Calculations are done in fixed point with 24 fractional bits.
@ -16,8 +17,8 @@ constexpr u64 scale_mask = scale_factor - 1;
/// Here we step over the input in steps of rate, until we consume all of the input.
/// Three adjacent samples are passed to fn each step.
template <typename Function>
static void StepOverSamples(State& state, StereoBuffer16& input, float rate,
DSP::HLE::StereoFrame16& output, size_t& outputi, Function fn) {
static void StepOverSamples(State& state, StereoBuffer16& input, float rate, StereoFrame16& output,
size_t& outputi, Function fn) {
ASSERT(rate > 0);
if (input.empty())
@ -50,14 +51,13 @@ static void StepOverSamples(State& state, StereoBuffer16& input, float rate,
input.erase(input.begin(), std::next(input.begin(), inputi + 2));
}
void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
size_t& outputi) {
void None(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, size_t& outputi) {
StepOverSamples(
state, input, rate, output, outputi,
[](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
}
void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
void Linear(State& state, StereoBuffer16& input, float rate, StereoFrame16& output,
size_t& outputi) {
// Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
StepOverSamples(state, input, rate, output, outputi,
@ -74,3 +74,4 @@ void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFra
}
} // namespace AudioInterp
} // namespace AudioCore

9
src/audio_core/interpolate.h
View file

@ -6,9 +6,10 @@
#include <array>
#include <deque>
#include "audio_core/hle/common.h"
#include "audio_core/audio_types.h"
#include "common/common_types.h"
namespace AudioCore {
namespace AudioInterp {
/// A variable length buffer of signed PCM16 stereo samples.
@ -31,8 +32,7 @@ struct State {
* @param output The resampled audio buffer.
* @param outputi The index of output to start writing to.
*/
void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
size_t& outputi);
void None(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, size_t& outputi);
/**
* Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
@ -43,7 +43,8 @@ void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame
* @param output The resampled audio buffer.
* @param outputi The index of output to start writing to.
*/
void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
void Linear(State& state, StereoBuffer16& input, float rate, StereoFrame16& output,
size_t& outputi);
} // namespace AudioInterp
} // namespace AudioCore

2
src/audio_core/null_sink.h
View file

@ -5,7 +5,7 @@
#pragma once
#include <cstddef>
#include "audio_core/audio_core.h"
#include "audio_core/audio_types.h"
#include "audio_core/sink.h"
namespace AudioCore {

2
src/audio_core/sdl2_sink.cpp
View file

@ -5,7 +5,7 @@
#include <list>
#include <numeric>
#include <SDL.h>
#include "audio_core/audio_core.h"
#include "audio_core/audio_types.h"
#include "audio_core/sdl2_sink.h"
#include "common/assert.h"
#include "common/logging/log.h"

1
src/audio_core/sink_details.cpp
View file

@ -4,6 +4,7 @@
#include <algorithm>
#include <memory>
#include <string>
#include <vector>
#include "audio_core/null_sink.h"
#include "audio_core/sink_details.h"

2
src/audio_core/time_stretch.cpp
View file

@ -6,7 +6,7 @@
#include <cmath>
#include <vector>
#include <SoundTouch.h>
#include "audio_core/audio_core.h"
#include "audio_core/audio_types.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/logging/log.h"

1
src/citra_qt/configuration/configure_audio.cpp
View file

@ -3,7 +3,6 @@
// Refer to the license.txt file included.
#include <memory>
#include "audio_core/audio_core.h"
#include "audio_core/sink.h"
#include "audio_core/sink_details.h"
#include "citra_qt/configuration/configure_audio.h"

20
src/core/core.cpp
View file

@ -4,7 +4,8 @@
#include <memory>
#include <utility>
#include "audio_core/audio_core.h"
#include "audio_core/dsp_interface.h"
#include "audio_core/hle/hle.h"
#include "common/logging/log.h"
#include "core/arm/arm_interface.h"
#ifdef ARCHITECTURE_x86_64
@ -149,6 +150,8 @@ void System::Reschedule() {
System::ResultStatus System::Init(EmuWindow* emu_window, u32 system_mode) {
LOG_DEBUG(HW_Memory, "initialized OK");
CoreTiming::Init();
if (Settings::values.use_cpu_jit) {
#ifdef ARCHITECTURE_x86_64
cpu_core = std::make_unique<ARM_Dynarmic>(USER32MODE);
@ -160,13 +163,15 @@ System::ResultStatus System::Init(EmuWindow* emu_window, u32 system_mode) {
cpu_core = std::make_unique<ARM_DynCom>(USER32MODE);
}
dsp_core = std::make_unique<AudioCore::DspHle>();
dsp_core->SetSink(Settings::values.sink_id);
dsp_core->EnableStretching(Settings::values.enable_audio_stretching);
telemetry_session = std::make_unique<Core::TelemetrySession>();
CoreTiming::Init();
HW::Init();
Kernel::Init(system_mode);
Service::Init();
AudioCore::Init();
GDBStub::Init();
Movie::GetInstance().Init();
@ -196,15 +201,16 @@ void System::Shutdown() {
// Shutdown emulation session
Movie::GetInstance().Shutdown();
GDBStub::Shutdown();
AudioCore::Shutdown();
VideoCore::Shutdown();
Service::Shutdown();
Kernel::Shutdown();
HW::Shutdown();
CoreTiming::Shutdown();
cpu_core = nullptr;
app_loader = nullptr;
telemetry_session = nullptr;
dsp_core = nullptr;
cpu_core = nullptr;
CoreTiming::Shutdown();
app_loader = nullptr;
if (auto room_member = Network::GetRoomMember().lock()) {
Network::GameInfo game_info{};
room_member->SendGameInfo(game_info);

19
src/core/core.h
View file

@ -15,6 +15,10 @@
class EmuWindow;
class ARM_Interface;
namespace AudioCore {
class DspInterface;
}
namespace Core {
class System {
@ -102,6 +106,14 @@ public:
return *cpu_core;
}
/**
* Gets a reference to the emulated DSP.
* @returns A reference to the emulated DSP.
*/
AudioCore::DspInterface& DSP() {
return *dsp_core;
}
PerfStats perf_stats;
FrameLimiter frame_limiter;
@ -138,6 +150,9 @@ private:
///< ARM11 CPU core
std::unique_ptr<ARM_Interface> cpu_core;
///< DSP core
std::unique_ptr<AudioCore::DspInterface> dsp_core;
/// When true, signals that a reschedule should happen
bool reschedule_pending{};
@ -154,6 +169,10 @@ inline ARM_Interface& CPU() {
return System::GetInstance().CPU();
}
inline AudioCore::DspInterface& DSP() {
return System::GetInstance().DSP();
}
inline TelemetrySession& Telemetry() {
return System::GetInstance().TelemetrySession();
}

46
src/core/hle/service/dsp_dsp.cpp
View file

@ -5,10 +5,12 @@
#include <algorithm>
#include <array>
#include <cinttypes>
#include "audio_core/hle/pipe.h"
#include "audio_core/audio_types.h"
#include "audio_core/dsp_interface.h"
#include "common/assert.h"
#include "common/hash.h"
#include "common/logging/log.h"
#include "core/core.h"
#include "core/hle/ipc.h"
#include "core/hle/kernel/event.h"
#include "core/hle/kernel/handle_table.h"
@ -16,7 +18,7 @@
#include "core/hle/service/dsp_dsp.h"
#include "core/memory.h"
using DspPipe = DSP::HLE::DspPipe;
using DspPipe = AudioCore::DspPipe;
namespace Service {
namespace DSP_DSP {
@ -44,7 +46,7 @@ public:
return one;
case InterruptType::Pipe: {
const size_t pipe_index = static_cast<size_t>(dsp_pipe);
ASSERT(pipe_index < DSP::HLE::NUM_DSP_PIPE);
ASSERT(pipe_index < AudioCore::num_dsp_pipe);
return pipe[pipe_index];
}
}
@ -73,7 +75,7 @@ private:
/// Currently unknown purpose
Kernel::SharedPtr<Kernel::Event> one = nullptr;
/// Each DSP pipe has an associated interrupt
std::array<Kernel::SharedPtr<Kernel::Event>, DSP::HLE::NUM_DSP_PIPE> pipe = {{}};
std::array<Kernel::SharedPtr<Kernel::Event>, AudioCore::num_dsp_pipe> pipe = {{}};
};
static InterruptEvents interrupt_events;
@ -216,7 +218,7 @@ static void RegisterInterruptEvents(Service::Interface* self) {
u32 pipe_index = cmd_buff[2];
u32 event_handle = cmd_buff[4];
ASSERT_MSG(type_index < NUM_INTERRUPT_TYPE && pipe_index < DSP::HLE::NUM_DSP_PIPE,
ASSERT_MSG(type_index < NUM_INTERRUPT_TYPE && pipe_index < AudioCore::num_dsp_pipe,
"Invalid type or pipe: type = %u, pipe = %u", type_index, pipe_index);
InterruptType type = static_cast<InterruptType>(cmd_buff[1]);
@ -289,7 +291,7 @@ static void WriteProcessPipe(Service::Interface* self) {
u32 size = cmd_buff[2];
u32 buffer = cmd_buff[4];
DSP::HLE::DspPipe pipe = static_cast<DSP::HLE::DspPipe>(pipe_index);
AudioCore::DspPipe pipe = static_cast<AudioCore::DspPipe>(pipe_index);
if (IPC::StaticBufferDesc(size, 1) != cmd_buff[3]) {
LOG_ERROR(Service_DSP, "IPC static buffer descriptor failed validation (0x%X). pipe=%u, "
@ -312,12 +314,12 @@ static void WriteProcessPipe(Service::Interface* self) {
// The likely reason for this is that games tend to pass in garbage at these bytes
// because they read random bytes off the stack.
switch (pipe) {
case DSP::HLE::DspPipe::Audio:
case AudioCore::DspPipe::Audio:
ASSERT(message.size() >= 4);
message[2] = 0;
message[3] = 0;
break;
case DSP::HLE::DspPipe::Binary:
case AudioCore::DspPipe::Binary:
ASSERT(message.size() >= 8);
message[4] = 1;
message[5] = 0;
@ -326,7 +328,7 @@ static void WriteProcessPipe(Service::Interface* self) {
break;
}
DSP::HLE::PipeWrite(pipe, message);
Core::DSP().PipeWrite(pipe, message);
cmd_buff[0] = IPC::MakeHeader(0xD, 1, 0);
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
@ -338,7 +340,7 @@ static void WriteProcessPipe(Service::Interface* self) {
* DSP_DSP::ReadPipeIfPossible service function
* A pipe is a means of communication between the ARM11 and DSP that occurs on
* hardware by writing to/reading from the DSP registers at 0x10203000.
* Pipes are used for initialisation. See also DSP::HLE::PipeRead.
* Pipes are used for initialisation. See also DspInterface::PipeRead.
* Inputs:
* 1 : Pipe Number
* 2 : Unknown
@ -356,7 +358,7 @@ static void ReadPipeIfPossible(Service::Interface* self) {
u32 size = cmd_buff[3] & 0xFFFF; // Lower 16 bits are size
VAddr addr = cmd_buff[0x41];
DSP::HLE::DspPipe pipe = static_cast<DSP::HLE::DspPipe>(pipe_index);
AudioCore::DspPipe pipe = static_cast<AudioCore::DspPipe>(pipe_index);
ASSERT_MSG(Memory::IsValidVirtualAddress(addr),
"Invalid addr: pipe=0x%08X, unknown=0x%08X, size=0x%X, buffer=0x%08X", pipe_index,
@ -364,8 +366,8 @@ static void ReadPipeIfPossible(Service::Interface* self) {
cmd_buff[0] = IPC::MakeHeader(0x10, 1, 2);
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
if (DSP::HLE::GetPipeReadableSize(pipe) >= size) {
std::vector<u8> response = DSP::HLE::PipeRead(pipe, size);
if (Core::DSP().GetPipeReadableSize(pipe) >= size) {
std::vector<u8> response = Core::DSP().PipeRead(pipe, size);
Memory::WriteBlock(addr, response.data(), response.size());
@ -400,14 +402,14 @@ static void ReadPipe(Service::Interface* self) {
u32 size = cmd_buff[3] & 0xFFFF; // Lower 16 bits are size
VAddr addr = cmd_buff[0x41];
DSP::HLE::DspPipe pipe = static_cast<DSP::HLE::DspPipe>(pipe_index);
AudioCore::DspPipe pipe = static_cast<AudioCore::DspPipe>(pipe_index);
ASSERT_MSG(Memory::IsValidVirtualAddress(addr),
"Invalid addr: pipe=0x%08X, unknown=0x%08X, size=0x%X, buffer=0x%08X", pipe_index,
unknown, size, addr);
if (DSP::HLE::GetPipeReadableSize(pipe) >= size) {
std::vector<u8> response = DSP::HLE::PipeRead(pipe, size);
if (Core::DSP().GetPipeReadableSize(pipe) >= size) {
std::vector<u8> response = Core::DSP().PipeRead(pipe, size);
Memory::WriteBlock(addr, response.data(), response.size());
@ -441,11 +443,11 @@ static void GetPipeReadableSize(Service::Interface* self) {
u32 pipe_index = cmd_buff[1];
u32 unknown = cmd_buff[2];
DSP::HLE::DspPipe pipe = static_cast<DSP::HLE::DspPipe>(pipe_index);
AudioCore::DspPipe pipe = static_cast<AudioCore::DspPipe>(pipe_index);
cmd_buff[0] = IPC::MakeHeader(0xF, 2, 0);
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
cmd_buff[2] = static_cast<u32>(DSP::HLE::GetPipeReadableSize(pipe));
cmd_buff[2] = static_cast<u32>(Core::DSP().GetPipeReadableSize(pipe));
LOG_DEBUG(Service_DSP, "pipe=%u, unknown=0x%08X, return cmd_buff[2]=0x%08X", pipe_index,
unknown, cmd_buff[2]);
@ -511,12 +513,12 @@ static void RecvData(Service::Interface* self) {
cmd_buff[0] = IPC::MakeHeader(0x1, 2, 0);
cmd_buff[1] = RESULT_SUCCESS.raw;
switch (DSP::HLE::GetDspState()) {
case DSP::HLE::DspState::On:
switch (Core::DSP().GetDspState()) {
case AudioCore::DspState::On:
cmd_buff[2] = 0;
break;
case DSP::HLE::DspState::Off:
case DSP::HLE::DspState::Sleeping:
case AudioCore::DspState::Off:
case AudioCore::DspState::Sleeping:
cmd_buff[2] = 1;
break;
default:

6
src/core/hle/service/dsp_dsp.h
View file

@ -7,11 +7,9 @@
#include <string>
#include "core/hle/service/service.h"
namespace DSP {
namespace HLE {
namespace AudioCore {
enum class DspPipe;
}
}
namespace Service {
namespace DSP_DSP {
@ -30,7 +28,7 @@ public:
* Signal a specific DSP related interrupt of type == InterruptType::Pipe, pipe == pipe.
* @param pipe The DSP pipe for which to signal an interrupt for.
*/
void SignalPipeInterrupt(DSP::HLE::DspPipe pipe);
void SignalPipeInterrupt(AudioCore::DspPipe pipe);
} // namespace DSP_DSP
} // namespace Service

4
src/core/memory.cpp
View file

@ -4,7 +4,7 @@
#include <array>
#include <cstring>
#include "audio_core/audio_core.h"
#include "audio_core/dsp_interface.h"
#include "common/assert.h"
#include "common/common_types.h"
#include "common/logging/log.h"
@ -311,7 +311,7 @@ u8* GetPhysicalPointer(PAddr address) {
target_pointer = vram.data() + offset_into_region;
break;
case DSP_RAM_PADDR:
target_pointer = AudioCore::GetDspMemory().data() + offset_into_region;
target_pointer = Core::DSP().GetDspMemory().data() + offset_into_region;
break;
case FCRAM_PADDR:
for (const auto& region : Kernel::memory_regions) {

9
src/core/settings.cpp
View file

@ -2,7 +2,8 @@
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include "audio_core/audio_core.h"
#include "audio_core/dsp_interface.h"
#include "core/core.h"
#include "core/gdbstub/gdbstub.h"
#include "core/hle/service/hid/hid.h"
#include "core/hle/service/ir/ir.h"
@ -28,8 +29,10 @@ void Apply() {
VideoCore::g_emu_window->UpdateCurrentFramebufferLayout(layout.width, layout.height);
}
AudioCore::SelectSink(values.sink_id);
AudioCore::EnableStretching(values.enable_audio_stretching);
if (Core::System::GetInstance().IsPoweredOn()) {
Core::DSP().SetSink(values.sink_id);
Core::DSP().EnableStretching(values.enable_audio_stretching);
}
Service::HID::ReloadInputDevices();
Service::IR::ReloadInputDevices();