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citra/src/audio_core/hle/shared_memory.h

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// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <array>
#include <cstddef>
#include <memory>
#include <type_traits>
#include <boost/serialization/access.hpp>
#include "audio_core/audio_types.h"
#include "audio_core/hle/common.h"
#include "common/bit_field.h"
#include "common/common_funcs.h"
#include "common/common_types.h"
#include "common/swap.h"
namespace AudioCore::HLE {
// The application-accessible region of DSP memory consists of two parts. Both are marked as IO and
// have Read/Write permissions.
//
// First Region: 0x1FF50000 (Size: 0x8000)
// Second Region: 0x1FF70000 (Size: 0x8000)
//
// The DSP reads from each region alternately based on the frame counter for each region much like a
// double-buffer. The frame counter is located as the very last u16 of each region and is
// incremented each audio tick.
constexpr u32 region0_offset = 0x50000;
constexpr u32 region1_offset = 0x70000;
/**
* 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
* layout of the ARM11. Hence from the ARM11's point of view the memory space appears to be
* middle-endian.
*
* Unusually this does not appear to be an issue for floating point numbers. The DSP makes the more
* sensible choice of keeping that little-endian. There are also some exceptions such as the
* IntermediateMixSamples structure, which is little-endian.
*
* This struct implements the conversion to and from this middle-endianness.
*/
struct u32_dsp {
u32_dsp() = default;
operator u32() const {
return Convert(storage);
}
void operator=(u32 new_value) {
storage = Convert(new_value);
}
private:
static constexpr u32 Convert(u32 value) {
return (value << 16) | (value >> 16);
}
u32_le storage;
template <class Archive>
void serialize(Archive& ar, const unsigned int) {
ar& storage;
}
friend class boost::serialization::access;
};
static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivially copyable");
// There are 15 structures in each memory region. A table of them in the order they appear in memory
// is presented below:
//
// # First Region DSP Address Purpose Control
// 5 0x8400 DSP Status DSP
// 9 0x8410 DSP Debug Info DSP
// 6 0x8540 Final Mix Samples DSP
// 2 0x8680 Source Status [24] DSP
// 8 0x8710 Compressor Table Application
// 4 0x9430 DSP Configuration Application
// 7 0x9492 Intermediate Mix Samples DSP + App
// 1 0x9E92 Source Configuration [24] Application
// 3 0xA792 Source ADPCM Coefficients [24] Application
// 10 0xA912 Surround Sound Related
// 11 0xAA12 Surround Sound Related
// 12 0xAAD2 Surround Sound Related
// 13 0xAC52 Surround Sound Related
// 14 0xAC5C Surround Sound Related
// 0 0xBFFF Frame Counter Application
//
// #: This refers to the order in which they appear in the DspPipe::Audio DSP pipe.
// 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
// implementation does and provides to applications.
//
// Application requests the DSP service to convert DSP addresses into ARM11 virtual addresses using
// the ConvertProcessAddressFromDspDram service call. Applications seem to derive the addresses for
// the second region via:
// second_region_dsp_addr = first_region_dsp_addr | 0x10000
//
// Applications maintain most of its own audio state, the memory region is used mainly for
// communication and not storage of state.
//
// In the documentation below, filter and effect transfer functions are specified in the z domain.
// (If you are more familiar with the Laplace transform, z = exp(sT). The z domain is the digital
// frequency domain, just like how the s domain is the analog frequency domain.)
#define INSERT_PADDING_DSPWORDS(num_words) INSERT_PADDING_BYTES(2 * (num_words))
#define ASSERT_DSP_STRUCT(name, size) \
static_assert(std::is_standard_layout<name>::value, \
"DSP structure " #name " doesn't use standard layout"); \
static_assert(std::is_trivially_copyable<name>::value, \
"DSP structure " #name " isn't trivially copyable"); \
static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
struct SourceConfiguration {
struct Configuration {
/// These dirty flags are set by the application when it updates the fields in this struct.
/// The DSP clears these each audio frame.
union {
u32_le dirty_raw;
BitField<0, 1, u32> format_dirty;
BitField<1, 1, u32> mono_or_stereo_dirty;
BitField<2, 1, u32> adpcm_coefficients_dirty;
/// Tends to be set when a looped buffer is queued.
BitField<3, 1, u32> partial_embedded_buffer_dirty;
BitField<4, 1, u32> partial_reset_flag;
BitField<16, 1, u32> enable_dirty;
BitField<17, 1, u32> interpolation_dirty;
BitField<18, 1, u32> rate_multiplier_dirty;
BitField<19, 1, u32> buffer_queue_dirty;
BitField<20, 1, u32> loop_related_dirty;
/// Tends to also be set when embedded buffer is updated.
BitField<21, 1, u32> play_position_dirty;
BitField<22, 1, u32> filters_enabled_dirty;
BitField<23, 1, u32> simple_filter_dirty;
BitField<24, 1, u32> biquad_filter_dirty;
BitField<25, 1, u32> gain_0_dirty;
BitField<26, 1, u32> gain_1_dirty;
BitField<27, 1, u32> gain_2_dirty;
BitField<28, 1, u32> sync_count_dirty;
BitField<29, 1, u32> reset_flag;
BitField<30, 1, u32> embedded_buffer_dirty;
};
// Gain control
/**
* Gain is between 0.0-1.0. This determines how much will this source appear on each of the
* 12 channels that feed into the intermediate mixers. Each of the three intermediate mixers
* is fed two left and two right channels.
*/
float_le gain[3][4];
// Interpolation
/// Multiplier for sample rate. Resampling occurs with the selected interpolation method.
float_le rate_multiplier;
enum class InterpolationMode : u8 {
Polyphase = 0,
Linear = 1,
None = 2,
};
InterpolationMode interpolation_mode;
INSERT_PADDING_BYTES(1); ///< Interpolation related
// Filters
/**
* This is the simplest normalized first-order digital recursive filter.
* The transfer function of this filter is:
* H(z) = b0 / (1 - a1 z^-1)
* Note the feedbackward coefficient is negated.
* Values are signed fixed point with 15 fractional bits.
*/
struct SimpleFilter {
s16_le b0;
s16_le a1;
};
/**
* This is a normalised biquad filter (second-order).
* The transfer function of this filter is:
* H(z) = (b0 + b1 z^-1 + b2 z^-2) / (1 - a1 z^-1 - a2 z^-2)
* Nintendo chose to negate the feedbackward coefficients. This differs from standard
* notation as in: https://ccrma.stanford.edu/~jos/filters/Direct_Form_I.html
* Values are signed fixed point with 14 fractional bits.
*/
struct BiquadFilter {
s16_le a2;
s16_le a1;
s16_le b2;
s16_le b1;
s16_le b0;
};
union {
u16_le filters_enabled;
BitField<0, 1, u16> simple_filter_enabled;
BitField<1, 1, u16> biquad_filter_enabled;
};
SimpleFilter simple_filter;
BiquadFilter biquad_filter;
// Buffer Queue
/// A buffer of audio data from the application, along with metadata about it.
struct Buffer {
/// Physical memory address of the start of the buffer
u32_dsp physical_address;
/// This is length in terms of samples.
/// Note that in different buffer formats a sample takes up different number of bytes.
u32_dsp length;
/// ADPCM Predictor (4 bits) and Scale (4 bits)
union {
u16_le adpcm_ps;
BitField<0, 4, u16> adpcm_scale;
BitField<4, 4, u16> adpcm_predictor;
};
/// ADPCM Historical Samples (y[n-1] and y[n-2])
u16_le adpcm_yn[2];
/// This is non-zero when the ADPCM values above are to be updated.
u8 adpcm_dirty;
/// Is a looping buffer.
u8 is_looping;
/// This value is shown in SourceStatus::previous_buffer_id when this buffer has
/// finished. This allows the emulated application to tell what buffer is currently
/// playing.
u16_le buffer_id;
INSERT_PADDING_DSPWORDS(1);
};
u16_le buffers_dirty; ///< Bitmap indicating which buffers are dirty (bit i -> buffers[i])
Buffer buffers[4]; ///< Queued Buffers
// Playback controls
u32_dsp loop_related;
u8 enable;
INSERT_PADDING_BYTES(1);
u16_le sync_count; ///< Application-side sync count (See also: SourceStatus::sync_count)
u32_dsp play_position; ///< Position. (Units: number of samples)
INSERT_PADDING_DSPWORDS(2);
// Embedded Buffer
// This buffer is often the first buffer to be used when initiating audio playback,
// after which the buffer queue is used.
u32_dsp physical_address;
/// This is length in terms of samples.
/// Note a sample takes up different number of bytes in different buffer formats.
u32_dsp length;
enum class MonoOrStereo : u16_le {
Mono = 1,
Stereo = 2,
};
enum class Format : u16_le {
PCM8 = 0,
PCM16 = 1,
ADPCM = 2,
};
union {
u16_le flags1_raw;
BitField<0, 2, MonoOrStereo> mono_or_stereo;
BitField<2, 2, Format> format;
BitField<5, 1, u16> fade_in;
};
/// ADPCM Predictor (4 bit) and Scale (4 bit)
union {
u16_le adpcm_ps;
BitField<0, 4, u16> adpcm_scale;
BitField<4, 4, u16> adpcm_predictor;
};
/// ADPCM Historical Samples (y[n-1] and y[n-2])
u16_le adpcm_yn[2];
union {
u16_le flags2_raw;
BitField<0, 1, u16> adpcm_dirty; ///< Has the ADPCM info above been changed?
BitField<1, 1, u16> is_looping; ///< Is this a looping buffer?
};
/// Buffer id of embedded buffer (used as a buffer id in SourceStatus to reference this
/// buffer).
u16_le buffer_id;
};
Configuration config[num_sources];
};
ASSERT_DSP_STRUCT(SourceConfiguration::Configuration, 192);
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 current_buffer_id_dirty; ///< Non-zero when current_buffer_id changes
u16_le sync_count; ///< Is set by the DSP to the value of SourceConfiguration::sync_count
u32_dsp buffer_position; ///< Number of samples into the current buffer
u16_le current_buffer_id; ///< Updated when a buffer finishes playing
u16_le last_buffer_id; ///< Updated when all buffers in the queue finish playing
};
Status status[num_sources];
};
ASSERT_DSP_STRUCT(SourceStatus::Status, 12);
struct DspConfiguration {
/// These dirty flags are set by the application when it updates the fields in this struct.
/// The DSP clears these each audio frame.
union {
u32_le dirty_raw;
BitField<6, 1, u32> aux_front_bypass_0_dirty;
BitField<7, 1, u32> aux_front_bypass_1_dirty;
BitField<8, 1, u32> aux_bus_enable_0_dirty;
BitField<9, 1, u32> aux_bus_enable_1_dirty;
BitField<10, 1, u32> delay_effect_0_dirty;
BitField<11, 1, u32> delay_effect_1_dirty;
BitField<12, 1, u32> reverb_effect_0_dirty;
BitField<13, 1, u32> reverb_effect_1_dirty;
BitField<15, 1, u32> output_buffer_count_dirty;
BitField<16, 1, u32> master_volume_dirty;
BitField<24, 1, u32> aux_return_volume_0_dirty;
BitField<25, 1, u32> aux_return_volume_1_dirty;
BitField<26, 1, u32> output_format_dirty;
BitField<27, 1, u32> clipping_mode_dirty;
BitField<28, 1, u32> headphones_connected_dirty;
BitField<29, 1, u32> surround_depth_dirty;
BitField<30, 1, u32> surround_speaker_position_dirty;
BitField<31, 1, u32> rear_ratio_dirty;
};
/// The DSP has three intermediate audio mixers. This controls the volume level (0.0-1.0) for
/// each at the final mixer.
float_le master_volume;
std::array<float_le, 2> aux_return_volume;
u16_le output_buffer_count;
INSERT_PADDING_DSPWORDS(2);
enum class OutputFormat : u16_le {
Mono = 0,
Stereo = 1,
Surround = 2,
};
OutputFormat output_format;
u16_le clipping_mode; ///< Not sure of the exact gain equation for the limiter.
u16_le headphones_connected; ///< Application updates the DSP on headphone status.
u16_le surround_depth;
u16_le surround_speaker_position;
INSERT_PADDING_DSPWORDS(1); ///< TODO: Surround sound related
u16_le rear_ratio;
std::array<u16_le, 2> aux_front_bypass;
std::array<u16_le, 2> aux_bus_enable;
/**
* This is delay with feedback.
* Transfer function:
* H(z) = a z^-N / (1 - b z^-1 + a g z^-N)
* where
* N = frame_count * samples_per_frame
* g, a and b are fixed point with 7 fractional bits
*/
struct DelayEffect {
/// These dirty flags are set by the application when it updates the fields in this struct.
/// The DSP clears these each audio frame.
union {
u16_le dirty_raw;
BitField<0, 1, u16> enable_dirty;
BitField<1, 1, u16> work_buffer_address_dirty;
BitField<2, 1, u16> other_dirty; ///< Set when anything else has been changed
};
u16_le enable;
INSERT_PADDING_DSPWORDS(1);
u16_le outputs;
/// The application allocates a block of memory for the DSP to use as a work buffer.
u32_dsp work_buffer_address;
/// Frames to delay by
u16_le frame_count;
// Coefficients
s16_le g; ///< Fixed point with 7 fractional bits
s16_le a; ///< Fixed point with 7 fractional bits
s16_le b; ///< Fixed point with 7 fractional bits
};
DelayEffect delay_effect[2];
struct ReverbEffect {
INSERT_PADDING_DSPWORDS(26); ///< TODO
};
ReverbEffect reverb_effect[2];
u16_le sync_mode;
INSERT_PADDING_DSPWORDS(1);
union {
u32_le dirty2_raw;
BitField<16, 1, u32> sync_mode_dirty;
};
};
ASSERT_DSP_STRUCT(DspConfiguration, 196);
ASSERT_DSP_STRUCT(DspConfiguration::DelayEffect, 20);
ASSERT_DSP_STRUCT(DspConfiguration::ReverbEffect, 52);
static_assert(offsetof(DspConfiguration, sync_mode) == 0xBC);
static_assert(offsetof(DspConfiguration, dirty2_raw) == 0xC0);
struct AdpcmCoefficients {
/// Coefficients are signed fixed point with 11 fractional bits.
/// Each source has 16 coefficients associated with it.
s16_le coeff[num_sources][16];
};
ASSERT_DSP_STRUCT(AdpcmCoefficients, 768);
struct DspStatus {
u16_le unknown;
u16_le dropped_frames;
INSERT_PADDING_DSPWORDS(0xE);
};
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[samples_per_frame][2];
};
ASSERT_DSP_STRUCT(FinalMixSamples, 640);
/// DSP writes output of intermediate mixers 1 and 2 here.
/// Writes to this region by the application edits the output of the intermediate mixers.
/// This seems to be intended to allow the application to do custom effects on the ARM11.
/// Values that exceed s16 range will be clipped by the DSP after further processing.
struct IntermediateMixSamples {
struct Samples {
s32_le pcm32[4][samples_per_frame]; ///< Little-endian as opposed to DSP middle-endian.
};
Samples mix1;
Samples mix2;
};
ASSERT_DSP_STRUCT(IntermediateMixSamples, 5120);
/// Compressor table
struct Compressor {
INSERT_PADDING_DSPWORDS(0xD20); ///< TODO
};
/// There is no easy way to implement this in a HLE implementation.
struct DspDebug {
INSERT_PADDING_DSPWORDS(0x130);
};
ASSERT_DSP_STRUCT(DspDebug, 0x260);
struct SharedMemory {
/// Padding
INSERT_PADDING_DSPWORDS(0x400);
DspStatus dsp_status;
DspDebug dsp_debug;
FinalMixSamples final_samples;
SourceStatus source_statuses;
Compressor compressor;
DspConfiguration dsp_configuration;
IntermediateMixSamples intermediate_mix_samples;
SourceConfiguration source_configurations;
AdpcmCoefficients adpcm_coefficients;
struct {
INSERT_PADDING_DSPWORDS(0x100);
} unknown10;
struct {
INSERT_PADDING_DSPWORDS(0xC0);
} unknown11;
struct {
INSERT_PADDING_DSPWORDS(0x180);
} unknown12;
struct {
INSERT_PADDING_DSPWORDS(0xA);
} unknown13;
struct {
INSERT_PADDING_DSPWORDS(0x13A3);
} unknown14;
u16_le frame_counter;
};
ASSERT_DSP_STRUCT(SharedMemory, 0x8000);
union DspMemory {
std::array<u8, 0x80000> raw_memory{};
struct {
u8 unused_0[0x50000];
SharedMemory region_0;
u8 unused_1[0x18000];
SharedMemory region_1;
u8 unused_2[0x8000];
};
};
static_assert(offsetof(DspMemory, region_0) == region0_offset,
"DSP region 0 is at the wrong offset");
static_assert(offsetof(DspMemory, region_1) == region1_offset,
"DSP region 1 is at the wrong offset");
// Structures must have an offset that is a multiple of two.
static_assert(offsetof(SharedMemory, frame_counter) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, source_statuses) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, adpcm_coefficients) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_configuration) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_status) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, final_samples) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, intermediate_mix_samples) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, compressor) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, dsp_debug) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown10) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown11) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown12) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown13) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
static_assert(offsetof(SharedMemory, unknown14) % 2 == 0,
"Structures in HLE::SharedMemory must be 2-byte aligned");
#undef INSERT_PADDING_DSPWORDS
#undef ASSERT_DSP_STRUCT
} // namespace AudioCore::HLE
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