// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <cstddef>
#include <type_traits>
#include "audio_core/audio_core.h"
#include "common/bit_field.h"
#include "common/common_funcs.h"
#include "common/common_types.h"
#include "common/swap.h"
namespace DSP {
namespace 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.
struct SharedMemory;
constexpr VAddr region0_base = 0x1FF50000;
extern SharedMemory g_region0;
constexpr VAddr region1_base = 0x1FF70000;
extern SharedMemory g_region1;
/**
* 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;
};
#if (__GNUC__ >= 5) || defined(__clang__) || defined(_MSC_VER)
static_assert(std::is_trivially_copyable<u32_dsp>::value, "u32_dsp isn't trivially copyable");
#endif
// There are 15 structures in each memory region. A table of them in the order they appear in memory
// is presented below
//
// Pipe 2 # 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
//
// 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))
// GCC versions < 5.0 do not implement std::is_trivially_copyable.
// Excluding MSVC because it has weird behaviour for std::is_trivially_copyable.
#if (__GNUC__ >= 5) || defined(__clang__)
#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)
#else
#define ASSERT_DSP_STRUCT(name, size) \
static_assert(std::is_standard_layout<name>::value, "DSP structure " #name " doesn't use standard layout"); \
static_assert(sizeof(name) == (size), "Unexpected struct size for DSP structure " #name)
#endif
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<2, 1, u32_le> adpcm_coefficients_dirty;
BitField<3, 1, u32_le> partial_embedded_buffer_dirty; ///< Tends to be set when a looped buffer is queued.
BitField<16, 1, u32_le> enable_dirty;
BitField<17, 1, u32_le> interpolation_dirty;
BitField<18, 1, u32_le> rate_multiplier_dirty;
BitField<19, 1, u32_le> buffer_queue_dirty;
BitField<20, 1, u32_le> loop_related_dirty;
BitField<21, 1, u32_le> play_position_dirty; ///< Tends to also be set when embedded buffer is updated.
BitField<22, 1, u32_le> filters_enabled_dirty;
BitField<23, 1, u32_le> simple_filter_dirty;
BitField<24, 1, u32_le> biquad_filter_dirty;
BitField<25, 1, u32_le> gain_0_dirty;
BitField<26, 1, u32_le> gain_1_dirty;
BitField<27, 1, u32_le> gain_2_dirty;
BitField<28, 1, u32_le> sync_dirty;
BitField<29, 1, u32_le> reset_flag;
BitField<31, 1, u32_le> 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 {
None = 0,
Linear = 1,
Polyphase = 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)
* 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 b0;
s16_le b1;
s16_le b2;
s16_le a1;
s16_le a2;
};
union {
u16_le filters_enabled;
BitField<0, 1, u16_le> simple_filter_enabled;
BitField<1, 1, u16_le> 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_le> adpcm_scale;
BitField<4, 4, u16_le> 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; ///< Application-side sync (See also: SourceStatus::sync)
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_le> fade_in;
};
/// ADPCM Predictor (4 bit) and Scale (4 bit)
union {
u16_le adpcm_ps;
BitField<0, 4, u16_le> adpcm_scale;
BitField<4, 4, u16_le> 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_le> adpcm_dirty; ///< Has the ADPCM info above been changed?
BitField<1, 1, u16_le> 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[AudioCore::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 previous_buffer_id_dirty; ///< Non-zero when previous_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
INSERT_PADDING_DSPWORDS(1);
};
Status status[AudioCore::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<8, 1, u32_le> mixer1_enabled_dirty;
BitField<9, 1, u32_le> mixer2_enabled_dirty;
BitField<10, 1, u32_le> delay_effect_0_dirty;
BitField<11, 1, u32_le> delay_effect_1_dirty;
BitField<12, 1, u32_le> reverb_effect_0_dirty;
BitField<13, 1, u32_le> reverb_effect_1_dirty;
BitField<16, 1, u32_le> volume_0_dirty;
BitField<24, 1, u32_le> volume_1_dirty;
BitField<25, 1, u32_le> volume_2_dirty;
BitField<26, 1, u32_le> output_format_dirty;
BitField<27, 1, u32_le> limiter_enabled_dirty;
BitField<28, 1, u32_le> headphones_connected_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 volume[3];
INSERT_PADDING_DSPWORDS(3);
enum class OutputFormat : u16_le {
Mono = 0,
Stereo = 1,
Surround = 2
};
OutputFormat output_format;
u16_le limiter_enabled; ///< Not sure of the exact gain equation for the limiter.
u16_le headphones_connected; ///< Application updates the DSP on headphone status.
INSERT_PADDING_DSPWORDS(4); ///< TODO: Surround sound related
INSERT_PADDING_DSPWORDS(2); ///< TODO: Intermediate mixer 1/2 related
u16_le mixer1_enabled;
u16_le mixer2_enabled;
/**
* 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_le> enable_dirty;
BitField<1, 1, u16_le> work_buffer_address_dirty;
BitField<2, 1, u16_le> other_dirty; ///< Set when anything else has been changed
};
u16_le enable;
INSERT_PADDING_DSPWORDS(1);
u16_le outputs;
u32_dsp work_buffer_address; ///< The application allocates a block of memory for the DSP to use as a work buffer.
u16_le frame_count; ///< Frames to delay by
// 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];
INSERT_PADDING_DSPWORDS(4);
};
ASSERT_DSP_STRUCT(DspConfiguration, 196);
ASSERT_DSP_STRUCT(DspConfiguration::DelayEffect, 20);
ASSERT_DSP_STRUCT(DspConfiguration::ReverbEffect, 52);
struct AdpcmCoefficients {
/// Coefficients are signed fixed point with 11 fractional bits.
/// Each source has 16 coefficients associated with it.
s16_le coeff[AudioCore::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[2 * AudioCore::samples_per_frame];
};
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][AudioCore::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;
/// Unknown 10-14 (Surround sound related)
INSERT_PADDING_DSPWORDS(0x16ED);
u16_le frame_counter;
};
ASSERT_DSP_STRUCT(SharedMemory, 0x8000);
#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();
/// Returns a mutable reference to the current region. Current region is selected based on the frame counter.
SharedMemory& CurrentRegion();
} // namespace HLE
} // namespace DSP
