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Diffstat (limited to 'src/audio_core/hle/dsp.h')
| -rw-r--r-- | src/audio_core/hle/dsp.h | 595 |
1 files changed, 0 insertions, 595 deletions
diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h deleted file mode 100644 index 94ce48863..000000000 --- a/src/audio_core/hle/dsp.h +++ /dev/null @@ -1,595 +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 <cstddef> -#include <memory> -#include <type_traits> -#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 { -class Sink; -} - -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. - -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; -}; -#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: -// -// # 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: DSP::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)) - -// 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<0, 1, u32_le> format_dirty; - BitField<1, 1, u32_le> mono_or_stereo_dirty; - BitField<2, 1, u32_le> adpcm_coefficients_dirty; - /// Tends to be set when a looped buffer is queued. - BitField<3, 1, u32_le> partial_embedded_buffer_dirty; - BitField<4, 1, u32_le> partial_reset_flag; - - 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; - /// Tends to also be set when embedded buffer is updated. - BitField<21, 1, u32_le> play_position_dirty; - 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<30, 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 { - 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_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[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; ///< Is set by the DSP to the value of SourceConfiguration::sync - u32_dsp buffer_position; ///< Number of samples into the current buffer - u16_le current_buffer_id; ///< Updated when a buffer finishes playing - INSERT_PADDING_DSPWORDS(1); - }; - - 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<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; - /// 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]; - - 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[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"); - -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"); -static_assert(offsetof(SharedMemory, source_configurations) % 2 == 0, - "Structures in DSP::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"); -static_assert(offsetof(SharedMemory, adpcm_coefficients) % 2 == 0, - "Structures in DSP::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"); -static_assert(offsetof(SharedMemory, dsp_status) % 2 == 0, - "Structures in DSP::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"); -static_assert(offsetof(SharedMemory, intermediate_mix_samples) % 2 == 0, - "Structures in DSP::HLE::SharedMemory must be 2-byte aligned"); -static_assert(offsetof(SharedMemory, compressor) % 2 == 0, - "Structures in DSP::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"); -static_assert(offsetof(SharedMemory, unknown10) % 2 == 0, - "Structures in DSP::HLE::SharedMemory must be 2-byte aligned"); -static_assert(offsetof(SharedMemory, unknown11) % 2 == 0, - "Structures in DSP::HLE::SharedMemory must be 2-byte aligned"); -static_assert(offsetof(SharedMemory, unknown12) % 2 == 0, - "Structures in DSP::HLE::SharedMemory must be 2-byte aligned"); -static_assert(offsetof(SharedMemory, unknown13) % 2 == 0, - "Structures in DSP::HLE::SharedMemory must be 2-byte aligned"); -static_assert(offsetof(SharedMemory, unknown14) % 2 == 0, - "Structures in DSP::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 |