diff options
Diffstat (limited to 'src/audio_core/hle')
| -rw-r--r-- | src/audio_core/hle/dsp.cpp | 42 | ||||
| -rw-r--r-- | src/audio_core/hle/dsp.h | 502 | ||||
| -rw-r--r-- | src/audio_core/hle/pipe.cpp | 55 | ||||
| -rw-r--r-- | src/audio_core/hle/pipe.h | 38 |
4 files changed, 637 insertions, 0 deletions
diff --git a/src/audio_core/hle/dsp.cpp b/src/audio_core/hle/dsp.cpp new file mode 100644 index 000000000..c89356edc --- /dev/null +++ b/src/audio_core/hle/dsp.cpp @@ -0,0 +1,42 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include "audio_core/hle/dsp.h" +#include "audio_core/hle/pipe.h" + +namespace DSP { +namespace HLE { + +SharedMemory g_region0; +SharedMemory g_region1; + +void Init() { + DSP::HLE::ResetPipes(); +} + +void Shutdown() { +} + +bool Tick() { + return true; +} + +SharedMemory& CurrentRegion() { + // The region with the higher frame counter is chosen unless there is wraparound. + + if (g_region0.frame_counter == 0xFFFFu && g_region1.frame_counter != 0xFFFEu) { + // Wraparound has occured. + return g_region1; + } + + if (g_region1.frame_counter == 0xFFFFu && g_region0.frame_counter != 0xFFFEu) { + // Wraparound has occured. + return g_region0; + } + + return (g_region0.frame_counter > g_region1.frame_counter) ? g_region0 : g_region1; +} + +} // namespace HLE +} // namespace DSP diff --git a/src/audio_core/hle/dsp.h b/src/audio_core/hle/dsp.h new file mode 100644 index 000000000..14c4000c6 --- /dev/null +++ b/src/audio_core/hle/dsp.h @@ -0,0 +1,502 @@ +// 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 diff --git a/src/audio_core/hle/pipe.cpp b/src/audio_core/hle/pipe.cpp new file mode 100644 index 000000000..6542c760c --- /dev/null +++ b/src/audio_core/hle/pipe.cpp @@ -0,0 +1,55 @@ +// 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/pipe.h" + +#include "common/common_types.h" +#include "common/logging/log.h" + +namespace DSP { +namespace HLE { + +static size_t pipe2position = 0; + +void ResetPipes() { + pipe2position = 0; +} + +std::vector<u8> PipeRead(u32 pipe_number, u32 length) { + if (pipe_number != 2) { + LOG_WARNING(Audio_DSP, "pipe_number = %u (!= 2), unimplemented", pipe_number); + return {}; // We currently don't handle anything other than the audio pipe. + } + + // Canned DSP responses that games expect. These were taken from HW by 3dmoo team. + // TODO: Our implementation will actually use a slightly different response than this one. + // TODO: Use offsetof on DSP structures instead for a proper response. + static const std::array<u8, 32> canned_response {{ + 0x0F, 0x00, 0xFF, 0xBF, 0x8E, 0x9E, 0x80, 0x86, 0x8E, 0xA7, 0x30, 0x94, 0x00, 0x84, 0x40, 0x85, + 0x8E, 0x94, 0x10, 0x87, 0x10, 0x84, 0x0E, 0xA9, 0x0E, 0xAA, 0xCE, 0xAA, 0x4E, 0xAC, 0x58, 0xAC + }}; + + // TODO: Move this into dsp::DSP service since it happens on the service side. + // Hardware observation: No data is returned if requested length reads beyond the end of the data in-pipe. + if (pipe2position + length > canned_response.size()) { + return {}; + } + + std::vector<u8> ret; + for (size_t i = 0; i < length; i++, pipe2position++) { + ret.emplace_back(canned_response[pipe2position]); + } + + return ret; +} + +void PipeWrite(u32 pipe_number, const std::vector<u8>& buffer) { + // TODO: proper pipe behaviour +} + +} // namespace HLE +} // namespace DSP diff --git a/src/audio_core/hle/pipe.h b/src/audio_core/hle/pipe.h new file mode 100644 index 000000000..ff6536950 --- /dev/null +++ b/src/audio_core/hle/pipe.h @@ -0,0 +1,38 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#pragma once + +#include <vector> + +#include "common/common_types.h" + +namespace DSP { +namespace HLE { + +/// Reset the pipes by setting pipe positions back to the beginning. +void ResetPipes(); + +/** + * Read a DSP pipe. + * Pipe IDs: + * pipe_number = 0: Debug + * pipe_number = 1: P-DMA + * pipe_number = 2: Audio + * pipe_number = 3: Binary + * @param pipe_number The Pipe ID + * @param length How much data to request. + * @return The data read from the pipe. The size of this vector can be less than the length requested. + */ +std::vector<u8> PipeRead(u32 pipe_number, u32 length); + +/** + * Write to a DSP pipe. + * @param pipe_number The Pipe ID + * @param buffer The data to write to the pipe. + */ +void PipeWrite(u32 pipe_number, const std::vector<u8>& buffer); + +} // namespace HLE +} // namespace DSP |