diff options
Diffstat (limited to 'src/audio_core/renderer/command/effect')
18 files changed, 2611 insertions, 0 deletions
diff --git a/src/audio_core/renderer/command/effect/aux_.cpp b/src/audio_core/renderer/command/effect/aux_.cpp new file mode 100644 index 000000000..e76db893f --- /dev/null +++ b/src/audio_core/renderer/command/effect/aux_.cpp @@ -0,0 +1,207 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/aux_.h" +#include "audio_core/renderer/effect/aux_.h" +#include "core/memory.h" + +namespace AudioCore::AudioRenderer { +/** + * Reset an AuxBuffer. + * + * @param memory - Core memory for writing. + * @param aux_info - Memory address pointing to the AuxInfo to reset. + */ +static void ResetAuxBufferDsp(Core::Memory::Memory& memory, const CpuAddr aux_info) { + if (aux_info == 0) { + LOG_ERROR(Service_Audio, "Aux info is 0!"); + return; + } + + auto info{reinterpret_cast<AuxInfo::AuxInfoDsp*>(memory.GetPointer(aux_info))}; + info->read_offset = 0; + info->write_offset = 0; + info->total_sample_count = 0; +} + +/** + * Write the given input mix buffer to the memory at send_buffer, and update send_info_ if + * update_count is set, to notify the game that an update happened. + * + * @param memory - Core memory for writing. + * @param send_info_ - Meta information for where to write the mix buffer. + * @param sample_count - Unused. + * @param send_buffer - Memory address to write the mix buffer to. + * @param count_max - Maximum number of samples in the receiving buffer. + * @param input - Input mix buffer to write. + * @param write_count_ - Number of samples to write. + * @param write_offset - Current offset to begin writing the receiving buffer at. + * @param update_count - If non-zero, send_info_ will be updated. + * @return Number of samples written. + */ +static u32 WriteAuxBufferDsp(Core::Memory::Memory& memory, const CpuAddr send_info_, + [[maybe_unused]] u32 sample_count, const CpuAddr send_buffer, + const u32 count_max, std::span<const s32> input, + const u32 write_count_, const u32 write_offset, + const u32 update_count) { + if (write_count_ > count_max) { + LOG_ERROR(Service_Audio, + "write_count must be smaller than count_max! write_count {}, count_max {}", + write_count_, count_max); + return 0; + } + + if (input.empty()) { + LOG_ERROR(Service_Audio, "input buffer is empty!"); + return 0; + } + + if (send_buffer == 0) { + LOG_ERROR(Service_Audio, "send_buffer is 0!"); + return 0; + } + + if (count_max == 0) { + return 0; + } + + AuxInfo::AuxInfoDsp send_info{}; + memory.ReadBlockUnsafe(send_info_, &send_info, sizeof(AuxInfo::AuxInfoDsp)); + + u32 target_write_offset{send_info.write_offset + write_offset}; + if (target_write_offset > count_max || write_count_ == 0) { + return 0; + } + + u32 write_count{write_count_}; + u32 write_pos{0}; + while (write_count > 0) { + u32 to_write{std::min(count_max - target_write_offset, write_count)}; + + if (to_write > 0) { + memory.WriteBlockUnsafe(send_buffer + target_write_offset * sizeof(s32), + &input[write_pos], to_write * sizeof(s32)); + } + + target_write_offset = (target_write_offset + to_write) % count_max; + write_count -= to_write; + write_pos += to_write; + } + + if (update_count) { + send_info.write_offset = (send_info.write_offset + update_count) % count_max; + } + + memory.WriteBlockUnsafe(send_info_, &send_info, sizeof(AuxInfo::AuxInfoDsp)); + + return write_count_; +} + +/** + * Read the given memory at return_buffer into the output mix buffer, and update return_info_ if + * update_count is set, to notify the game that an update happened. + * + * @param memory - Core memory for writing. + * @param return_info_ - Meta information for where to read the mix buffer. + * @param return_buffer - Memory address to read the samples from. + * @param count_max - Maximum number of samples in the receiving buffer. + * @param output - Output mix buffer which will receive the samples. + * @param count_ - Number of samples to read. + * @param read_offset - Current offset to begin reading the return_buffer at. + * @param update_count - If non-zero, send_info_ will be updated. + * @return Number of samples read. + */ +static u32 ReadAuxBufferDsp(Core::Memory::Memory& memory, const CpuAddr return_info_, + const CpuAddr return_buffer, const u32 count_max, std::span<s32> output, + const u32 count_, const u32 read_offset, const u32 update_count) { + if (count_max == 0) { + return 0; + } + + if (count_ > count_max) { + LOG_ERROR(Service_Audio, "count must be smaller than count_max! count {}, count_max {}", + count_, count_max); + return 0; + } + + if (output.empty()) { + LOG_ERROR(Service_Audio, "output buffer is empty!"); + return 0; + } + + if (return_buffer == 0) { + LOG_ERROR(Service_Audio, "return_buffer is 0!"); + return 0; + } + + AuxInfo::AuxInfoDsp return_info{}; + memory.ReadBlockUnsafe(return_info_, &return_info, sizeof(AuxInfo::AuxInfoDsp)); + + u32 target_read_offset{return_info.read_offset + read_offset}; + if (target_read_offset > count_max) { + return 0; + } + + u32 read_count{count_}; + u32 read_pos{0}; + while (read_count > 0) { + u32 to_read{std::min(count_max - target_read_offset, read_count)}; + + if (to_read > 0) { + memory.ReadBlockUnsafe(return_buffer + target_read_offset * sizeof(s32), + &output[read_pos], to_read * sizeof(s32)); + } + + target_read_offset = (target_read_offset + to_read) % count_max; + read_count -= to_read; + read_pos += to_read; + } + + if (update_count) { + return_info.read_offset = (return_info.read_offset + update_count) % count_max; + } + + memory.WriteBlockUnsafe(return_info_, &return_info, sizeof(AuxInfo::AuxInfoDsp)); + + return count_; +} + +void AuxCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format("AuxCommand\n\tenabled {} input {:02X} output {:02X}\n", effect_enabled, + input, output); +} + +void AuxCommand::Process(const ADSP::CommandListProcessor& processor) { + auto input_buffer{ + processor.mix_buffers.subspan(input * processor.sample_count, processor.sample_count)}; + auto output_buffer{ + processor.mix_buffers.subspan(output * processor.sample_count, processor.sample_count)}; + + if (effect_enabled) { + WriteAuxBufferDsp(*processor.memory, send_buffer_info, processor.sample_count, send_buffer, + count_max, input_buffer, processor.sample_count, write_offset, + update_count); + + auto read{ReadAuxBufferDsp(*processor.memory, return_buffer_info, return_buffer, count_max, + output_buffer, processor.sample_count, write_offset, + update_count)}; + + if (read != processor.sample_count) { + std::memset(&output_buffer[read], 0, processor.sample_count - read); + } + } else { + ResetAuxBufferDsp(*processor.memory, send_buffer_info); + ResetAuxBufferDsp(*processor.memory, return_buffer_info); + if (input != output) { + std::memcpy(output_buffer.data(), input_buffer.data(), output_buffer.size_bytes()); + } + } +} + +bool AuxCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/aux_.h b/src/audio_core/renderer/command/effect/aux_.h new file mode 100644 index 000000000..825c93732 --- /dev/null +++ b/src/audio_core/renderer/command/effect/aux_.h @@ -0,0 +1,66 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command to read and write an auxiliary buffer, writing the input mix buffer to game + * memory, and reading into the output buffer from game memory. + */ +struct AuxCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer index + s16 input; + /// Output mix buffer index + s16 output; + /// Meta info for writing + CpuAddr send_buffer_info; + /// Meta info for reading + CpuAddr return_buffer_info; + /// Game memory write buffer + CpuAddr send_buffer; + /// Game memory read buffer + CpuAddr return_buffer; + /// Max samples to read/write + u32 count_max; + /// Current read/write offset + u32 write_offset; + /// Number of samples to update per call + u32 update_count; + /// is this effect enabled? + bool effect_enabled; +}; + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/biquad_filter.cpp b/src/audio_core/renderer/command/effect/biquad_filter.cpp new file mode 100644 index 000000000..1baae74fd --- /dev/null +++ b/src/audio_core/renderer/command/effect/biquad_filter.cpp @@ -0,0 +1,118 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/biquad_filter.h" +#include "audio_core/renderer/voice/voice_state.h" + +namespace AudioCore::AudioRenderer { +/** + * Biquad filter float implementation. + * + * @param output - Output container for filtered samples. + * @param input - Input container for samples to be filtered. + * @param b - Feedforward coefficients. + * @param a - Feedback coefficients. + * @param state - State to track previous samples between calls. + * @param sample_count - Number of samples to process. + */ +void ApplyBiquadFilterFloat(std::span<s32> output, std::span<const s32> input, + std::array<s16, 3>& b_, std::array<s16, 2>& a_, + VoiceState::BiquadFilterState& state, const u32 sample_count) { + constexpr s64 min{std::numeric_limits<s32>::min()}; + constexpr s64 max{std::numeric_limits<s32>::max()}; + std::array<f64, 3> b{Common::FixedPoint<50, 14>::from_base(b_[0]).to_double(), + Common::FixedPoint<50, 14>::from_base(b_[1]).to_double(), + Common::FixedPoint<50, 14>::from_base(b_[2]).to_double()}; + std::array<f64, 2> a{Common::FixedPoint<50, 14>::from_base(a_[0]).to_double(), + Common::FixedPoint<50, 14>::from_base(a_[1]).to_double()}; + std::array<f64, 4> s{state.s0.to_double(), state.s1.to_double(), state.s2.to_double(), + state.s3.to_double()}; + + for (u32 i = 0; i < sample_count; i++) { + f64 in_sample{static_cast<f64>(input[i])}; + auto sample{in_sample * b[0] + s[0] * b[1] + s[1] * b[2] + s[2] * a[0] + s[3] * a[1]}; + + output[i] = static_cast<s32>(std::clamp(static_cast<s64>(sample), min, max)); + + s[1] = s[0]; + s[0] = in_sample; + s[3] = s[2]; + s[2] = sample; + } + + state.s0 = s[0]; + state.s1 = s[1]; + state.s2 = s[2]; + state.s3 = s[3]; +} + +/** + * Biquad filter s32 implementation. + * + * @param output - Output container for filtered samples. + * @param input - Input container for samples to be filtered. + * @param b - Feedforward coefficients. + * @param a - Feedback coefficients. + * @param state - State to track previous samples between calls. + * @param sample_count - Number of samples to process. + */ +static void ApplyBiquadFilterInt(std::span<s32> output, std::span<const s32> input, + std::array<s16, 3>& b_, std::array<s16, 2>& a_, + VoiceState::BiquadFilterState& state, const u32 sample_count) { + constexpr s64 min{std::numeric_limits<s32>::min()}; + constexpr s64 max{std::numeric_limits<s32>::max()}; + std::array<Common::FixedPoint<50, 14>, 3> b{ + Common::FixedPoint<50, 14>::from_base(b_[0]), + Common::FixedPoint<50, 14>::from_base(b_[1]), + Common::FixedPoint<50, 14>::from_base(b_[2]), + }; + std::array<Common::FixedPoint<50, 14>, 3> a{ + Common::FixedPoint<50, 14>::from_base(a_[0]), + Common::FixedPoint<50, 14>::from_base(a_[1]), + }; + + for (u32 i = 0; i < sample_count; i++) { + s64 in_sample{input[i]}; + auto sample{in_sample * b[0] + state.s0}; + const auto out_sample{std::clamp(sample.to_long(), min, max)}; + + output[i] = static_cast<s32>(out_sample); + + state.s0 = state.s1 + b[1] * in_sample + a[0] * out_sample; + state.s1 = 0 + b[2] * in_sample + a[1] * out_sample; + } +} + +void BiquadFilterCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format( + "BiquadFilterCommand\n\tinput {:02X} output {:02X} needs_init {} use_float_processing {}\n", + input, output, needs_init, use_float_processing); +} + +void BiquadFilterCommand::Process(const ADSP::CommandListProcessor& processor) { + auto state_{reinterpret_cast<VoiceState::BiquadFilterState*>(state)}; + if (needs_init) { + std::memset(state_, 0, sizeof(VoiceState::BiquadFilterState)); + } + + auto input_buffer{ + processor.mix_buffers.subspan(input * processor.sample_count, processor.sample_count)}; + auto output_buffer{ + processor.mix_buffers.subspan(output * processor.sample_count, processor.sample_count)}; + + if (use_float_processing) { + ApplyBiquadFilterFloat(output_buffer, input_buffer, biquad.b, biquad.a, *state_, + processor.sample_count); + } else { + ApplyBiquadFilterInt(output_buffer, input_buffer, biquad.b, biquad.a, *state_, + processor.sample_count); + } +} + +bool BiquadFilterCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/biquad_filter.h b/src/audio_core/renderer/command/effect/biquad_filter.h new file mode 100644 index 000000000..4c9c42d29 --- /dev/null +++ b/src/audio_core/renderer/command/effect/biquad_filter.h @@ -0,0 +1,74 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "audio_core/renderer/voice/voice_info.h" +#include "audio_core/renderer/voice/voice_state.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for applying a biquad filter to the input mix buffer, saving the results to + * the output mix buffer. + */ +struct BiquadFilterCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer index + s16 input; + /// Output mix buffer index + s16 output; + /// Input parameters for biquad + VoiceInfo::BiquadFilterParameter biquad; + /// Biquad state, updated each call + CpuAddr state; + /// If true, reset the state + bool needs_init; + /// If true, use float processing rather than int + bool use_float_processing; +}; + +/** + * Biquad filter float implementation. + * + * @param output - Output container for filtered samples. + * @param input - Input container for samples to be filtered. + * @param b - Feedforward coefficients. + * @param a - Feedback coefficients. + * @param state - State to track previous samples. + * @param sample_count - Number of samples to process. + */ +void ApplyBiquadFilterFloat(std::span<s32> output, std::span<const s32> input, + std::array<s16, 3>& b, std::array<s16, 2>& a, + VoiceState::BiquadFilterState& state, const u32 sample_count); + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/capture.cpp b/src/audio_core/renderer/command/effect/capture.cpp new file mode 100644 index 000000000..042fd286e --- /dev/null +++ b/src/audio_core/renderer/command/effect/capture.cpp @@ -0,0 +1,142 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/capture.h" +#include "audio_core/renderer/effect/aux_.h" +#include "core/memory.h" + +namespace AudioCore::AudioRenderer { +/** + * Reset an AuxBuffer. + * + * @param memory - Core memory for writing. + * @param aux_info - Memory address pointing to the AuxInfo to reset. + */ +static void ResetAuxBufferDsp(Core::Memory::Memory& memory, const CpuAddr aux_info) { + if (aux_info == 0) { + LOG_ERROR(Service_Audio, "Aux info is 0!"); + return; + } + + memory.Write32(VAddr(aux_info + offsetof(AuxInfo::AuxInfoDsp, read_offset)), 0); + memory.Write32(VAddr(aux_info + offsetof(AuxInfo::AuxInfoDsp, write_offset)), 0); + memory.Write32(VAddr(aux_info + offsetof(AuxInfo::AuxInfoDsp, total_sample_count)), 0); +} + +/** + * Write the given input mix buffer to the memory at send_buffer, and update send_info_ if + * update_count is set, to notify the game that an update happened. + * + * @param memory - Core memory for writing. + * @param send_info_ - Header information for where to write the mix buffer. + * @param send_buffer - Memory address to write the mix buffer to. + * @param count_max - Maximum number of samples in the receiving buffer. + * @param input - Input mix buffer to write. + * @param write_count_ - Number of samples to write. + * @param write_offset - Current offset to begin writing the receiving buffer at. + * @param update_count - If non-zero, send_info_ will be updated. + * @return Number of samples written. + */ +static u32 WriteAuxBufferDsp(Core::Memory::Memory& memory, const CpuAddr send_info_, + const CpuAddr send_buffer, u32 count_max, std::span<const s32> input, + const u32 write_count_, const u32 write_offset, + const u32 update_count) { + if (write_count_ > count_max) { + LOG_ERROR(Service_Audio, + "write_count must be smaller than count_max! write_count {}, count_max {}", + write_count_, count_max); + return 0; + } + + if (send_info_ == 0) { + LOG_ERROR(Service_Audio, "send_info is 0!"); + return 0; + } + + if (input.empty()) { + LOG_ERROR(Service_Audio, "input buffer is empty!"); + return 0; + } + + if (send_buffer == 0) { + LOG_ERROR(Service_Audio, "send_buffer is 0!"); + return 0; + } + + if (count_max == 0) { + return 0; + } + + AuxInfo::AuxBufferInfo send_info{}; + memory.ReadBlockUnsafe(send_info_, &send_info, sizeof(AuxInfo::AuxBufferInfo)); + + u32 target_write_offset{send_info.dsp_info.write_offset + write_offset}; + if (target_write_offset > count_max || write_count_ == 0) { + return 0; + } + + u32 write_count{write_count_}; + u32 write_pos{0}; + while (write_count > 0) { + u32 to_write{std::min(count_max - target_write_offset, write_count)}; + + if (to_write > 0) { + memory.WriteBlockUnsafe(send_buffer + target_write_offset * sizeof(s32), + &input[write_pos], to_write * sizeof(s32)); + } + + target_write_offset = (target_write_offset + to_write) % count_max; + write_count -= to_write; + write_pos += to_write; + } + + if (update_count) { + const auto count_diff{send_info.dsp_info.total_sample_count - + send_info.cpu_info.total_sample_count}; + if (count_diff >= count_max) { + auto dsp_lost_count{send_info.dsp_info.lost_sample_count + update_count}; + if (dsp_lost_count - send_info.cpu_info.lost_sample_count < + send_info.dsp_info.lost_sample_count - send_info.cpu_info.lost_sample_count) { + dsp_lost_count = send_info.cpu_info.lost_sample_count - 1; + } + send_info.dsp_info.lost_sample_count = dsp_lost_count; + } + + send_info.dsp_info.write_offset = + (send_info.dsp_info.write_offset + update_count + count_max) % count_max; + + auto new_sample_count{send_info.dsp_info.total_sample_count + update_count}; + if (new_sample_count - send_info.cpu_info.total_sample_count < count_diff) { + new_sample_count = send_info.cpu_info.total_sample_count - 1; + } + send_info.dsp_info.total_sample_count = new_sample_count; + } + + memory.WriteBlockUnsafe(send_info_, &send_info, sizeof(AuxInfo::AuxBufferInfo)); + + return write_count_; +} + +void CaptureCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format("CaptureCommand\n\tenabled {} input {:02X} output {:02X}", effect_enabled, + input, output); +} + +void CaptureCommand::Process(const ADSP::CommandListProcessor& processor) { + if (effect_enabled) { + auto input_buffer{ + processor.mix_buffers.subspan(input * processor.sample_count, processor.sample_count)}; + WriteAuxBufferDsp(*processor.memory, send_buffer_info, send_buffer, count_max, input_buffer, + processor.sample_count, write_offset, update_count); + } else { + ResetAuxBufferDsp(*processor.memory, send_buffer_info); + } +} + +bool CaptureCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/capture.h b/src/audio_core/renderer/command/effect/capture.h new file mode 100644 index 000000000..8670acb24 --- /dev/null +++ b/src/audio_core/renderer/command/effect/capture.h @@ -0,0 +1,62 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for capturing a mix buffer. That is, writing it back to a given game memory + * address. + */ +struct CaptureCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer index + s16 input; + /// Output mix buffer index + s16 output; + /// Meta info for writing + CpuAddr send_buffer_info; + /// Game memory write buffer + CpuAddr send_buffer; + /// Max samples to read/write + u32 count_max; + /// Current read/write offset + u32 write_offset; + /// Number of samples to update per call + u32 update_count; + /// is this effect enabled? + bool effect_enabled; +}; + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/compressor.cpp b/src/audio_core/renderer/command/effect/compressor.cpp new file mode 100644 index 000000000..2ebc140f1 --- /dev/null +++ b/src/audio_core/renderer/command/effect/compressor.cpp @@ -0,0 +1,156 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include <cmath> +#include <span> +#include <vector> + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/compressor.h" +#include "audio_core/renderer/effect/compressor.h" + +namespace AudioCore::AudioRenderer { + +static void SetCompressorEffectParameter(CompressorInfo::ParameterVersion2& params, + CompressorInfo::State& state) { + const auto ratio{1.0f / params.compressor_ratio}; + auto makeup_gain{0.0f}; + if (params.makeup_gain_enabled) { + makeup_gain = (params.threshold * 0.5f) * (ratio - 1.0f) - 3.0f; + } + state.makeup_gain = makeup_gain; + state.unk_18 = params.unk_28; + + const auto a{(params.out_gain + makeup_gain) / 20.0f * 3.3219f}; + const auto b{(a - std::trunc(a)) * 0.69315f}; + const auto c{std::pow(2.0f, b)}; + + state.unk_0C = (1.0f - ratio) / 6.0f; + state.unk_14 = params.threshold + 1.5f; + state.unk_10 = params.threshold - 1.5f; + state.unk_20 = c; +} + +static void InitializeCompressorEffect(CompressorInfo::ParameterVersion2& params, + CompressorInfo::State& state) { + std::memset(&state, 0, sizeof(CompressorInfo::State)); + + state.unk_00 = 0; + state.unk_04 = 1.0f; + state.unk_08 = 1.0f; + + SetCompressorEffectParameter(params, state); +} + +static void ApplyCompressorEffect(CompressorInfo::ParameterVersion2& params, + CompressorInfo::State& state, bool enabled, + std::vector<std::span<const s32>> input_buffers, + std::vector<std::span<s32>> output_buffers, u32 sample_count) { + if (enabled) { + auto state_00{state.unk_00}; + auto state_04{state.unk_04}; + auto state_08{state.unk_08}; + auto state_18{state.unk_18}; + + for (u32 i = 0; i < sample_count; i++) { + auto a{0.0f}; + for (s16 channel = 0; channel < params.channel_count; channel++) { + const auto input_sample{Common::FixedPoint<49, 15>(input_buffers[channel][i])}; + a += (input_sample * input_sample).to_float(); + } + + state_00 += params.unk_24 * ((a / params.channel_count) - state.unk_00); + + auto b{-100.0f}; + auto c{0.0f}; + if (state_00 >= 1.0e-10) { + b = std::log10(state_00) * 10.0f; + c = 1.0f; + } + + if (b >= state.unk_10) { + const auto d{b >= state.unk_14 + ? ((1.0f / params.compressor_ratio) - 1.0f) * + (b - params.threshold) + : (b - state.unk_10) * (b - state.unk_10) * -state.unk_0C}; + const auto e{d / 20.0f * 3.3219f}; + const auto f{(e - std::trunc(e)) * 0.69315f}; + c = std::pow(2.0f, f); + } + + state_18 = params.unk_28; + auto tmp{c}; + if ((state_04 - c) <= 0.08f) { + state_18 = params.unk_2C; + if (((state_04 - c) >= -0.08f) && (std::abs(state_08 - c) >= 0.001f)) { + tmp = state_04; + } + } + + state_04 = tmp; + state_08 += (c - state_08) * state_18; + + for (s16 channel = 0; channel < params.channel_count; channel++) { + output_buffers[channel][i] = static_cast<s32>( + static_cast<f32>(input_buffers[channel][i]) * state_08 * state.unk_20); + } + } + + state.unk_00 = state_00; + state.unk_04 = state_04; + state.unk_08 = state_08; + state.unk_18 = state_18; + } else { + for (s16 channel = 0; channel < params.channel_count; channel++) { + if (params.inputs[channel] != params.outputs[channel]) { + std::memcpy((char*)output_buffers[channel].data(), + (char*)input_buffers[channel].data(), + output_buffers[channel].size_bytes()); + } + } + } +} + +void CompressorCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format("CompressorCommand\n\tenabled {} \n\tinputs: ", effect_enabled); + for (s16 i = 0; i < parameter.channel_count; i++) { + string += fmt::format("{:02X}, ", inputs[i]); + } + string += "\n\toutputs: "; + for (s16 i = 0; i < parameter.channel_count; i++) { + string += fmt::format("{:02X}, ", outputs[i]); + } + string += "\n"; +} + +void CompressorCommand::Process(const ADSP::CommandListProcessor& processor) { + std::vector<std::span<const s32>> input_buffers(parameter.channel_count); + std::vector<std::span<s32>> output_buffers(parameter.channel_count); + + for (s16 i = 0; i < parameter.channel_count; i++) { + input_buffers[i] = processor.mix_buffers.subspan(inputs[i] * processor.sample_count, + processor.sample_count); + output_buffers[i] = processor.mix_buffers.subspan(outputs[i] * processor.sample_count, + processor.sample_count); + } + + auto state_{reinterpret_cast<CompressorInfo::State*>(state)}; + + if (effect_enabled) { + if (parameter.state == CompressorInfo::ParameterState::Updating) { + SetCompressorEffectParameter(parameter, *state_); + } else if (parameter.state == CompressorInfo::ParameterState::Initialized) { + InitializeCompressorEffect(parameter, *state_); + } + } + + ApplyCompressorEffect(parameter, *state_, effect_enabled, input_buffers, output_buffers, + processor.sample_count); +} + +bool CompressorCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/compressor.h b/src/audio_core/renderer/command/effect/compressor.h new file mode 100644 index 000000000..f8e96cb43 --- /dev/null +++ b/src/audio_core/renderer/command/effect/compressor.h @@ -0,0 +1,60 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <array> +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "audio_core/renderer/effect/compressor.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for limiting volume between a high and low threshold. + * Version 1. + */ +struct CompressorCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer offsets for each channel + std::array<s16, MaxChannels> inputs; + /// Output mix buffer offsets for each channel + std::array<s16, MaxChannels> outputs; + /// Input parameters + CompressorInfo::ParameterVersion2 parameter; + /// State, updated each call + CpuAddr state; + /// Game-supplied workbuffer (Unused) + CpuAddr workbuffer; + /// Is this effect enabled? + bool effect_enabled; +}; + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/delay.cpp b/src/audio_core/renderer/command/effect/delay.cpp new file mode 100644 index 000000000..a4e408d40 --- /dev/null +++ b/src/audio_core/renderer/command/effect/delay.cpp @@ -0,0 +1,238 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/delay.h" + +namespace AudioCore::AudioRenderer { +/** + * Update the DelayInfo state according to the given parameters. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + */ +static void SetDelayEffectParameter(const DelayInfo::ParameterVersion1& params, + DelayInfo::State& state) { + auto channel_spread{params.channel_spread}; + state.feedback_gain = params.feedback_gain * 0.97998046875f; + state.delay_feedback_gain = state.feedback_gain * (1.0f - channel_spread); + if (params.channel_count == 4 || params.channel_count == 6) { + channel_spread >>= 1; + } + state.delay_feedback_cross_gain = channel_spread * state.feedback_gain; + state.lowpass_feedback_gain = params.lowpass_amount * 0.949951171875f; + state.lowpass_gain = 1.0f - state.lowpass_feedback_gain; +} + +/** + * Initialize a new DelayInfo state according to the given parameters. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + * @param workbuffer - Game-supplied memory for the state. (Unused) + */ +static void InitializeDelayEffect(const DelayInfo::ParameterVersion1& params, + DelayInfo::State& state, + [[maybe_unused]] const CpuAddr workbuffer) { + state = {}; + + for (u32 channel = 0; channel < params.channel_count; channel++) { + Common::FixedPoint<32, 32> sample_count_max{0.064f}; + sample_count_max *= params.sample_rate.to_int_floor() * params.delay_time_max; + + Common::FixedPoint<18, 14> delay_time{params.delay_time}; + delay_time *= params.sample_rate / 1000; + Common::FixedPoint<32, 32> sample_count{delay_time}; + + if (sample_count > sample_count_max) { + sample_count = sample_count_max; + } + + state.delay_lines[channel].sample_count_max = sample_count_max.to_int_floor(); + state.delay_lines[channel].sample_count = sample_count.to_int_floor(); + state.delay_lines[channel].buffer.resize(state.delay_lines[channel].sample_count, 0); + if (state.delay_lines[channel].buffer.size() == 0) { + state.delay_lines[channel].buffer.push_back(0); + } + state.delay_lines[channel].buffer_pos = 0; + state.delay_lines[channel].decay_rate = 1.0f; + } + + SetDelayEffectParameter(params, state); +} + +/** + * Delay effect impl, according to the parameters and current state, on the input mix buffers, + * saving the results to the output mix buffers. + * + * @tparam NumChannels - Number of channels to process. 1-6. + * @param params - Input parameters to use. + * @param state - State to use, must be initialized (see InitializeDelayEffect). + * @param inputs - Input mix buffers to performan the delay on. + * @param outputs - Output mix buffers to receive the delayed samples. + * @param sample_count - Number of samples to process. + */ +template <size_t NumChannels> +static void ApplyDelay(const DelayInfo::ParameterVersion1& params, DelayInfo::State& state, + std::vector<std::span<const s32>>& inputs, + std::vector<std::span<s32>>& outputs, const u32 sample_count) { + for (u32 sample_index = 0; sample_index < sample_count; sample_index++) { + std::array<Common::FixedPoint<50, 14>, NumChannels> input_samples{}; + for (u32 channel = 0; channel < NumChannels; channel++) { + input_samples[channel] = inputs[channel][sample_index] * 64; + } + + std::array<Common::FixedPoint<50, 14>, NumChannels> delay_samples{}; + for (u32 channel = 0; channel < NumChannels; channel++) { + delay_samples[channel] = state.delay_lines[channel].Read(); + } + + // clang-format off + std::array<std::array<Common::FixedPoint<18, 14>, NumChannels>, NumChannels> matrix{}; + if constexpr (NumChannels == 1) { + matrix = {{ + {state.feedback_gain}, + }}; + } else if constexpr (NumChannels == 2) { + matrix = {{ + {state.delay_feedback_gain, state.delay_feedback_cross_gain}, + {state.delay_feedback_cross_gain, state.delay_feedback_gain}, + }}; + } else if constexpr (NumChannels == 4) { + matrix = {{ + {state.delay_feedback_gain, state.delay_feedback_cross_gain, state.delay_feedback_cross_gain, 0.0f}, + {state.delay_feedback_cross_gain, state.delay_feedback_gain, 0.0f, state.delay_feedback_cross_gain}, + {state.delay_feedback_cross_gain, 0.0f, state.delay_feedback_gain, state.delay_feedback_cross_gain}, + {0.0f, state.delay_feedback_cross_gain, state.delay_feedback_cross_gain, state.delay_feedback_gain}, + }}; + } else if constexpr (NumChannels == 6) { + matrix = {{ + {state.delay_feedback_gain, 0.0f, state.delay_feedback_cross_gain, 0.0f, state.delay_feedback_cross_gain, 0.0f}, + {0.0f, state.delay_feedback_gain, state.delay_feedback_cross_gain, 0.0f, 0.0f, state.delay_feedback_cross_gain}, + {state.delay_feedback_cross_gain, state.delay_feedback_cross_gain, state.delay_feedback_gain, 0.0f, 0.0f, 0.0f}, + {0.0f, 0.0f, 0.0f, params.feedback_gain, 0.0f, 0.0f}, + {state.delay_feedback_cross_gain, 0.0f, 0.0f, 0.0f, state.delay_feedback_gain, state.delay_feedback_cross_gain}, + {0.0f, state.delay_feedback_cross_gain, 0.0f, 0.0f, state.delay_feedback_cross_gain, state.delay_feedback_gain}, + }}; + } + // clang-format on + + std::array<Common::FixedPoint<50, 14>, NumChannels> gained_samples{}; + for (u32 channel = 0; channel < NumChannels; channel++) { + Common::FixedPoint<50, 14> delay{}; + for (u32 j = 0; j < NumChannels; j++) { + delay += delay_samples[j] * matrix[j][channel]; + } + gained_samples[channel] = input_samples[channel] * params.in_gain + delay; + } + + for (u32 channel = 0; channel < NumChannels; channel++) { + state.lowpass_z[channel] = gained_samples[channel] * state.lowpass_gain + + state.lowpass_z[channel] * state.lowpass_feedback_gain; + state.delay_lines[channel].Write(state.lowpass_z[channel]); + } + + for (u32 channel = 0; channel < NumChannels; channel++) { + outputs[channel][sample_index] = (input_samples[channel] * params.dry_gain + + delay_samples[channel] * params.wet_gain) + .to_int_floor() / + 64; + } + } +} + +/** + * Apply a delay effect if enabled, according to the parameters and current state, on the input mix + * buffers, saving the results to the output mix buffers. + * + * @param params - Input parameters to use. + * @param state - State to use, must be initialized (see InitializeDelayEffect). + * @param enabled - If enabled, delay will be applied, otherwise input is copied to output. + * @param inputs - Input mix buffers to performan the delay on. + * @param outputs - Output mix buffers to receive the delayed samples. + * @param sample_count - Number of samples to process. + */ +static void ApplyDelayEffect(const DelayInfo::ParameterVersion1& params, DelayInfo::State& state, + const bool enabled, std::vector<std::span<const s32>>& inputs, + std::vector<std::span<s32>>& outputs, const u32 sample_count) { + + if (!IsChannelCountValid(params.channel_count)) { + LOG_ERROR(Service_Audio, "Invalid delay channels {}", params.channel_count); + return; + } + + if (enabled) { + switch (params.channel_count) { + case 1: + ApplyDelay<1>(params, state, inputs, outputs, sample_count); + break; + case 2: + ApplyDelay<2>(params, state, inputs, outputs, sample_count); + break; + case 4: + ApplyDelay<4>(params, state, inputs, outputs, sample_count); + break; + case 6: + ApplyDelay<6>(params, state, inputs, outputs, sample_count); + break; + default: + for (u32 channel = 0; channel < params.channel_count; channel++) { + if (inputs[channel].data() != outputs[channel].data()) { + std::memcpy(outputs[channel].data(), inputs[channel].data(), + sample_count * sizeof(s32)); + } + } + break; + } + } else { + for (u32 channel = 0; channel < params.channel_count; channel++) { + if (inputs[channel].data() != outputs[channel].data()) { + std::memcpy(outputs[channel].data(), inputs[channel].data(), + sample_count * sizeof(s32)); + } + } + } +} + +void DelayCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format("DelayCommand\n\tenabled {} \n\tinputs: ", effect_enabled); + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", inputs[i]); + } + string += "\n\toutputs: "; + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", outputs[i]); + } + string += "\n"; +} + +void DelayCommand::Process(const ADSP::CommandListProcessor& processor) { + std::vector<std::span<const s32>> input_buffers(parameter.channel_count); + std::vector<std::span<s32>> output_buffers(parameter.channel_count); + + for (s16 i = 0; i < parameter.channel_count; i++) { + input_buffers[i] = processor.mix_buffers.subspan(inputs[i] * processor.sample_count, + processor.sample_count); + output_buffers[i] = processor.mix_buffers.subspan(outputs[i] * processor.sample_count, + processor.sample_count); + } + + auto state_{reinterpret_cast<DelayInfo::State*>(state)}; + + if (effect_enabled) { + if (parameter.state == DelayInfo::ParameterState::Updating) { + SetDelayEffectParameter(parameter, *state_); + } else if (parameter.state == DelayInfo::ParameterState::Initialized) { + InitializeDelayEffect(parameter, *state_, workbuffer); + } + } + ApplyDelayEffect(parameter, *state_, effect_enabled, input_buffers, output_buffers, + processor.sample_count); +} + +bool DelayCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/delay.h b/src/audio_core/renderer/command/effect/delay.h new file mode 100644 index 000000000..b7a15ae6b --- /dev/null +++ b/src/audio_core/renderer/command/effect/delay.h @@ -0,0 +1,60 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <array> +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "audio_core/renderer/effect/delay.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for a delay effect. Delays inputs mix buffers according to the parameters + * and state, outputs receives the delayed samples. + */ +struct DelayCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer offsets for each channel + std::array<s16, MaxChannels> inputs; + /// Output mix buffer offsets for each channel + std::array<s16, MaxChannels> outputs; + /// Input parameters + DelayInfo::ParameterVersion1 parameter; + /// State, updated each call + CpuAddr state; + /// Game-supplied workbuffer (Unused) + CpuAddr workbuffer; + /// Is this effect enabled? + bool effect_enabled; +}; + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/i3dl2_reverb.cpp b/src/audio_core/renderer/command/effect/i3dl2_reverb.cpp new file mode 100644 index 000000000..c4bf3943a --- /dev/null +++ b/src/audio_core/renderer/command/effect/i3dl2_reverb.cpp @@ -0,0 +1,437 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include <numbers> + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/i3dl2_reverb.h" + +namespace AudioCore::AudioRenderer { + +constexpr std::array<f32, I3dl2ReverbInfo::MaxDelayLines> MinDelayLineTimes{ + 5.0f, + 6.0f, + 13.0f, + 14.0f, +}; +constexpr std::array<f32, I3dl2ReverbInfo::MaxDelayLines> MaxDelayLineTimes{ + 45.7042007446f, + 82.7817001343f, + 149.938293457f, + 271.575805664f, +}; +constexpr std::array<f32, I3dl2ReverbInfo::MaxDelayLines> Decay0MaxDelayLineTimes{17.0f, 13.0f, + 9.0f, 7.0f}; +constexpr std::array<f32, I3dl2ReverbInfo::MaxDelayLines> Decay1MaxDelayLineTimes{19.0f, 11.0f, + 10.0f, 6.0f}; +constexpr std::array<f32, I3dl2ReverbInfo::MaxDelayTaps> EarlyTapTimes{ + 0.0171360000968f, + 0.0591540001333f, + 0.161733001471f, + 0.390186011791f, + 0.425262004137f, + 0.455410987139f, + 0.689737021923f, + 0.74590998888f, + 0.833844006062f, + 0.859502017498f, + 0.0f, + 0.0750240013003f, + 0.168788000941f, + 0.299901008606f, + 0.337442994118f, + 0.371903002262f, + 0.599011003971f, + 0.716741025448f, + 0.817858994007f, + 0.85166400671f, +}; + +constexpr std::array<f32, I3dl2ReverbInfo::MaxDelayTaps> EarlyGains{ + 0.67096f, 0.61027f, 1.0f, 0.3568f, 0.68361f, 0.65978f, 0.51939f, + 0.24712f, 0.45945f, 0.45021f, 0.64196f, 0.54879f, 0.92925f, 0.3827f, + 0.72867f, 0.69794f, 0.5464f, 0.24563f, 0.45214f, 0.44042f}; + +/** + * Update the I3dl2ReverbInfo state according to the given parameters. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + * @param reset - If enabled, the state buffers will be reset. Only set this on initialize. + */ +static void UpdateI3dl2ReverbEffectParameter(const I3dl2ReverbInfo::ParameterVersion1& params, + I3dl2ReverbInfo::State& state, const bool reset) { + const auto pow_10 = [](f32 val) -> f32 { + return (val >= 0.0f) ? 1.0f : (val <= -5.3f) ? 0.0f : std::pow(10.0f, val); + }; + const auto sin = [](f32 degrees) -> f32 { + return std::sin(degrees * std::numbers::pi_v<f32> / 180.0f); + }; + const auto cos = [](f32 degrees) -> f32 { + return std::cos(degrees * std::numbers::pi_v<f32> / 180.0f); + }; + + Common::FixedPoint<50, 14> delay{static_cast<f32>(params.sample_rate) / 1000.0f}; + + state.dry_gain = params.dry_gain; + Common::FixedPoint<50, 14> early_gain{ + std::min(params.room_gain + params.reflection_gain, 5000.0f) / 2000.0f}; + state.early_gain = pow_10(early_gain.to_float()); + Common::FixedPoint<50, 14> late_gain{std::min(params.room_gain + params.reverb_gain, 5000.0f) / + 2000.0f}; + state.late_gain = pow_10(late_gain.to_float()); + + Common::FixedPoint<50, 14> hf_gain{pow_10(params.room_HF_gain / 2000.0f)}; + if (hf_gain >= 1.0f) { + state.lowpass_1 = 0.0f; + state.lowpass_2 = 1.0f; + } else { + const auto reference_hf{(params.reference_HF * 256.0f) / + static_cast<f32>(params.sample_rate)}; + const Common::FixedPoint<50, 14> a{1.0f - hf_gain.to_float()}; + const Common::FixedPoint<50, 14> b{2.0f + (-cos(reference_hf) * (hf_gain * 2.0f))}; + const Common::FixedPoint<50, 14> c{ + std::sqrt(std::pow(b.to_float(), 2.0f) + (std::pow(a.to_float(), 2.0f) * -4.0f))}; + + state.lowpass_1 = std::min(((b - c) / (a * 2.0f)).to_float(), 0.99723f); + state.lowpass_2 = 1.0f - state.lowpass_1; + } + + state.early_to_late_taps = + (((params.reflection_delay + params.late_reverb_delay_time) * 1000.0f) * delay).to_int(); + state.last_reverb_echo = params.late_reverb_diffusion * 0.6f * 0.01f; + + for (u32 i = 0; i < I3dl2ReverbInfo::MaxDelayLines; i++) { + auto curr_delay{ + ((MinDelayLineTimes[i] + (params.late_reverb_density / 100.0f) * + (MaxDelayLineTimes[i] - MinDelayLineTimes[i])) * + delay) + .to_int()}; + state.fdn_delay_lines[i].SetDelay(curr_delay); + + const auto a{ + (static_cast<f32>(state.fdn_delay_lines[i].delay + state.decay_delay_lines0[i].delay + + state.decay_delay_lines1[i].delay) * + -60.0f) / + (params.late_reverb_decay_time * static_cast<f32>(params.sample_rate))}; + const auto b{a / params.late_reverb_HF_decay_ratio}; + const auto c{ + cos(((params.reference_HF * 0.5f) * 128.0f) / static_cast<f32>(params.sample_rate)) / + sin(((params.reference_HF * 0.5f) * 128.0f) / static_cast<f32>(params.sample_rate))}; + const auto d{pow_10((b - a) / 40.0f)}; + const auto e{pow_10((b + a) / 40.0f) * 0.7071f}; + + state.lowpass_coeff[i][0] = ((c * d + 1.0f) * e) / (c + d); + state.lowpass_coeff[i][1] = ((1.0f - (c * d)) * e) / (c + d); + state.lowpass_coeff[i][2] = (c - d) / (c + d); + + state.decay_delay_lines0[i].wet_gain = state.last_reverb_echo; + state.decay_delay_lines1[i].wet_gain = state.last_reverb_echo * -0.9f; + } + + if (reset) { + state.shelf_filter.fill(0.0f); + state.lowpass_0 = 0.0f; + for (u32 i = 0; i < I3dl2ReverbInfo::MaxDelayLines; i++) { + std::ranges::fill(state.fdn_delay_lines[i].buffer, 0); + std::ranges::fill(state.decay_delay_lines0[i].buffer, 0); + std::ranges::fill(state.decay_delay_lines1[i].buffer, 0); + } + std::ranges::fill(state.center_delay_line.buffer, 0); + std::ranges::fill(state.early_delay_line.buffer, 0); + } + + const auto reflection_time{(params.late_reverb_delay_time * 0.9998f + 0.02f) * 1000.0f}; + const auto reflection_delay{params.reflection_delay * 1000.0f}; + for (u32 i = 0; i < I3dl2ReverbInfo::MaxDelayTaps; i++) { + auto length{((reflection_delay + reflection_time * EarlyTapTimes[i]) * delay).to_int()}; + if (length >= state.early_delay_line.max_delay) { + length = state.early_delay_line.max_delay; + } + state.early_tap_steps[i] = length; + } +} + +/** + * Initialize a new I3dl2ReverbInfo state according to the given parameters. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + * @param workbuffer - Game-supplied memory for the state. (Unused) + */ +static void InitializeI3dl2ReverbEffect(const I3dl2ReverbInfo::ParameterVersion1& params, + I3dl2ReverbInfo::State& state, const CpuAddr workbuffer) { + state = {}; + Common::FixedPoint<50, 14> delay{static_cast<f32>(params.sample_rate) / 1000}; + + for (u32 i = 0; i < I3dl2ReverbInfo::MaxDelayLines; i++) { + auto fdn_delay_time{(MaxDelayLineTimes[i] * delay).to_uint_floor()}; + state.fdn_delay_lines[i].Initialize(fdn_delay_time); + + auto decay0_delay_time{(Decay0MaxDelayLineTimes[i] * delay).to_uint_floor()}; + state.decay_delay_lines0[i].Initialize(decay0_delay_time); + + auto decay1_delay_time{(Decay1MaxDelayLineTimes[i] * delay).to_uint_floor()}; + state.decay_delay_lines1[i].Initialize(decay1_delay_time); + } + + const auto center_delay_time{(5 * delay).to_uint_floor()}; + state.center_delay_line.Initialize(center_delay_time); + + const auto early_delay_time{(400 * delay).to_uint_floor()}; + state.early_delay_line.Initialize(early_delay_time); + + UpdateI3dl2ReverbEffectParameter(params, state, true); +} + +/** + * Pass-through the effect, copying input to output directly, with no reverb applied. + * + * @param inputs - Array of input mix buffers to copy. + * @param outputs - Array of output mix buffers to receive copy. + * @param channel_count - Number of channels in inputs and outputs. + * @param sample_count - Number of samples within each channel (unused). + */ +static void ApplyI3dl2ReverbEffectBypass(std::span<std::span<const s32>> inputs, + std::span<std::span<s32>> outputs, const u32 channel_count, + [[maybe_unused]] const u32 sample_count) { + for (u32 i = 0; i < channel_count; i++) { + if (inputs[i].data() != outputs[i].data()) { + std::memcpy(outputs[i].data(), inputs[i].data(), outputs[i].size_bytes()); + } + } +} + +/** + * Tick the delay lines, reading and returning their current output, and writing a new decaying + * sample (mix). + * + * @param decay0 - The first decay line. + * @param decay1 - The second decay line. + * @param fdn - Feedback delay network. + * @param mix - The new calculated sample to be written and decayed. + * @return The next delayed and decayed sample. + */ +static Common::FixedPoint<50, 14> Axfx2AllPassTick(I3dl2ReverbInfo::I3dl2DelayLine& decay0, + I3dl2ReverbInfo::I3dl2DelayLine& decay1, + I3dl2ReverbInfo::I3dl2DelayLine& fdn, + const Common::FixedPoint<50, 14> mix) { + auto val{decay0.Read()}; + auto mixed{mix - (val * decay0.wet_gain)}; + auto out{decay0.Tick(mixed) + (mixed * decay0.wet_gain)}; + + val = decay1.Read(); + mixed = out - (val * decay1.wet_gain); + out = decay1.Tick(mixed) + (mixed * decay1.wet_gain); + + fdn.Tick(out); + return out; +} + +/** + * Impl. Apply a I3DL2 reverb according to the current state, on the input mix buffers, + * saving the results to the output mix buffers. + * + * @tparam NumChannels - Number of channels to process. 1-6. + Inputs/outputs should have this many buffers. + * @param state - State to use, must be initialized (see InitializeI3dl2ReverbEffect). + * @param inputs - Input mix buffers to perform the reverb on. + * @param outputs - Output mix buffers to receive the reverbed samples. + * @param sample_count - Number of samples to process. + */ +template <size_t NumChannels> +static void ApplyI3dl2ReverbEffect(I3dl2ReverbInfo::State& state, + std::span<std::span<const s32>> inputs, + std::span<std::span<s32>> outputs, const u32 sample_count) { + constexpr std::array<u8, I3dl2ReverbInfo::MaxDelayTaps> OutTapIndexes1Ch{ + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + }; + constexpr std::array<u8, I3dl2ReverbInfo::MaxDelayTaps> OutTapIndexes2Ch{ + 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 1, 1, 1, 0, 0, 0, 0, 1, 1, 1, + }; + constexpr std::array<u8, I3dl2ReverbInfo::MaxDelayTaps> OutTapIndexes4Ch{ + 0, 0, 0, 1, 1, 1, 1, 2, 2, 2, 1, 1, 1, 0, 0, 0, 0, 3, 3, 3, + }; + constexpr std::array<u8, I3dl2ReverbInfo::MaxDelayTaps> OutTapIndexes6Ch{ + 2, 0, 0, 1, 1, 1, 1, 4, 4, 4, 1, 1, 1, 0, 0, 0, 0, 5, 5, 5, + }; + + std::span<const u8> tap_indexes{}; + if constexpr (NumChannels == 1) { + tap_indexes = OutTapIndexes1Ch; + } else if constexpr (NumChannels == 2) { + tap_indexes = OutTapIndexes2Ch; + } else if constexpr (NumChannels == 4) { + tap_indexes = OutTapIndexes4Ch; + } else if constexpr (NumChannels == 6) { + tap_indexes = OutTapIndexes6Ch; + } + + for (u32 sample_index = 0; sample_index < sample_count; sample_index++) { + Common::FixedPoint<50, 14> early_to_late_tap{ + state.early_delay_line.TapOut(state.early_to_late_taps)}; + std::array<Common::FixedPoint<50, 14>, NumChannels> output_samples{}; + + for (u32 early_tap = 0; early_tap < I3dl2ReverbInfo::MaxDelayTaps; early_tap++) { + output_samples[tap_indexes[early_tap]] += + state.early_delay_line.TapOut(state.early_tap_steps[early_tap]) * + EarlyGains[early_tap]; + if constexpr (NumChannels == 6) { + output_samples[static_cast<u32>(Channels::LFE)] += + state.early_delay_line.TapOut(state.early_tap_steps[early_tap]) * + EarlyGains[early_tap]; + } + } + + Common::FixedPoint<50, 14> current_sample{}; + for (u32 channel = 0; channel < NumChannels; channel++) { + current_sample += inputs[channel][sample_index]; + } + + state.lowpass_0 = + (current_sample * state.lowpass_2 + state.lowpass_0 * state.lowpass_1).to_float(); + state.early_delay_line.Tick(state.lowpass_0); + + for (u32 channel = 0; channel < NumChannels; channel++) { + output_samples[channel] *= state.early_gain; + } + + std::array<Common::FixedPoint<50, 14>, I3dl2ReverbInfo::MaxDelayLines> filtered_samples{}; + for (u32 delay_line = 0; delay_line < I3dl2ReverbInfo::MaxDelayLines; delay_line++) { + filtered_samples[delay_line] = + state.fdn_delay_lines[delay_line].Read() * state.lowpass_coeff[delay_line][0] + + state.shelf_filter[delay_line]; + state.shelf_filter[delay_line] = + (filtered_samples[delay_line] * state.lowpass_coeff[delay_line][2] + + state.fdn_delay_lines[delay_line].Read() * state.lowpass_coeff[delay_line][1]) + .to_float(); + } + + const std::array<Common::FixedPoint<50, 14>, I3dl2ReverbInfo::MaxDelayLines> mix_matrix{ + filtered_samples[1] + filtered_samples[2] + early_to_late_tap * state.late_gain, + -filtered_samples[0] - filtered_samples[3] + early_to_late_tap * state.late_gain, + filtered_samples[0] - filtered_samples[3] + early_to_late_tap * state.late_gain, + filtered_samples[1] - filtered_samples[2] + early_to_late_tap * state.late_gain, + }; + + std::array<Common::FixedPoint<50, 14>, I3dl2ReverbInfo::MaxDelayLines> allpass_samples{}; + for (u32 delay_line = 0; delay_line < I3dl2ReverbInfo::MaxDelayLines; delay_line++) { + allpass_samples[delay_line] = Axfx2AllPassTick( + state.decay_delay_lines0[delay_line], state.decay_delay_lines1[delay_line], + state.fdn_delay_lines[delay_line], mix_matrix[delay_line]); + } + + if constexpr (NumChannels == 6) { + const std::array<Common::FixedPoint<50, 14>, MaxChannels> allpass_outputs{ + allpass_samples[0], allpass_samples[1], allpass_samples[2] - allpass_samples[3], + allpass_samples[3], allpass_samples[2], allpass_samples[3], + }; + + for (u32 channel = 0; channel < NumChannels; channel++) { + Common::FixedPoint<50, 14> allpass{}; + + if (channel == static_cast<u32>(Channels::Center)) { + allpass = state.center_delay_line.Tick(allpass_outputs[channel] * 0.5f); + } else { + allpass = allpass_outputs[channel]; + } + + auto out_sample{output_samples[channel] + allpass + + state.dry_gain * static_cast<f32>(inputs[channel][sample_index])}; + + outputs[channel][sample_index] = + static_cast<s32>(std::clamp(out_sample.to_float(), -8388600.0f, 8388600.0f)); + } + } else { + for (u32 channel = 0; channel < NumChannels; channel++) { + auto out_sample{output_samples[channel] + allpass_samples[channel] + + state.dry_gain * static_cast<f32>(inputs[channel][sample_index])}; + outputs[channel][sample_index] = + static_cast<s32>(std::clamp(out_sample.to_float(), -8388600.0f, 8388600.0f)); + } + } + } +} + +/** + * Apply a I3DL2 reverb if enabled, according to the current state, on the input mix buffers, + * saving the results to the output mix buffers. + * + * @param params - Input parameters to use. + * @param state - State to use, must be initialized (see InitializeI3dl2ReverbEffect). + * @param enabled - If enabled, delay will be applied, otherwise input is copied to output. + * @param inputs - Input mix buffers to performan the delay on. + * @param outputs - Output mix buffers to receive the delayed samples. + * @param sample_count - Number of samples to process. + */ +static void ApplyI3dl2ReverbEffect(const I3dl2ReverbInfo::ParameterVersion1& params, + I3dl2ReverbInfo::State& state, const bool enabled, + std::span<std::span<const s32>> inputs, + std::span<std::span<s32>> outputs, const u32 sample_count) { + if (enabled) { + switch (params.channel_count) { + case 0: + return; + case 1: + ApplyI3dl2ReverbEffect<1>(state, inputs, outputs, sample_count); + break; + case 2: + ApplyI3dl2ReverbEffect<2>(state, inputs, outputs, sample_count); + break; + case 4: + ApplyI3dl2ReverbEffect<4>(state, inputs, outputs, sample_count); + break; + case 6: + ApplyI3dl2ReverbEffect<6>(state, inputs, outputs, sample_count); + break; + default: + ApplyI3dl2ReverbEffectBypass(inputs, outputs, params.channel_count, sample_count); + break; + } + } else { + ApplyI3dl2ReverbEffectBypass(inputs, outputs, params.channel_count, sample_count); + } +} + +void I3dl2ReverbCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format("I3dl2ReverbCommand\n\tenabled {} \n\tinputs: ", effect_enabled); + for (u32 i = 0; i < parameter.channel_count; i++) { + string += fmt::format("{:02X}, ", inputs[i]); + } + string += "\n\toutputs: "; + for (u32 i = 0; i < parameter.channel_count; i++) { + string += fmt::format("{:02X}, ", outputs[i]); + } + string += "\n"; +} + +void I3dl2ReverbCommand::Process(const ADSP::CommandListProcessor& processor) { + std::vector<std::span<const s32>> input_buffers(parameter.channel_count); + std::vector<std::span<s32>> output_buffers(parameter.channel_count); + + for (u32 i = 0; i < parameter.channel_count; i++) { + input_buffers[i] = processor.mix_buffers.subspan(inputs[i] * processor.sample_count, + processor.sample_count); + output_buffers[i] = processor.mix_buffers.subspan(outputs[i] * processor.sample_count, + processor.sample_count); + } + + auto state_{reinterpret_cast<I3dl2ReverbInfo::State*>(state)}; + + if (effect_enabled) { + if (parameter.state == I3dl2ReverbInfo::ParameterState::Updating) { + UpdateI3dl2ReverbEffectParameter(parameter, *state_, false); + } else if (parameter.state == I3dl2ReverbInfo::ParameterState::Initialized) { + InitializeI3dl2ReverbEffect(parameter, *state_, workbuffer); + } + } + ApplyI3dl2ReverbEffect(parameter, *state_, effect_enabled, input_buffers, output_buffers, + processor.sample_count); +} + +bool I3dl2ReverbCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/i3dl2_reverb.h b/src/audio_core/renderer/command/effect/i3dl2_reverb.h new file mode 100644 index 000000000..243877056 --- /dev/null +++ b/src/audio_core/renderer/command/effect/i3dl2_reverb.h @@ -0,0 +1,60 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <array> +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "audio_core/renderer/effect/i3dl2.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for a I3DL2Reverb effect. Apply a reverb to inputs mix buffer according to + * the I3DL2 spec, outputs receives the results. + */ +struct I3dl2ReverbCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer offsets for each channel + std::array<s16, MaxChannels> inputs; + /// Output mix buffer offsets for each channel + std::array<s16, MaxChannels> outputs; + /// Input parameters + I3dl2ReverbInfo::ParameterVersion1 parameter; + /// State, updated each call + CpuAddr state; + /// Game-supplied workbuffer (Unused) + CpuAddr workbuffer; + /// Is this effect enabled? + bool effect_enabled; +}; + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/light_limiter.cpp b/src/audio_core/renderer/command/effect/light_limiter.cpp new file mode 100644 index 000000000..e8fb0e2fc --- /dev/null +++ b/src/audio_core/renderer/command/effect/light_limiter.cpp @@ -0,0 +1,222 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/light_limiter.h" + +namespace AudioCore::AudioRenderer { +/** + * Update the LightLimiterInfo state according to the given parameters. + * A no-op. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + */ +static void UpdateLightLimiterEffectParameter(const LightLimiterInfo::ParameterVersion2& params, + LightLimiterInfo::State& state) {} + +/** + * Initialize a new LightLimiterInfo state according to the given parameters. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + * @param workbuffer - Game-supplied memory for the state. (Unused) + */ +static void InitializeLightLimiterEffect(const LightLimiterInfo::ParameterVersion2& params, + LightLimiterInfo::State& state, const CpuAddr workbuffer) { + state = {}; + state.samples_average.fill(0.0f); + state.compression_gain.fill(1.0f); + state.look_ahead_sample_offsets.fill(0); + for (u32 i = 0; i < params.channel_count; i++) { + state.look_ahead_sample_buffers[i].resize(params.look_ahead_samples_max, 0.0f); + } +} + +/** + * Apply a light limiter effect if enabled, according to the current state, on the input mix + * buffers, saving the results to the output mix buffers. + * + * @param params - Input parameters to use. + * @param state - State to use, must be initialized (see InitializeLightLimiterEffect). + * @param enabled - If enabled, limiter will be applied, otherwise input is copied to output. + * @param inputs - Input mix buffers to perform the limiter on. + * @param outputs - Output mix buffers to receive the limited samples. + * @param sample_count - Number of samples to process. + * @params statistics - Optional output statistics, only used with version 2. + */ +static void ApplyLightLimiterEffect(const LightLimiterInfo::ParameterVersion2& params, + LightLimiterInfo::State& state, const bool enabled, + std::vector<std::span<const s32>>& inputs, + std::vector<std::span<s32>>& outputs, const u32 sample_count, + LightLimiterInfo::StatisticsInternal* statistics) { + constexpr s64 min{std::numeric_limits<s32>::min()}; + constexpr s64 max{std::numeric_limits<s32>::max()}; + + const auto recip_estimate = [](f64 a) -> f64 { + s32 q, s; + f64 r; + q = (s32)(a * 512.0); /* a in units of 1/512 rounded down */ + r = 1.0 / (((f64)q + 0.5) / 512.0); /* reciprocal r */ + s = (s32)(256.0 * r + 0.5); /* r in units of 1/256 rounded to nearest */ + return ((f64)s / 256.0); + }; + + if (enabled) { + if (statistics && params.statistics_reset_required) { + for (u32 i = 0; i < params.channel_count; i++) { + statistics->channel_compression_gain_min[i] = 1.0f; + statistics->channel_max_sample[i] = 0; + } + } + + for (u32 sample_index = 0; sample_index < sample_count; sample_index++) { + for (u32 channel = 0; channel < params.channel_count; channel++) { + auto sample{(Common::FixedPoint<49, 15>(inputs[channel][sample_index]) / + Common::FixedPoint<49, 15>::one) * + params.input_gain}; + auto abs_sample{sample}; + if (sample < 0.0f) { + abs_sample = -sample; + } + auto coeff{abs_sample > state.samples_average[channel] ? params.attack_coeff + : params.release_coeff}; + state.samples_average[channel] += + ((abs_sample - state.samples_average[channel]) * coeff).to_float(); + + // Reciprocal estimate + auto new_average_sample{Common::FixedPoint<49, 15>( + recip_estimate(state.samples_average[channel].to_double()))}; + if (params.processing_mode != LightLimiterInfo::ProcessingMode::Mode1) { + // Two Newton-Raphson steps + auto temp{2.0 - (state.samples_average[channel] * new_average_sample)}; + new_average_sample = 2.0 - (state.samples_average[channel] * temp); + } + + auto above_threshold{state.samples_average[channel] > params.threshold}; + auto attenuation{above_threshold ? params.threshold * new_average_sample : 1.0f}; + coeff = attenuation < state.compression_gain[channel] ? params.attack_coeff + : params.release_coeff; + state.compression_gain[channel] += + (attenuation - state.compression_gain[channel]) * coeff; + + auto lookahead_sample{ + state.look_ahead_sample_buffers[channel] + [state.look_ahead_sample_offsets[channel]]}; + + state.look_ahead_sample_buffers[channel][state.look_ahead_sample_offsets[channel]] = + sample; + state.look_ahead_sample_offsets[channel] = + (state.look_ahead_sample_offsets[channel] + 1) % params.look_ahead_samples_min; + + outputs[channel][sample_index] = static_cast<s32>( + std::clamp((lookahead_sample * state.compression_gain[channel] * + params.output_gain * Common::FixedPoint<49, 15>::one) + .to_long(), + min, max)); + + if (statistics) { + statistics->channel_max_sample[channel] = + std::max(statistics->channel_max_sample[channel], abs_sample.to_float()); + statistics->channel_compression_gain_min[channel] = + std::min(statistics->channel_compression_gain_min[channel], + state.compression_gain[channel].to_float()); + } + } + } + } else { + for (u32 i = 0; i < params.channel_count; i++) { + if (params.inputs[i] != params.outputs[i]) { + std::memcpy(outputs[i].data(), inputs[i].data(), outputs[i].size_bytes()); + } + } + } +} + +void LightLimiterVersion1Command::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format("LightLimiterVersion1Command\n\tinputs: "); + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", inputs[i]); + } + string += "\n\toutputs: "; + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", outputs[i]); + } + string += "\n"; +} + +void LightLimiterVersion1Command::Process(const ADSP::CommandListProcessor& processor) { + std::vector<std::span<const s32>> input_buffers(parameter.channel_count); + std::vector<std::span<s32>> output_buffers(parameter.channel_count); + + for (u32 i = 0; i < parameter.channel_count; i++) { + input_buffers[i] = processor.mix_buffers.subspan(inputs[i] * processor.sample_count, + processor.sample_count); + output_buffers[i] = processor.mix_buffers.subspan(outputs[i] * processor.sample_count, + processor.sample_count); + } + + auto state_{reinterpret_cast<LightLimiterInfo::State*>(state)}; + + if (effect_enabled) { + if (parameter.state == LightLimiterInfo::ParameterState::Updating) { + UpdateLightLimiterEffectParameter(parameter, *state_); + } else if (parameter.state == LightLimiterInfo::ParameterState::Initialized) { + InitializeLightLimiterEffect(parameter, *state_, workbuffer); + } + } + + LightLimiterInfo::StatisticsInternal* statistics{nullptr}; + ApplyLightLimiterEffect(parameter, *state_, effect_enabled, input_buffers, output_buffers, + processor.sample_count, statistics); +} + +bool LightLimiterVersion1Command::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +void LightLimiterVersion2Command::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format("LightLimiterVersion2Command\n\tinputs: \n"); + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", inputs[i]); + } + string += "\n\toutputs: "; + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", outputs[i]); + } + string += "\n"; +} + +void LightLimiterVersion2Command::Process(const ADSP::CommandListProcessor& processor) { + std::vector<std::span<const s32>> input_buffers(parameter.channel_count); + std::vector<std::span<s32>> output_buffers(parameter.channel_count); + + for (u32 i = 0; i < parameter.channel_count; i++) { + input_buffers[i] = processor.mix_buffers.subspan(inputs[i] * processor.sample_count, + processor.sample_count); + output_buffers[i] = processor.mix_buffers.subspan(outputs[i] * processor.sample_count, + processor.sample_count); + } + + auto state_{reinterpret_cast<LightLimiterInfo::State*>(state)}; + + if (effect_enabled) { + if (parameter.state == LightLimiterInfo::ParameterState::Updating) { + UpdateLightLimiterEffectParameter(parameter, *state_); + } else if (parameter.state == LightLimiterInfo::ParameterState::Initialized) { + InitializeLightLimiterEffect(parameter, *state_, workbuffer); + } + } + + auto statistics{reinterpret_cast<LightLimiterInfo::StatisticsInternal*>(result_state)}; + ApplyLightLimiterEffect(parameter, *state_, effect_enabled, input_buffers, output_buffers, + processor.sample_count, statistics); +} + +bool LightLimiterVersion2Command::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/light_limiter.h b/src/audio_core/renderer/command/effect/light_limiter.h new file mode 100644 index 000000000..5d98272c7 --- /dev/null +++ b/src/audio_core/renderer/command/effect/light_limiter.h @@ -0,0 +1,103 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <array> +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "audio_core/renderer/effect/light_limiter.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for limiting volume between a high and low threshold. + * Version 1. + */ +struct LightLimiterVersion1Command : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer offsets for each channel + std::array<s16, MaxChannels> inputs; + /// Output mix buffer offsets for each channel + std::array<s16, MaxChannels> outputs; + /// Input parameters + LightLimiterInfo::ParameterVersion2 parameter; + /// State, updated each call + CpuAddr state; + /// Game-supplied workbuffer (Unused) + CpuAddr workbuffer; + /// Is this effect enabled? + bool effect_enabled; +}; + +/** + * AudioRenderer command for limiting volume between a high and low threshold. + * Version 2 with output statistics. + */ +struct LightLimiterVersion2Command : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer offsets for each channel + std::array<s16, MaxChannels> inputs; + /// Output mix buffer offsets for each channel + std::array<s16, MaxChannels> outputs; + /// Input parameters + LightLimiterInfo::ParameterVersion2 parameter; + /// State, updated each call + CpuAddr state; + /// Game-supplied workbuffer (Unused) + CpuAddr workbuffer; + /// Optional statistics, sent back to the sysmodule + CpuAddr result_state; + /// Is this effect enabled? + bool effect_enabled; +}; + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/multi_tap_biquad_filter.cpp b/src/audio_core/renderer/command/effect/multi_tap_biquad_filter.cpp new file mode 100644 index 000000000..b3c3ba4ba --- /dev/null +++ b/src/audio_core/renderer/command/effect/multi_tap_biquad_filter.cpp @@ -0,0 +1,45 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/biquad_filter.h" +#include "audio_core/renderer/command/effect/multi_tap_biquad_filter.h" + +namespace AudioCore::AudioRenderer { + +void MultiTapBiquadFilterCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format( + "MultiTapBiquadFilterCommand\n\tinput {:02X}\n\toutput {:02X}\n\tneeds_init ({}, {})\n", + input, output, needs_init[0], needs_init[1]); +} + +void MultiTapBiquadFilterCommand::Process(const ADSP::CommandListProcessor& processor) { + if (filter_tap_count > MaxBiquadFilters) { + LOG_ERROR(Service_Audio, "Too many filter taps! {}", filter_tap_count); + filter_tap_count = MaxBiquadFilters; + } + + auto input_buffer{ + processor.mix_buffers.subspan(input * processor.sample_count, processor.sample_count)}; + auto output_buffer{ + processor.mix_buffers.subspan(output * processor.sample_count, processor.sample_count)}; + + // TODO: Fix this, currently just applies the filter to the input twice, + // and doesn't chain the biquads together at all. + for (u32 i = 0; i < filter_tap_count; i++) { + auto state{reinterpret_cast<VoiceState::BiquadFilterState*>(states[i])}; + if (needs_init[i]) { + std::memset(state, 0, sizeof(VoiceState::BiquadFilterState)); + } + + ApplyBiquadFilterFloat(output_buffer, input_buffer, biquads[i].b, biquads[i].a, *state, + processor.sample_count); + } +} + +bool MultiTapBiquadFilterCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/multi_tap_biquad_filter.h b/src/audio_core/renderer/command/effect/multi_tap_biquad_filter.h new file mode 100644 index 000000000..99c2c0830 --- /dev/null +++ b/src/audio_core/renderer/command/effect/multi_tap_biquad_filter.h @@ -0,0 +1,59 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <array> +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "audio_core/renderer/voice/voice_info.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for applying multiple biquads at once. + */ +struct MultiTapBiquadFilterCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer index + s16 input; + /// Output mix buffer index + s16 output; + /// Biquad parameters + std::array<VoiceInfo::BiquadFilterParameter, MaxBiquadFilters> biquads; + /// Biquad states, updated each call + std::array<CpuAddr, MaxBiquadFilters> states; + /// If each biquad needs initialisation + std::array<bool, MaxBiquadFilters> needs_init; + /// Number of active biquads + u8 filter_tap_count; +}; + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/reverb.cpp b/src/audio_core/renderer/command/effect/reverb.cpp new file mode 100644 index 000000000..fe2b1eb43 --- /dev/null +++ b/src/audio_core/renderer/command/effect/reverb.cpp @@ -0,0 +1,440 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include <numbers> +#include <ranges> + +#include "audio_core/renderer/adsp/command_list_processor.h" +#include "audio_core/renderer/command/effect/reverb.h" + +namespace AudioCore::AudioRenderer { + +constexpr std::array<f32, ReverbInfo::MaxDelayLines> FdnMaxDelayLineTimes = { + 53.9532470703125f, + 79.19256591796875f, + 116.23876953125f, + 170.61529541015625f, +}; + +constexpr std::array<f32, ReverbInfo::MaxDelayLines> DecayMaxDelayLineTimes = { + 7.0f, + 9.0f, + 13.0f, + 17.0f, +}; + +constexpr std::array<std::array<f32, ReverbInfo::MaxDelayTaps + 1>, ReverbInfo::NumEarlyModes> + EarlyDelayTimes = { + {{0.000000f, 3.500000f, 2.799988f, 3.899963f, 2.699951f, 13.399963f, 7.899963f, 8.399963f, + 9.899963f, 12.000000f, 12.500000f}, + {0.000000f, 11.799988f, 5.500000f, 11.199951f, 10.399963f, 38.099976f, 22.199951f, + 29.599976f, 21.199951f, 24.799988f, 40.000000f}, + {0.000000f, 41.500000f, 20.500000f, 41.299988f, 0.000000f, 29.500000f, 33.799988f, + 45.199951f, 46.799988f, 0.000000f, 50.000000f}, + {33.099976f, 43.299988f, 22.799988f, 37.899963f, 14.899963f, 35.299988f, 17.899963f, + 34.199951f, 0.000000f, 43.299988f, 50.000000f}, + {0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, + 0.000000f, 0.000000f, 0.000000f}}, +}; + +constexpr std::array<std::array<f32, ReverbInfo::MaxDelayTaps>, ReverbInfo::NumEarlyModes> + EarlyDelayGains = {{ + {0.699951f, 0.679993f, 0.699951f, 0.679993f, 0.699951f, 0.679993f, 0.699951f, 0.679993f, + 0.679993f, 0.679993f}, + {0.699951f, 0.679993f, 0.699951f, 0.679993f, 0.699951f, 0.679993f, 0.679993f, 0.679993f, + 0.679993f, 0.679993f}, + {0.500000f, 0.699951f, 0.699951f, 0.679993f, 0.500000f, 0.679993f, 0.679993f, 0.699951f, + 0.679993f, 0.000000f}, + {0.929993f, 0.919983f, 0.869995f, 0.859985f, 0.939941f, 0.809998f, 0.799988f, 0.769958f, + 0.759949f, 0.649963f}, + {0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, 0.000000f, + 0.000000f, 0.000000f}, + }}; + +constexpr std::array<std::array<f32, ReverbInfo::MaxDelayLines>, ReverbInfo::NumLateModes> + FdnDelayTimes = {{ + {53.953247f, 79.192566f, 116.238770f, 130.615295f}, + {53.953247f, 79.192566f, 116.238770f, 170.615295f}, + {5.000000f, 10.000000f, 5.000000f, 10.000000f}, + {47.029968f, 71.000000f, 103.000000f, 170.000000f}, + {53.953247f, 79.192566f, 116.238770f, 170.615295f}, + }}; + +constexpr std::array<std::array<f32, ReverbInfo::MaxDelayLines>, ReverbInfo::NumLateModes> + DecayDelayTimes = {{ + {7.000000f, 9.000000f, 13.000000f, 17.000000f}, + {7.000000f, 9.000000f, 13.000000f, 17.000000f}, + {1.000000f, 1.000000f, 1.000000f, 1.000000f}, + {7.000000f, 7.000000f, 13.000000f, 9.000000f}, + {7.000000f, 9.000000f, 13.000000f, 17.000000f}, + }}; + +/** + * Update the ReverbInfo state according to the given parameters. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + */ +static void UpdateReverbEffectParameter(const ReverbInfo::ParameterVersion2& params, + ReverbInfo::State& state) { + const auto pow_10 = [](f32 val) -> f32 { + return (val >= 0.0f) ? 1.0f : (val <= -5.3f) ? 0.0f : std::pow(10.0f, val); + }; + const auto cos = [](f32 degrees) -> f32 { + return std::cos(degrees * std::numbers::pi_v<f32> / 180.0f); + }; + + static bool unk_initialized{false}; + static Common::FixedPoint<50, 14> unk_value{}; + + const auto sample_rate{Common::FixedPoint<50, 14>::from_base(params.sample_rate)}; + const auto pre_delay_time{Common::FixedPoint<50, 14>::from_base(params.pre_delay)}; + + for (u32 i = 0; i < ReverbInfo::MaxDelayTaps; i++) { + auto early_delay{ + ((pre_delay_time + EarlyDelayTimes[params.early_mode][i]) * sample_rate).to_int()}; + early_delay = std::min(early_delay, state.pre_delay_line.sample_count_max); + state.early_delay_times[i] = early_delay + 1; + state.early_gains[i] = Common::FixedPoint<50, 14>::from_base(params.early_gain) * + EarlyDelayGains[params.early_mode][i]; + } + + if (params.channel_count == 2) { + state.early_gains[4] * 0.5f; + state.early_gains[5] * 0.5f; + } + + auto pre_time{ + ((pre_delay_time + EarlyDelayTimes[params.early_mode][10]) * sample_rate).to_int()}; + state.pre_delay_time = std::min(pre_time, state.pre_delay_line.sample_count_max); + + if (!unk_initialized) { + unk_value = cos((1280.0f / sample_rate).to_float()); + unk_initialized = true; + } + + for (u32 i = 0; i < ReverbInfo::MaxDelayLines; i++) { + const auto fdn_delay{(FdnDelayTimes[params.late_mode][i] * sample_rate).to_int()}; + state.fdn_delay_lines[i].sample_count = + std::min(fdn_delay, state.fdn_delay_lines[i].sample_count_max); + state.fdn_delay_lines[i].buffer_end = + &state.fdn_delay_lines[i].buffer[state.fdn_delay_lines[i].sample_count - 1]; + + const auto decay_delay{(DecayDelayTimes[params.late_mode][i] * sample_rate).to_int()}; + state.decay_delay_lines[i].sample_count = + std::min(decay_delay, state.decay_delay_lines[i].sample_count_max); + state.decay_delay_lines[i].buffer_end = + &state.decay_delay_lines[i].buffer[state.decay_delay_lines[i].sample_count - 1]; + + state.decay_delay_lines[i].decay = + 0.5999755859375f * (1.0f - Common::FixedPoint<50, 14>::from_base(params.colouration)); + + auto a{(Common::FixedPoint<50, 14>(state.fdn_delay_lines[i].sample_count_max) + + state.decay_delay_lines[i].sample_count_max) * + -3}; + auto b{a / (Common::FixedPoint<50, 14>::from_base(params.decay_time) * sample_rate)}; + Common::FixedPoint<50, 14> c{0.0f}; + Common::FixedPoint<50, 14> d{0.0f}; + auto hf_decay_ratio{Common::FixedPoint<50, 14>::from_base(params.high_freq_decay_ratio)}; + + if (hf_decay_ratio > 0.99493408203125f) { + c = 0.0f; + d = 1.0f; + } else { + const auto e{ + pow_10(((((1.0f / hf_decay_ratio) - 1.0f) * 2) / 100 * (b / 10)).to_float())}; + const auto f{1.0f - e}; + const auto g{2.0f - (unk_value * e * 2)}; + const auto h{std::sqrt(std::pow(g.to_float(), 2.0f) - (std::pow(f, 2.0f) * 4))}; + + c = (g - h) / (f * 2.0f); + d = 1.0f - c; + } + + state.hf_decay_prev_gain[i] = c; + state.hf_decay_gain[i] = pow_10((b / 1000).to_float()) * d * 0.70709228515625f; + state.prev_feedback_output[i] = 0; + } +} + +/** + * Initialize a new ReverbInfo state according to the given parameters. + * + * @param params - Input parameters to update the state. + * @param state - State to be updated. + * @param workbuffer - Game-supplied memory for the state. (Unused) + * @param long_size_pre_delay_supported - Use a longer pre-delay time before reverb begins. + */ +static void InitializeReverbEffect(const ReverbInfo::ParameterVersion2& params, + ReverbInfo::State& state, const CpuAddr workbuffer, + const bool long_size_pre_delay_supported) { + state = {}; + + auto delay{Common::FixedPoint<50, 14>::from_base(params.sample_rate)}; + + for (u32 i = 0; i < ReverbInfo::MaxDelayLines; i++) { + auto fdn_delay_time{(FdnMaxDelayLineTimes[i] * delay).to_uint_floor()}; + state.fdn_delay_lines[i].Initialize(fdn_delay_time, 1.0f); + + auto decay_delay_time{(DecayMaxDelayLineTimes[i] * delay).to_uint_floor()}; + state.decay_delay_lines[i].Initialize(decay_delay_time, 0.0f); + } + + const auto pre_delay{long_size_pre_delay_supported ? 350.0f : 150.0f}; + const auto pre_delay_line{(pre_delay * delay).to_uint_floor()}; + state.pre_delay_line.Initialize(pre_delay_line, 1.0f); + + const auto center_delay_time{(5 * delay).to_uint_floor()}; + state.center_delay_line.Initialize(center_delay_time, 1.0f); + + UpdateReverbEffectParameter(params, state); + + for (u32 i = 0; i < ReverbInfo::MaxDelayLines; i++) { + std::ranges::fill(state.fdn_delay_lines[i].buffer, 0); + std::ranges::fill(state.decay_delay_lines[i].buffer, 0); + } + std::ranges::fill(state.center_delay_line.buffer, 0); + std::ranges::fill(state.pre_delay_line.buffer, 0); +} + +/** + * Pass-through the effect, copying input to output directly, with no reverb applied. + * + * @param inputs - Array of input mix buffers to copy. + * @param outputs - Array of output mix buffers to receive copy. + * @param channel_count - Number of channels in inputs and outputs. + * @param sample_count - Number of samples within each channel. + */ +static void ApplyReverbEffectBypass(std::span<std::span<const s32>> inputs, + std::span<std::span<s32>> outputs, const u32 channel_count, + const u32 sample_count) { + for (u32 i = 0; i < channel_count; i++) { + if (inputs[i].data() != outputs[i].data()) { + std::memcpy(outputs[i].data(), inputs[i].data(), outputs[i].size_bytes()); + } + } +} + +/** + * Tick the delay lines, reading and returning their current output, and writing a new decaying + * sample (mix). + * + * @param decay - The decay line. + * @param fdn - Feedback delay network. + * @param mix - The new calculated sample to be written and decayed. + * @return The next delayed and decayed sample. + */ +static Common::FixedPoint<50, 14> Axfx2AllPassTick(ReverbInfo::ReverbDelayLine& decay, + ReverbInfo::ReverbDelayLine& fdn, + const Common::FixedPoint<50, 14> mix) { + const auto val{decay.Read()}; + const auto mixed{mix - (val * decay.decay)}; + const auto out{decay.Tick(mixed) + (mixed * decay.decay)}; + + fdn.Tick(out); + return out; +} + +/** + * Impl. Apply a Reverb according to the current state, on the input mix buffers, + * saving the results to the output mix buffers. + * + * @tparam NumChannels - Number of channels to process. 1-6. + Inputs/outputs should have this many buffers. + * @param params - Input parameters to update the state. + * @param state - State to use, must be initialized (see InitializeReverbEffect). + * @param inputs - Input mix buffers to perform the reverb on. + * @param outputs - Output mix buffers to receive the reverbed samples. + * @param sample_count - Number of samples to process. + */ +template <size_t NumChannels> +static void ApplyReverbEffect(const ReverbInfo::ParameterVersion2& params, ReverbInfo::State& state, + std::vector<std::span<const s32>>& inputs, + std::vector<std::span<s32>>& outputs, const u32 sample_count) { + constexpr std::array<u8, ReverbInfo::MaxDelayTaps> OutTapIndexes1Ch{ + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + }; + constexpr std::array<u8, ReverbInfo::MaxDelayTaps> OutTapIndexes2Ch{ + 0, 0, 1, 1, 0, 1, 0, 0, 1, 1, + }; + constexpr std::array<u8, ReverbInfo::MaxDelayTaps> OutTapIndexes4Ch{ + 0, 0, 1, 1, 0, 1, 2, 2, 3, 3, + }; + constexpr std::array<u8, ReverbInfo::MaxDelayTaps> OutTapIndexes6Ch{ + 0, 0, 1, 1, 2, 2, 4, 4, 5, 5, + }; + + std::span<const u8> tap_indexes{}; + if constexpr (NumChannels == 1) { + tap_indexes = OutTapIndexes1Ch; + } else if constexpr (NumChannels == 2) { + tap_indexes = OutTapIndexes2Ch; + } else if constexpr (NumChannels == 4) { + tap_indexes = OutTapIndexes4Ch; + } else if constexpr (NumChannels == 6) { + tap_indexes = OutTapIndexes6Ch; + } + + for (u32 sample_index = 0; sample_index < sample_count; sample_index++) { + std::array<Common::FixedPoint<50, 14>, NumChannels> output_samples{}; + + for (u32 early_tap = 0; early_tap < ReverbInfo::MaxDelayTaps; early_tap++) { + const auto sample{state.pre_delay_line.TapOut(state.early_delay_times[early_tap]) * + state.early_gains[early_tap]}; + output_samples[tap_indexes[early_tap]] += sample; + if constexpr (NumChannels == 6) { + output_samples[static_cast<u32>(Channels::LFE)] += sample; + } + } + + if constexpr (NumChannels == 6) { + output_samples[static_cast<u32>(Channels::LFE)] *= 0.2f; + } + + Common::FixedPoint<50, 14> input_sample{}; + for (u32 channel = 0; channel < NumChannels; channel++) { + input_sample += inputs[channel][sample_index]; + } + + input_sample *= 64; + input_sample *= Common::FixedPoint<50, 14>::from_base(params.base_gain); + state.pre_delay_line.Write(input_sample); + + for (u32 i = 0; i < ReverbInfo::MaxDelayLines; i++) { + state.prev_feedback_output[i] = + state.prev_feedback_output[i] * state.hf_decay_prev_gain[i] + + state.fdn_delay_lines[i].Read() * state.hf_decay_gain[i]; + } + + Common::FixedPoint<50, 14> pre_delay_sample{ + state.pre_delay_line.Read() * Common::FixedPoint<50, 14>::from_base(params.late_gain)}; + + std::array<Common::FixedPoint<50, 14>, ReverbInfo::MaxDelayLines> mix_matrix{ + state.prev_feedback_output[2] + state.prev_feedback_output[1] + pre_delay_sample, + -state.prev_feedback_output[0] - state.prev_feedback_output[3] + pre_delay_sample, + state.prev_feedback_output[0] - state.prev_feedback_output[3] + pre_delay_sample, + state.prev_feedback_output[1] - state.prev_feedback_output[2] + pre_delay_sample, + }; + + std::array<Common::FixedPoint<50, 14>, ReverbInfo::MaxDelayLines> allpass_samples{}; + for (u32 i = 0; i < ReverbInfo::MaxDelayLines; i++) { + allpass_samples[i] = Axfx2AllPassTick(state.decay_delay_lines[i], + state.fdn_delay_lines[i], mix_matrix[i]); + } + + const auto dry_gain{Common::FixedPoint<50, 14>::from_base(params.dry_gain)}; + const auto wet_gain{Common::FixedPoint<50, 14>::from_base(params.wet_gain)}; + + if constexpr (NumChannels == 6) { + const std::array<Common::FixedPoint<50, 14>, MaxChannels> allpass_outputs{ + allpass_samples[0], allpass_samples[1], allpass_samples[2] - allpass_samples[3], + allpass_samples[3], allpass_samples[2], allpass_samples[3], + }; + + for (u32 channel = 0; channel < NumChannels; channel++) { + auto in_sample{inputs[channel][sample_index] * dry_gain}; + + Common::FixedPoint<50, 14> allpass{}; + if (channel == static_cast<u32>(Channels::Center)) { + allpass = state.center_delay_line.Tick(allpass_outputs[channel] * 0.5f); + } else { + allpass = allpass_outputs[channel]; + } + + auto out_sample{((output_samples[channel] + allpass) * wet_gain) / 64}; + outputs[channel][sample_index] = (in_sample + out_sample).to_int(); + } + } else { + for (u32 channel = 0; channel < NumChannels; channel++) { + auto in_sample{inputs[channel][sample_index] * dry_gain}; + auto out_sample{((output_samples[channel] + allpass_samples[channel]) * wet_gain) / + 64}; + outputs[channel][sample_index] = (in_sample + out_sample).to_int(); + } + } + } +} + +/** + * Apply a Reverb if enabled, according to the current state, on the input mix buffers, + * saving the results to the output mix buffers. + * + * @param params - Input parameters to use. + * @param state - State to use, must be initialized (see InitializeReverbEffect). + * @param enabled - If enabled, delay will be applied, otherwise input is copied to output. + * @param inputs - Input mix buffers to performan the reverb on. + * @param outputs - Output mix buffers to receive the reverbed samples. + * @param sample_count - Number of samples to process. + */ +static void ApplyReverbEffect(const ReverbInfo::ParameterVersion2& params, ReverbInfo::State& state, + const bool enabled, std::vector<std::span<const s32>>& inputs, + std::vector<std::span<s32>>& outputs, const u32 sample_count) { + if (enabled) { + switch (params.channel_count) { + case 0: + return; + case 1: + ApplyReverbEffect<1>(params, state, inputs, outputs, sample_count); + break; + case 2: + ApplyReverbEffect<2>(params, state, inputs, outputs, sample_count); + break; + case 4: + ApplyReverbEffect<4>(params, state, inputs, outputs, sample_count); + break; + case 6: + ApplyReverbEffect<6>(params, state, inputs, outputs, sample_count); + break; + default: + ApplyReverbEffectBypass(inputs, outputs, params.channel_count, sample_count); + break; + } + } else { + ApplyReverbEffectBypass(inputs, outputs, params.channel_count, sample_count); + } +} + +void ReverbCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor, + std::string& string) { + string += fmt::format( + "ReverbCommand\n\tenabled {} long_size_pre_delay_supported {}\n\tinputs: ", effect_enabled, + long_size_pre_delay_supported); + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", inputs[i]); + } + string += "\n\toutputs: "; + for (u32 i = 0; i < MaxChannels; i++) { + string += fmt::format("{:02X}, ", outputs[i]); + } + string += "\n"; +} + +void ReverbCommand::Process(const ADSP::CommandListProcessor& processor) { + std::vector<std::span<const s32>> input_buffers(parameter.channel_count); + std::vector<std::span<s32>> output_buffers(parameter.channel_count); + + for (u32 i = 0; i < parameter.channel_count; i++) { + input_buffers[i] = processor.mix_buffers.subspan(inputs[i] * processor.sample_count, + processor.sample_count); + output_buffers[i] = processor.mix_buffers.subspan(outputs[i] * processor.sample_count, + processor.sample_count); + } + + auto state_{reinterpret_cast<ReverbInfo::State*>(state)}; + + if (effect_enabled) { + if (parameter.state == ReverbInfo::ParameterState::Updating) { + UpdateReverbEffectParameter(parameter, *state_); + } else if (parameter.state == ReverbInfo::ParameterState::Initialized) { + InitializeReverbEffect(parameter, *state_, workbuffer, long_size_pre_delay_supported); + } + } + ApplyReverbEffect(parameter, *state_, effect_enabled, input_buffers, output_buffers, + processor.sample_count); +} + +bool ReverbCommand::Verify(const ADSP::CommandListProcessor& processor) { + return true; +} + +} // namespace AudioCore::AudioRenderer diff --git a/src/audio_core/renderer/command/effect/reverb.h b/src/audio_core/renderer/command/effect/reverb.h new file mode 100644 index 000000000..328756150 --- /dev/null +++ b/src/audio_core/renderer/command/effect/reverb.h @@ -0,0 +1,62 @@ +// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <array> +#include <string> + +#include "audio_core/renderer/command/icommand.h" +#include "audio_core/renderer/effect/reverb.h" +#include "common/common_types.h" + +namespace AudioCore::AudioRenderer { +namespace ADSP { +class CommandListProcessor; +} + +/** + * AudioRenderer command for a Reverb effect. Apply a reverb to inputs mix buffer, outputs receives + * the results. + */ +struct ReverbCommand : ICommand { + /** + * Print this command's information to a string. + * + * @param processor - The CommandListProcessor processing this command. + * @param string - The string to print into. + */ + void Dump(const ADSP::CommandListProcessor& processor, std::string& string) override; + + /** + * Process this command. + * + * @param processor - The CommandListProcessor processing this command. + */ + void Process(const ADSP::CommandListProcessor& processor) override; + + /** + * Verify this command's data is valid. + * + * @param processor - The CommandListProcessor processing this command. + * @return True if the command is valid, otherwise false. + */ + bool Verify(const ADSP::CommandListProcessor& processor) override; + + /// Input mix buffer offsets for each channel + std::array<s16, MaxChannels> inputs; + /// Output mix buffer offsets for each channel + std::array<s16, MaxChannels> outputs; + /// Input parameters + ReverbInfo::ParameterVersion2 parameter; + /// State, updated each call + CpuAddr state; + /// Game-supplied workbuffer (Unused) + CpuAddr workbuffer; + /// Is this effect enabled? + bool effect_enabled; + /// Is a longer pre-delay time supported? + bool long_size_pre_delay_supported; +}; + +} // namespace AudioCore::AudioRenderer |