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mgthepro
2022-11-05 13:58:44 +01:00
parent 4a9f2bbf2a
commit 9f63fbe700
2002 changed files with 671171 additions and 671092 deletions
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src/audio_core/renderer/command/command_buffer.cpp
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src/audio_core/renderer/command/command_buffer.h
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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/renderer/command/commands.h"
#include "audio_core/renderer/effect/light_limiter.h"
#include "audio_core/renderer/performance/performance_manager.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
struct UpsamplerInfo;
struct VoiceState;
class EffectInfoBase;
class ICommandProcessingTimeEstimator;
class MixInfo;
class MemoryPoolInfo;
class SinkInfoBase;
class VoiceInfo;
/**
* Utility functions to generate and add commands into the current command list.
*/
class CommandBuffer {
public:
/**
* Generate a PCM s16 version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param memory_pool - Memory pool for translating buffer addresses to the DSP.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmInt16Version1Command(s32 node_id, const MemoryPoolInfo& memory_pool,
VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel);
/**
* Generate a PCM s16 version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmInt16Version2Command(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count,
s8 channel);
/**
* Generate a PCM f32 version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param memory_pool - Memory pool for translating buffer addresses to the DSP.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmFloatVersion1Command(s32 node_id, const MemoryPoolInfo& memory_pool,
VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel);
/**
* Generate a PCM f32 version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmFloatVersion2Command(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count,
s8 channel);
/**
* Generate an ADPCM version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param memory_pool - Memory pool for translating buffer addresses to the DSP.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GenerateAdpcmVersion1Command(s32 node_id, const MemoryPoolInfo& memory_pool,
VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel);
/**
* Generate an ADPCM version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GenerateAdpcmVersion2Command(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count, s8 channel);
/**
* Generate a volume command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer index to generate this command at.
* @param input_index - Channel index and mix buffer offset for this command.
* @param volume - Mix volume added to the input samples.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateVolumeCommand(s32 node_id, s16 buffer_offset, s16 input_index, f32 volume,
u8 precision);
/**
* Generate a volume ramp command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command takes its volumes from.
* @param buffer_count - Number of active mix buffers, command will generate at this index.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateVolumeRampCommand(s32 node_id, VoiceInfo& voice_info, s16 buffer_count,
u8 precision);
/**
* Generate a biquad filter command from a voice, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command takes biquad parameters from.
* @param voice_state - Used by the AudioRenderer to track previous samples.
* @param buffer_count - Number of active mix buffers,
* command will generate at this index + channel.
* @param channel - Channel index for this filter to work on.
* @param biquad_index - Which biquad filter to use for this command (0-1).
* @param use_float_processing - Should int or float processing be used?
*/
void GenerateBiquadFilterCommand(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count, s8 channel,
u32 biquad_index, bool use_float_processing);
/**
* Generate a biquad filter effect command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - The effect info this command takes biquad parameters from.
* @param buffer_offset - Mix buffer offset this command will use,
* command will generate at this index + channel.
* @param channel - Channel index for this filter to work on.
* @param needs_init - True if the biquad state needs initialisation.
* @param use_float_processing - Should int or float processing be used?
*/
void GenerateBiquadFilterCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset,
s8 channel, bool needs_init, bool use_float_processing);
/**
* Generate a mix command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param input_index - Input mix buffer index for this command.
* Added to the buffer offset.
* @param output_index - Output mix buffer index for this command.
* Added to the buffer offset.
* @param buffer_offset - Mix buffer offset this command will use.
* @param volume - Volume to be applied to the input.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixCommand(s32 node_id, s16 input_index, s16 output_index, s16 buffer_offset,
f32 volume, u8 precision);
/**
* Generate a mix ramp command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_count - Number of active mix buffers.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_count.
* @param output_index - Output mix buffer index for this command.
* Added to buffer_count.
* @param volume - Current mix volume used for calculating the ramp.
* @param prev_volume - Previous mix volume, used for calculating the ramp,
* also applied to the input.
* @param prev_samples - Previous sample buffer. Used for depopping.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixRampCommand(s32 node_id, s16 buffer_count, s16 input_index, s16 output_index,
f32 volume, f32 prev_volume, CpuAddr prev_samples, u8 precision);
/**
* Generate a mix ramp grouped command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_count - Number of active mix buffers.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_count.
* @param output_index - Output mix buffer index for this command.
* Added to buffer_count.
* @param volumes - Current mix volumes used for calculating the ramp.
* @param prev_volumes - Previous mix volumes, used for calculating the ramp,
* also applied to the input.
* @param prev_samples - Previous sample buffer. Used for depopping.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixRampGroupedCommand(s32 node_id, s16 buffer_count, s16 input_index,
s16 output_index, std::span<const f32> volumes,
std::span<const f32> prev_volumes, CpuAddr prev_samples,
u8 precision);
/**
* Generate a depop prepare command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_state - State to track the previous depop samples for each mix buffer.
* @param buffer - State to track the current depop samples for each mix buffer.
* @param buffer_count - Number of active mix buffers.
* @param buffer_offset - Base mix buffer index to generate the channel depops at.
* @param was_playing - Command only needs to work if the voice was previously playing.
*/
void GenerateDepopPrepareCommand(s32 node_id, const VoiceState& voice_state,
std::span<const s32> buffer, s16 buffer_count,
s16 buffer_offset, bool was_playing);
/**
* Generate a depop command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param mix_info - Mix info to get the buffer count and base offsets from.
* @param depop_buffer - Buffer of current depop sample values to be added to the input
* channels.
*/
void GenerateDepopForMixBuffersCommand(s32 node_id, const MixInfo& mix_info,
std::span<const s32> depop_buffer);
/**
* Generate a delay command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Delay effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to apply the apply the delay.
*/
void GenerateDelayCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset);
/**
* Generate an upsample command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to upsample.
* @param upsampler_info - Upsampler info to control the upsampling.
* @param input_count - Number of input channels to upsample.
* @param inputs - Input mix buffer indexes.
* @param buffer_count - Number of active mix buffers.
* @param sample_count - Source sample count of the input.
* @param sample_rate - Source sample rate of the input.
*/
void GenerateUpsampleCommand(s32 node_id, s16 buffer_offset, UpsamplerInfo& upsampler_info,
u32 input_count, std::span<const s8> inputs, s16 buffer_count,
u32 sample_count, u32 sample_rate);
/**
* Generate a downmix 6 -> 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param inputs - Input mix buffer indexes.
* @param buffer_offset - Base mix buffer offset of the channels to downmix.
* @param downmix_coeff - Downmixing coefficients.
*/
void GenerateDownMix6chTo2chCommand(s32 node_id, std::span<const s8> inputs, s16 buffer_offset,
std::span<const f32> downmix_coeff);
/**
* Generate an aux buffer command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Aux effect info to generate this command from.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_offset.
* @param output_index - Output mix buffer index for this command.
* Added to buffer_offset.
* @param buffer_offset - Base mix buffer offset to use.
* @param update_count - Number of samples to write back to the game as updated, can be 0.
* @param count_max - Maximum number of samples to read or write.
* @param write_offset - Current read or write offset within the buffer.
*/
void GenerateAuxCommand(s32 node_id, EffectInfoBase& effect_info, s16 input_index,
s16 output_index, s16 buffer_offset, u32 update_count, u32 count_max,
u32 write_offset);
/**
* Generate a device sink command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to use.
* @param sink_info - The sink_info to generate this command from.
* @param session_id - System session id this command is generated from.
* @param samples_buffer - The buffer to be sent to the sink if upsampling is not used.
*/
void GenerateDeviceSinkCommand(s32 node_id, s16 buffer_offset, SinkInfoBase& sink_info,
u32 session_id, std::span<s32> samples_buffer);
/**
* Generate a circular buffer sink command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param sink_info - The sink_info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
*/
void GenerateCircularBufferSinkCommand(s32 node_id, SinkInfoBase& sink_info, s16 buffer_offset);
/**
* Generate a reverb command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Reverb effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
* @param long_size_pre_delay_supported - Should a longer pre-delay time be used before reverb
* begins?
*/
void GenerateReverbCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset,
bool long_size_pre_delay_supported);
/**
* Generate an I3DL2 reverb command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - I3DL2Reverb effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
*/
void GenerateI3dl2ReverbCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset);
/**
* Generate a performance command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param state - State of the performance.
* @param entry_addresses - The addresses to be filled in by the AudioRenderer.
*/
void GeneratePerformanceCommand(s32 node_id, PerformanceState state,
const PerformanceEntryAddresses& entry_addresses);
/**
* Generate a clear mix command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateClearMixCommand(s32 node_id);
/**
* Generate a copy mix command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - BiquadFilter effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
* @param channel - Index to the effect's parameters input indexes for this command.
*/
void GenerateCopyMixBufferCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset,
s8 channel);
/**
* Generate a light limiter version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to use.
* @param parameter - Effect parameter to generate from.
* @param state - State used by the AudioRenderer between commands.
* @param enabled - Is this command enabled?
* @param workbuffer - Game-supplied memory for the state.
*/
void GenerateLightLimiterCommand(s32 node_id, s16 buffer_offset,
const LightLimiterInfo::ParameterVersion1& parameter,
const LightLimiterInfo::State& state, bool enabled,
CpuAddr workbuffer);
/**
* Generate a light limiter version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to use.
* @param parameter - Effect parameter to generate from.
* @param statistics - Statistics reported by the AudioRenderer on the limiter's state.
* @param state - State used by the AudioRenderer between commands.
* @param enabled - Is this command enabled?
* @param workbuffer - Game-supplied memory for the state.
*/
void GenerateLightLimiterCommand(s32 node_id, s16 buffer_offset,
const LightLimiterInfo::ParameterVersion2& parameter,
const LightLimiterInfo::StatisticsInternal& statistics,
const LightLimiterInfo::State& state, bool enabled,
CpuAddr workbuffer);
/**
* Generate a multitap biquad filter command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command takes biquad parameters from.
* @param voice_state - Used by the AudioRenderer to track previous samples.
* @param buffer_count - Number of active mix buffers,
* command will generate at this index + channel.
* @param channel - Channel index for this filter to work on.
*/
void GenerateMultitapBiquadFilterCommand(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count,
s8 channel);
/**
* Generate a capture command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Capture effect info to generate this command from.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_offset.
* @param output_index - Output mix buffer index for this command (unused).
* Added to buffer_offset.
* @param buffer_offset - Base mix buffer offset to use.
* @param update_count - Number of samples to write back to the game as updated, can be 0.
* @param count_max - Maximum number of samples to read or write.
* @param write_offset - Current read or write offset within the buffer.
*/
void GenerateCaptureCommand(s32 node_id, EffectInfoBase& effect_info, s16 input_index,
s16 output_index, s16 buffer_offset, u32 update_count,
u32 count_max, u32 write_offset);
/**
* Generate a compressor command, adding it to the command list.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Capture effect info to generate this command from.
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateCompressorCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/// Command list buffer generated commands will be added to
std::span<u8> command_list{};
/// Input sample count, unused
u32 sample_count{};
/// Input sample rate, unused
u32 sample_rate{};
/// Current size of the command buffer
u64 size{};
/// Current number of commands added
u32 count{};
/// Current estimated processing time for all commands
u32 estimated_process_time{};
/// Used for mapping buffers for the AudioRenderer
MemoryPoolInfo* memory_pool{};
/// Used for estimating command process times
ICommandProcessingTimeEstimator* time_estimator{};
/// Used to check which rendering features are currently enabled
BehaviorInfo* behavior{};
private:
template <typename T, CommandId Id>
T& GenerateStart(const s32 node_id);
template <typename T>
void GenerateEnd(T& cmd);
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/renderer/command/commands.h"
#include "audio_core/renderer/effect/light_limiter.h"
#include "audio_core/renderer/performance/performance_manager.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
struct UpsamplerInfo;
struct VoiceState;
class EffectInfoBase;
class ICommandProcessingTimeEstimator;
class MixInfo;
class MemoryPoolInfo;
class SinkInfoBase;
class VoiceInfo;
/**
* Utility functions to generate and add commands into the current command list.
*/
class CommandBuffer {
public:
/**
* Generate a PCM s16 version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param memory_pool - Memory pool for translating buffer addresses to the DSP.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmInt16Version1Command(s32 node_id, const MemoryPoolInfo& memory_pool,
VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel);
/**
* Generate a PCM s16 version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmInt16Version2Command(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count,
s8 channel);
/**
* Generate a PCM f32 version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param memory_pool - Memory pool for translating buffer addresses to the DSP.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmFloatVersion1Command(s32 node_id, const MemoryPoolInfo& memory_pool,
VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel);
/**
* Generate a PCM f32 version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GeneratePcmFloatVersion2Command(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count,
s8 channel);
/**
* Generate an ADPCM version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param memory_pool - Memory pool for translating buffer addresses to the DSP.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GenerateAdpcmVersion1Command(s32 node_id, const MemoryPoolInfo& memory_pool,
VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel);
/**
* Generate an ADPCM version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command is generated from.
* @param voice_state - The voice state the DSP will use for this command.
* @param buffer_count - Number of mix buffers in use,
* data will be read into this index + channel.
* @param channel - Channel index for this command.
*/
void GenerateAdpcmVersion2Command(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count, s8 channel);
/**
* Generate a volume command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer index to generate this command at.
* @param input_index - Channel index and mix buffer offset for this command.
* @param volume - Mix volume added to the input samples.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateVolumeCommand(s32 node_id, s16 buffer_offset, s16 input_index, f32 volume,
u8 precision);
/**
* Generate a volume ramp command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command takes its volumes from.
* @param buffer_count - Number of active mix buffers, command will generate at this index.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateVolumeRampCommand(s32 node_id, VoiceInfo& voice_info, s16 buffer_count,
u8 precision);
/**
* Generate a biquad filter command from a voice, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command takes biquad parameters from.
* @param voice_state - Used by the AudioRenderer to track previous samples.
* @param buffer_count - Number of active mix buffers,
* command will generate at this index + channel.
* @param channel - Channel index for this filter to work on.
* @param biquad_index - Which biquad filter to use for this command (0-1).
* @param use_float_processing - Should int or float processing be used?
*/
void GenerateBiquadFilterCommand(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count, s8 channel,
u32 biquad_index, bool use_float_processing);
/**
* Generate a biquad filter effect command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - The effect info this command takes biquad parameters from.
* @param buffer_offset - Mix buffer offset this command will use,
* command will generate at this index + channel.
* @param channel - Channel index for this filter to work on.
* @param needs_init - True if the biquad state needs initialisation.
* @param use_float_processing - Should int or float processing be used?
*/
void GenerateBiquadFilterCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset,
s8 channel, bool needs_init, bool use_float_processing);
/**
* Generate a mix command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param input_index - Input mix buffer index for this command.
* Added to the buffer offset.
* @param output_index - Output mix buffer index for this command.
* Added to the buffer offset.
* @param buffer_offset - Mix buffer offset this command will use.
* @param volume - Volume to be applied to the input.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixCommand(s32 node_id, s16 input_index, s16 output_index, s16 buffer_offset,
f32 volume, u8 precision);
/**
* Generate a mix ramp command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_count - Number of active mix buffers.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_count.
* @param output_index - Output mix buffer index for this command.
* Added to buffer_count.
* @param volume - Current mix volume used for calculating the ramp.
* @param prev_volume - Previous mix volume, used for calculating the ramp,
* also applied to the input.
* @param prev_samples - Previous sample buffer. Used for depopping.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixRampCommand(s32 node_id, s16 buffer_count, s16 input_index, s16 output_index,
f32 volume, f32 prev_volume, CpuAddr prev_samples, u8 precision);
/**
* Generate a mix ramp grouped command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_count - Number of active mix buffers.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_count.
* @param output_index - Output mix buffer index for this command.
* Added to buffer_count.
* @param volumes - Current mix volumes used for calculating the ramp.
* @param prev_volumes - Previous mix volumes, used for calculating the ramp,
* also applied to the input.
* @param prev_samples - Previous sample buffer. Used for depopping.
* @param precision - Number of decimal bits for fixed point operations.
*/
void GenerateMixRampGroupedCommand(s32 node_id, s16 buffer_count, s16 input_index,
s16 output_index, std::span<const f32> volumes,
std::span<const f32> prev_volumes, CpuAddr prev_samples,
u8 precision);
/**
* Generate a depop prepare command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_state - State to track the previous depop samples for each mix buffer.
* @param buffer - State to track the current depop samples for each mix buffer.
* @param buffer_count - Number of active mix buffers.
* @param buffer_offset - Base mix buffer index to generate the channel depops at.
* @param was_playing - Command only needs to work if the voice was previously playing.
*/
void GenerateDepopPrepareCommand(s32 node_id, const VoiceState& voice_state,
std::span<const s32> buffer, s16 buffer_count,
s16 buffer_offset, bool was_playing);
/**
* Generate a depop command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param mix_info - Mix info to get the buffer count and base offsets from.
* @param depop_buffer - Buffer of current depop sample values to be added to the input
* channels.
*/
void GenerateDepopForMixBuffersCommand(s32 node_id, const MixInfo& mix_info,
std::span<const s32> depop_buffer);
/**
* Generate a delay command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Delay effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to apply the apply the delay.
*/
void GenerateDelayCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset);
/**
* Generate an upsample command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to upsample.
* @param upsampler_info - Upsampler info to control the upsampling.
* @param input_count - Number of input channels to upsample.
* @param inputs - Input mix buffer indexes.
* @param buffer_count - Number of active mix buffers.
* @param sample_count - Source sample count of the input.
* @param sample_rate - Source sample rate of the input.
*/
void GenerateUpsampleCommand(s32 node_id, s16 buffer_offset, UpsamplerInfo& upsampler_info,
u32 input_count, std::span<const s8> inputs, s16 buffer_count,
u32 sample_count, u32 sample_rate);
/**
* Generate a downmix 6 -> 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param inputs - Input mix buffer indexes.
* @param buffer_offset - Base mix buffer offset of the channels to downmix.
* @param downmix_coeff - Downmixing coefficients.
*/
void GenerateDownMix6chTo2chCommand(s32 node_id, std::span<const s8> inputs, s16 buffer_offset,
std::span<const f32> downmix_coeff);
/**
* Generate an aux buffer command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Aux effect info to generate this command from.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_offset.
* @param output_index - Output mix buffer index for this command.
* Added to buffer_offset.
* @param buffer_offset - Base mix buffer offset to use.
* @param update_count - Number of samples to write back to the game as updated, can be 0.
* @param count_max - Maximum number of samples to read or write.
* @param write_offset - Current read or write offset within the buffer.
*/
void GenerateAuxCommand(s32 node_id, EffectInfoBase& effect_info, s16 input_index,
s16 output_index, s16 buffer_offset, u32 update_count, u32 count_max,
u32 write_offset);
/**
* Generate a device sink command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to use.
* @param sink_info - The sink_info to generate this command from.
* @param session_id - System session id this command is generated from.
* @param samples_buffer - The buffer to be sent to the sink if upsampling is not used.
*/
void GenerateDeviceSinkCommand(s32 node_id, s16 buffer_offset, SinkInfoBase& sink_info,
u32 session_id, std::span<s32> samples_buffer);
/**
* Generate a circular buffer sink command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param sink_info - The sink_info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
*/
void GenerateCircularBufferSinkCommand(s32 node_id, SinkInfoBase& sink_info, s16 buffer_offset);
/**
* Generate a reverb command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Reverb effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
* @param long_size_pre_delay_supported - Should a longer pre-delay time be used before reverb
* begins?
*/
void GenerateReverbCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset,
bool long_size_pre_delay_supported);
/**
* Generate an I3DL2 reverb command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - I3DL2Reverb effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
*/
void GenerateI3dl2ReverbCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset);
/**
* Generate a performance command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param state - State of the performance.
* @param entry_addresses - The addresses to be filled in by the AudioRenderer.
*/
void GeneratePerformanceCommand(s32 node_id, PerformanceState state,
const PerformanceEntryAddresses& entry_addresses);
/**
* Generate a clear mix command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateClearMixCommand(s32 node_id);
/**
* Generate a copy mix command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - BiquadFilter effect info to generate this command from.
* @param buffer_offset - Base mix buffer offset to use.
* @param channel - Index to the effect's parameters input indexes for this command.
*/
void GenerateCopyMixBufferCommand(s32 node_id, EffectInfoBase& effect_info, s16 buffer_offset,
s8 channel);
/**
* Generate a light limiter version 1 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to use.
* @param parameter - Effect parameter to generate from.
* @param state - State used by the AudioRenderer between commands.
* @param enabled - Is this command enabled?
* @param workbuffer - Game-supplied memory for the state.
*/
void GenerateLightLimiterCommand(s32 node_id, s16 buffer_offset,
const LightLimiterInfo::ParameterVersion1& parameter,
const LightLimiterInfo::State& state, bool enabled,
CpuAddr workbuffer);
/**
* Generate a light limiter version 2 command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param buffer_offset - Base mix buffer offset to use.
* @param parameter - Effect parameter to generate from.
* @param statistics - Statistics reported by the AudioRenderer on the limiter's state.
* @param state - State used by the AudioRenderer between commands.
* @param enabled - Is this command enabled?
* @param workbuffer - Game-supplied memory for the state.
*/
void GenerateLightLimiterCommand(s32 node_id, s16 buffer_offset,
const LightLimiterInfo::ParameterVersion2& parameter,
const LightLimiterInfo::StatisticsInternal& statistics,
const LightLimiterInfo::State& state, bool enabled,
CpuAddr workbuffer);
/**
* Generate a multitap biquad filter command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param voice_info - The voice info this command takes biquad parameters from.
* @param voice_state - Used by the AudioRenderer to track previous samples.
* @param buffer_count - Number of active mix buffers,
* command will generate at this index + channel.
* @param channel - Channel index for this filter to work on.
*/
void GenerateMultitapBiquadFilterCommand(s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state, s16 buffer_count,
s8 channel);
/**
* Generate a capture command, adding it to the command list.
*
* @param node_id - Node id of the voice this command is generated for.
* @param effect_info - Capture effect info to generate this command from.
* @param input_index - Input mix buffer index for this command.
* Added to buffer_offset.
* @param output_index - Output mix buffer index for this command (unused).
* Added to buffer_offset.
* @param buffer_offset - Base mix buffer offset to use.
* @param update_count - Number of samples to write back to the game as updated, can be 0.
* @param count_max - Maximum number of samples to read or write.
* @param write_offset - Current read or write offset within the buffer.
*/
void GenerateCaptureCommand(s32 node_id, EffectInfoBase& effect_info, s16 input_index,
s16 output_index, s16 buffer_offset, u32 update_count,
u32 count_max, u32 write_offset);
/**
* Generate a compressor command, adding it to the command list.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Capture effect info to generate this command from.
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateCompressorCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/// Command list buffer generated commands will be added to
std::span<u8> command_list{};
/// Input sample count, unused
u32 sample_count{};
/// Input sample rate, unused
u32 sample_rate{};
/// Current size of the command buffer
u64 size{};
/// Current number of commands added
u32 count{};
/// Current estimated processing time for all commands
u32 estimated_process_time{};
/// Used for mapping buffers for the AudioRenderer
MemoryPoolInfo* memory_pool{};
/// Used for estimating command process times
ICommandProcessingTimeEstimator* time_estimator{};
/// Used to check which rendering features are currently enabled
BehaviorInfo* behavior{};
private:
template <typename T, CommandId Id>
T& GenerateStart(const s32 node_id);
template <typename T>
void GenerateEnd(T& cmd);
};
} // namespace AudioCore::AudioRenderer

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/renderer/command/commands.h"
#include "audio_core/renderer/performance/performance_manager.h"
#include "common/common_types.h"
namespace AudioCore {
struct AudioRendererSystemContext;
namespace AudioRenderer {
class CommandBuffer;
struct CommandListHeader;
class VoiceContext;
class MixContext;
class EffectContext;
class SplitterContext;
class SinkContext;
class BehaviorInfo;
class VoiceInfo;
struct VoiceState;
class MixInfo;
class SinkInfoBase;
/**
* Generates all commands to build up a command list, which are sent to the AudioRender for
* processing.
*/
class CommandGenerator {
public:
explicit CommandGenerator(CommandBuffer& command_buffer,
const CommandListHeader& command_list_header,
const AudioRendererSystemContext& render_context,
VoiceContext& voice_context, MixContext& mix_context,
EffectContext& effect_context, SinkContext& sink_context,
SplitterContext& splitter_context,
PerformanceManager* performance_manager);
/**
* Calculate the buffer size needed for commands.
*
* @param behavior - Used to check what features are enabled.
* @param params - Input rendering parameters for numbers of voices/mixes/sinks etc.
*/
static u64 CalculateCommandBufferSize(const BehaviorInfo& behavior,
const AudioRendererParameterInternal& params) {
u64 size{0};
// Effects
size += params.effects * sizeof(EffectInfoBase);
// Voices
u64 voice_size{0};
if (behavior.IsWaveBufferVer2Supported()) {
voice_size = std::max(std::max(sizeof(AdpcmDataSourceVersion2Command),
sizeof(PcmInt16DataSourceVersion2Command)),
sizeof(PcmFloatDataSourceVersion2Command));
} else {
voice_size = std::max(std::max(sizeof(AdpcmDataSourceVersion1Command),
sizeof(PcmInt16DataSourceVersion1Command)),
sizeof(PcmFloatDataSourceVersion1Command));
}
voice_size += sizeof(BiquadFilterCommand) * MaxBiquadFilters;
voice_size += sizeof(VolumeRampCommand);
voice_size += sizeof(MixRampGroupedCommand);
size += params.voices * (params.splitter_infos * sizeof(DepopPrepareCommand) + voice_size);
// Sub mixes
size += sizeof(DepopForMixBuffersCommand) +
(sizeof(MixCommand) * MaxMixBuffers) * MaxMixBuffers;
// Final mix
size += sizeof(DepopForMixBuffersCommand) + sizeof(VolumeCommand) * MaxMixBuffers;
// Splitters
size += params.splitter_destinations * sizeof(MixRampCommand) * MaxMixBuffers;
// Sinks
size +=
params.sinks * std::max(sizeof(DeviceSinkCommand), sizeof(CircularBufferSinkCommand));
// Performance
size += (params.effects + params.voices + params.sinks + params.sub_mixes + 1 +
PerformanceManager::MaxDetailEntries) *
sizeof(PerformanceCommand);
return size;
}
/**
* Get the current command buffer used to generate commands.
*
* @return The command buffer.
*/
CommandBuffer& GetCommandBuffer() {
return command_buffer;
}
/**
* Get the current performance manager,
*
* @return The performance manager. May be nullptr.
*/
PerformanceManager* GetPerformanceManager() {
return performance_manager;
}
/**
* Generate a data source command.
* These are the basis for all audio output.
*
* @param voice_info - Generate the command from this voice.
* @param voice_state - State used by the AudioRenderer across calls.
* @param channel - Channel index to generate the command into.
*/
void GenerateDataSourceCommand(VoiceInfo& voice_info, const VoiceState& voice_state,
s8 channel);
/**
* Generate voice mixing commands.
* These are used to mix buffers together, to mix one input to many outputs,
* and also used as copy commands to move data around and prevent it being accidentally
* overwritten, e.g by another data source command into the same channel.
*
* @param mix_volumes - Current volumes of the mix.
* @param prev_mix_volumes - Previous volumes of the mix.
* @param voice_state - State used by the AudioRenderer across calls.
* @param output_index - Output mix buffer index.
* @param buffer_count - Number of active mix buffers.
* @param input_index - Input mix buffer index.
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateVoiceMixCommand(std::span<const f32> mix_volumes,
std::span<const f32> prev_mix_volumes,
const VoiceState& voice_state, s16 output_index, s16 buffer_count,
s16 input_index, s32 node_id);
/**
* Generate a biquad filter command for a voice.
*
* @param voice_info - Voice info this command is generated from.
* @param voice_state - State used by the AudioRenderer across calls.
* @param buffer_count - Number of active mix buffers.
* @param channel - Channel index of this command.
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateBiquadFilterCommandForVoice(VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel, s32 node_id);
/**
* Generate commands for a voice.
* Includes a data source, biquad filter, volume and mixing.
*
* @param voice_info - Voice info these commands are generated from.
*/
void GenerateVoiceCommand(VoiceInfo& voice_info);
/**
* Generate commands for all voices.
*/
void GenerateVoiceCommands();
/**
* Generate a mixing command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info_base - BufferMixer effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateBufferMixerCommand(s16 buffer_offset, EffectInfoBase& effect_info_base,
s32 node_id);
/**
* Generate a delay effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info_base - Delay effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateDelayCommand(s16 buffer_offset, EffectInfoBase& effect_info_base, s32 node_id);
/**
* Generate a reverb effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info_base - Reverb effect info.
* @param node_id - Node id of the mix this command is generated for.
* @param long_size_pre_delay_supported - Use a longer pre-delay time before reverb starts.
*/
void GenerateReverbCommand(s16 buffer_offset, EffectInfoBase& effect_info_base, s32 node_id,
bool long_size_pre_delay_supported);
/**
* Generate an I3DL2 reverb effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - I3DL2Reverb effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateI3dl2ReverbEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id);
/**
* Generate an aux effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Aux effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateAuxCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate a biquad filter effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Aux effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateBiquadFilterEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id);
/**
* Generate a light limiter effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Limiter effect info.
* @param node_id - Node id of the mix this command is generated for.
* @param effect_index - Index for the statistics state.
*/
void GenerateLightLimiterEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id, u32 effect_index);
/**
* Generate a capture effect command.
* Writes a mix buffer back to game memory.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Capture effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateCaptureCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate a compressor effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Compressor effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateCompressorCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate all effect commands for a mix.
*
* @param mix_info - Mix to generate effects from.
*/
void GenerateEffectCommand(MixInfo& mix_info);
/**
* Generate all mix commands.
*
* @param mix_info - Mix to generate effects from.
*/
void GenerateMixCommands(MixInfo& mix_info);
/**
* Generate a submix command.
* Generates all effects and all mixing commands.
*
* @param mix_info - Mix to generate effects from.
*/
void GenerateSubMixCommand(MixInfo& mix_info);
/**
* Generate all submix command.
*/
void GenerateSubMixCommands();
/**
* Generate the final mix.
*/
void GenerateFinalMixCommand();
/**
* Generate the final mix commands.
*/
void GenerateFinalMixCommands();
/**
* Generate all sink commands.
*/
void GenerateSinkCommands();
/**
* Generate a sink command.
* Sends samples out to the backend, or a game-supplied circular buffer.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param sink_info - Sink info to generate the commands from.
*/
void GenerateSinkCommand(s16 buffer_offset, SinkInfoBase& sink_info);
/**
* Generate a device sink command.
* Sends samples out to the backend.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param sink_info - Sink info to generate the commands from.
*/
void GenerateDeviceSinkCommand(s16 buffer_offset, SinkInfoBase& sink_info);
/**
* Generate a performance command.
* Used to report performance metrics of the AudioRenderer back to the game.
*
* @param node_id - Node ID of the mix this command is generated for
* @param state - Output state of the generated performance command
* @param entry_addresses - Addresses to be written
*/
void GeneratePerformanceCommand(s32 node_id, PerformanceState state,
const PerformanceEntryAddresses& entry_addresses);
private:
/// Commands will be written by this buffer
CommandBuffer& command_buffer;
/// Header information for the commands generated
const CommandListHeader& command_header;
/// Various things to control generation
const AudioRendererSystemContext& render_context;
/// Used for generating voices
VoiceContext& voice_context;
/// Used for generating mixes
MixContext& mix_context;
/// Used for generating effects
EffectContext& effect_context;
/// Used for generating sinks
SinkContext& sink_context;
/// Used for generating submixes
SplitterContext& splitter_context;
/// Used for generating performance
PerformanceManager* performance_manager;
};
} // namespace AudioRenderer
} // namespace AudioCore
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/renderer/command/commands.h"
#include "audio_core/renderer/performance/performance_manager.h"
#include "common/common_types.h"
namespace AudioCore {
struct AudioRendererSystemContext;
namespace AudioRenderer {
class CommandBuffer;
struct CommandListHeader;
class VoiceContext;
class MixContext;
class EffectContext;
class SplitterContext;
class SinkContext;
class BehaviorInfo;
class VoiceInfo;
struct VoiceState;
class MixInfo;
class SinkInfoBase;
/**
* Generates all commands to build up a command list, which are sent to the AudioRender for
* processing.
*/
class CommandGenerator {
public:
explicit CommandGenerator(CommandBuffer& command_buffer,
const CommandListHeader& command_list_header,
const AudioRendererSystemContext& render_context,
VoiceContext& voice_context, MixContext& mix_context,
EffectContext& effect_context, SinkContext& sink_context,
SplitterContext& splitter_context,
PerformanceManager* performance_manager);
/**
* Calculate the buffer size needed for commands.
*
* @param behavior - Used to check what features are enabled.
* @param params - Input rendering parameters for numbers of voices/mixes/sinks etc.
*/
static u64 CalculateCommandBufferSize(const BehaviorInfo& behavior,
const AudioRendererParameterInternal& params) {
u64 size{0};
// Effects
size += params.effects * sizeof(EffectInfoBase);
// Voices
u64 voice_size{0};
if (behavior.IsWaveBufferVer2Supported()) {
voice_size = std::max(std::max(sizeof(AdpcmDataSourceVersion2Command),
sizeof(PcmInt16DataSourceVersion2Command)),
sizeof(PcmFloatDataSourceVersion2Command));
} else {
voice_size = std::max(std::max(sizeof(AdpcmDataSourceVersion1Command),
sizeof(PcmInt16DataSourceVersion1Command)),
sizeof(PcmFloatDataSourceVersion1Command));
}
voice_size += sizeof(BiquadFilterCommand) * MaxBiquadFilters;
voice_size += sizeof(VolumeRampCommand);
voice_size += sizeof(MixRampGroupedCommand);
size += params.voices * (params.splitter_infos * sizeof(DepopPrepareCommand) + voice_size);
// Sub mixes
size += sizeof(DepopForMixBuffersCommand) +
(sizeof(MixCommand) * MaxMixBuffers) * MaxMixBuffers;
// Final mix
size += sizeof(DepopForMixBuffersCommand) + sizeof(VolumeCommand) * MaxMixBuffers;
// Splitters
size += params.splitter_destinations * sizeof(MixRampCommand) * MaxMixBuffers;
// Sinks
size +=
params.sinks * std::max(sizeof(DeviceSinkCommand), sizeof(CircularBufferSinkCommand));
// Performance
size += (params.effects + params.voices + params.sinks + params.sub_mixes + 1 +
PerformanceManager::MaxDetailEntries) *
sizeof(PerformanceCommand);
return size;
}
/**
* Get the current command buffer used to generate commands.
*
* @return The command buffer.
*/
CommandBuffer& GetCommandBuffer() {
return command_buffer;
}
/**
* Get the current performance manager,
*
* @return The performance manager. May be nullptr.
*/
PerformanceManager* GetPerformanceManager() {
return performance_manager;
}
/**
* Generate a data source command.
* These are the basis for all audio output.
*
* @param voice_info - Generate the command from this voice.
* @param voice_state - State used by the AudioRenderer across calls.
* @param channel - Channel index to generate the command into.
*/
void GenerateDataSourceCommand(VoiceInfo& voice_info, const VoiceState& voice_state,
s8 channel);
/**
* Generate voice mixing commands.
* These are used to mix buffers together, to mix one input to many outputs,
* and also used as copy commands to move data around and prevent it being accidentally
* overwritten, e.g by another data source command into the same channel.
*
* @param mix_volumes - Current volumes of the mix.
* @param prev_mix_volumes - Previous volumes of the mix.
* @param voice_state - State used by the AudioRenderer across calls.
* @param output_index - Output mix buffer index.
* @param buffer_count - Number of active mix buffers.
* @param input_index - Input mix buffer index.
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateVoiceMixCommand(std::span<const f32> mix_volumes,
std::span<const f32> prev_mix_volumes,
const VoiceState& voice_state, s16 output_index, s16 buffer_count,
s16 input_index, s32 node_id);
/**
* Generate a biquad filter command for a voice.
*
* @param voice_info - Voice info this command is generated from.
* @param voice_state - State used by the AudioRenderer across calls.
* @param buffer_count - Number of active mix buffers.
* @param channel - Channel index of this command.
* @param node_id - Node id of the voice this command is generated for.
*/
void GenerateBiquadFilterCommandForVoice(VoiceInfo& voice_info, const VoiceState& voice_state,
s16 buffer_count, s8 channel, s32 node_id);
/**
* Generate commands for a voice.
* Includes a data source, biquad filter, volume and mixing.
*
* @param voice_info - Voice info these commands are generated from.
*/
void GenerateVoiceCommand(VoiceInfo& voice_info);
/**
* Generate commands for all voices.
*/
void GenerateVoiceCommands();
/**
* Generate a mixing command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info_base - BufferMixer effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateBufferMixerCommand(s16 buffer_offset, EffectInfoBase& effect_info_base,
s32 node_id);
/**
* Generate a delay effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info_base - Delay effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateDelayCommand(s16 buffer_offset, EffectInfoBase& effect_info_base, s32 node_id);
/**
* Generate a reverb effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info_base - Reverb effect info.
* @param node_id - Node id of the mix this command is generated for.
* @param long_size_pre_delay_supported - Use a longer pre-delay time before reverb starts.
*/
void GenerateReverbCommand(s16 buffer_offset, EffectInfoBase& effect_info_base, s32 node_id,
bool long_size_pre_delay_supported);
/**
* Generate an I3DL2 reverb effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - I3DL2Reverb effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateI3dl2ReverbEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id);
/**
* Generate an aux effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Aux effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateAuxCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate a biquad filter effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Aux effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateBiquadFilterEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id);
/**
* Generate a light limiter effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Limiter effect info.
* @param node_id - Node id of the mix this command is generated for.
* @param effect_index - Index for the statistics state.
*/
void GenerateLightLimiterEffectCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id, u32 effect_index);
/**
* Generate a capture effect command.
* Writes a mix buffer back to game memory.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Capture effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateCaptureCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate a compressor effect command.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param effect_info - Compressor effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateCompressorCommand(s16 buffer_offset, EffectInfoBase& effect_info, s32 node_id);
/**
* Generate all effect commands for a mix.
*
* @param mix_info - Mix to generate effects from.
*/
void GenerateEffectCommand(MixInfo& mix_info);
/**
* Generate all mix commands.
*
* @param mix_info - Mix to generate effects from.
*/
void GenerateMixCommands(MixInfo& mix_info);
/**
* Generate a submix command.
* Generates all effects and all mixing commands.
*
* @param mix_info - Mix to generate effects from.
*/
void GenerateSubMixCommand(MixInfo& mix_info);
/**
* Generate all submix command.
*/
void GenerateSubMixCommands();
/**
* Generate the final mix.
*/
void GenerateFinalMixCommand();
/**
* Generate the final mix commands.
*/
void GenerateFinalMixCommands();
/**
* Generate all sink commands.
*/
void GenerateSinkCommands();
/**
* Generate a sink command.
* Sends samples out to the backend, or a game-supplied circular buffer.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param sink_info - Sink info to generate the commands from.
*/
void GenerateSinkCommand(s16 buffer_offset, SinkInfoBase& sink_info);
/**
* Generate a device sink command.
* Sends samples out to the backend.
*
* @param buffer_offset - Base mix buffer offset to use.
* @param sink_info - Sink info to generate the commands from.
*/
void GenerateDeviceSinkCommand(s16 buffer_offset, SinkInfoBase& sink_info);
/**
* Generate a performance command.
* Used to report performance metrics of the AudioRenderer back to the game.
*
* @param node_id - Node ID of the mix this command is generated for
* @param state - Output state of the generated performance command
* @param entry_addresses - Addresses to be written
*/
void GeneratePerformanceCommand(s32 node_id, PerformanceState state,
const PerformanceEntryAddresses& entry_addresses);
private:
/// Commands will be written by this buffer
CommandBuffer& command_buffer;
/// Header information for the commands generated
const CommandListHeader& command_header;
/// Various things to control generation
const AudioRendererSystemContext& render_context;
/// Used for generating voices
VoiceContext& voice_context;
/// Used for generating mixes
MixContext& mix_context;
/// Used for generating effects
EffectContext& effect_context;
/// Used for generating sinks
SinkContext& sink_context;
/// Used for generating submixes
SplitterContext& splitter_context;
/// Used for generating performance
PerformanceManager* performance_manager;
};
} // namespace AudioRenderer
} // namespace AudioCore

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@@ -1,22 +1,22 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/common/common.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
struct CommandListHeader {
u64 buffer_size;
u32 command_count;
std::span<s32> samples_buffer;
s16 buffer_count;
u32 sample_count;
u32 sample_rate;
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/common/common.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
struct CommandListHeader {
u64 buffer_size;
u32 command_count;
std::span<s32> samples_buffer;
s16 buffer_count;
u32 sample_count;
u32 sample_rate;
};
} // namespace AudioCore::AudioRenderer

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@@ -1,254 +1,254 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "audio_core/renderer/command/commands.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
/**
* Estimate the processing time required for all commands.
*/
class ICommandProcessingTimeEstimator {
public:
virtual ~ICommandProcessingTimeEstimator() = default;
virtual u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const = 0;
virtual u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const = 0;
virtual u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const = 0;
virtual u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const = 0;
virtual u32 Estimate(const AdpcmDataSourceVersion1Command& command) const = 0;
virtual u32 Estimate(const AdpcmDataSourceVersion2Command& command) const = 0;
virtual u32 Estimate(const VolumeCommand& command) const = 0;
virtual u32 Estimate(const VolumeRampCommand& command) const = 0;
virtual u32 Estimate(const BiquadFilterCommand& command) const = 0;
virtual u32 Estimate(const MixCommand& command) const = 0;
virtual u32 Estimate(const MixRampCommand& command) const = 0;
virtual u32 Estimate(const MixRampGroupedCommand& command) const = 0;
virtual u32 Estimate(const DepopPrepareCommand& command) const = 0;
virtual u32 Estimate(const DepopForMixBuffersCommand& command) const = 0;
virtual u32 Estimate(const DelayCommand& command) const = 0;
virtual u32 Estimate(const UpsampleCommand& command) const = 0;
virtual u32 Estimate(const DownMix6chTo2chCommand& command) const = 0;
virtual u32 Estimate(const AuxCommand& command) const = 0;
virtual u32 Estimate(const DeviceSinkCommand& command) const = 0;
virtual u32 Estimate(const CircularBufferSinkCommand& command) const = 0;
virtual u32 Estimate(const ReverbCommand& command) const = 0;
virtual u32 Estimate(const I3dl2ReverbCommand& command) const = 0;
virtual u32 Estimate(const PerformanceCommand& command) const = 0;
virtual u32 Estimate(const ClearMixBufferCommand& command) const = 0;
virtual u32 Estimate(const CopyMixBufferCommand& command) const = 0;
virtual u32 Estimate(const LightLimiterVersion1Command& command) const = 0;
virtual u32 Estimate(const LightLimiterVersion2Command& command) const = 0;
virtual u32 Estimate(const MultiTapBiquadFilterCommand& command) const = 0;
virtual u32 Estimate(const CaptureCommand& command) const = 0;
virtual u32 Estimate(const CompressorCommand& command) const = 0;
};
class CommandProcessingTimeEstimatorVersion1 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion1(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion2 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion2(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion3 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion3(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion4 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion4(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion5 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion5(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "audio_core/renderer/command/commands.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
/**
* Estimate the processing time required for all commands.
*/
class ICommandProcessingTimeEstimator {
public:
virtual ~ICommandProcessingTimeEstimator() = default;
virtual u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const = 0;
virtual u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const = 0;
virtual u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const = 0;
virtual u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const = 0;
virtual u32 Estimate(const AdpcmDataSourceVersion1Command& command) const = 0;
virtual u32 Estimate(const AdpcmDataSourceVersion2Command& command) const = 0;
virtual u32 Estimate(const VolumeCommand& command) const = 0;
virtual u32 Estimate(const VolumeRampCommand& command) const = 0;
virtual u32 Estimate(const BiquadFilterCommand& command) const = 0;
virtual u32 Estimate(const MixCommand& command) const = 0;
virtual u32 Estimate(const MixRampCommand& command) const = 0;
virtual u32 Estimate(const MixRampGroupedCommand& command) const = 0;
virtual u32 Estimate(const DepopPrepareCommand& command) const = 0;
virtual u32 Estimate(const DepopForMixBuffersCommand& command) const = 0;
virtual u32 Estimate(const DelayCommand& command) const = 0;
virtual u32 Estimate(const UpsampleCommand& command) const = 0;
virtual u32 Estimate(const DownMix6chTo2chCommand& command) const = 0;
virtual u32 Estimate(const AuxCommand& command) const = 0;
virtual u32 Estimate(const DeviceSinkCommand& command) const = 0;
virtual u32 Estimate(const CircularBufferSinkCommand& command) const = 0;
virtual u32 Estimate(const ReverbCommand& command) const = 0;
virtual u32 Estimate(const I3dl2ReverbCommand& command) const = 0;
virtual u32 Estimate(const PerformanceCommand& command) const = 0;
virtual u32 Estimate(const ClearMixBufferCommand& command) const = 0;
virtual u32 Estimate(const CopyMixBufferCommand& command) const = 0;
virtual u32 Estimate(const LightLimiterVersion1Command& command) const = 0;
virtual u32 Estimate(const LightLimiterVersion2Command& command) const = 0;
virtual u32 Estimate(const MultiTapBiquadFilterCommand& command) const = 0;
virtual u32 Estimate(const CaptureCommand& command) const = 0;
virtual u32 Estimate(const CompressorCommand& command) const = 0;
};
class CommandProcessingTimeEstimatorVersion1 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion1(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion2 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion2(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion3 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion3(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion4 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion4(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
class CommandProcessingTimeEstimatorVersion5 final : public ICommandProcessingTimeEstimator {
public:
CommandProcessingTimeEstimatorVersion5(u32 sample_count_, u32 buffer_count_)
: sample_count{sample_count_}, buffer_count{buffer_count_} {}
u32 Estimate(const PcmInt16DataSourceVersion1Command& command) const override;
u32 Estimate(const PcmInt16DataSourceVersion2Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion1Command& command) const override;
u32 Estimate(const PcmFloatDataSourceVersion2Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion1Command& command) const override;
u32 Estimate(const AdpcmDataSourceVersion2Command& command) const override;
u32 Estimate(const VolumeCommand& command) const override;
u32 Estimate(const VolumeRampCommand& command) const override;
u32 Estimate(const BiquadFilterCommand& command) const override;
u32 Estimate(const MixCommand& command) const override;
u32 Estimate(const MixRampCommand& command) const override;
u32 Estimate(const MixRampGroupedCommand& command) const override;
u32 Estimate(const DepopPrepareCommand& command) const override;
u32 Estimate(const DepopForMixBuffersCommand& command) const override;
u32 Estimate(const DelayCommand& command) const override;
u32 Estimate(const UpsampleCommand& command) const override;
u32 Estimate(const DownMix6chTo2chCommand& command) const override;
u32 Estimate(const AuxCommand& command) const override;
u32 Estimate(const DeviceSinkCommand& command) const override;
u32 Estimate(const CircularBufferSinkCommand& command) const override;
u32 Estimate(const ReverbCommand& command) const override;
u32 Estimate(const I3dl2ReverbCommand& command) const override;
u32 Estimate(const PerformanceCommand& command) const override;
u32 Estimate(const ClearMixBufferCommand& command) const override;
u32 Estimate(const CopyMixBufferCommand& command) const override;
u32 Estimate(const LightLimiterVersion1Command& command) const override;
u32 Estimate(const LightLimiterVersion2Command& command) const override;
u32 Estimate(const MultiTapBiquadFilterCommand& command) const override;
u32 Estimate(const CaptureCommand& command) const override;
u32 Estimate(const CompressorCommand& command) const override;
private:
u32 sample_count{};
u32 buffer_count{};
};
} // namespace AudioCore::AudioRenderer

64
src/audio_core/renderer/command/commands.h
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@@ -1,32 +1,32 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "audio_core/renderer/command/data_source/adpcm.h"
#include "audio_core/renderer/command/data_source/pcm_float.h"
#include "audio_core/renderer/command/data_source/pcm_int16.h"
#include "audio_core/renderer/command/effect/aux_.h"
#include "audio_core/renderer/command/effect/biquad_filter.h"
#include "audio_core/renderer/command/effect/capture.h"
#include "audio_core/renderer/command/effect/compressor.h"
#include "audio_core/renderer/command/effect/delay.h"
#include "audio_core/renderer/command/effect/i3dl2_reverb.h"
#include "audio_core/renderer/command/effect/light_limiter.h"
#include "audio_core/renderer/command/effect/multi_tap_biquad_filter.h"
#include "audio_core/renderer/command/effect/reverb.h"
#include "audio_core/renderer/command/icommand.h"
#include "audio_core/renderer/command/mix/clear_mix.h"
#include "audio_core/renderer/command/mix/copy_mix.h"
#include "audio_core/renderer/command/mix/depop_for_mix_buffers.h"
#include "audio_core/renderer/command/mix/depop_prepare.h"
#include "audio_core/renderer/command/mix/mix.h"
#include "audio_core/renderer/command/mix/mix_ramp.h"
#include "audio_core/renderer/command/mix/mix_ramp_grouped.h"
#include "audio_core/renderer/command/mix/volume.h"
#include "audio_core/renderer/command/mix/volume_ramp.h"
#include "audio_core/renderer/command/performance/performance.h"
#include "audio_core/renderer/command/resample/downmix_6ch_to_2ch.h"
#include "audio_core/renderer/command/resample/upsample.h"
#include "audio_core/renderer/command/sink/circular_buffer.h"
#include "audio_core/renderer/command/sink/device.h"
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "audio_core/renderer/command/data_source/adpcm.h"
#include "audio_core/renderer/command/data_source/pcm_float.h"
#include "audio_core/renderer/command/data_source/pcm_int16.h"
#include "audio_core/renderer/command/effect/aux_.h"
#include "audio_core/renderer/command/effect/biquad_filter.h"
#include "audio_core/renderer/command/effect/capture.h"
#include "audio_core/renderer/command/effect/compressor.h"
#include "audio_core/renderer/command/effect/delay.h"
#include "audio_core/renderer/command/effect/i3dl2_reverb.h"
#include "audio_core/renderer/command/effect/light_limiter.h"
#include "audio_core/renderer/command/effect/multi_tap_biquad_filter.h"
#include "audio_core/renderer/command/effect/reverb.h"
#include "audio_core/renderer/command/icommand.h"
#include "audio_core/renderer/command/mix/clear_mix.h"
#include "audio_core/renderer/command/mix/copy_mix.h"
#include "audio_core/renderer/command/mix/depop_for_mix_buffers.h"
#include "audio_core/renderer/command/mix/depop_prepare.h"
#include "audio_core/renderer/command/mix/mix.h"
#include "audio_core/renderer/command/mix/mix_ramp.h"
#include "audio_core/renderer/command/mix/mix_ramp_grouped.h"
#include "audio_core/renderer/command/mix/volume.h"
#include "audio_core/renderer/command/mix/volume_ramp.h"
#include "audio_core/renderer/command/performance/performance.h"
#include "audio_core/renderer/command/resample/downmix_6ch_to_2ch.h"
#include "audio_core/renderer/command/resample/upsample.h"
#include "audio_core/renderer/command/sink/circular_buffer.h"
#include "audio_core/renderer/command/sink/device.h"

168
src/audio_core/renderer/command/data_source/adpcm.cpp
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@@ -1,84 +1,84 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <span>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/data_source/adpcm.h"
#include "audio_core/renderer/command/data_source/decode.h"
namespace AudioCore::AudioRenderer {
void AdpcmDataSourceVersion1Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("AdpcmDataSourceVersion1Command\n\toutput_index {:02X} source sample "
"rate {} target sample rate {} src quality {}\n",
output_index, sample_rate, processor.target_sample_rate, src_quality);
}
void AdpcmDataSourceVersion1Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::Adpcm},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{0},
.channel_count{1},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{data_address},
.data_size{data_size},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool AdpcmDataSourceVersion1Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
void AdpcmDataSourceVersion2Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("AdpcmDataSourceVersion2Command\n\toutput_index {:02X} source sample "
"rate {} target sample rate {} src quality {}\n",
output_index, sample_rate, processor.target_sample_rate, src_quality);
}
void AdpcmDataSourceVersion2Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::Adpcm},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{0},
.channel_count{1},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{data_address},
.data_size{data_size},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool AdpcmDataSourceVersion2Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <span>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/data_source/adpcm.h"
#include "audio_core/renderer/command/data_source/decode.h"
namespace AudioCore::AudioRenderer {
void AdpcmDataSourceVersion1Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("AdpcmDataSourceVersion1Command\n\toutput_index {:02X} source sample "
"rate {} target sample rate {} src quality {}\n",
output_index, sample_rate, processor.target_sample_rate, src_quality);
}
void AdpcmDataSourceVersion1Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::Adpcm},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{0},
.channel_count{1},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{data_address},
.data_size{data_size},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool AdpcmDataSourceVersion1Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
void AdpcmDataSourceVersion2Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("AdpcmDataSourceVersion2Command\n\toutput_index {:02X} source sample "
"rate {} target sample rate {} src quality {}\n",
output_index, sample_rate, processor.target_sample_rate, src_quality);
}
void AdpcmDataSourceVersion2Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::Adpcm},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{0},
.channel_count{1},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{data_address},
.data_size{data_size},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool AdpcmDataSourceVersion2Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

238
src/audio_core/renderer/command/data_source/adpcm.h
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@@ -1,119 +1,119 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <string>
#include "audio_core/common/common.h"
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command to decode ADPCM-encoded version 1 wavebuffers
* into the output_index mix buffer.
*/
struct AdpcmDataSourceVersion1Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
/// Coefficients data address
CpuAddr data_address;
/// Coefficients data size
u64 data_size;
};
/**
* AudioRenderer command to decode ADPCM-encoded version 2 wavebuffers
* into the output_index mix buffer.
*/
struct AdpcmDataSourceVersion2Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
/// Coefficients data address
CpuAddr data_address;
/// Coefficients data size
u64 data_size;
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <string>
#include "audio_core/common/common.h"
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command to decode ADPCM-encoded version 1 wavebuffers
* into the output_index mix buffer.
*/
struct AdpcmDataSourceVersion1Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
/// Coefficients data address
CpuAddr data_address;
/// Coefficients data size
u64 data_size;
};
/**
* AudioRenderer command to decode ADPCM-encoded version 2 wavebuffers
* into the output_index mix buffer.
*/
struct AdpcmDataSourceVersion2Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
/// Coefficients data address
CpuAddr data_address;
/// Coefficients data size
u64 data_size;
};
} // namespace AudioCore::AudioRenderer

856
src/audio_core/renderer/command/data_source/decode.cpp
View File

@@ -1,428 +1,428 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <array>
#include <vector>
#include "audio_core/renderer/command/data_source/decode.h"
#include "audio_core/renderer/command/resample/resample.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
#include "core/memory.h"
namespace AudioCore::AudioRenderer {
constexpr u32 TempBufferSize = 0x3F00;
constexpr std::array<u8, 3> PitchBySrcQuality = {4, 8, 4};
/**
* Decode PCM data. Only s16 or f32 is supported.
*
* @tparam T - Type to decode. Only s16 and f32 are supported.
* @param memory - Core memory for reading samples.
* @param out_buffer - Output mix buffer to receive the samples.
* @param req - Information for how to decode.
* @return Number of samples decoded.
*/
template <typename T>
static u32 DecodePcm(Core::Memory::Memory& memory, std::span<s16> out_buffer,
const DecodeArg& req) {
constexpr s32 min{std::numeric_limits<s16>::min()};
constexpr s32 max{std::numeric_limits<s16>::max()};
if (req.buffer == 0 || req.buffer_size == 0) {
return 0;
}
if (req.start_offset >= req.end_offset) {
return 0;
}
auto samples_to_decode{
std::min(req.samples_to_read, req.end_offset - req.start_offset - req.offset)};
u32 channel_count{static_cast<u32>(req.channel_count)};
switch (req.channel_count) {
default: {
const VAddr source{req.buffer +
(((req.start_offset + req.offset) * channel_count) * sizeof(T))};
const u64 size{channel_count * samples_to_decode};
const u64 size_bytes{size * sizeof(T)};
std::vector<T> samples(size);
memory.ReadBlockUnsafe(source, samples.data(), size_bytes);
if constexpr (std::is_floating_point_v<T>) {
for (u32 i = 0; i < samples_to_decode; i++) {
auto sample{static_cast<s32>(samples[i * channel_count + req.target_channel] *
std::numeric_limits<s16>::max())};
out_buffer[i] = static_cast<s16>(std::clamp(sample, min, max));
}
} else {
for (u32 i = 0; i < samples_to_decode; i++) {
out_buffer[i] = samples[i * channel_count + req.target_channel];
}
}
} break;
case 1:
if (req.target_channel != 0) {
LOG_ERROR(Service_Audio, "Invalid target channel, expected 0, got {}",
req.target_channel);
return 0;
}
const VAddr source{req.buffer + ((req.start_offset + req.offset) * sizeof(T))};
std::vector<T> samples(samples_to_decode);
memory.ReadBlockUnsafe(source, samples.data(), samples_to_decode * sizeof(T));
if constexpr (std::is_floating_point_v<T>) {
for (u32 i = 0; i < samples_to_decode; i++) {
auto sample{static_cast<s32>(samples[i * channel_count + req.target_channel] *
std::numeric_limits<s16>::max())};
out_buffer[i] = static_cast<s16>(std::clamp(sample, min, max));
}
} else {
std::memcpy(out_buffer.data(), samples.data(), samples_to_decode * sizeof(s16));
}
break;
}
return samples_to_decode;
}
/**
* Decode ADPCM data.
*
* @param memory - Core memory for reading samples.
* @param out_buffer - Output mix buffer to receive the samples.
* @param req - Information for how to decode.
* @return Number of samples decoded.
*/
static u32 DecodeAdpcm(Core::Memory::Memory& memory, std::span<s16> out_buffer,
const DecodeArg& req) {
constexpr u32 SamplesPerFrame{14};
constexpr u32 NibblesPerFrame{16};
if (req.buffer == 0 || req.buffer_size == 0) {
return 0;
}
if (req.end_offset < req.start_offset) {
return 0;
}
auto end{(req.end_offset % SamplesPerFrame) +
NibblesPerFrame * (req.end_offset / SamplesPerFrame)};
if (req.end_offset % SamplesPerFrame) {
end += 3;
} else {
end += 1;
}
if (req.buffer_size < end / 2) {
return 0;
}
auto samples_to_process{
std::min(req.end_offset - req.start_offset - req.offset, req.samples_to_read)};
auto samples_to_read{samples_to_process};
auto start_pos{req.start_offset + req.offset};
auto samples_remaining_in_frame{start_pos % SamplesPerFrame};
auto position_in_frame{(start_pos / SamplesPerFrame) * NibblesPerFrame +
samples_remaining_in_frame};
if (samples_remaining_in_frame) {
position_in_frame += 2;
}
const auto size{std::max((samples_to_process / 8U) * SamplesPerFrame, 8U)};
std::vector<u8> wavebuffer(size);
memory.ReadBlockUnsafe(req.buffer + position_in_frame / 2, wavebuffer.data(),
wavebuffer.size());
auto context{req.adpcm_context};
auto header{context->header};
u8 coeff_index{static_cast<u8>((header >> 4U) & 0xFU)};
u8 scale{static_cast<u8>(header & 0xFU)};
s32 coeff0{req.coefficients[coeff_index * 2 + 0]};
s32 coeff1{req.coefficients[coeff_index * 2 + 1]};
auto yn0{context->yn0};
auto yn1{context->yn1};
static constexpr std::array<s32, 16> Steps{
0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1,
};
const auto decode_sample = [&](const s32 code) -> s16 {
const auto xn = code * (1 << scale);
const auto prediction = coeff0 * yn0 + coeff1 * yn1;
const auto sample = ((xn << 11) + 0x400 + prediction) >> 11;
const auto saturated = std::clamp<s32>(sample, -0x8000, 0x7FFF);
yn1 = yn0;
yn0 = static_cast<s16>(saturated);
return yn0;
};
u32 read_index{0};
u32 write_index{0};
while (samples_to_read > 0) {
// Are we at a new frame?
if ((position_in_frame % NibblesPerFrame) == 0) {
header = wavebuffer[read_index++];
coeff_index = (header >> 4) & 0xF;
scale = header & 0xF;
coeff0 = req.coefficients[coeff_index * 2 + 0];
coeff1 = req.coefficients[coeff_index * 2 + 1];
position_in_frame += 2;
// Can we consume all of this frame's samples?
if (samples_to_read >= SamplesPerFrame) {
// Can grab all samples until the next header
for (u32 i = 0; i < SamplesPerFrame / 2; i++) {
auto code0{Steps[(wavebuffer[read_index] >> 4) & 0xF]};
auto code1{Steps[wavebuffer[read_index] & 0xF]};
read_index++;
out_buffer[write_index++] = decode_sample(code0);
out_buffer[write_index++] = decode_sample(code1);
}
position_in_frame += SamplesPerFrame;
samples_to_read -= SamplesPerFrame;
continue;
}
}
// Decode a single sample
auto code{wavebuffer[read_index]};
if (position_in_frame & 1) {
code &= 0xF;
read_index++;
} else {
code >>= 4;
}
out_buffer[write_index++] = decode_sample(Steps[code]);
position_in_frame++;
samples_to_read--;
}
context->header = header;
context->yn0 = yn0;
context->yn1 = yn1;
return samples_to_process;
}
/**
* Decode implementation.
* Decode wavebuffers according to the given args.
*
* @param memory - Core memory to read data from.
* @param args - The wavebuffer data, and information for how to decode it.
*/
void DecodeFromWaveBuffers(Core::Memory::Memory& memory, const DecodeFromWaveBuffersArgs& args) {
auto& voice_state{*args.voice_state};
auto remaining_sample_count{args.sample_count};
auto fraction{voice_state.fraction};
const auto sample_rate_ratio{
(Common::FixedPoint<49, 15>(args.source_sample_rate) / args.target_sample_rate) *
args.pitch};
const auto size_required{fraction + remaining_sample_count * sample_rate_ratio};
if (size_required < 0) {
return;
}
auto pitch{PitchBySrcQuality[static_cast<u32>(args.src_quality)]};
if (static_cast<u32>(pitch + size_required.to_int_floor()) > TempBufferSize) {
return;
}
auto max_remaining_sample_count{
((Common::FixedPoint<17, 15>(TempBufferSize) - fraction) / sample_rate_ratio)
.to_uint_floor()};
max_remaining_sample_count = std::min(max_remaining_sample_count, remaining_sample_count);
auto wavebuffers_consumed{voice_state.wave_buffers_consumed};
auto wavebuffer_index{voice_state.wave_buffer_index};
auto played_sample_count{voice_state.played_sample_count};
bool is_buffer_starved{false};
u32 offset{voice_state.offset};
auto output_buffer{args.output};
std::vector<s16> temp_buffer(TempBufferSize, 0);
while (remaining_sample_count > 0) {
const auto samples_to_write{std::min(remaining_sample_count, max_remaining_sample_count)};
const auto samples_to_read{
(fraction + samples_to_write * sample_rate_ratio).to_uint_floor()};
u32 temp_buffer_pos{0};
if (!args.IsVoicePitchAndSrcSkippedSupported) {
for (u32 i = 0; i < pitch; i++) {
temp_buffer[i] = voice_state.sample_history[i];
}
temp_buffer_pos = pitch;
}
u32 samples_read{0};
while (samples_read < samples_to_read) {
if (wavebuffer_index >= MaxWaveBuffers) {
LOG_ERROR(Service_Audio, "Invalid wavebuffer index! {}", wavebuffer_index);
wavebuffer_index = 0;
voice_state.wave_buffer_valid.fill(false);
wavebuffers_consumed = MaxWaveBuffers;
}
if (!voice_state.wave_buffer_valid[wavebuffer_index]) {
is_buffer_starved = true;
break;
}
auto& wavebuffer{args.wave_buffers[wavebuffer_index]};
if (offset == 0 && args.sample_format == SampleFormat::Adpcm &&
wavebuffer.context != 0) {
memory.ReadBlockUnsafe(wavebuffer.context, &voice_state.adpcm_context,
wavebuffer.context_size);
}
auto start_offset{wavebuffer.start_offset};
auto end_offset{wavebuffer.end_offset};
if (wavebuffer.loop && voice_state.loop_count > 0 &&
wavebuffer.loop_start_offset != 0 && wavebuffer.loop_end_offset != 0 &&
wavebuffer.loop_start_offset <= wavebuffer.loop_end_offset) {
start_offset = wavebuffer.loop_start_offset;
end_offset = wavebuffer.loop_end_offset;
}
DecodeArg decode_arg{.buffer{wavebuffer.buffer},
.buffer_size{wavebuffer.buffer_size},
.start_offset{start_offset},
.end_offset{end_offset},
.channel_count{args.channel_count},
.coefficients{},
.adpcm_context{nullptr},
.target_channel{args.channel},
.offset{offset},
.samples_to_read{samples_to_read - samples_read}};
s32 samples_decoded{0};
switch (args.sample_format) {
case SampleFormat::PcmInt16:
samples_decoded = DecodePcm<s16>(
memory, {&temp_buffer[temp_buffer_pos], TempBufferSize - temp_buffer_pos},
decode_arg);
break;
case SampleFormat::PcmFloat:
samples_decoded = DecodePcm<f32>(
memory, {&temp_buffer[temp_buffer_pos], TempBufferSize - temp_buffer_pos},
decode_arg);
break;
case SampleFormat::Adpcm: {
decode_arg.adpcm_context = &voice_state.adpcm_context;
memory.ReadBlockUnsafe(args.data_address, &decode_arg.coefficients, args.data_size);
samples_decoded = DecodeAdpcm(
memory, {&temp_buffer[temp_buffer_pos], TempBufferSize - temp_buffer_pos},
decode_arg);
} break;
default:
LOG_ERROR(Service_Audio, "Invalid sample format to decode {}",
static_cast<u32>(args.sample_format));
samples_decoded = 0;
break;
}
played_sample_count += samples_decoded;
samples_read += samples_decoded;
temp_buffer_pos += samples_decoded;
offset += samples_decoded;
if (samples_decoded == 0 || offset >= end_offset - start_offset) {
offset = 0;
if (!wavebuffer.loop) {
voice_state.wave_buffer_valid[wavebuffer_index] = false;
voice_state.loop_count = 0;
if (wavebuffer.stream_ended) {
played_sample_count = 0;
}
wavebuffer_index = (wavebuffer_index + 1) % MaxWaveBuffers;
wavebuffers_consumed++;
} else {
voice_state.loop_count++;
if (wavebuffer.loop_count > 0 &&
(voice_state.loop_count > wavebuffer.loop_count || samples_decoded == 0)) {
voice_state.wave_buffer_valid[wavebuffer_index] = false;
voice_state.loop_count = 0;
if (wavebuffer.stream_ended) {
played_sample_count = 0;
}
wavebuffer_index = (wavebuffer_index + 1) % MaxWaveBuffers;
wavebuffers_consumed++;
}
if (samples_decoded == 0) {
is_buffer_starved = true;
break;
}
if (args.IsVoicePlayedSampleCountResetAtLoopPointSupported) {
played_sample_count = 0;
}
}
}
}
if (args.IsVoicePitchAndSrcSkippedSupported) {
if (samples_read > output_buffer.size()) {
LOG_ERROR(Service_Audio, "Attempting to write past the end of output buffer!");
}
for (u32 i = 0; i < samples_read; i++) {
output_buffer[i] = temp_buffer[i];
}
} else {
std::memset(&temp_buffer[temp_buffer_pos], 0,
(samples_to_read - samples_read) * sizeof(s16));
Resample(output_buffer, temp_buffer, sample_rate_ratio, fraction, samples_to_write,
args.src_quality);
std::memcpy(voice_state.sample_history.data(), &temp_buffer[samples_to_read],
pitch * sizeof(s16));
}
remaining_sample_count -= samples_to_write;
if (remaining_sample_count != 0 && is_buffer_starved) {
LOG_ERROR(Service_Audio, "Samples remaining but buffer is starving??");
break;
}
output_buffer = output_buffer.subspan(samples_to_write);
}
voice_state.wave_buffers_consumed = wavebuffers_consumed;
voice_state.played_sample_count = played_sample_count;
voice_state.wave_buffer_index = wavebuffer_index;
voice_state.offset = offset;
voice_state.fraction = fraction;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <array>
#include <vector>
#include "audio_core/renderer/command/data_source/decode.h"
#include "audio_core/renderer/command/resample/resample.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
#include "core/memory.h"
namespace AudioCore::AudioRenderer {
constexpr u32 TempBufferSize = 0x3F00;
constexpr std::array<u8, 3> PitchBySrcQuality = {4, 8, 4};
/**
* Decode PCM data. Only s16 or f32 is supported.
*
* @tparam T - Type to decode. Only s16 and f32 are supported.
* @param memory - Core memory for reading samples.
* @param out_buffer - Output mix buffer to receive the samples.
* @param req - Information for how to decode.
* @return Number of samples decoded.
*/
template <typename T>
static u32 DecodePcm(Core::Memory::Memory& memory, std::span<s16> out_buffer,
const DecodeArg& req) {
constexpr s32 min{std::numeric_limits<s16>::min()};
constexpr s32 max{std::numeric_limits<s16>::max()};
if (req.buffer == 0 || req.buffer_size == 0) {
return 0;
}
if (req.start_offset >= req.end_offset) {
return 0;
}
auto samples_to_decode{
std::min(req.samples_to_read, req.end_offset - req.start_offset - req.offset)};
u32 channel_count{static_cast<u32>(req.channel_count)};
switch (req.channel_count) {
default: {
const VAddr source{req.buffer +
(((req.start_offset + req.offset) * channel_count) * sizeof(T))};
const u64 size{channel_count * samples_to_decode};
const u64 size_bytes{size * sizeof(T)};
std::vector<T> samples(size);
memory.ReadBlockUnsafe(source, samples.data(), size_bytes);
if constexpr (std::is_floating_point_v<T>) {
for (u32 i = 0; i < samples_to_decode; i++) {
auto sample{static_cast<s32>(samples[i * channel_count + req.target_channel] *
std::numeric_limits<s16>::max())};
out_buffer[i] = static_cast<s16>(std::clamp(sample, min, max));
}
} else {
for (u32 i = 0; i < samples_to_decode; i++) {
out_buffer[i] = samples[i * channel_count + req.target_channel];
}
}
} break;
case 1:
if (req.target_channel != 0) {
LOG_ERROR(Service_Audio, "Invalid target channel, expected 0, got {}",
req.target_channel);
return 0;
}
const VAddr source{req.buffer + ((req.start_offset + req.offset) * sizeof(T))};
std::vector<T> samples(samples_to_decode);
memory.ReadBlockUnsafe(source, samples.data(), samples_to_decode * sizeof(T));
if constexpr (std::is_floating_point_v<T>) {
for (u32 i = 0; i < samples_to_decode; i++) {
auto sample{static_cast<s32>(samples[i * channel_count + req.target_channel] *
std::numeric_limits<s16>::max())};
out_buffer[i] = static_cast<s16>(std::clamp(sample, min, max));
}
} else {
std::memcpy(out_buffer.data(), samples.data(), samples_to_decode * sizeof(s16));
}
break;
}
return samples_to_decode;
}
/**
* Decode ADPCM data.
*
* @param memory - Core memory for reading samples.
* @param out_buffer - Output mix buffer to receive the samples.
* @param req - Information for how to decode.
* @return Number of samples decoded.
*/
static u32 DecodeAdpcm(Core::Memory::Memory& memory, std::span<s16> out_buffer,
const DecodeArg& req) {
constexpr u32 SamplesPerFrame{14};
constexpr u32 NibblesPerFrame{16};
if (req.buffer == 0 || req.buffer_size == 0) {
return 0;
}
if (req.end_offset < req.start_offset) {
return 0;
}
auto end{(req.end_offset % SamplesPerFrame) +
NibblesPerFrame * (req.end_offset / SamplesPerFrame)};
if (req.end_offset % SamplesPerFrame) {
end += 3;
} else {
end += 1;
}
if (req.buffer_size < end / 2) {
return 0;
}
auto samples_to_process{
std::min(req.end_offset - req.start_offset - req.offset, req.samples_to_read)};
auto samples_to_read{samples_to_process};
auto start_pos{req.start_offset + req.offset};
auto samples_remaining_in_frame{start_pos % SamplesPerFrame};
auto position_in_frame{(start_pos / SamplesPerFrame) * NibblesPerFrame +
samples_remaining_in_frame};
if (samples_remaining_in_frame) {
position_in_frame += 2;
}
const auto size{std::max((samples_to_process / 8U) * SamplesPerFrame, 8U)};
std::vector<u8> wavebuffer(size);
memory.ReadBlockUnsafe(req.buffer + position_in_frame / 2, wavebuffer.data(),
wavebuffer.size());
auto context{req.adpcm_context};
auto header{context->header};
u8 coeff_index{static_cast<u8>((header >> 4U) & 0xFU)};
u8 scale{static_cast<u8>(header & 0xFU)};
s32 coeff0{req.coefficients[coeff_index * 2 + 0]};
s32 coeff1{req.coefficients[coeff_index * 2 + 1]};
auto yn0{context->yn0};
auto yn1{context->yn1};
static constexpr std::array<s32, 16> Steps{
0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1,
};
const auto decode_sample = [&](const s32 code) -> s16 {
const auto xn = code * (1 << scale);
const auto prediction = coeff0 * yn0 + coeff1 * yn1;
const auto sample = ((xn << 11) + 0x400 + prediction) >> 11;
const auto saturated = std::clamp<s32>(sample, -0x8000, 0x7FFF);
yn1 = yn0;
yn0 = static_cast<s16>(saturated);
return yn0;
};
u32 read_index{0};
u32 write_index{0};
while (samples_to_read > 0) {
// Are we at a new frame?
if ((position_in_frame % NibblesPerFrame) == 0) {
header = wavebuffer[read_index++];
coeff_index = (header >> 4) & 0xF;
scale = header & 0xF;
coeff0 = req.coefficients[coeff_index * 2 + 0];
coeff1 = req.coefficients[coeff_index * 2 + 1];
position_in_frame += 2;
// Can we consume all of this frame's samples?
if (samples_to_read >= SamplesPerFrame) {
// Can grab all samples until the next header
for (u32 i = 0; i < SamplesPerFrame / 2; i++) {
auto code0{Steps[(wavebuffer[read_index] >> 4) & 0xF]};
auto code1{Steps[wavebuffer[read_index] & 0xF]};
read_index++;
out_buffer[write_index++] = decode_sample(code0);
out_buffer[write_index++] = decode_sample(code1);
}
position_in_frame += SamplesPerFrame;
samples_to_read -= SamplesPerFrame;
continue;
}
}
// Decode a single sample
auto code{wavebuffer[read_index]};
if (position_in_frame & 1) {
code &= 0xF;
read_index++;
} else {
code >>= 4;
}
out_buffer[write_index++] = decode_sample(Steps[code]);
position_in_frame++;
samples_to_read--;
}
context->header = header;
context->yn0 = yn0;
context->yn1 = yn1;
return samples_to_process;
}
/**
* Decode implementation.
* Decode wavebuffers according to the given args.
*
* @param memory - Core memory to read data from.
* @param args - The wavebuffer data, and information for how to decode it.
*/
void DecodeFromWaveBuffers(Core::Memory::Memory& memory, const DecodeFromWaveBuffersArgs& args) {
auto& voice_state{*args.voice_state};
auto remaining_sample_count{args.sample_count};
auto fraction{voice_state.fraction};
const auto sample_rate_ratio{
(Common::FixedPoint<49, 15>(args.source_sample_rate) / args.target_sample_rate) *
args.pitch};
const auto size_required{fraction + remaining_sample_count * sample_rate_ratio};
if (size_required < 0) {
return;
}
auto pitch{PitchBySrcQuality[static_cast<u32>(args.src_quality)]};
if (static_cast<u32>(pitch + size_required.to_int_floor()) > TempBufferSize) {
return;
}
auto max_remaining_sample_count{
((Common::FixedPoint<17, 15>(TempBufferSize) - fraction) / sample_rate_ratio)
.to_uint_floor()};
max_remaining_sample_count = std::min(max_remaining_sample_count, remaining_sample_count);
auto wavebuffers_consumed{voice_state.wave_buffers_consumed};
auto wavebuffer_index{voice_state.wave_buffer_index};
auto played_sample_count{voice_state.played_sample_count};
bool is_buffer_starved{false};
u32 offset{voice_state.offset};
auto output_buffer{args.output};
std::vector<s16> temp_buffer(TempBufferSize, 0);
while (remaining_sample_count > 0) {
const auto samples_to_write{std::min(remaining_sample_count, max_remaining_sample_count)};
const auto samples_to_read{
(fraction + samples_to_write * sample_rate_ratio).to_uint_floor()};
u32 temp_buffer_pos{0};
if (!args.IsVoicePitchAndSrcSkippedSupported) {
for (u32 i = 0; i < pitch; i++) {
temp_buffer[i] = voice_state.sample_history[i];
}
temp_buffer_pos = pitch;
}
u32 samples_read{0};
while (samples_read < samples_to_read) {
if (wavebuffer_index >= MaxWaveBuffers) {
LOG_ERROR(Service_Audio, "Invalid wavebuffer index! {}", wavebuffer_index);
wavebuffer_index = 0;
voice_state.wave_buffer_valid.fill(false);
wavebuffers_consumed = MaxWaveBuffers;
}
if (!voice_state.wave_buffer_valid[wavebuffer_index]) {
is_buffer_starved = true;
break;
}
auto& wavebuffer{args.wave_buffers[wavebuffer_index]};
if (offset == 0 && args.sample_format == SampleFormat::Adpcm &&
wavebuffer.context != 0) {
memory.ReadBlockUnsafe(wavebuffer.context, &voice_state.adpcm_context,
wavebuffer.context_size);
}
auto start_offset{wavebuffer.start_offset};
auto end_offset{wavebuffer.end_offset};
if (wavebuffer.loop && voice_state.loop_count > 0 &&
wavebuffer.loop_start_offset != 0 && wavebuffer.loop_end_offset != 0 &&
wavebuffer.loop_start_offset <= wavebuffer.loop_end_offset) {
start_offset = wavebuffer.loop_start_offset;
end_offset = wavebuffer.loop_end_offset;
}
DecodeArg decode_arg{.buffer{wavebuffer.buffer},
.buffer_size{wavebuffer.buffer_size},
.start_offset{start_offset},
.end_offset{end_offset},
.channel_count{args.channel_count},
.coefficients{},
.adpcm_context{nullptr},
.target_channel{args.channel},
.offset{offset},
.samples_to_read{samples_to_read - samples_read}};
s32 samples_decoded{0};
switch (args.sample_format) {
case SampleFormat::PcmInt16:
samples_decoded = DecodePcm<s16>(
memory, {&temp_buffer[temp_buffer_pos], TempBufferSize - temp_buffer_pos},
decode_arg);
break;
case SampleFormat::PcmFloat:
samples_decoded = DecodePcm<f32>(
memory, {&temp_buffer[temp_buffer_pos], TempBufferSize - temp_buffer_pos},
decode_arg);
break;
case SampleFormat::Adpcm: {
decode_arg.adpcm_context = &voice_state.adpcm_context;
memory.ReadBlockUnsafe(args.data_address, &decode_arg.coefficients, args.data_size);
samples_decoded = DecodeAdpcm(
memory, {&temp_buffer[temp_buffer_pos], TempBufferSize - temp_buffer_pos},
decode_arg);
} break;
default:
LOG_ERROR(Service_Audio, "Invalid sample format to decode {}",
static_cast<u32>(args.sample_format));
samples_decoded = 0;
break;
}
played_sample_count += samples_decoded;
samples_read += samples_decoded;
temp_buffer_pos += samples_decoded;
offset += samples_decoded;
if (samples_decoded == 0 || offset >= end_offset - start_offset) {
offset = 0;
if (!wavebuffer.loop) {
voice_state.wave_buffer_valid[wavebuffer_index] = false;
voice_state.loop_count = 0;
if (wavebuffer.stream_ended) {
played_sample_count = 0;
}
wavebuffer_index = (wavebuffer_index + 1) % MaxWaveBuffers;
wavebuffers_consumed++;
} else {
voice_state.loop_count++;
if (wavebuffer.loop_count > 0 &&
(voice_state.loop_count > wavebuffer.loop_count || samples_decoded == 0)) {
voice_state.wave_buffer_valid[wavebuffer_index] = false;
voice_state.loop_count = 0;
if (wavebuffer.stream_ended) {
played_sample_count = 0;
}
wavebuffer_index = (wavebuffer_index + 1) % MaxWaveBuffers;
wavebuffers_consumed++;
}
if (samples_decoded == 0) {
is_buffer_starved = true;
break;
}
if (args.IsVoicePlayedSampleCountResetAtLoopPointSupported) {
played_sample_count = 0;
}
}
}
}
if (args.IsVoicePitchAndSrcSkippedSupported) {
if (samples_read > output_buffer.size()) {
LOG_ERROR(Service_Audio, "Attempting to write past the end of output buffer!");
}
for (u32 i = 0; i < samples_read; i++) {
output_buffer[i] = temp_buffer[i];
}
} else {
std::memset(&temp_buffer[temp_buffer_pos], 0,
(samples_to_read - samples_read) * sizeof(s16));
Resample(output_buffer, temp_buffer, sample_rate_ratio, fraction, samples_to_write,
args.src_quality);
std::memcpy(voice_state.sample_history.data(), &temp_buffer[samples_to_read],
pitch * sizeof(s16));
}
remaining_sample_count -= samples_to_write;
if (remaining_sample_count != 0 && is_buffer_starved) {
LOG_ERROR(Service_Audio, "Samples remaining but buffer is starving??");
break;
}
output_buffer = output_buffer.subspan(samples_to_write);
}
voice_state.wave_buffers_consumed = wavebuffers_consumed;
voice_state.played_sample_count = played_sample_count;
voice_state.wave_buffer_index = wavebuffer_index;
voice_state.offset = offset;
voice_state.fraction = fraction;
}
} // namespace AudioCore::AudioRenderer

118
src/audio_core/renderer/command/data_source/decode.h
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@@ -1,59 +1,59 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <span>
#include "audio_core/common/common.h"
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/voice/voice_state.h"
#include "common/common_types.h"
namespace Core::Memory {
class Memory;
}
namespace AudioCore::AudioRenderer {
struct DecodeFromWaveBuffersArgs {
SampleFormat sample_format;
std::span<s32> output;
VoiceState* voice_state;
std::span<WaveBufferVersion2> wave_buffers;
s8 channel;
s8 channel_count;
SrcQuality src_quality;
f32 pitch;
u32 source_sample_rate;
u32 target_sample_rate;
u32 sample_count;
CpuAddr data_address;
u64 data_size;
bool IsVoicePlayedSampleCountResetAtLoopPointSupported;
bool IsVoicePitchAndSrcSkippedSupported;
};
struct DecodeArg {
CpuAddr buffer;
u64 buffer_size;
u32 start_offset;
u32 end_offset;
s8 channel_count;
std::array<s16, 16> coefficients;
VoiceState::AdpcmContext* adpcm_context;
s8 target_channel;
u32 offset;
u32 samples_to_read;
};
/**
* Decode wavebuffers according to the given args.
*
* @param memory - Core memory to read data from.
* @param args - The wavebuffer data, and information for how to decode it.
*/
void DecodeFromWaveBuffers(Core::Memory::Memory& memory, const DecodeFromWaveBuffersArgs& args);
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <span>
#include "audio_core/common/common.h"
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/voice/voice_state.h"
#include "common/common_types.h"
namespace Core::Memory {
class Memory;
}
namespace AudioCore::AudioRenderer {
struct DecodeFromWaveBuffersArgs {
SampleFormat sample_format;
std::span<s32> output;
VoiceState* voice_state;
std::span<WaveBufferVersion2> wave_buffers;
s8 channel;
s8 channel_count;
SrcQuality src_quality;
f32 pitch;
u32 source_sample_rate;
u32 target_sample_rate;
u32 sample_count;
CpuAddr data_address;
u64 data_size;
bool IsVoicePlayedSampleCountResetAtLoopPointSupported;
bool IsVoicePitchAndSrcSkippedSupported;
};
struct DecodeArg {
CpuAddr buffer;
u64 buffer_size;
u32 start_offset;
u32 end_offset;
s8 channel_count;
std::array<s16, 16> coefficients;
VoiceState::AdpcmContext* adpcm_context;
s8 target_channel;
u32 offset;
u32 samples_to_read;
};
/**
* Decode wavebuffers according to the given args.
*
* @param memory - Core memory to read data from.
* @param args - The wavebuffer data, and information for how to decode it.
*/
void DecodeFromWaveBuffers(Core::Memory::Memory& memory, const DecodeFromWaveBuffersArgs& args);
} // namespace AudioCore::AudioRenderer

172
src/audio_core/renderer/command/data_source/pcm_float.cpp
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@@ -1,86 +1,86 @@
// 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/data_source/decode.h"
#include "audio_core/renderer/command/data_source/pcm_float.h"
namespace AudioCore::AudioRenderer {
void PcmFloatDataSourceVersion1Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmFloatDataSourceVersion1Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmFloatDataSourceVersion1Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmFloat},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmFloatDataSourceVersion1Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
void PcmFloatDataSourceVersion2Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmFloatDataSourceVersion2Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmFloatDataSourceVersion2Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmFloat},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmFloatDataSourceVersion2Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/data_source/decode.h"
#include "audio_core/renderer/command/data_source/pcm_float.h"
namespace AudioCore::AudioRenderer {
void PcmFloatDataSourceVersion1Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmFloatDataSourceVersion1Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmFloatDataSourceVersion1Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmFloat},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmFloatDataSourceVersion1Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
void PcmFloatDataSourceVersion2Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmFloatDataSourceVersion2Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmFloatDataSourceVersion2Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmFloat},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmFloatDataSourceVersion2Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

226
src/audio_core/renderer/command/data_source/pcm_float.h
View File

@@ -1,113 +1,113 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <string>
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command to decode PCM float-encoded version 1 wavebuffers
* into the output_index mix buffer.
*/
struct PcmFloatDataSourceVersion1Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
/**
* AudioRenderer command to decode PCM float-encoded version 2 wavebuffers
* into the output_index mix buffer.
*/
struct PcmFloatDataSourceVersion2Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <string>
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command to decode PCM float-encoded version 1 wavebuffers
* into the output_index mix buffer.
*/
struct PcmFloatDataSourceVersion1Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
/**
* AudioRenderer command to decode PCM float-encoded version 2 wavebuffers
* into the output_index mix buffer.
*/
struct PcmFloatDataSourceVersion2Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
} // namespace AudioCore::AudioRenderer

174
src/audio_core/renderer/command/data_source/pcm_int16.cpp
View File

@@ -1,87 +1,87 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <span>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/data_source/decode.h"
#include "audio_core/renderer/command/data_source/pcm_int16.h"
namespace AudioCore::AudioRenderer {
void PcmInt16DataSourceVersion1Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmInt16DataSourceVersion1Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmInt16DataSourceVersion1Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmInt16},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmInt16DataSourceVersion1Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
void PcmInt16DataSourceVersion2Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmInt16DataSourceVersion2Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmInt16DataSourceVersion2Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmInt16},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmInt16DataSourceVersion2Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <span>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/data_source/decode.h"
#include "audio_core/renderer/command/data_source/pcm_int16.h"
namespace AudioCore::AudioRenderer {
void PcmInt16DataSourceVersion1Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmInt16DataSourceVersion1Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmInt16DataSourceVersion1Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmInt16},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmInt16DataSourceVersion1Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
void PcmInt16DataSourceVersion2Command::Dump(const ADSP::CommandListProcessor& processor,
std::string& string) {
string +=
fmt::format("PcmInt16DataSourceVersion2Command\n\toutput_index {:02X} channel {} "
"channel count {} source sample rate {} target sample rate {} src quality {}\n",
output_index, channel_index, channel_count, sample_rate,
processor.target_sample_rate, src_quality);
}
void PcmInt16DataSourceVersion2Command::Process(const ADSP::CommandListProcessor& processor) {
auto out_buffer = processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count);
DecodeFromWaveBuffersArgs args{
.sample_format{SampleFormat::PcmInt16},
.output{out_buffer},
.voice_state{reinterpret_cast<VoiceState*>(voice_state)},
.wave_buffers{wave_buffers},
.channel{channel_index},
.channel_count{channel_count},
.src_quality{src_quality},
.pitch{pitch},
.source_sample_rate{sample_rate},
.target_sample_rate{processor.target_sample_rate},
.sample_count{processor.sample_count},
.data_address{0},
.data_size{0},
.IsVoicePlayedSampleCountResetAtLoopPointSupported{(flags & 1) != 0},
.IsVoicePitchAndSrcSkippedSupported{(flags & 2) != 0},
};
DecodeFromWaveBuffers(*processor.memory, args);
}
bool PcmInt16DataSourceVersion2Command::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

220
src/audio_core/renderer/command/data_source/pcm_int16.h
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@@ -1,110 +1,110 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <string>
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command to decode PCM s16-encoded version 1 wavebuffers
* into the output_index mix buffer.
*/
struct PcmInt16DataSourceVersion1Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
/**
* AudioRenderer command to decode PCM s16-encoded version 2 wavebuffers
* into the output_index mix buffer.
*/
struct PcmInt16DataSourceVersion2Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <string>
#include "audio_core/common/wave_buffer.h"
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command to decode PCM s16-encoded version 1 wavebuffers
* into the output_index mix buffer.
*/
struct PcmInt16DataSourceVersion1Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
/**
* AudioRenderer command to decode PCM s16-encoded version 2 wavebuffers
* into the output_index mix buffer.
*/
struct PcmInt16DataSourceVersion2Command : 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;
/// Quality used for sample rate conversion
SrcQuality src_quality;
/// Mix buffer index for decoded samples
s16 output_index;
/// Flags to control decoding (see AudioCore::AudioRenderer::VoiceInfo::Flags)
u16 flags;
/// Wavebuffer sample rate
u32 sample_rate;
/// Pitch used for sample rate conversion
f32 pitch;
/// Target channel to read within the wavebuffer
s8 channel_index;
/// Number of channels within the wavebuffer
s8 channel_count;
/// Wavebuffers containing the wavebuffer address, context address, looping information etc
std::array<WaveBufferVersion2, MaxWaveBuffers> wave_buffers;
/// Voice state, updated each call and written back to game
CpuAddr voice_state;
};
} // namespace AudioCore::AudioRenderer

414
src/audio_core/renderer/command/effect/aux_.cpp
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@@ -1,207 +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
// 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

132
src/audio_core/renderer/command/effect/aux_.h
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@@ -1,66 +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
// 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

236
src/audio_core/renderer/command/effect/biquad_filter.cpp
View File

@@ -1,118 +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) {
*state_ = {};
}
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
// 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) {
*state_ = {};
}
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

148
src/audio_core/renderer/command/effect/biquad_filter.h
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@@ -1,74 +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
// 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

284
src/audio_core/renderer/command/effect/capture.cpp
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@@ -1,142 +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
// 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

124
src/audio_core/renderer/command/effect/capture.h
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@@ -1,62 +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
// 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

310
src/audio_core/renderer/command/effect/compressor.cpp
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@@ -1,155 +1,155 @@
// 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(const 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(const CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state) {
state = {};
state.unk_00 = 0;
state.unk_04 = 1.0f;
state.unk_08 = 1.0f;
SetCompressorEffectParameter(params, state);
}
static void ApplyCompressorEffect(const 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(output_buffers[channel].data(), 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
// 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(const 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(const CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state) {
state = {};
state.unk_00 = 0;
state.unk_04 = 1.0f;
state.unk_08 = 1.0f;
SetCompressorEffectParameter(params, state);
}
static void ApplyCompressorEffect(const 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(output_buffers[channel].data(), 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

120
src/audio_core/renderer/command/effect/compressor.h
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@@ -1,60 +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
// 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

476
src/audio_core/renderer/command/effect/delay.cpp
View File

@@ -1,238 +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
// 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

120
src/audio_core/renderer/command/effect/delay.h
View File

@@ -1,60 +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
// 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

874
src/audio_core/renderer/command/effect/i3dl2_reverb.cpp
View File

@@ -1,437 +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
// 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

120
src/audio_core/renderer/command/effect/i3dl2_reverb.h
View File

@@ -1,60 +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
// 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

444
src/audio_core/renderer/command/effect/light_limiter.cpp
View File

@@ -1,222 +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
// 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

206
src/audio_core/renderer/command/effect/light_limiter.h
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@@ -1,103 +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
// 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

90
src/audio_core/renderer/command/effect/multi_tap_biquad_filter.cpp
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@@ -1,45 +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]) {
*state = {};
}
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
// 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]) {
*state = {};
}
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

118
src/audio_core/renderer/command/effect/multi_tap_biquad_filter.h
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@@ -1,59 +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
// 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

880
src/audio_core/renderer/command/effect/reverb.cpp
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@@ -1,440 +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
// 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

124
src/audio_core/renderer/command/effect/reverb.h
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@@ -1,62 +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
// 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

186
src/audio_core/renderer/command/icommand.h
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@@ -1,93 +1,93 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "audio_core/common/common.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
enum class CommandId : u8 {
/* 0x00 */ Invalid,
/* 0x01 */ DataSourcePcmInt16Version1,
/* 0x02 */ DataSourcePcmInt16Version2,
/* 0x03 */ DataSourcePcmFloatVersion1,
/* 0x04 */ DataSourcePcmFloatVersion2,
/* 0x05 */ DataSourceAdpcmVersion1,
/* 0x06 */ DataSourceAdpcmVersion2,
/* 0x07 */ Volume,
/* 0x08 */ VolumeRamp,
/* 0x09 */ BiquadFilter,
/* 0x0A */ Mix,
/* 0x0B */ MixRamp,
/* 0x0C */ MixRampGrouped,
/* 0x0D */ DepopPrepare,
/* 0x0E */ DepopForMixBuffers,
/* 0x0F */ Delay,
/* 0x10 */ Upsample,
/* 0x11 */ DownMix6chTo2ch,
/* 0x12 */ Aux,
/* 0x13 */ DeviceSink,
/* 0x14 */ CircularBufferSink,
/* 0x15 */ Reverb,
/* 0x16 */ I3dl2Reverb,
/* 0x17 */ Performance,
/* 0x18 */ ClearMixBuffer,
/* 0x19 */ CopyMixBuffer,
/* 0x1A */ LightLimiterVersion1,
/* 0x1B */ LightLimiterVersion2,
/* 0x1C */ MultiTapBiquadFilter,
/* 0x1D */ Capture,
/* 0x1E */ Compressor,
};
constexpr u32 CommandMagic{0xCAFEBABE};
/**
* A command, generated by the host, and processed by the ADSP's AudioRenderer.
*/
struct ICommand {
virtual ~ICommand() = default;
/**
* Print this command's information to a string.
*
* @param processor - The CommandListProcessor processing this command.
* @param string - The string to print into.
*/
virtual void Dump(const ADSP::CommandListProcessor& processor, std::string& string) = 0;
/**
* Process this command.
*
* @param processor - The CommandListProcessor processing this command.
*/
virtual void Process(const ADSP::CommandListProcessor& processor) = 0;
/**
* Verify this command's data is valid.
*
* @param processor - The CommandListProcessor processing this command.
* @return True if the command is valid, otherwise false.
*/
virtual bool Verify(const ADSP::CommandListProcessor& processor) = 0;
/// Command magic 0xCAFEBABE
u32 magic{};
/// Command enabled
bool enabled{};
/// Type of this command (see CommandId)
CommandId type{};
/// Size of this command
s16 size{};
/// Estimated processing time for this command
u32 estimated_process_time{};
/// Node id of the voice or mix this command was generated from
u32 node_id{};
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "audio_core/common/common.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
enum class CommandId : u8 {
/* 0x00 */ Invalid,
/* 0x01 */ DataSourcePcmInt16Version1,
/* 0x02 */ DataSourcePcmInt16Version2,
/* 0x03 */ DataSourcePcmFloatVersion1,
/* 0x04 */ DataSourcePcmFloatVersion2,
/* 0x05 */ DataSourceAdpcmVersion1,
/* 0x06 */ DataSourceAdpcmVersion2,
/* 0x07 */ Volume,
/* 0x08 */ VolumeRamp,
/* 0x09 */ BiquadFilter,
/* 0x0A */ Mix,
/* 0x0B */ MixRamp,
/* 0x0C */ MixRampGrouped,
/* 0x0D */ DepopPrepare,
/* 0x0E */ DepopForMixBuffers,
/* 0x0F */ Delay,
/* 0x10 */ Upsample,
/* 0x11 */ DownMix6chTo2ch,
/* 0x12 */ Aux,
/* 0x13 */ DeviceSink,
/* 0x14 */ CircularBufferSink,
/* 0x15 */ Reverb,
/* 0x16 */ I3dl2Reverb,
/* 0x17 */ Performance,
/* 0x18 */ ClearMixBuffer,
/* 0x19 */ CopyMixBuffer,
/* 0x1A */ LightLimiterVersion1,
/* 0x1B */ LightLimiterVersion2,
/* 0x1C */ MultiTapBiquadFilter,
/* 0x1D */ Capture,
/* 0x1E */ Compressor,
};
constexpr u32 CommandMagic{0xCAFEBABE};
/**
* A command, generated by the host, and processed by the ADSP's AudioRenderer.
*/
struct ICommand {
virtual ~ICommand() = default;
/**
* Print this command's information to a string.
*
* @param processor - The CommandListProcessor processing this command.
* @param string - The string to print into.
*/
virtual void Dump(const ADSP::CommandListProcessor& processor, std::string& string) = 0;
/**
* Process this command.
*
* @param processor - The CommandListProcessor processing this command.
*/
virtual void Process(const ADSP::CommandListProcessor& processor) = 0;
/**
* Verify this command's data is valid.
*
* @param processor - The CommandListProcessor processing this command.
* @return True if the command is valid, otherwise false.
*/
virtual bool Verify(const ADSP::CommandListProcessor& processor) = 0;
/// Command magic 0xCAFEBABE
u32 magic{};
/// Command enabled
bool enabled{};
/// Type of this command (see CommandId)
CommandId type{};
/// Size of this command
s16 size{};
/// Estimated processing time for this command
u32 estimated_process_time{};
/// Node id of the voice or mix this command was generated from
u32 node_id{};
};
} // namespace AudioCore::AudioRenderer

48
src/audio_core/renderer/command/mix/clear_mix.cpp
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@@ -1,24 +1,24 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <string>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/mix/clear_mix.h"
namespace AudioCore::AudioRenderer {
void ClearMixBufferCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("ClearMixBufferCommand\n");
}
void ClearMixBufferCommand::Process(const ADSP::CommandListProcessor& processor) {
memset(processor.mix_buffers.data(), 0, processor.mix_buffers.size_bytes());
}
bool ClearMixBufferCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <string>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/mix/clear_mix.h"
namespace AudioCore::AudioRenderer {
void ClearMixBufferCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("ClearMixBufferCommand\n");
}
void ClearMixBufferCommand::Process(const ADSP::CommandListProcessor& processor) {
memset(processor.mix_buffers.data(), 0, processor.mix_buffers.size_bytes());
}
bool ClearMixBufferCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

90
src/audio_core/renderer/command/mix/clear_mix.h
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@@ -1,45 +1,45 @@
// 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 a clearing the mix buffers.
* Used at the start of each command list.
*/
struct ClearMixBufferCommand : 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;
};
} // namespace AudioCore::AudioRenderer
// 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 a clearing the mix buffers.
* Used at the start of each command list.
*/
struct ClearMixBufferCommand : 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;
};
} // namespace AudioCore::AudioRenderer

54
src/audio_core/renderer/command/mix/copy_mix.cpp
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@@ -1,27 +1,27 @@
// 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/mix/copy_mix.h"
namespace AudioCore::AudioRenderer {
void CopyMixBufferCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("CopyMixBufferCommand\n\tinput {:02X} output {:02X}\n", input_index,
output_index);
}
void CopyMixBufferCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
std::memcpy(output.data(), input.data(), processor.sample_count * sizeof(s32));
}
bool CopyMixBufferCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/mix/copy_mix.h"
namespace AudioCore::AudioRenderer {
void CopyMixBufferCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("CopyMixBufferCommand\n\tinput {:02X} output {:02X}\n", input_index,
output_index);
}
void CopyMixBufferCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
std::memcpy(output.data(), input.data(), processor.sample_count * sizeof(s32));
}
bool CopyMixBufferCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

98
src/audio_core/renderer/command/mix/copy_mix.h
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@@ -1,49 +1,49 @@
// 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 a copying a mix buffer from input to output.
*/
struct CopyMixBufferCommand : 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_index;
/// Output mix buffer index
s16 output_index;
};
} // namespace AudioCore::AudioRenderer
// 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 a copying a mix buffer from input to output.
*/
struct CopyMixBufferCommand : 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_index;
/// Output mix buffer index
s16 output_index;
};
} // namespace AudioCore::AudioRenderer

128
src/audio_core/renderer/command/mix/depop_for_mix_buffers.cpp
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@@ -1,64 +1,64 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/common/common.h"
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/mix/depop_for_mix_buffers.h"
namespace AudioCore::AudioRenderer {
/**
* Apply depopping. Add the depopped sample to each incoming new sample, decaying it each time
* according to decay.
*
* @param output - Output buffer to be depopped.
* @param depop_sample - Depopped sample to apply to output samples.
* @param decay_ - Amount to decay the depopped sample for every output sample.
* @param sample_count - Samples to process.
* @return Final decayed depop sample.
*/
static s32 ApplyDepopMix(std::span<s32> output, const s32 depop_sample,
Common::FixedPoint<49, 15>& decay_, const u32 sample_count) {
auto sample{std::abs(depop_sample)};
auto decay{decay_.to_raw()};
if (depop_sample <= 0) {
for (u32 i = 0; i < sample_count; i++) {
sample = static_cast<s32>((static_cast<s64>(sample) * decay) >> 15);
output[i] -= sample;
}
return -sample;
} else {
for (u32 i = 0; i < sample_count; i++) {
sample = static_cast<s32>((static_cast<s64>(sample) * decay) >> 15);
output[i] += sample;
}
return sample;
}
}
void DepopForMixBuffersCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DepopForMixBuffersCommand\n\tinput {:02X} count {} decay {}\n", input,
count, decay.to_float());
}
void DepopForMixBuffersCommand::Process(const ADSP::CommandListProcessor& processor) {
auto end_index{std::min(processor.buffer_count, input + count)};
std::span<s32> depop_buff{reinterpret_cast<s32*>(depop_buffer), end_index};
for (u32 index = input; index < end_index; index++) {
const auto depop_sample{depop_buff[index]};
if (depop_sample != 0) {
auto input_buffer{processor.mix_buffers.subspan(index * processor.sample_count,
processor.sample_count)};
depop_buff[index] =
ApplyDepopMix(input_buffer, depop_sample, decay, processor.sample_count);
}
}
}
bool DepopForMixBuffersCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/common/common.h"
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/mix/depop_for_mix_buffers.h"
namespace AudioCore::AudioRenderer {
/**
* Apply depopping. Add the depopped sample to each incoming new sample, decaying it each time
* according to decay.
*
* @param output - Output buffer to be depopped.
* @param depop_sample - Depopped sample to apply to output samples.
* @param decay_ - Amount to decay the depopped sample for every output sample.
* @param sample_count - Samples to process.
* @return Final decayed depop sample.
*/
static s32 ApplyDepopMix(std::span<s32> output, const s32 depop_sample,
Common::FixedPoint<49, 15>& decay_, const u32 sample_count) {
auto sample{std::abs(depop_sample)};
auto decay{decay_.to_raw()};
if (depop_sample <= 0) {
for (u32 i = 0; i < sample_count; i++) {
sample = static_cast<s32>((static_cast<s64>(sample) * decay) >> 15);
output[i] -= sample;
}
return -sample;
} else {
for (u32 i = 0; i < sample_count; i++) {
sample = static_cast<s32>((static_cast<s64>(sample) * decay) >> 15);
output[i] += sample;
}
return sample;
}
}
void DepopForMixBuffersCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DepopForMixBuffersCommand\n\tinput {:02X} count {} decay {}\n", input,
count, decay.to_float());
}
void DepopForMixBuffersCommand::Process(const ADSP::CommandListProcessor& processor) {
auto end_index{std::min(processor.buffer_count, input + count)};
std::span<s32> depop_buff{reinterpret_cast<s32*>(depop_buffer), end_index};
for (u32 index = input; index < end_index; index++) {
const auto depop_sample{depop_buff[index]};
if (depop_sample != 0) {
auto input_buffer{processor.mix_buffers.subspan(index * processor.sample_count,
processor.sample_count)};
depop_buff[index] =
ApplyDepopMix(input_buffer, depop_sample, decay, processor.sample_count);
}
}
}
bool DepopForMixBuffersCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

110
src/audio_core/renderer/command/mix/depop_for_mix_buffers.h
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@@ -1,55 +1,55 @@
// 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"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for depopping a mix buffer.
* Adds a cumulation of previous samples to the current mix buffer with a decay.
*/
struct DepopForMixBuffersCommand : 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;
/// Starting input mix buffer index
u32 input;
/// Number of mix buffers to depop
u32 count;
/// Amount to decay the depop sample for each new sample
Common::FixedPoint<49, 15> decay;
/// Address of the depop buffer, holding the last sample for every mix buffer
CpuAddr depop_buffer;
};
} // namespace AudioCore::AudioRenderer
// 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"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for depopping a mix buffer.
* Adds a cumulation of previous samples to the current mix buffer with a decay.
*/
struct DepopForMixBuffersCommand : 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;
/// Starting input mix buffer index
u32 input;
/// Number of mix buffers to depop
u32 count;
/// Amount to decay the depop sample for each new sample
Common::FixedPoint<49, 15> decay;
/// Address of the depop buffer, holding the last sample for every mix buffer
CpuAddr depop_buffer;
};
} // namespace AudioCore::AudioRenderer

72
src/audio_core/renderer/command/mix/depop_prepare.cpp
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@@ -1,36 +1,36 @@
// 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/mix/depop_prepare.h"
#include "audio_core/renderer/voice/voice_state.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
void DepopPrepareCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DepopPrepareCommand\n\tinputs: ");
for (u32 i = 0; i < buffer_count; i++) {
string += fmt::format("{:02X}, ", inputs[i]);
}
string += "\n";
}
void DepopPrepareCommand::Process(const ADSP::CommandListProcessor& processor) {
auto samples{reinterpret_cast<s32*>(previous_samples)};
auto buffer{reinterpret_cast<s32*>(depop_buffer)};
for (u32 i = 0; i < buffer_count; i++) {
if (samples[i]) {
buffer[inputs[i]] += samples[i];
samples[i] = 0;
}
}
}
bool DepopPrepareCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/mix/depop_prepare.h"
#include "audio_core/renderer/voice/voice_state.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
void DepopPrepareCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DepopPrepareCommand\n\tinputs: ");
for (u32 i = 0; i < buffer_count; i++) {
string += fmt::format("{:02X}, ", inputs[i]);
}
string += "\n";
}
void DepopPrepareCommand::Process(const ADSP::CommandListProcessor& processor) {
auto samples{reinterpret_cast<s32*>(previous_samples)};
auto buffer{reinterpret_cast<s32*>(depop_buffer)};
for (u32 i = 0; i < buffer_count; i++) {
if (samples[i]) {
buffer[inputs[i]] += samples[i];
samples[i] = 0;
}
}
}
bool DepopPrepareCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

108
src/audio_core/renderer/command/mix/depop_prepare.h
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@@ -1,54 +1,54 @@
// 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 preparing depop.
* Adds the previusly output last samples to the depop buffer.
*/
struct DepopPrepareCommand : 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;
/// Depop buffer offset for each mix buffer
std::array<s16, MaxMixBuffers> inputs;
/// Pointer to the previous mix buffer samples
CpuAddr previous_samples;
/// Number of mix buffers to use
u32 buffer_count;
/// Pointer to the current depop values
CpuAddr depop_buffer;
};
} // namespace AudioCore::AudioRenderer
// 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 preparing depop.
* Adds the previusly output last samples to the depop buffer.
*/
struct DepopPrepareCommand : 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;
/// Depop buffer offset for each mix buffer
std::array<s16, MaxMixBuffers> inputs;
/// Pointer to the previous mix buffer samples
CpuAddr previous_samples;
/// Number of mix buffers to use
u32 buffer_count;
/// Pointer to the current depop values
CpuAddr depop_buffer;
};
} // namespace AudioCore::AudioRenderer

140
src/audio_core/renderer/command/mix/mix.cpp
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@@ -1,70 +1,70 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include <limits>
#include <span>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/mix/mix.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
/**
* Mix input mix buffer into output mix buffer, with volume applied to the input.
*
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffer.
* @param input - Input mix buffer.
* @param volume - Volume applied to the input.
* @param sample_count - Number of samples to process.
*/
template <size_t Q>
static void ApplyMix(std::span<s32> output, std::span<const s32> input, const f32 volume_,
const u32 sample_count) {
const Common::FixedPoint<64 - Q, Q> volume{volume_};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (output[i] + input[i] * volume).to_int();
}
}
void MixCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("MixCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += "\n";
}
void MixCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
// If volume is 0, nothing will be added to the output, so just skip.
if (volume == 0.0f) {
return;
}
switch (precision) {
case 15:
ApplyMix<15>(output, input, volume, processor.sample_count);
break;
case 23:
ApplyMix<23>(output, input, volume, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool MixCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include <limits>
#include <span>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/mix/mix.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
/**
* Mix input mix buffer into output mix buffer, with volume applied to the input.
*
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffer.
* @param input - Input mix buffer.
* @param volume - Volume applied to the input.
* @param sample_count - Number of samples to process.
*/
template <size_t Q>
static void ApplyMix(std::span<s32> output, std::span<const s32> input, const f32 volume_,
const u32 sample_count) {
const Common::FixedPoint<64 - Q, Q> volume{volume_};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (output[i] + input[i] * volume).to_int();
}
}
void MixCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("MixCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += "\n";
}
void MixCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
// If volume is 0, nothing will be added to the output, so just skip.
if (volume == 0.0f) {
return;
}
switch (precision) {
case 15:
ApplyMix<15>(output, input, volume, processor.sample_count);
break;
case 23:
ApplyMix<23>(output, input, volume, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool MixCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

108
src/audio_core/renderer/command/mix/mix.h
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@@ -1,54 +1,54 @@
// 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 mixing an input mix buffer to an output mix buffer, with a volume
* applied to the input.
*/
struct MixCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Mix volume applied to the input
f32 volume;
};
} // namespace AudioCore::AudioRenderer
// 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 mixing an input mix buffer to an output mix buffer, with a volume
* applied to the input.
*/
struct MixCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Mix volume applied to the input
f32 volume;
};
} // namespace AudioCore::AudioRenderer

164
src/audio_core/renderer/command/mix/mix_ramp.cpp
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@@ -1,82 +1,82 @@
// 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/mix/mix_ramp.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
namespace AudioCore::AudioRenderer {
template <size_t Q>
s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, const f32 volume_,
const f32 ramp_, const u32 sample_count) {
Common::FixedPoint<64 - Q, Q> volume{volume_};
Common::FixedPoint<64 - Q, Q> sample{0};
if (ramp_ == 0.0f) {
for (u32 i = 0; i < sample_count; i++) {
sample = input[i] * volume;
output[i] = (output[i] + sample).to_int();
}
} else {
Common::FixedPoint<64 - Q, Q> ramp{ramp_};
for (u32 i = 0; i < sample_count; i++) {
sample = input[i] * volume;
output[i] = (output[i] + sample).to_int();
volume += ramp;
}
}
return sample.to_int();
}
template s32 ApplyMixRamp<15>(std::span<s32>, std::span<const s32>, f32, f32, u32);
template s32 ApplyMixRamp<23>(std::span<s32>, std::span<const s32>, f32, f32, u32);
void MixRampCommand::Dump(const ADSP::CommandListProcessor& processor, std::string& string) {
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
string += fmt::format("MixRampCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += fmt::format("\n\tprev_volume {:.8f}", prev_volume);
string += fmt::format("\n\tramp {:.8f}", ramp);
string += "\n";
}
void MixRampCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
auto prev_sample_ptr{reinterpret_cast<s32*>(previous_sample)};
// If previous volume and ramp are both 0, nothing will be added to the output, so just skip.
if (prev_volume == 0.0f && ramp == 0.0f) {
*prev_sample_ptr = 0;
return;
}
switch (precision) {
case 15:
*prev_sample_ptr =
ApplyMixRamp<15>(output, input, prev_volume, ramp, processor.sample_count);
break;
case 23:
*prev_sample_ptr =
ApplyMixRamp<23>(output, input, prev_volume, ramp, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool MixRampCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/mix/mix_ramp.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
namespace AudioCore::AudioRenderer {
template <size_t Q>
s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, const f32 volume_,
const f32 ramp_, const u32 sample_count) {
Common::FixedPoint<64 - Q, Q> volume{volume_};
Common::FixedPoint<64 - Q, Q> sample{0};
if (ramp_ == 0.0f) {
for (u32 i = 0; i < sample_count; i++) {
sample = input[i] * volume;
output[i] = (output[i] + sample).to_int();
}
} else {
Common::FixedPoint<64 - Q, Q> ramp{ramp_};
for (u32 i = 0; i < sample_count; i++) {
sample = input[i] * volume;
output[i] = (output[i] + sample).to_int();
volume += ramp;
}
}
return sample.to_int();
}
template s32 ApplyMixRamp<15>(std::span<s32>, std::span<const s32>, f32, f32, u32);
template s32 ApplyMixRamp<23>(std::span<s32>, std::span<const s32>, f32, f32, u32);
void MixRampCommand::Dump(const ADSP::CommandListProcessor& processor, std::string& string) {
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
string += fmt::format("MixRampCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += fmt::format("\n\tprev_volume {:.8f}", prev_volume);
string += fmt::format("\n\tramp {:.8f}", ramp);
string += "\n";
}
void MixRampCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
auto prev_sample_ptr{reinterpret_cast<s32*>(previous_sample)};
// If previous volume and ramp are both 0, nothing will be added to the output, so just skip.
if (prev_volume == 0.0f && ramp == 0.0f) {
*prev_sample_ptr = 0;
return;
}
switch (precision) {
case 15:
*prev_sample_ptr =
ApplyMixRamp<15>(output, input, prev_volume, ramp, processor.sample_count);
break;
case 23:
*prev_sample_ptr =
ApplyMixRamp<23>(output, input, prev_volume, ramp, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool MixRampCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

146
src/audio_core/renderer/command/mix/mix_ramp.h
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@@ -1,73 +1,73 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include <string>
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for mixing an input mix buffer to an output mix buffer, with a volume
* applied to the input, and volume ramping to smooth out the transition.
*/
struct MixRampCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Previous mix volume
f32 prev_volume;
/// Current mix volume
f32 volume;
/// Pointer to the previous sample buffer, used for depopping
CpuAddr previous_sample;
};
/**
* Mix input mix buffer into output mix buffer, with volume applied to the input.
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffer.
* @param input - Input mix buffer.
* @param volume_ - Volume applied to the input.
* @param ramp_ - Ramp applied to volume every sample.
* @param sample_count - Number of samples to process.
* @return The final gained input sample, used for depopping.
*/
template <size_t Q>
s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, f32 volume_, f32 ramp_,
u32 sample_count);
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include <string>
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for mixing an input mix buffer to an output mix buffer, with a volume
* applied to the input, and volume ramping to smooth out the transition.
*/
struct MixRampCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Previous mix volume
f32 prev_volume;
/// Current mix volume
f32 volume;
/// Pointer to the previous sample buffer, used for depopping
CpuAddr previous_sample;
};
/**
* Mix input mix buffer into output mix buffer, with volume applied to the input.
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffer.
* @param input - Input mix buffer.
* @param volume_ - Volume applied to the input.
* @param ramp_ - Ramp applied to volume every sample.
* @param sample_count - Number of samples to process.
* @return The final gained input sample, used for depopping.
*/
template <size_t Q>
s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, f32 volume_, f32 ramp_,
u32 sample_count);
} // namespace AudioCore::AudioRenderer

130
src/audio_core/renderer/command/mix/mix_ramp_grouped.cpp
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@@ -1,65 +1,65 @@
// 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/mix/mix_ramp.h"
#include "audio_core/renderer/command/mix/mix_ramp_grouped.h"
namespace AudioCore::AudioRenderer {
void MixRampGroupedCommand::Dump(const ADSP::CommandListProcessor& processor, std::string& string) {
string += "MixRampGroupedCommand";
for (u32 i = 0; i < buffer_count; i++) {
string += fmt::format("\n\t{}", i);
const auto ramp{(volumes[i] - prev_volumes[i]) / static_cast<f32>(processor.sample_count)};
string += fmt::format("\n\t\tinput {:02X}", inputs[i]);
string += fmt::format("\n\t\toutput {:02X}", outputs[i]);
string += fmt::format("\n\t\tvolume {:.8f}", volumes[i]);
string += fmt::format("\n\t\tprev_volume {:.8f}", prev_volumes[i]);
string += fmt::format("\n\t\tramp {:.8f}", ramp);
string += "\n";
}
}
void MixRampGroupedCommand::Process(const ADSP::CommandListProcessor& processor) {
std::span<s32> prev_samples = {reinterpret_cast<s32*>(previous_samples), MaxMixBuffers};
for (u32 i = 0; i < buffer_count; i++) {
auto last_sample{0};
if (prev_volumes[i] != 0.0f || volumes[i] != 0.0f) {
const auto output{processor.mix_buffers.subspan(outputs[i] * processor.sample_count,
processor.sample_count)};
const auto input{processor.mix_buffers.subspan(inputs[i] * processor.sample_count,
processor.sample_count)};
const auto ramp{(volumes[i] - prev_volumes[i]) /
static_cast<f32>(processor.sample_count)};
if (prev_volumes[i] == 0.0f && ramp == 0.0f) {
prev_samples[i] = 0;
continue;
}
switch (precision) {
case 15:
last_sample =
ApplyMixRamp<15>(output, input, prev_volumes[i], ramp, processor.sample_count);
break;
case 23:
last_sample =
ApplyMixRamp<23>(output, input, prev_volumes[i], ramp, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
prev_samples[i] = last_sample;
}
}
bool MixRampGroupedCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/mix/mix_ramp.h"
#include "audio_core/renderer/command/mix/mix_ramp_grouped.h"
namespace AudioCore::AudioRenderer {
void MixRampGroupedCommand::Dump(const ADSP::CommandListProcessor& processor, std::string& string) {
string += "MixRampGroupedCommand";
for (u32 i = 0; i < buffer_count; i++) {
string += fmt::format("\n\t{}", i);
const auto ramp{(volumes[i] - prev_volumes[i]) / static_cast<f32>(processor.sample_count)};
string += fmt::format("\n\t\tinput {:02X}", inputs[i]);
string += fmt::format("\n\t\toutput {:02X}", outputs[i]);
string += fmt::format("\n\t\tvolume {:.8f}", volumes[i]);
string += fmt::format("\n\t\tprev_volume {:.8f}", prev_volumes[i]);
string += fmt::format("\n\t\tramp {:.8f}", ramp);
string += "\n";
}
}
void MixRampGroupedCommand::Process(const ADSP::CommandListProcessor& processor) {
std::span<s32> prev_samples = {reinterpret_cast<s32*>(previous_samples), MaxMixBuffers};
for (u32 i = 0; i < buffer_count; i++) {
auto last_sample{0};
if (prev_volumes[i] != 0.0f || volumes[i] != 0.0f) {
const auto output{processor.mix_buffers.subspan(outputs[i] * processor.sample_count,
processor.sample_count)};
const auto input{processor.mix_buffers.subspan(inputs[i] * processor.sample_count,
processor.sample_count)};
const auto ramp{(volumes[i] - prev_volumes[i]) /
static_cast<f32>(processor.sample_count)};
if (prev_volumes[i] == 0.0f && ramp == 0.0f) {
prev_samples[i] = 0;
continue;
}
switch (precision) {
case 15:
last_sample =
ApplyMixRamp<15>(output, input, prev_volumes[i], ramp, processor.sample_count);
break;
case 23:
last_sample =
ApplyMixRamp<23>(output, input, prev_volumes[i], ramp, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
prev_samples[i] = last_sample;
}
}
bool MixRampGroupedCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

122
src/audio_core/renderer/command/mix/mix_ramp_grouped.h
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@@ -1,61 +1,61 @@
// 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 "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for mixing multiple input mix buffers to multiple output mix buffers, with
* a volume applied to the input, and volume ramping to smooth out the transition.
*/
struct MixRampGroupedCommand : 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;
/// Fixed point precision
u8 precision;
/// Number of mix buffers to mix
u32 buffer_count;
/// Input mix buffer indexes for each mix buffer
std::array<s16, MaxMixBuffers> inputs;
/// Output mix buffer indexes for each mix buffer
std::array<s16, MaxMixBuffers> outputs;
/// Previous mix volumes for each mix buffer
std::array<f32, MaxMixBuffers> prev_volumes;
/// Current mix volumes for each mix buffer
std::array<f32, MaxMixBuffers> volumes;
/// Pointer to the previous sample buffer, used for depop
CpuAddr previous_samples;
};
} // namespace AudioCore::AudioRenderer
// 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 "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for mixing multiple input mix buffers to multiple output mix buffers, with
* a volume applied to the input, and volume ramping to smooth out the transition.
*/
struct MixRampGroupedCommand : 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;
/// Fixed point precision
u8 precision;
/// Number of mix buffers to mix
u32 buffer_count;
/// Input mix buffer indexes for each mix buffer
std::array<s16, MaxMixBuffers> inputs;
/// Output mix buffer indexes for each mix buffer
std::array<s16, MaxMixBuffers> outputs;
/// Previous mix volumes for each mix buffer
std::array<f32, MaxMixBuffers> prev_volumes;
/// Current mix volumes for each mix buffer
std::array<f32, MaxMixBuffers> volumes;
/// Pointer to the previous sample buffer, used for depop
CpuAddr previous_samples;
};
} // namespace AudioCore::AudioRenderer

144
src/audio_core/renderer/command/mix/volume.cpp
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@@ -1,72 +1,72 @@
// 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/mix/volume.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
namespace AudioCore::AudioRenderer {
/**
* Apply volume to the input mix buffer, saving to the output buffer.
*
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffer.
* @param input - Input mix buffer.
* @param volume - Volume applied to the input.
* @param sample_count - Number of samples to process.
*/
template <size_t Q>
static void ApplyUniformGain(std::span<s32> output, std::span<const s32> input, const f32 volume,
const u32 sample_count) {
if (volume == 1.0f) {
std::memcpy(output.data(), input.data(), input.size_bytes());
} else {
const Common::FixedPoint<64 - Q, Q> gain{volume};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (input[i] * gain).to_int();
}
}
}
void VolumeCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("VolumeCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += "\n";
}
void VolumeCommand::Process(const ADSP::CommandListProcessor& processor) {
// If input and output buffers are the same, and the volume is 1.0f, this won't do
// anything, so just skip.
if (input_index == output_index && volume == 1.0f) {
return;
}
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
switch (precision) {
case 15:
ApplyUniformGain<15>(output, input, volume, processor.sample_count);
break;
case 23:
ApplyUniformGain<23>(output, input, volume, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool VolumeCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/mix/volume.h"
#include "common/fixed_point.h"
#include "common/logging/log.h"
namespace AudioCore::AudioRenderer {
/**
* Apply volume to the input mix buffer, saving to the output buffer.
*
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffer.
* @param input - Input mix buffer.
* @param volume - Volume applied to the input.
* @param sample_count - Number of samples to process.
*/
template <size_t Q>
static void ApplyUniformGain(std::span<s32> output, std::span<const s32> input, const f32 volume,
const u32 sample_count) {
if (volume == 1.0f) {
std::memcpy(output.data(), input.data(), input.size_bytes());
} else {
const Common::FixedPoint<64 - Q, Q> gain{volume};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (input[i] * gain).to_int();
}
}
}
void VolumeCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("VolumeCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += "\n";
}
void VolumeCommand::Process(const ADSP::CommandListProcessor& processor) {
// If input and output buffers are the same, and the volume is 1.0f, this won't do
// anything, so just skip.
if (input_index == output_index && volume == 1.0f) {
return;
}
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
switch (precision) {
case 15:
ApplyUniformGain<15>(output, input, volume, processor.sample_count);
break;
case 23:
ApplyUniformGain<23>(output, input, volume, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool VolumeCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

106
src/audio_core/renderer/command/mix/volume.h
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@@ -1,53 +1,53 @@
// 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 applying volume to a mix buffer.
*/
struct VolumeCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Mix volume applied to the input
f32 volume;
};
} // namespace AudioCore::AudioRenderer
// 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 applying volume to a mix buffer.
*/
struct VolumeCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Mix volume applied to the input
f32 volume;
};
} // namespace AudioCore::AudioRenderer

168
src/audio_core/renderer/command/mix/volume_ramp.cpp
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@@ -1,84 +1,84 @@
// 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/mix/volume_ramp.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
/**
* Apply volume with ramping to the input mix buffer, saving to the output buffer.
*
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffers.
* @param input - Input mix buffers.
* @param volume - Volume applied to the input.
* @param ramp - Ramp applied to volume every sample.
* @param sample_count - Number of samples to process.
*/
template <size_t Q>
static void ApplyLinearEnvelopeGain(std::span<s32> output, std::span<const s32> input,
const f32 volume, const f32 ramp_, const u32 sample_count) {
if (volume == 0.0f && ramp_ == 0.0f) {
std::memset(output.data(), 0, output.size_bytes());
} else if (volume == 1.0f && ramp_ == 0.0f) {
std::memcpy(output.data(), input.data(), output.size_bytes());
} else if (ramp_ == 0.0f) {
const Common::FixedPoint<64 - Q, Q> gain{volume};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (input[i] * gain).to_int();
}
} else {
Common::FixedPoint<64 - Q, Q> gain{volume};
const Common::FixedPoint<64 - Q, Q> ramp{ramp_};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (input[i] * gain).to_int();
gain += ramp;
}
}
}
void VolumeRampCommand::Dump(const ADSP::CommandListProcessor& processor, std::string& string) {
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
string += fmt::format("VolumeRampCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += fmt::format("\n\tprev_volume {:.8f}", prev_volume);
string += fmt::format("\n\tramp {:.8f}", ramp);
string += "\n";
}
void VolumeRampCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
// If input and output buffers are the same, and the volume is 1.0f, and there's no ramping,
// this won't do anything, so just skip.
if (input_index == output_index && prev_volume == 1.0f && ramp == 0.0f) {
return;
}
switch (precision) {
case 15:
ApplyLinearEnvelopeGain<15>(output, input, prev_volume, ramp, processor.sample_count);
break;
case 23:
ApplyLinearEnvelopeGain<23>(output, input, prev_volume, ramp, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool VolumeRampCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/mix/volume_ramp.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
/**
* Apply volume with ramping to the input mix buffer, saving to the output buffer.
*
* @tparam Q - Number of bits for fixed point operations.
* @param output - Output mix buffers.
* @param input - Input mix buffers.
* @param volume - Volume applied to the input.
* @param ramp - Ramp applied to volume every sample.
* @param sample_count - Number of samples to process.
*/
template <size_t Q>
static void ApplyLinearEnvelopeGain(std::span<s32> output, std::span<const s32> input,
const f32 volume, const f32 ramp_, const u32 sample_count) {
if (volume == 0.0f && ramp_ == 0.0f) {
std::memset(output.data(), 0, output.size_bytes());
} else if (volume == 1.0f && ramp_ == 0.0f) {
std::memcpy(output.data(), input.data(), output.size_bytes());
} else if (ramp_ == 0.0f) {
const Common::FixedPoint<64 - Q, Q> gain{volume};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (input[i] * gain).to_int();
}
} else {
Common::FixedPoint<64 - Q, Q> gain{volume};
const Common::FixedPoint<64 - Q, Q> ramp{ramp_};
for (u32 i = 0; i < sample_count; i++) {
output[i] = (input[i] * gain).to_int();
gain += ramp;
}
}
}
void VolumeRampCommand::Dump(const ADSP::CommandListProcessor& processor, std::string& string) {
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
string += fmt::format("VolumeRampCommand");
string += fmt::format("\n\tinput {:02X}", input_index);
string += fmt::format("\n\toutput {:02X}", output_index);
string += fmt::format("\n\tvolume {:.8f}", volume);
string += fmt::format("\n\tprev_volume {:.8f}", prev_volume);
string += fmt::format("\n\tramp {:.8f}", ramp);
string += "\n";
}
void VolumeRampCommand::Process(const ADSP::CommandListProcessor& processor) {
auto output{processor.mix_buffers.subspan(output_index * processor.sample_count,
processor.sample_count)};
auto input{processor.mix_buffers.subspan(input_index * processor.sample_count,
processor.sample_count)};
const auto ramp{(volume - prev_volume) / static_cast<f32>(processor.sample_count)};
// If input and output buffers are the same, and the volume is 1.0f, and there's no ramping,
// this won't do anything, so just skip.
if (input_index == output_index && prev_volume == 1.0f && ramp == 0.0f) {
return;
}
switch (precision) {
case 15:
ApplyLinearEnvelopeGain<15>(output, input, prev_volume, ramp, processor.sample_count);
break;
case 23:
ApplyLinearEnvelopeGain<23>(output, input, prev_volume, ramp, processor.sample_count);
break;
default:
LOG_ERROR(Service_Audio, "Invalid precision {}", precision);
break;
}
}
bool VolumeRampCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

112
src/audio_core/renderer/command/mix/volume_ramp.h
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@@ -1,56 +1,56 @@
// 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 applying volume to a mix buffer, with ramping for the volume to smooth
* out the transition.
*/
struct VolumeRampCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Previous mix volume applied to the input
f32 prev_volume;
/// Current mix volume applied to the input
f32 volume;
};
} // namespace AudioCore::AudioRenderer
// 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 applying volume to a mix buffer, with ramping for the volume to smooth
* out the transition.
*/
struct VolumeRampCommand : 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;
/// Fixed point precision
u8 precision;
/// Input mix buffer index
s16 input_index;
/// Output mix buffer index
s16 output_index;
/// Previous mix volume applied to the input
f32 prev_volume;
/// Current mix volume applied to the input
f32 volume;
};
} // namespace AudioCore::AudioRenderer

86
src/audio_core/renderer/command/performance/performance.cpp
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@@ -1,43 +1,43 @@
// 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/performance/performance.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/core_timing_util.h"
namespace AudioCore::AudioRenderer {
void PerformanceCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("PerformanceCommand\n\tstate {}\n", static_cast<u32>(state));
}
void PerformanceCommand::Process(const ADSP::CommandListProcessor& processor) {
auto base{entry_address.translated_address};
if (state == PerformanceState::Start) {
auto start_time_ptr{reinterpret_cast<u32*>(base + entry_address.entry_start_time_offset)};
*start_time_ptr = static_cast<u32>(
Core::Timing::CyclesToUs(processor.system->CoreTiming().GetClockTicks() -
processor.start_time - processor.current_processing_time)
.count());
} else if (state == PerformanceState::Stop) {
auto processed_time_ptr{
reinterpret_cast<u32*>(base + entry_address.entry_processed_time_offset)};
auto entry_count_ptr{
reinterpret_cast<u32*>(base + entry_address.header_entry_count_offset)};
*processed_time_ptr = static_cast<u32>(
Core::Timing::CyclesToUs(processor.system->CoreTiming().GetClockTicks() -
processor.start_time - processor.current_processing_time)
.count());
(*entry_count_ptr)++;
}
}
bool PerformanceCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/performance/performance.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/core_timing_util.h"
namespace AudioCore::AudioRenderer {
void PerformanceCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("PerformanceCommand\n\tstate {}\n", static_cast<u32>(state));
}
void PerformanceCommand::Process(const ADSP::CommandListProcessor& processor) {
auto base{entry_address.translated_address};
if (state == PerformanceState::Start) {
auto start_time_ptr{reinterpret_cast<u32*>(base + entry_address.entry_start_time_offset)};
*start_time_ptr = static_cast<u32>(
Core::Timing::CyclesToUs(processor.system->CoreTiming().GetClockTicks() -
processor.start_time - processor.current_processing_time)
.count());
} else if (state == PerformanceState::Stop) {
auto processed_time_ptr{
reinterpret_cast<u32*>(base + entry_address.entry_processed_time_offset)};
auto entry_count_ptr{
reinterpret_cast<u32*>(base + entry_address.header_entry_count_offset)};
*processed_time_ptr = static_cast<u32>(
Core::Timing::CyclesToUs(processor.system->CoreTiming().GetClockTicks() -
processor.start_time - processor.current_processing_time)
.count());
(*entry_count_ptr)++;
}
}
bool PerformanceCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

102
src/audio_core/renderer/command/performance/performance.h
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@@ -1,51 +1,51 @@
// 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/performance/performance_entry_addresses.h"
#include "audio_core/renderer/performance/performance_manager.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for writing AudioRenderer performance metrics back to the sysmodule.
*/
struct PerformanceCommand : 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;
/// State of the performance
PerformanceState state;
/// Pointers to be written
PerformanceEntryAddresses entry_address;
};
} // namespace AudioCore::AudioRenderer
// 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/performance/performance_entry_addresses.h"
#include "audio_core/renderer/performance/performance_manager.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for writing AudioRenderer performance metrics back to the sysmodule.
*/
struct PerformanceCommand : 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;
/// State of the performance
PerformanceState state;
/// Pointers to be written
PerformanceEntryAddresses entry_address;
};
} // namespace AudioCore::AudioRenderer

148
src/audio_core/renderer/command/resample/downmix_6ch_to_2ch.cpp
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@@ -1,74 +1,74 @@
// 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/resample/downmix_6ch_to_2ch.h"
namespace AudioCore::AudioRenderer {
void DownMix6chTo2chCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DownMix6chTo2chCommand\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 DownMix6chTo2chCommand::Process(const ADSP::CommandListProcessor& processor) {
auto in_front_left{
processor.mix_buffers.subspan(inputs[0] * processor.sample_count, processor.sample_count)};
auto in_front_right{
processor.mix_buffers.subspan(inputs[1] * processor.sample_count, processor.sample_count)};
auto in_center{
processor.mix_buffers.subspan(inputs[2] * processor.sample_count, processor.sample_count)};
auto in_lfe{
processor.mix_buffers.subspan(inputs[3] * processor.sample_count, processor.sample_count)};
auto in_back_left{
processor.mix_buffers.subspan(inputs[4] * processor.sample_count, processor.sample_count)};
auto in_back_right{
processor.mix_buffers.subspan(inputs[5] * processor.sample_count, processor.sample_count)};
auto out_front_left{
processor.mix_buffers.subspan(outputs[0] * processor.sample_count, processor.sample_count)};
auto out_front_right{
processor.mix_buffers.subspan(outputs[1] * processor.sample_count, processor.sample_count)};
auto out_center{
processor.mix_buffers.subspan(outputs[2] * processor.sample_count, processor.sample_count)};
auto out_lfe{
processor.mix_buffers.subspan(outputs[3] * processor.sample_count, processor.sample_count)};
auto out_back_left{
processor.mix_buffers.subspan(outputs[4] * processor.sample_count, processor.sample_count)};
auto out_back_right{
processor.mix_buffers.subspan(outputs[5] * processor.sample_count, processor.sample_count)};
for (u32 i = 0; i < processor.sample_count; i++) {
const auto left_sample{(in_front_left[i] * down_mix_coeff[0] +
in_center[i] * down_mix_coeff[1] + in_lfe[i] * down_mix_coeff[2] +
in_back_left[i] * down_mix_coeff[3])
.to_int()};
const auto right_sample{(in_front_right[i] * down_mix_coeff[0] +
in_center[i] * down_mix_coeff[1] + in_lfe[i] * down_mix_coeff[2] +
in_back_right[i] * down_mix_coeff[3])
.to_int()};
out_front_left[i] = left_sample;
out_front_right[i] = right_sample;
}
std::memset(out_center.data(), 0, out_center.size_bytes());
std::memset(out_lfe.data(), 0, out_lfe.size_bytes());
std::memset(out_back_left.data(), 0, out_back_left.size_bytes());
std::memset(out_back_right.data(), 0, out_back_right.size_bytes());
}
bool DownMix6chTo2chCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// 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/resample/downmix_6ch_to_2ch.h"
namespace AudioCore::AudioRenderer {
void DownMix6chTo2chCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DownMix6chTo2chCommand\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 DownMix6chTo2chCommand::Process(const ADSP::CommandListProcessor& processor) {
auto in_front_left{
processor.mix_buffers.subspan(inputs[0] * processor.sample_count, processor.sample_count)};
auto in_front_right{
processor.mix_buffers.subspan(inputs[1] * processor.sample_count, processor.sample_count)};
auto in_center{
processor.mix_buffers.subspan(inputs[2] * processor.sample_count, processor.sample_count)};
auto in_lfe{
processor.mix_buffers.subspan(inputs[3] * processor.sample_count, processor.sample_count)};
auto in_back_left{
processor.mix_buffers.subspan(inputs[4] * processor.sample_count, processor.sample_count)};
auto in_back_right{
processor.mix_buffers.subspan(inputs[5] * processor.sample_count, processor.sample_count)};
auto out_front_left{
processor.mix_buffers.subspan(outputs[0] * processor.sample_count, processor.sample_count)};
auto out_front_right{
processor.mix_buffers.subspan(outputs[1] * processor.sample_count, processor.sample_count)};
auto out_center{
processor.mix_buffers.subspan(outputs[2] * processor.sample_count, processor.sample_count)};
auto out_lfe{
processor.mix_buffers.subspan(outputs[3] * processor.sample_count, processor.sample_count)};
auto out_back_left{
processor.mix_buffers.subspan(outputs[4] * processor.sample_count, processor.sample_count)};
auto out_back_right{
processor.mix_buffers.subspan(outputs[5] * processor.sample_count, processor.sample_count)};
for (u32 i = 0; i < processor.sample_count; i++) {
const auto left_sample{(in_front_left[i] * down_mix_coeff[0] +
in_center[i] * down_mix_coeff[1] + in_lfe[i] * down_mix_coeff[2] +
in_back_left[i] * down_mix_coeff[3])
.to_int()};
const auto right_sample{(in_front_right[i] * down_mix_coeff[0] +
in_center[i] * down_mix_coeff[1] + in_lfe[i] * down_mix_coeff[2] +
in_back_right[i] * down_mix_coeff[3])
.to_int()};
out_front_left[i] = left_sample;
out_front_right[i] = right_sample;
}
std::memset(out_center.data(), 0, out_center.size_bytes());
std::memset(out_lfe.data(), 0, out_lfe.size_bytes());
std::memset(out_back_left.data(), 0, out_back_left.size_bytes());
std::memset(out_back_right.data(), 0, out_back_right.size_bytes());
}
bool DownMix6chTo2chCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

118
src/audio_core/renderer/command/resample/downmix_6ch_to_2ch.h
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@@ -1,59 +1,59 @@
// 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"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for downmixing 6 channels to 2.
* Channel layout (SMPTE):
* 0 - front left
* 1 - front right
* 2 - center
* 3 - lfe
* 4 - back left
* 5 - back right
*/
struct DownMix6chTo2chCommand : 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;
/// Coefficients used for downmixing
std::array<Common::FixedPoint<48, 16>, 4> down_mix_coeff;
};
} // namespace AudioCore::AudioRenderer
// 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"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for downmixing 6 channels to 2.
* Channel layout (SMPTE):
* 0 - front left
* 1 - front right
* 2 - center
* 3 - lfe
* 4 - back left
* 5 - back right
*/
struct DownMix6chTo2chCommand : 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;
/// Coefficients used for downmixing
std::array<Common::FixedPoint<48, 16>, 4> down_mix_coeff;
};
} // namespace AudioCore::AudioRenderer

1766
src/audio_core/renderer/command/resample/resample.cpp
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58
src/audio_core/renderer/command/resample/resample.h
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@@ -1,29 +1,29 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/common/common.h"
#include "common/common_types.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
/**
* Resample an input buffer into an output buffer, according to the sample_rate_ratio.
*
* @param output - Output buffer.
* @param input - Input buffer.
* @param sample_rate_ratio - Ratio for resampling.
e.g 32000/48000 = 0.666 input samples read per output.
* @param fraction - Current read fraction, written to and should be passed back in for
* multiple calls.
* @param samples_to_write - Number of samples to write.
* @param src_quality - Resampling quality.
*/
void Resample(std::span<s32> output, std::span<const s16> input,
const Common::FixedPoint<49, 15>& sample_rate_ratio,
Common::FixedPoint<49, 15>& fraction, u32 samples_to_write, SrcQuality src_quality);
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "audio_core/common/common.h"
#include "common/common_types.h"
#include "common/fixed_point.h"
namespace AudioCore::AudioRenderer {
/**
* Resample an input buffer into an output buffer, according to the sample_rate_ratio.
*
* @param output - Output buffer.
* @param input - Input buffer.
* @param sample_rate_ratio - Ratio for resampling.
e.g 32000/48000 = 0.666 input samples read per output.
* @param fraction - Current read fraction, written to and should be passed back in for
* multiple calls.
* @param samples_to_write - Number of samples to write.
* @param src_quality - Resampling quality.
*/
void Resample(std::span<s32> output, std::span<const s16> input,
const Common::FixedPoint<49, 15>& sample_rate_ratio,
Common::FixedPoint<49, 15>& fraction, u32 samples_to_write, SrcQuality src_quality);
} // namespace AudioCore::AudioRenderer

524
src/audio_core/renderer/command/resample/upsample.cpp
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@@ -1,262 +1,262 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <array>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/resample/upsample.h"
#include "audio_core/renderer/upsampler/upsampler_info.h"
namespace AudioCore::AudioRenderer {
/**
* Upsampling impl. Input must be 8K, 16K or 32K, output is 48K.
*
* @param output - Output buffer.
* @param input - Input buffer.
* @param target_sample_count - Number of samples for output.
* @param state - Upsampler state, updated each call.
*/
static void SrcProcessFrame(std::span<s32> output, std::span<const s32> input,
const u32 target_sample_count, const u32 source_sample_count,
UpsamplerState* state) {
constexpr u32 WindowSize = 10;
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow1{
51.93359375f, -18.80078125f, 9.73046875f, -5.33203125f, 2.84375f,
-1.41015625f, 0.62109375f, -0.2265625f, 0.0625f, -0.00390625f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow2{
105.35546875f, -24.52734375f, 11.9609375f, -6.515625f, 3.52734375f,
-1.796875f, 0.828125f, -0.32421875f, 0.1015625f, -0.015625f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow3{
122.08203125f, -16.47656250f, 7.68359375f, -4.15625000f, 2.26171875f,
-1.16796875f, 0.54687500f, -0.22265625f, 0.07421875f, -0.01171875f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow4{
23.73437500f, -9.62109375f, 5.07812500f, -2.78125000f, 1.46875000f,
-0.71484375f, 0.30859375f, -0.10546875f, 0.02734375f, 0.00000000f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow5{
80.62500000f, -24.67187500f, 12.44921875f, -6.80859375f, 3.66406250f,
-1.83984375f, 0.83203125f, -0.31640625f, 0.09375000f, -0.01171875f,
};
if (!state->initialized) {
switch (source_sample_count) {
case 40:
state->window_size = WindowSize;
state->ratio = 6.0f;
state->history.fill(0);
break;
case 80:
state->window_size = WindowSize;
state->ratio = 3.0f;
state->history.fill(0);
break;
case 160:
state->window_size = WindowSize;
state->ratio = 1.5f;
state->history.fill(0);
break;
default:
LOG_ERROR(Service_Audio, "Invalid upsampling source count {}!", source_sample_count);
// This continues anyway, but let's assume 160 for sanity
state->window_size = WindowSize;
state->ratio = 1.5f;
state->history.fill(0);
break;
}
state->history_input_index = 0;
state->history_output_index = 9;
state->history_start_index = 0;
state->history_end_index = UpsamplerState::HistorySize - 1;
state->initialized = true;
}
if (target_sample_count == 0) {
return;
}
u32 read_index{0};
auto increment = [&]() -> void {
state->history[state->history_input_index] = input[read_index++];
state->history_input_index =
static_cast<u16>((state->history_input_index + 1) % UpsamplerState::HistorySize);
state->history_output_index =
static_cast<u16>((state->history_output_index + 1) % UpsamplerState::HistorySize);
};
auto calculate_sample = [&state](std::span<const Common::FixedPoint<24, 8>> coeffs1,
std::span<const Common::FixedPoint<24, 8>> coeffs2) -> s32 {
auto output_index{state->history_output_index};
auto start_pos{output_index - state->history_start_index + 1U};
auto end_pos{10U};
if (start_pos < 10) {
end_pos = start_pos;
}
u64 prev_contrib{0};
u32 coeff_index{0};
for (; coeff_index < end_pos; coeff_index++, output_index--) {
prev_contrib += static_cast<u64>(state->history[output_index].to_raw()) *
coeffs1[coeff_index].to_raw();
}
auto end_index{state->history_end_index};
for (; start_pos < 9; start_pos++, coeff_index++, end_index--) {
prev_contrib += static_cast<u64>(state->history[end_index].to_raw()) *
coeffs1[coeff_index].to_raw();
}
output_index =
static_cast<u16>((state->history_output_index + 1) % UpsamplerState::HistorySize);
start_pos = state->history_end_index - output_index + 1U;
end_pos = 10U;
if (start_pos < 10) {
end_pos = start_pos;
}
u64 next_contrib{0};
coeff_index = 0;
for (; coeff_index < end_pos; coeff_index++, output_index++) {
next_contrib += static_cast<u64>(state->history[output_index].to_raw()) *
coeffs2[coeff_index].to_raw();
}
auto start_index{state->history_start_index};
for (; start_pos < 9; start_pos++, start_index++, coeff_index++) {
next_contrib += static_cast<u64>(state->history[start_index].to_raw()) *
coeffs2[coeff_index].to_raw();
}
return static_cast<s32>(((prev_contrib >> 15) + (next_contrib >> 15)) >> 8);
};
switch (state->ratio.to_int_floor()) {
// 40 -> 240
case 6:
for (u32 write_index = 0; write_index < target_sample_count; write_index++) {
switch (state->sample_index) {
case 0:
increment();
output[write_index] = state->history[state->history_output_index].to_int_floor();
break;
case 1:
output[write_index] = calculate_sample(SincWindow3, SincWindow4);
break;
case 2:
output[write_index] = calculate_sample(SincWindow2, SincWindow1);
break;
case 3:
output[write_index] = calculate_sample(SincWindow5, SincWindow5);
break;
case 4:
output[write_index] = calculate_sample(SincWindow1, SincWindow2);
break;
case 5:
output[write_index] = calculate_sample(SincWindow4, SincWindow3);
break;
}
state->sample_index = static_cast<u8>((state->sample_index + 1) % 6);
}
break;
// 80 -> 240
case 3:
for (u32 write_index = 0; write_index < target_sample_count; write_index++) {
switch (state->sample_index) {
case 0:
increment();
output[write_index] = state->history[state->history_output_index].to_int_floor();
break;
case 1:
output[write_index] = calculate_sample(SincWindow2, SincWindow1);
break;
case 2:
output[write_index] = calculate_sample(SincWindow1, SincWindow2);
break;
}
state->sample_index = static_cast<u8>((state->sample_index + 1) % 3);
}
break;
// 160 -> 240
default:
for (u32 write_index = 0; write_index < target_sample_count; write_index++) {
switch (state->sample_index) {
case 0:
increment();
output[write_index] = state->history[state->history_output_index].to_int_floor();
break;
case 1:
output[write_index] = calculate_sample(SincWindow1, SincWindow2);
break;
case 2:
increment();
output[write_index] = calculate_sample(SincWindow2, SincWindow1);
break;
}
state->sample_index = static_cast<u8>((state->sample_index + 1) % 3);
}
break;
}
}
auto UpsampleCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) -> void {
string += fmt::format("UpsampleCommand\n\tsource_sample_count {} source_sample_rate {}",
source_sample_count, source_sample_rate);
const auto upsampler{reinterpret_cast<UpsamplerInfo*>(upsampler_info)};
if (upsampler != nullptr) {
string += fmt::format("\n\tUpsampler\n\t\tenabled {} sample count {}\n\tinputs: ",
upsampler->enabled, upsampler->sample_count);
for (u32 i = 0; i < upsampler->input_count; i++) {
string += fmt::format("{:02X}, ", upsampler->inputs[i]);
}
}
string += "\n";
}
void UpsampleCommand::Process(const ADSP::CommandListProcessor& processor) {
const auto info{reinterpret_cast<UpsamplerInfo*>(upsampler_info)};
const auto input_count{std::min(info->input_count, buffer_count)};
const std::span<const s16> inputs_{reinterpret_cast<const s16*>(inputs), input_count};
for (u32 i = 0; i < input_count; i++) {
const auto channel{inputs_[i]};
if (channel >= 0 && channel < static_cast<s16>(processor.buffer_count)) {
auto state{&info->states[i]};
std::span<s32> output{
reinterpret_cast<s32*>(samples_buffer + info->sample_count * channel * sizeof(s32)),
info->sample_count};
auto input{processor.mix_buffers.subspan(channel * processor.sample_count,
processor.sample_count)};
SrcProcessFrame(output, input, info->sample_count, source_sample_count, state);
}
}
}
bool UpsampleCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <array>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/resample/upsample.h"
#include "audio_core/renderer/upsampler/upsampler_info.h"
namespace AudioCore::AudioRenderer {
/**
* Upsampling impl. Input must be 8K, 16K or 32K, output is 48K.
*
* @param output - Output buffer.
* @param input - Input buffer.
* @param target_sample_count - Number of samples for output.
* @param state - Upsampler state, updated each call.
*/
static void SrcProcessFrame(std::span<s32> output, std::span<const s32> input,
const u32 target_sample_count, const u32 source_sample_count,
UpsamplerState* state) {
constexpr u32 WindowSize = 10;
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow1{
51.93359375f, -18.80078125f, 9.73046875f, -5.33203125f, 2.84375f,
-1.41015625f, 0.62109375f, -0.2265625f, 0.0625f, -0.00390625f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow2{
105.35546875f, -24.52734375f, 11.9609375f, -6.515625f, 3.52734375f,
-1.796875f, 0.828125f, -0.32421875f, 0.1015625f, -0.015625f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow3{
122.08203125f, -16.47656250f, 7.68359375f, -4.15625000f, 2.26171875f,
-1.16796875f, 0.54687500f, -0.22265625f, 0.07421875f, -0.01171875f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow4{
23.73437500f, -9.62109375f, 5.07812500f, -2.78125000f, 1.46875000f,
-0.71484375f, 0.30859375f, -0.10546875f, 0.02734375f, 0.00000000f,
};
constexpr std::array<Common::FixedPoint<24, 8>, WindowSize> SincWindow5{
80.62500000f, -24.67187500f, 12.44921875f, -6.80859375f, 3.66406250f,
-1.83984375f, 0.83203125f, -0.31640625f, 0.09375000f, -0.01171875f,
};
if (!state->initialized) {
switch (source_sample_count) {
case 40:
state->window_size = WindowSize;
state->ratio = 6.0f;
state->history.fill(0);
break;
case 80:
state->window_size = WindowSize;
state->ratio = 3.0f;
state->history.fill(0);
break;
case 160:
state->window_size = WindowSize;
state->ratio = 1.5f;
state->history.fill(0);
break;
default:
LOG_ERROR(Service_Audio, "Invalid upsampling source count {}!", source_sample_count);
// This continues anyway, but let's assume 160 for sanity
state->window_size = WindowSize;
state->ratio = 1.5f;
state->history.fill(0);
break;
}
state->history_input_index = 0;
state->history_output_index = 9;
state->history_start_index = 0;
state->history_end_index = UpsamplerState::HistorySize - 1;
state->initialized = true;
}
if (target_sample_count == 0) {
return;
}
u32 read_index{0};
auto increment = [&]() -> void {
state->history[state->history_input_index] = input[read_index++];
state->history_input_index =
static_cast<u16>((state->history_input_index + 1) % UpsamplerState::HistorySize);
state->history_output_index =
static_cast<u16>((state->history_output_index + 1) % UpsamplerState::HistorySize);
};
auto calculate_sample = [&state](std::span<const Common::FixedPoint<24, 8>> coeffs1,
std::span<const Common::FixedPoint<24, 8>> coeffs2) -> s32 {
auto output_index{state->history_output_index};
auto start_pos{output_index - state->history_start_index + 1U};
auto end_pos{10U};
if (start_pos < 10) {
end_pos = start_pos;
}
u64 prev_contrib{0};
u32 coeff_index{0};
for (; coeff_index < end_pos; coeff_index++, output_index--) {
prev_contrib += static_cast<u64>(state->history[output_index].to_raw()) *
coeffs1[coeff_index].to_raw();
}
auto end_index{state->history_end_index};
for (; start_pos < 9; start_pos++, coeff_index++, end_index--) {
prev_contrib += static_cast<u64>(state->history[end_index].to_raw()) *
coeffs1[coeff_index].to_raw();
}
output_index =
static_cast<u16>((state->history_output_index + 1) % UpsamplerState::HistorySize);
start_pos = state->history_end_index - output_index + 1U;
end_pos = 10U;
if (start_pos < 10) {
end_pos = start_pos;
}
u64 next_contrib{0};
coeff_index = 0;
for (; coeff_index < end_pos; coeff_index++, output_index++) {
next_contrib += static_cast<u64>(state->history[output_index].to_raw()) *
coeffs2[coeff_index].to_raw();
}
auto start_index{state->history_start_index};
for (; start_pos < 9; start_pos++, start_index++, coeff_index++) {
next_contrib += static_cast<u64>(state->history[start_index].to_raw()) *
coeffs2[coeff_index].to_raw();
}
return static_cast<s32>(((prev_contrib >> 15) + (next_contrib >> 15)) >> 8);
};
switch (state->ratio.to_int_floor()) {
// 40 -> 240
case 6:
for (u32 write_index = 0; write_index < target_sample_count; write_index++) {
switch (state->sample_index) {
case 0:
increment();
output[write_index] = state->history[state->history_output_index].to_int_floor();
break;
case 1:
output[write_index] = calculate_sample(SincWindow3, SincWindow4);
break;
case 2:
output[write_index] = calculate_sample(SincWindow2, SincWindow1);
break;
case 3:
output[write_index] = calculate_sample(SincWindow5, SincWindow5);
break;
case 4:
output[write_index] = calculate_sample(SincWindow1, SincWindow2);
break;
case 5:
output[write_index] = calculate_sample(SincWindow4, SincWindow3);
break;
}
state->sample_index = static_cast<u8>((state->sample_index + 1) % 6);
}
break;
// 80 -> 240
case 3:
for (u32 write_index = 0; write_index < target_sample_count; write_index++) {
switch (state->sample_index) {
case 0:
increment();
output[write_index] = state->history[state->history_output_index].to_int_floor();
break;
case 1:
output[write_index] = calculate_sample(SincWindow2, SincWindow1);
break;
case 2:
output[write_index] = calculate_sample(SincWindow1, SincWindow2);
break;
}
state->sample_index = static_cast<u8>((state->sample_index + 1) % 3);
}
break;
// 160 -> 240
default:
for (u32 write_index = 0; write_index < target_sample_count; write_index++) {
switch (state->sample_index) {
case 0:
increment();
output[write_index] = state->history[state->history_output_index].to_int_floor();
break;
case 1:
output[write_index] = calculate_sample(SincWindow1, SincWindow2);
break;
case 2:
increment();
output[write_index] = calculate_sample(SincWindow2, SincWindow1);
break;
}
state->sample_index = static_cast<u8>((state->sample_index + 1) % 3);
}
break;
}
}
auto UpsampleCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) -> void {
string += fmt::format("UpsampleCommand\n\tsource_sample_count {} source_sample_rate {}",
source_sample_count, source_sample_rate);
const auto upsampler{reinterpret_cast<UpsamplerInfo*>(upsampler_info)};
if (upsampler != nullptr) {
string += fmt::format("\n\tUpsampler\n\t\tenabled {} sample count {}\n\tinputs: ",
upsampler->enabled, upsampler->sample_count);
for (u32 i = 0; i < upsampler->input_count; i++) {
string += fmt::format("{:02X}, ", upsampler->inputs[i]);
}
}
string += "\n";
}
void UpsampleCommand::Process(const ADSP::CommandListProcessor& processor) {
const auto info{reinterpret_cast<UpsamplerInfo*>(upsampler_info)};
const auto input_count{std::min(info->input_count, buffer_count)};
const std::span<const s16> inputs_{reinterpret_cast<const s16*>(inputs), input_count};
for (u32 i = 0; i < input_count; i++) {
const auto channel{inputs_[i]};
if (channel >= 0 && channel < static_cast<s16>(processor.buffer_count)) {
auto state{&info->states[i]};
std::span<s32> output{
reinterpret_cast<s32*>(samples_buffer + info->sample_count * channel * sizeof(s32)),
info->sample_count};
auto input{processor.mix_buffers.subspan(channel * processor.sample_count,
processor.sample_count)};
SrcProcessFrame(output, input, info->sample_count, source_sample_count, state);
}
}
}
bool UpsampleCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

120
src/audio_core/renderer/command/resample/upsample.h
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@@ -1,60 +1,60 @@
// 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 upsampling a mix buffer to 48Khz.
* Input must be 8Khz, 16Khz or 32Khz, and output will be 48Khz.
*/
struct UpsampleCommand : 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;
/// Pointer to the output samples buffer.
CpuAddr samples_buffer;
/// Pointer to input mix buffer indexes.
CpuAddr inputs;
/// Number of input mix buffers.
u32 buffer_count;
/// Unknown, unused.
u32 unk_20;
/// Source data sample count.
u32 source_sample_count;
/// Source data sample rate.
u32 source_sample_rate;
/// Pointer to the upsampler info for this command.
CpuAddr upsampler_info;
};
} // namespace AudioCore::AudioRenderer
// 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 upsampling a mix buffer to 48Khz.
* Input must be 8Khz, 16Khz or 32Khz, and output will be 48Khz.
*/
struct UpsampleCommand : 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;
/// Pointer to the output samples buffer.
CpuAddr samples_buffer;
/// Pointer to input mix buffer indexes.
CpuAddr inputs;
/// Number of input mix buffers.
u32 buffer_count;
/// Unknown, unused.
u32 unk_20;
/// Source data sample count.
u32 source_sample_count;
/// Source data sample rate.
u32 source_sample_rate;
/// Pointer to the upsampler info for this command.
CpuAddr upsampler_info;
};
} // namespace AudioCore::AudioRenderer

96
src/audio_core/renderer/command/sink/circular_buffer.cpp
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@@ -1,48 +1,48 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <vector>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/sink/circular_buffer.h"
#include "core/memory.h"
namespace AudioCore::AudioRenderer {
void CircularBufferSinkCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format(
"CircularBufferSinkCommand\n\tinput_count {} ring size {:04X} ring pos {:04X}\n\tinputs: ",
input_count, size, pos);
for (u32 i = 0; i < input_count; i++) {
string += fmt::format("{:02X}, ", inputs[i]);
}
string += "\n";
}
void CircularBufferSinkCommand::Process(const ADSP::CommandListProcessor& processor) {
constexpr s32 min{std::numeric_limits<s16>::min()};
constexpr s32 max{std::numeric_limits<s16>::max()};
std::vector<s16> output(processor.sample_count);
for (u32 channel = 0; channel < input_count; channel++) {
auto input{processor.mix_buffers.subspan(inputs[channel] * processor.sample_count,
processor.sample_count)};
for (u32 sample_index = 0; sample_index < processor.sample_count; sample_index++) {
output[sample_index] = static_cast<s16>(std::clamp(input[sample_index], min, max));
}
processor.memory->WriteBlockUnsafe(address + pos, output.data(),
output.size() * sizeof(s16));
pos += static_cast<u32>(processor.sample_count * sizeof(s16));
if (pos >= size) {
pos = 0;
}
}
}
bool CircularBufferSinkCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <vector>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/sink/circular_buffer.h"
#include "core/memory.h"
namespace AudioCore::AudioRenderer {
void CircularBufferSinkCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format(
"CircularBufferSinkCommand\n\tinput_count {} ring size {:04X} ring pos {:04X}\n\tinputs: ",
input_count, size, pos);
for (u32 i = 0; i < input_count; i++) {
string += fmt::format("{:02X}, ", inputs[i]);
}
string += "\n";
}
void CircularBufferSinkCommand::Process(const ADSP::CommandListProcessor& processor) {
constexpr s32 min{std::numeric_limits<s16>::min()};
constexpr s32 max{std::numeric_limits<s16>::max()};
std::vector<s16> output(processor.sample_count);
for (u32 channel = 0; channel < input_count; channel++) {
auto input{processor.mix_buffers.subspan(inputs[channel] * processor.sample_count,
processor.sample_count)};
for (u32 sample_index = 0; sample_index < processor.sample_count; sample_index++) {
output[sample_index] = static_cast<s16>(std::clamp(input[sample_index], min, max));
}
processor.memory->WriteBlockUnsafe(address + pos, output.data(),
output.size() * sizeof(s16));
pos += static_cast<u32>(processor.sample_count * sizeof(s16));
if (pos >= size) {
pos = 0;
}
}
}
bool CircularBufferSinkCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

110
src/audio_core/renderer/command/sink/circular_buffer.h
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@@ -1,55 +1,55 @@
// 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 sinking samples to a circular buffer.
*/
struct CircularBufferSinkCommand : 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;
/// Number of input mix buffers
u32 input_count;
/// Input mix buffer indexes
std::array<s16, MaxChannels> inputs;
/// Circular buffer address
CpuAddr address;
/// Circular buffer size
u32 size;
/// Current buffer offset
u32 pos;
};
} // namespace AudioCore::AudioRenderer
// 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 sinking samples to a circular buffer.
*/
struct CircularBufferSinkCommand : 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;
/// Number of input mix buffers
u32 input_count;
/// Input mix buffer indexes
std::array<s16, MaxChannels> inputs;
/// Circular buffer address
CpuAddr address;
/// Circular buffer size
u32 size;
/// Current buffer offset
u32 pos;
};
} // namespace AudioCore::AudioRenderer

118
src/audio_core/renderer/command/sink/device.cpp
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@@ -1,59 +1,59 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/sink/device.h"
#include "audio_core/sink/sink.h"
namespace AudioCore::AudioRenderer {
void DeviceSinkCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DeviceSinkCommand\n\t{} session {} input_count {}\n\tinputs: ",
std::string_view(name), session_id, input_count);
for (u32 i = 0; i < input_count; i++) {
string += fmt::format("{:02X}, ", inputs[i]);
}
string += "\n";
}
void DeviceSinkCommand::Process(const ADSP::CommandListProcessor& processor) {
constexpr s32 min = std::numeric_limits<s16>::min();
constexpr s32 max = std::numeric_limits<s16>::max();
auto stream{processor.GetOutputSinkStream()};
stream->SetSystemChannels(input_count);
Sink::SinkBuffer out_buffer{
.frames{TargetSampleCount},
.frames_played{0},
.tag{0},
.consumed{false},
};
std::vector<s16> samples(out_buffer.frames * input_count);
for (u32 channel = 0; channel < input_count; channel++) {
const auto offset{inputs[channel] * out_buffer.frames};
for (u32 index = 0; index < out_buffer.frames; index++) {
samples[index * input_count + channel] =
static_cast<s16>(std::clamp(sample_buffer[offset + index], min, max));
}
}
out_buffer.tag = reinterpret_cast<u64>(samples.data());
stream->AppendBuffer(out_buffer, samples);
if (stream->IsPaused()) {
stream->Start();
}
}
bool DeviceSinkCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/sink/device.h"
#include "audio_core/sink/sink.h"
namespace AudioCore::AudioRenderer {
void DeviceSinkCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("DeviceSinkCommand\n\t{} session {} input_count {}\n\tinputs: ",
std::string_view(name), session_id, input_count);
for (u32 i = 0; i < input_count; i++) {
string += fmt::format("{:02X}, ", inputs[i]);
}
string += "\n";
}
void DeviceSinkCommand::Process(const ADSP::CommandListProcessor& processor) {
constexpr s32 min = std::numeric_limits<s16>::min();
constexpr s32 max = std::numeric_limits<s16>::max();
auto stream{processor.GetOutputSinkStream()};
stream->SetSystemChannels(input_count);
Sink::SinkBuffer out_buffer{
.frames{TargetSampleCount},
.frames_played{0},
.tag{0},
.consumed{false},
};
std::vector<s16> samples(out_buffer.frames * input_count);
for (u32 channel = 0; channel < input_count; channel++) {
const auto offset{inputs[channel] * out_buffer.frames};
for (u32 index = 0; index < out_buffer.frames; index++) {
samples[index * input_count + channel] =
static_cast<s16>(std::clamp(sample_buffer[offset + index], min, max));
}
}
out_buffer.tag = reinterpret_cast<u64>(samples.data());
stream->AppendBuffer(out_buffer, samples);
if (stream->IsPaused()) {
stream->Start();
}
}
bool DeviceSinkCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

114
src/audio_core/renderer/command/sink/device.h
View File

@@ -1,57 +1,57 @@
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <span>
#include <string>
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for sinking samples to an output device.
*/
struct DeviceSinkCommand : 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;
/// Device name
char name[0x100];
/// System session id (unused)
s32 session_id;
/// Sample buffer to sink
std::span<s32> sample_buffer;
/// Number of input channels
u32 input_count;
/// Mix buffer indexes for each channel
std::array<s16, MaxChannels> inputs;
};
} // namespace AudioCore::AudioRenderer
// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <span>
#include <string>
#include "audio_core/renderer/command/icommand.h"
#include "common/common_types.h"
namespace AudioCore::AudioRenderer {
namespace ADSP {
class CommandListProcessor;
}
/**
* AudioRenderer command for sinking samples to an output device.
*/
struct DeviceSinkCommand : 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;
/// Device name
char name[0x100];
/// System session id (unused)
s32 session_id;
/// Sample buffer to sink
std::span<s32> sample_buffer;
/// Number of input channels
u32 input_count;
/// Mix buffer indexes for each channel
std::array<s16, MaxChannels> inputs;
};
} // namespace AudioCore::AudioRenderer