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Merge pull request #8545 from Kelebek1/Audio

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Project Andio
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liamwhite 2022-07-23 15:20:39 -04:00 committed by GitHub
parent 6c4e48dac4 458da8a948
commit 97729fd8e9
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GPG key ID: 4AEE18F83AFDEB23
270 changed files with 33712 additions and 8445 deletions
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2
.gitmodules vendored
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@ -3,7 +3,7 @@
url = https://github.com/benhoyt/inih.git url = https://github.com/benhoyt/inih.git
[submodule "cubeb"] [submodule "cubeb"]
path = externals/cubeb path = externals/cubeb
url = https://github.com/kinetiknz/cubeb.git url = https://github.com/mozilla/cubeb.git
[submodule "dynarmic"] [submodule "dynarmic"]
path = externals/dynarmic path = externals/dynarmic
url = https://github.com/MerryMage/dynarmic.git url = https://github.com/MerryMage/dynarmic.git

253
src/audio_core/CMakeLists.txt
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@ -1,54 +1,218 @@
add_library(audio_core STATIC add_library(audio_core STATIC
algorithm/filter.cpp audio_core.cpp
algorithm/filter.h audio_core.h
algorithm/interpolate.cpp audio_event.h
algorithm/interpolate.h audio_event.cpp
audio_out.cpp audio_render_manager.cpp
audio_out.h audio_render_manager.h
audio_renderer.cpp audio_in_manager.cpp
audio_renderer.h audio_in_manager.h
behavior_info.cpp audio_out_manager.cpp
behavior_info.h audio_out_manager.h
buffer.h audio_manager.cpp
codec.cpp audio_manager.h
codec.h common/audio_renderer_parameter.h
command_generator.cpp common/common.h
command_generator.h common/feature_support.h
common.h common/wave_buffer.h
delay_line.cpp common/workbuffer_allocator.h
delay_line.h device/audio_buffer.h
effect_context.cpp device/audio_buffers.h
effect_context.h device/device_session.cpp
info_updater.cpp device/device_session.h
info_updater.h in/audio_in.cpp
memory_pool.cpp in/audio_in.h
memory_pool.h in/audio_in_system.cpp
mix_context.cpp in/audio_in_system.h
mix_context.h out/audio_out.cpp
null_sink.h out/audio_out.h
sink.h out/audio_out_system.cpp
sink_context.cpp out/audio_out_system.h
sink_context.h renderer/adsp/adsp.cpp
sink_details.cpp renderer/adsp/adsp.h
sink_details.h renderer/adsp/audio_renderer.cpp
sink_stream.h renderer/adsp/audio_renderer.h
splitter_context.cpp renderer/adsp/command_buffer.h
splitter_context.h renderer/adsp/command_list_processor.cpp
stream.cpp renderer/adsp/command_list_processor.h
stream.h renderer/audio_device.cpp
voice_context.cpp renderer/audio_device.h
voice_context.h renderer/audio_renderer.h
renderer/audio_renderer.cpp
$<$<BOOL:${ENABLE_CUBEB}>:cubeb_sink.cpp cubeb_sink.h> renderer/behavior/behavior_info.cpp
$<$<BOOL:${ENABLE_SDL2}>:sdl2_sink.cpp sdl2_sink.h> renderer/behavior/behavior_info.h
renderer/behavior/info_updater.cpp
renderer/behavior/info_updater.h
renderer/command/data_source/adpcm.cpp
renderer/command/data_source/adpcm.h
renderer/command/data_source/decode.cpp
renderer/command/data_source/decode.h
renderer/command/data_source/pcm_float.cpp
renderer/command/data_source/pcm_float.h
renderer/command/data_source/pcm_int16.cpp
renderer/command/data_source/pcm_int16.h
renderer/command/effect/aux_.cpp
renderer/command/effect/aux_.h
renderer/command/effect/biquad_filter.cpp
renderer/command/effect/biquad_filter.h
renderer/command/effect/capture.cpp
renderer/command/effect/capture.h
renderer/command/effect/compressor.cpp
renderer/command/effect/compressor.h
renderer/command/effect/delay.cpp
renderer/command/effect/delay.h
renderer/command/effect/i3dl2_reverb.cpp
renderer/command/effect/i3dl2_reverb.h
renderer/command/effect/light_limiter.cpp
renderer/command/effect/light_limiter.h
renderer/command/effect/multi_tap_biquad_filter.cpp
renderer/command/effect/multi_tap_biquad_filter.h
renderer/command/effect/reverb.cpp
renderer/command/effect/reverb.h
renderer/command/mix/clear_mix.cpp
renderer/command/mix/clear_mix.h
renderer/command/mix/copy_mix.cpp
renderer/command/mix/copy_mix.h
renderer/command/mix/depop_for_mix_buffers.cpp
renderer/command/mix/depop_for_mix_buffers.h
renderer/command/mix/depop_prepare.cpp
renderer/command/mix/depop_prepare.h
renderer/command/mix/mix.cpp
renderer/command/mix/mix.h
renderer/command/mix/mix_ramp.cpp
renderer/command/mix/mix_ramp.h
renderer/command/mix/mix_ramp_grouped.cpp
renderer/command/mix/mix_ramp_grouped.h
renderer/command/mix/volume.cpp
renderer/command/mix/volume.h
renderer/command/mix/volume_ramp.cpp
renderer/command/mix/volume_ramp.h
renderer/command/performance/performance.cpp
renderer/command/performance/performance.h
renderer/command/resample/downmix_6ch_to_2ch.cpp
renderer/command/resample/downmix_6ch_to_2ch.h
renderer/command/resample/resample.h
renderer/command/resample/resample.cpp
renderer/command/resample/upsample.cpp
renderer/command/resample/upsample.h
renderer/command/sink/device.cpp
renderer/command/sink/device.h
renderer/command/sink/circular_buffer.cpp
renderer/command/sink/circular_buffer.h
renderer/command/command_buffer.cpp
renderer/command/command_buffer.h
renderer/command/command_generator.cpp
renderer/command/command_generator.h
renderer/command/command_list_header.h
renderer/command/command_processing_time_estimator.cpp
renderer/command/command_processing_time_estimator.h
renderer/command/commands.h
renderer/command/icommand.h
renderer/effect/aux_.cpp
renderer/effect/aux_.h
renderer/effect/biquad_filter.cpp
renderer/effect/biquad_filter.h
renderer/effect/buffer_mixer.cpp
renderer/effect/buffer_mixer.h
renderer/effect/capture.cpp
renderer/effect/capture.h
renderer/effect/compressor.cpp
renderer/effect/compressor.h
renderer/effect/delay.cpp
renderer/effect/delay.h
renderer/effect/effect_context.cpp
renderer/effect/effect_context.h
renderer/effect/effect_info_base.h
renderer/effect/effect_reset.h
renderer/effect/effect_result_state.h
renderer/effect/i3dl2.cpp
renderer/effect/i3dl2.h
renderer/effect/light_limiter.cpp
renderer/effect/light_limiter.h
renderer/effect/reverb.h
renderer/effect/reverb.cpp
renderer/mix/mix_context.cpp
renderer/mix/mix_context.h
renderer/mix/mix_info.cpp
renderer/mix/mix_info.h
renderer/memory/address_info.h
renderer/memory/memory_pool_info.cpp
renderer/memory/memory_pool_info.h
renderer/memory/pool_mapper.cpp
renderer/memory/pool_mapper.h
renderer/nodes/bit_array.h
renderer/nodes/edge_matrix.cpp
renderer/nodes/edge_matrix.h
renderer/nodes/node_states.cpp
renderer/nodes/node_states.h
renderer/performance/detail_aspect.cpp
renderer/performance/detail_aspect.h
renderer/performance/entry_aspect.cpp
renderer/performance/entry_aspect.h
renderer/performance/performance_detail.h
renderer/performance/performance_entry.h
renderer/performance/performance_entry_addresses.h
renderer/performance/performance_frame_header.h
renderer/performance/performance_manager.cpp
renderer/performance/performance_manager.h
renderer/sink/circular_buffer_sink_info.cpp
renderer/sink/circular_buffer_sink_info.h
renderer/sink/device_sink_info.cpp
renderer/sink/device_sink_info.h
renderer/sink/sink_context.cpp
renderer/sink/sink_context.h
renderer/sink/sink_info_base.cpp
renderer/sink/sink_info_base.h
renderer/splitter/splitter_context.cpp
renderer/splitter/splitter_context.h
renderer/splitter/splitter_destinations_data.cpp
renderer/splitter/splitter_destinations_data.h
renderer/splitter/splitter_info.cpp
renderer/splitter/splitter_info.h
renderer/system.cpp
renderer/system.h
renderer/system_manager.cpp
renderer/system_manager.h
renderer/upsampler/upsampler_info.h
renderer/upsampler/upsampler_manager.cpp
renderer/upsampler/upsampler_manager.h
renderer/upsampler/upsampler_state.h
renderer/voice/voice_channel_resource.h
renderer/voice/voice_context.cpp
renderer/voice/voice_context.h
renderer/voice/voice_info.cpp
renderer/voice/voice_info.h
renderer/voice/voice_state.h
sink/cubeb_sink.cpp
sink/cubeb_sink.h
sink/null_sink.h
sink/sdl2_sink.cpp
sink/sdl2_sink.h
sink/sink.h
sink/sink_details.cpp
sink/sink_details.h
sink/sink_stream.h
) )
create_target_directory_groups(audio_core) create_target_directory_groups(audio_core)
if (NOT MSVC) if (MSVC)
target_compile_options(audio_core PRIVATE
/we4242 # 'identifier': conversion from 'type1' to 'type2', possible loss of data
/we4244 # 'conversion': conversion from 'type1' to 'type2', possible loss of data
/we4245 # 'conversion': conversion from 'type1' to 'type2', signed/unsigned mismatch
/we4254 # 'operator': conversion from 'type1:field_bits' to 'type2:field_bits', possible loss of data
/we4456 # Declaration of 'identifier' hides previous local declaration
/we4457 # Declaration of 'identifier' hides function parameter
/we4458 # Declaration of 'identifier' hides class member
/we4459 # Declaration of 'identifier' hides global declaration
)
else()
target_compile_options(audio_core PRIVATE target_compile_options(audio_core PRIVATE
-Werror=conversion -Werror=conversion
-Werror=ignored-qualifiers -Werror=ignored-qualifiers
-Werror=shadow
-Werror=unused-variable
$<$<CXX_COMPILER_ID:GNU>:-Werror=unused-but-set-parameter> $<$<CXX_COMPILER_ID:GNU>:-Werror=unused-but-set-parameter>
$<$<CXX_COMPILER_ID:GNU>:-Werror=unused-but-set-variable> $<$<CXX_COMPILER_ID:GNU>:-Werror=unused-but-set-variable>
@ -58,6 +222,9 @@ if (NOT MSVC)
endif() endif()
target_link_libraries(audio_core PUBLIC common core) target_link_libraries(audio_core PUBLIC common core)
if (ARCHITECTURE_x86_64)
target_link_libraries(audio_core PRIVATE dynarmic)
endif()
if(ENABLE_CUBEB) if(ENABLE_CUBEB)
target_link_libraries(audio_core PRIVATE cubeb) target_link_libraries(audio_core PRIVATE cubeb)

79
src/audio_core/algorithm/filter.cpp
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@ -1,79 +0,0 @@
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#define _USE_MATH_DEFINES
#include <algorithm>
#include <array>
#include <cmath>
#include <vector>
#include "audio_core/algorithm/filter.h"
#include "common/common_types.h"
namespace AudioCore {
Filter Filter::LowPass(double cutoff, double Q) {
const double w0 = 2.0 * M_PI * cutoff;
const double sin_w0 = std::sin(w0);
const double cos_w0 = std::cos(w0);
const double alpha = sin_w0 / (2 * Q);
const double a0 = 1 + alpha;
const double a1 = -2.0 * cos_w0;
const double a2 = 1 - alpha;
const double b0 = 0.5 * (1 - cos_w0);
const double b1 = 1.0 * (1 - cos_w0);
const double b2 = 0.5 * (1 - cos_w0);
return {a0, a1, a2, b0, b1, b2};
}
Filter::Filter() : Filter(1.0, 0.0, 0.0, 1.0, 0.0, 0.0) {}
Filter::Filter(double a0_, double a1_, double a2_, double b0_, double b1_, double b2_)
: a1(a1_ / a0_), a2(a2_ / a0_), b0(b0_ / a0_), b1(b1_ / a0_), b2(b2_ / a0_) {}
void Filter::Process(std::vector<s16>& signal) {
const std::size_t num_frames = signal.size() / 2;
for (std::size_t i = 0; i < num_frames; i++) {
std::rotate(in.begin(), in.end() - 1, in.end());
std::rotate(out.begin(), out.end() - 1, out.end());
for (std::size_t ch = 0; ch < channel_count; ch++) {
in[0][ch] = signal[i * channel_count + ch];
out[0][ch] = b0 * in[0][ch] + b1 * in[1][ch] + b2 * in[2][ch] - a1 * out[1][ch] -
a2 * out[2][ch];
signal[i * 2 + ch] = static_cast<s16>(std::clamp(out[0][ch], -32768.0, 32767.0));
}
}
}
/// Calculates the appropriate Q for each biquad in a cascading filter.
/// @param total_count The total number of biquads to be cascaded.
/// @param index 0-index of the biquad to calculate the Q value for.
static double CascadingBiquadQ(std::size_t total_count, std::size_t index) {
const auto pole =
M_PI * static_cast<double>(2 * index + 1) / (4.0 * static_cast<double>(total_count));
return 1.0 / (2.0 * std::cos(pole));
}
CascadingFilter CascadingFilter::LowPass(double cutoff, std::size_t cascade_size) {
std::vector<Filter> cascade(cascade_size);
for (std::size_t i = 0; i < cascade_size; i++) {
cascade[i] = Filter::LowPass(cutoff, CascadingBiquadQ(cascade_size, i));
}
return CascadingFilter{std::move(cascade)};
}
CascadingFilter::CascadingFilter() = default;
CascadingFilter::CascadingFilter(std::vector<Filter> filters_) : filters(std::move(filters_)) {}
void CascadingFilter::Process(std::vector<s16>& signal) {
for (auto& filter : filters) {
filter.Process(signal);
}
}
} // namespace AudioCore

61
src/audio_core/algorithm/filter.h
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@ -1,61 +0,0 @@
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <vector>
#include "common/common_types.h"
namespace AudioCore {
/// Digital biquad filter:
///
/// b0 + b1 z^-1 + b2 z^-2
/// H(z) = ------------------------
/// a0 + a1 z^-1 + b2 z^-2
class Filter {
public:
/// Creates a low-pass filter.
/// @param cutoff Determines the cutoff frequency. A value from 0.0 to 1.0.
/// @param Q Determines the quality factor of this filter.
static Filter LowPass(double cutoff, double Q = 0.7071);
/// Passthrough filter.
Filter();
Filter(double a0_, double a1_, double a2_, double b0_, double b1_, double b2_);
void Process(std::vector<s16>& signal);
private:
static constexpr std::size_t channel_count = 2;
/// Coefficients are in normalized form (a0 = 1.0).
double a1, a2, b0, b1, b2;
/// Input History
std::array<std::array<double, channel_count>, 3> in;
/// Output History
std::array<std::array<double, channel_count>, 3> out;
};
/// Cascade filters to build up higher-order filters from lower-order ones.
class CascadingFilter {
public:
/// Creates a cascading low-pass filter.
/// @param cutoff Determines the cutoff frequency. A value from 0.0 to 1.0.
/// @param cascade_size Number of biquads in cascade.
static CascadingFilter LowPass(double cutoff, std::size_t cascade_size);
/// Passthrough.
CascadingFilter();
explicit CascadingFilter(std::vector<Filter> filters_);
void Process(std::vector<s16>& signal);
private:
std::vector<Filter> filters;
};
} // namespace AudioCore

232
src/audio_core/algorithm/interpolate.cpp
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@ -1,232 +0,0 @@
// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#define _USE_MATH_DEFINES
#include <algorithm>
#include <climits>
#include <cmath>
#include <vector>
#include "audio_core/algorithm/interpolate.h"
#include "common/common_types.h"
#include "common/logging/log.h"
namespace AudioCore {
constexpr std::array<s16, 512> curve_lut0{
6600, 19426, 6722, 3, 6479, 19424, 6845, 9, 6359, 19419, 6968, 15, 6239,
19412, 7093, 22, 6121, 19403, 7219, 28, 6004, 19391, 7345, 34, 5888, 19377,
7472, 41, 5773, 19361, 7600, 48, 5659, 19342, 7728, 55, 5546, 19321, 7857,
62, 5434, 19298, 7987, 69, 5323, 19273, 8118, 77, 5213, 19245, 8249, 84,
5104, 19215, 8381, 92, 4997, 19183, 8513, 101, 4890, 19148, 8646, 109, 4785,
19112, 8780, 118, 4681, 19073, 8914, 127, 4579, 19031, 9048, 137, 4477, 18988,
9183, 147, 4377, 18942, 9318, 157, 4277, 18895, 9454, 168, 4179, 18845, 9590,
179, 4083, 18793, 9726, 190, 3987, 18738, 9863, 202, 3893, 18682, 10000, 215,
3800, 18624, 10137, 228, 3709, 18563, 10274, 241, 3618, 18500, 10411, 255, 3529,
18436, 10549, 270, 3441, 18369, 10687, 285, 3355, 18300, 10824, 300, 3269, 18230,
10962, 317, 3186, 18157, 11100, 334, 3103, 18082, 11238, 351, 3022, 18006, 11375,
369, 2942, 17927, 11513, 388, 2863, 17847, 11650, 408, 2785, 17765, 11788, 428,
2709, 17681, 11925, 449, 2635, 17595, 12062, 471, 2561, 17507, 12198, 494, 2489,
17418, 12334, 517, 2418, 17327, 12470, 541, 2348, 17234, 12606, 566, 2280, 17140,
12741, 592, 2213, 17044, 12876, 619, 2147, 16946, 13010, 647, 2083, 16846, 13144,
675, 2020, 16745, 13277, 704, 1958, 16643, 13409, 735, 1897, 16539, 13541, 766,
1838, 16434, 13673, 798, 1780, 16327, 13803, 832, 1723, 16218, 13933, 866, 1667,
16109, 14062, 901, 1613, 15998, 14191, 937, 1560, 15885, 14318, 975, 1508, 15772,
14445, 1013, 1457, 15657, 14571, 1052, 1407, 15540, 14695, 1093, 1359, 15423, 14819,
1134, 1312, 15304, 14942, 1177, 1266, 15185, 15064, 1221, 1221, 15064, 15185, 1266,
1177, 14942, 15304, 1312, 1134, 14819, 15423, 1359, 1093, 14695, 15540, 1407, 1052,
14571, 15657, 1457, 1013, 14445, 15772, 1508, 975, 14318, 15885, 1560, 937, 14191,
15998, 1613, 901, 14062, 16109, 1667, 866, 13933, 16218, 1723, 832, 13803, 16327,
1780, 798, 13673, 16434, 1838, 766, 13541, 16539, 1897, 735, 13409, 16643, 1958,
704, 13277, 16745, 2020, 675, 13144, 16846, 2083, 647, 13010, 16946, 2147, 619,
12876, 17044, 2213, 592, 12741, 17140, 2280, 566, 12606, 17234, 2348, 541, 12470,
17327, 2418, 517, 12334, 17418, 2489, 494, 12198, 17507, 2561, 471, 12062, 17595,
2635, 449, 11925, 17681, 2709, 428, 11788, 17765, 2785, 408, 11650, 17847, 2863,
388, 11513, 17927, 2942, 369, 11375, 18006, 3022, 351, 11238, 18082, 3103, 334,
11100, 18157, 3186, 317, 10962, 18230, 3269, 300, 10824, 18300, 3355, 285, 10687,
18369, 3441, 270, 10549, 18436, 3529, 255, 10411, 18500, 3618, 241, 10274, 18563,
3709, 228, 10137, 18624, 3800, 215, 10000, 18682, 3893, 202, 9863, 18738, 3987,
190, 9726, 18793, 4083, 179, 9590, 18845, 4179, 168, 9454, 18895, 4277, 157,
9318, 18942, 4377, 147, 9183, 18988, 4477, 137, 9048, 19031, 4579, 127, 8914,
19073, 4681, 118, 8780, 19112, 4785, 109, 8646, 19148, 4890, 101, 8513, 19183,
4997, 92, 8381, 19215, 5104, 84, 8249, 19245, 5213, 77, 8118, 19273, 5323,
69, 7987, 19298, 5434, 62, 7857, 19321, 5546, 55, 7728, 19342, 5659, 48,
7600, 19361, 5773, 41, 7472, 19377, 5888, 34, 7345, 19391, 6004, 28, 7219,
19403, 6121, 22, 7093, 19412, 6239, 15, 6968, 19419, 6359, 9, 6845, 19424,
6479, 3, 6722, 19426, 6600};
constexpr std::array<s16, 512> curve_lut1{
-68, 32639, 69, -5, -200, 32630, 212, -15, -328, 32613, 359, -26, -450,
32586, 512, -36, -568, 32551, 669, -47, -680, 32507, 832, -58, -788, 32454,
1000, -69, -891, 32393, 1174, -80, -990, 32323, 1352, -92, -1084, 32244, 1536,
-103, -1173, 32157, 1724, -115, -1258, 32061, 1919, -128, -1338, 31956, 2118, -140,
-1414, 31844, 2322, -153, -1486, 31723, 2532, -167, -1554, 31593, 2747, -180, -1617,
31456, 2967, -194, -1676, 31310, 3192, -209, -1732, 31157, 3422, -224, -1783, 30995,
3657, -240, -1830, 30826, 3897, -256, -1874, 30649, 4143, -272, -1914, 30464, 4393,
-289, -1951, 30272, 4648, -307, -1984, 30072, 4908, -325, -2014, 29866, 5172, -343,
-2040, 29652, 5442, -362, -2063, 29431, 5716, -382, -2083, 29203, 5994, -403, -2100,
28968, 6277, -424, -2114, 28727, 6565, -445, -2125, 28480, 6857, -468, -2133, 28226,
7153, -490, -2139, 27966, 7453, -514, -2142, 27700, 7758, -538, -2142, 27428, 8066,
-563, -2141, 27151, 8378, -588, -2136, 26867, 8694, -614, -2130, 26579, 9013, -641,
-2121, 26285, 9336, -668, -2111, 25987, 9663, -696, -2098, 25683, 9993, -724, -2084,
25375, 10326, -753, -2067, 25063, 10662, -783, -2049, 24746, 11000, -813, -2030, 24425,
11342, -844, -2009, 24100, 11686, -875, -1986, 23771, 12033, -907, -1962, 23438, 12382,
-939, -1937, 23103, 12733, -972, -1911, 22764, 13086, -1005, -1883, 22422, 13441, -1039,
-1855, 22077, 13798, -1072, -1825, 21729, 14156, -1107, -1795, 21380, 14516, -1141, -1764,
21027, 14877, -1176, -1732, 20673, 15239, -1211, -1700, 20317, 15602, -1246, -1667, 19959,
15965, -1282, -1633, 19600, 16329, -1317, -1599, 19239, 16694, -1353, -1564, 18878, 17058,
-1388, -1530, 18515, 17423, -1424, -1495, 18151, 17787, -1459, -1459, 17787, 18151, -1495,
-1424, 17423, 18515, -1530, -1388, 17058, 18878, -1564, -1353, 16694, 19239, -1599, -1317,
16329, 19600, -1633, -1282, 15965, 19959, -1667, -1246, 15602, 20317, -1700, -1211, 15239,
20673, -1732, -1176, 14877, 21027, -1764, -1141, 14516, 21380, -1795, -1107, 14156, 21729,
-1825, -1072, 13798, 22077, -1855, -1039, 13441, 22422, -1883, -1005, 13086, 22764, -1911,
-972, 12733, 23103, -1937, -939, 12382, 23438, -1962, -907, 12033, 23771, -1986, -875,
11686, 24100, -2009, -844, 11342, 24425, -2030, -813, 11000, 24746, -2049, -783, 10662,
25063, -2067, -753, 10326, 25375, -2084, -724, 9993, 25683, -2098, -696, 9663, 25987,
-2111, -668, 9336, 26285, -2121, -641, 9013, 26579, -2130, -614, 8694, 26867, -2136,
-588, 8378, 27151, -2141, -563, 8066, 27428, -2142, -538, 7758, 27700, -2142, -514,
7453, 27966, -2139, -490, 7153, 28226, -2133, -468, 6857, 28480, -2125, -445, 6565,
28727, -2114, -424, 6277, 28968, -2100, -403, 5994, 29203, -2083, -382, 5716, 29431,
-2063, -362, 5442, 29652, -2040, -343, 5172, 29866, -2014, -325, 4908, 30072, -1984,
-307, 4648, 30272, -1951, -289, 4393, 30464, -1914, -272, 4143, 30649, -1874, -256,
3897, 30826, -1830, -240, 3657, 30995, -1783, -224, 3422, 31157, -1732, -209, 3192,
31310, -1676, -194, 2967, 31456, -1617, -180, 2747, 31593, -1554, -167, 2532, 31723,
-1486, -153, 2322, 31844, -1414, -140, 2118, 31956, -1338, -128, 1919, 32061, -1258,
-115, 1724, 32157, -1173, -103, 1536, 32244, -1084, -92, 1352, 32323, -990, -80,
1174, 32393, -891, -69, 1000, 32454, -788, -58, 832, 32507, -680, -47, 669,
32551, -568, -36, 512, 32586, -450, -26, 359, 32613, -328, -15, 212, 32630,
-200, -5, 69, 32639, -68};
constexpr std::array<s16, 512> curve_lut2{
3195, 26287, 3329, -32, 3064, 26281, 3467, -34, 2936, 26270, 3608, -38, 2811,
26253, 3751, -42, 2688, 26230, 3897, -46, 2568, 26202, 4046, -50, 2451, 26169,
4199, -54, 2338, 26130, 4354, -58, 2227, 26085, 4512, -63, 2120, 26035, 4673,
-67, 2015, 25980, 4837, -72, 1912, 25919, 5004, -76, 1813, 25852, 5174, -81,
1716, 25780, 5347, -87, 1622, 25704, 5522, -92, 1531, 25621, 5701, -98, 1442,
25533, 5882, -103, 1357, 25440, 6066, -109, 1274, 25342, 6253, -115, 1193, 25239,
6442, -121, 1115, 25131, 6635, -127, 1040, 25018, 6830, -133, 967, 24899, 7027,
-140, 897, 24776, 7227, -146, 829, 24648, 7430, -153, 764, 24516, 7635, -159,
701, 24379, 7842, -166, 641, 24237, 8052, -174, 583, 24091, 8264, -181, 526,
23940, 8478, -187, 472, 23785, 8695, -194, 420, 23626, 8914, -202, 371, 23462,
9135, -209, 324, 23295, 9358, -215, 279, 23123, 9583, -222, 236, 22948, 9809,
-230, 194, 22769, 10038, -237, 154, 22586, 10269, -243, 117, 22399, 10501, -250,
81, 22208, 10735, -258, 47, 22015, 10970, -265, 15, 21818, 11206, -271, -16,
21618, 11444, -277, -44, 21415, 11684, -283, -71, 21208, 11924, -290, -97, 20999,
12166, -296, -121, 20786, 12409, -302, -143, 20571, 12653, -306, -163, 20354, 12898,
-311, -183, 20134, 13143, -316, -201, 19911, 13389, -321, -218, 19686, 13635, -325,
-234, 19459, 13882, -328, -248, 19230, 14130, -332, -261, 18998, 14377, -335, -273,
18765, 14625, -337, -284, 18531, 14873, -339, -294, 18295, 15121, -341, -302, 18057,
15369, -341, -310, 17817, 15617, -341, -317, 17577, 15864, -340, -323, 17335, 16111,
-340, -328, 17092, 16357, -338, -332, 16848, 16603, -336, -336, 16603, 16848, -332,
-338, 16357, 17092, -328, -340, 16111, 17335, -323, -340, 15864, 17577, -317, -341,
15617, 17817, -310, -341, 15369, 18057, -302, -341, 15121, 18295, -294, -339, 14873,
18531, -284, -337, 14625, 18765, -273, -335, 14377, 18998, -261, -332, 14130, 19230,
-248, -328, 13882, 19459, -234, -325, 13635, 19686, -218, -321, 13389, 19911, -201,
-316, 13143, 20134, -183, -311, 12898, 20354, -163, -306, 12653, 20571, -143, -302,
12409, 20786, -121, -296, 12166, 20999, -97, -290, 11924, 21208, -71, -283, 11684,
21415, -44, -277, 11444, 21618, -16, -271, 11206, 21818, 15, -265, 10970, 22015,
47, -258, 10735, 22208, 81, -250, 10501, 22399, 117, -243, 10269, 22586, 154,
-237, 10038, 22769, 194, -230, 9809, 22948, 236, -222, 9583, 23123, 279, -215,
9358, 23295, 324, -209, 9135, 23462, 371, -202, 8914, 23626, 420, -194, 8695,
23785, 472, -187, 8478, 23940, 526, -181, 8264, 24091, 583, -174, 8052, 24237,
641, -166, 7842, 24379, 701, -159, 7635, 24516, 764, -153, 7430, 24648, 829,
-146, 7227, 24776, 897, -140, 7027, 24899, 967, -133, 6830, 25018, 1040, -127,
6635, 25131, 1115, -121, 6442, 25239, 1193, -115, 6253, 25342, 1274, -109, 6066,
25440, 1357, -103, 5882, 25533, 1442, -98, 5701, 25621, 1531, -92, 5522, 25704,
1622, -87, 5347, 25780, 1716, -81, 5174, 25852, 1813, -76, 5004, 25919, 1912,
-72, 4837, 25980, 2015, -67, 4673, 26035, 2120, -63, 4512, 26085, 2227, -58,
4354, 26130, 2338, -54, 4199, 26169, 2451, -50, 4046, 26202, 2568, -46, 3897,
26230, 2688, -42, 3751, 26253, 2811, -38, 3608, 26270, 2936, -34, 3467, 26281,
3064, -32, 3329, 26287, 3195};
std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input, double ratio) {
if (input.size() < 2)
return {};
if (ratio <= 0) {
LOG_ERROR(Audio, "Nonsensical interpolation ratio {}", ratio);
return input;
}
const s32 step{static_cast<s32>(ratio * 0x8000)};
const std::array<s16, 512>& lut = [step] {
if (step > 0xaaaa) {
return curve_lut0;
}
if (step <= 0x8000) {
return curve_lut1;
}
return curve_lut2;
}();
const std::size_t num_frames{input.size() / 2};
std::vector<s16> output;
output.reserve(static_cast<std::size_t>(static_cast<double>(input.size()) / ratio +
InterpolationState::taps));
for (std::size_t frame{}; frame < num_frames; ++frame) {
const std::size_t lut_index{(state.fraction >> 8) * InterpolationState::taps};
std::rotate(state.history.begin(), state.history.end() - 1, state.history.end());
state.history[0][0] = input[frame * 2 + 0];
state.history[0][1] = input[frame * 2 + 1];
while (state.position <= 1.0) {
const s32 left{state.history[0][0] * lut[lut_index + 0] +
state.history[1][0] * lut[lut_index + 1] +
state.history[2][0] * lut[lut_index + 2] +
state.history[3][0] * lut[lut_index + 3]};
const s32 right{state.history[0][1] * lut[lut_index + 0] +
state.history[1][1] * lut[lut_index + 1] +
state.history[2][1] * lut[lut_index + 2] +
state.history[3][1] * lut[lut_index + 3]};
const s32 new_offset{state.fraction + step};
state.fraction = new_offset & 0x7fff;
output.emplace_back(static_cast<s16>(std::clamp(left >> 15, SHRT_MIN, SHRT_MAX)));
output.emplace_back(static_cast<s16>(std::clamp(right >> 15, SHRT_MIN, SHRT_MAX)));
state.position += ratio;
}
state.position -= 1.0;
}
return output;
}
void Resample(s32* output, const s32* input, s32 pitch, s32& fraction, std::size_t sample_count) {
const std::array<s16, 512>& lut = [pitch] {
if (pitch > 0xaaaa) {
return curve_lut0;
}
if (pitch <= 0x8000) {
return curve_lut1;
}
return curve_lut2;
}();
std::size_t index{};
for (std::size_t i = 0; i < sample_count; i++) {
const std::size_t lut_index{(static_cast<std::size_t>(fraction) >> 8) * 4};
const auto l0 = lut[lut_index + 0];
const auto l1 = lut[lut_index + 1];
const auto l2 = lut[lut_index + 2];
const auto l3 = lut[lut_index + 3];
const auto s0 = static_cast<s32>(input[index + 0]);
const auto s1 = static_cast<s32>(input[index + 1]);
const auto s2 = static_cast<s32>(input[index + 2]);
const auto s3 = static_cast<s32>(input[index + 3]);
output[i] = (l0 * s0 + l1 * s1 + l2 * s2 + l3 * s3) >> 15;
fraction += pitch;
index += (fraction >> 15);
fraction &= 0x7fff;
}
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <vector>
#include "common/common_types.h"
namespace AudioCore {
struct InterpolationState {
static constexpr std::size_t taps{4};
static constexpr std::size_t history_size{taps * 2 - 1};
std::array<std::array<s16, 2>, history_size> history{};
double position{};
s32 fraction{};
};
/// Interpolates input signal to produce output signal.
/// @param input The signal to interpolate.
/// @param ratio Interpolation ratio.
/// ratio > 1.0 results in fewer output samples.
/// ratio < 1.0 results in more output samples.
/// @returns Output signal.
std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input, double ratio);
/// Interpolates input signal to produce output signal.
/// @param input The signal to interpolate.
/// @param input_rate The sample rate of input.
/// @param output_rate The desired sample rate of the output.
/// @returns Output signal.
inline std::vector<s16> Interpolate(InterpolationState& state, std::vector<s16> input,
u32 input_rate, u32 output_rate) {
const double ratio = static_cast<double>(input_rate) / static_cast<double>(output_rate);
return Interpolate(state, std::move(input), ratio);
}
/// Nintendo Switchs DSP resampling algorithm. Based on a single channel
void Resample(s32* output, const s32* input, s32 pitch, s32& fraction, std::size_t sample_count);
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_core.h"
#include "audio_core/sink/sink_details.h"
#include "common/settings.h"
#include "core/core.h"
namespace AudioCore {
AudioCore::AudioCore(Core::System& system) : audio_manager{std::make_unique<AudioManager>(system)} {
CreateSinks();
// Must be created after the sinks
adsp = std::make_unique<AudioRenderer::ADSP::ADSP>(system, *output_sink);
}
AudioCore ::~AudioCore() {
Shutdown();
}
void AudioCore::CreateSinks() {
const auto& sink_id{Settings::values.sink_id};
const auto& audio_output_device_id{Settings::values.audio_output_device_id};
const auto& audio_input_device_id{Settings::values.audio_input_device_id};
output_sink = Sink::CreateSinkFromID(sink_id.GetValue(), audio_output_device_id.GetValue());
input_sink = Sink::CreateSinkFromID(sink_id.GetValue(), audio_input_device_id.GetValue());
}
void AudioCore::Shutdown() {
audio_manager->Shutdown();
}
AudioManager& AudioCore::GetAudioManager() {
return *audio_manager;
}
Sink::Sink& AudioCore::GetOutputSink() {
return *output_sink;
}
Sink::Sink& AudioCore::GetInputSink() {
return *input_sink;
}
AudioRenderer::ADSP::ADSP& AudioCore::GetADSP() {
return *adsp;
}
void AudioCore::PauseSinks(const bool pausing) const {
if (pausing) {
output_sink->PauseStreams();
input_sink->PauseStreams();
} else {
output_sink->UnpauseStreams();
input_sink->UnpauseStreams();
}
}
u32 AudioCore::GetStreamQueue() const {
return estimated_queue.load();
}
void AudioCore::SetStreamQueue(u32 size) {
estimated_queue.store(size);
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <memory>
#include "audio_core/audio_manager.h"
#include "audio_core/renderer/adsp/adsp.h"
#include "audio_core/sink/sink.h"
namespace Core {
class System;
}
namespace AudioCore {
class AudioManager;
/**
* Main audio class, sotred inside the core, and holding the audio manager, all sinks, and the ADSP.
*/
class AudioCore {
public:
explicit AudioCore(Core::System& system);
~AudioCore();
/**
* Shutdown the audio core.
*/
void Shutdown();
/**
* Get a reference to the audio manager.
*
* @return Ref to the audio manager.
*/
AudioManager& GetAudioManager();
/**
* Get the audio output sink currently in use.
*
* @return Ref to the sink.
*/
Sink::Sink& GetOutputSink();
/**
* Get the audio input sink currently in use.
*
* @return Ref to the sink.
*/
Sink::Sink& GetInputSink();
/**
* Get the ADSP.
*
* @return Ref to the ADSP.
*/
AudioRenderer::ADSP::ADSP& GetADSP();
/**
* Pause the sink. Called from the core.
*
* @param pausing - Is this pause due to an actual pause, or shutdown?
* Unfortunately, shutdown also pauses streams, which can cause issues.
*/
void PauseSinks(bool pausing) const;
/**
* Get the size of the current stream queue.
*
* @return Current stream queue size.
*/
u32 GetStreamQueue() const;
/**
* Get the size of the current stream queue.
*
* @param size - New stream size.
*/
void SetStreamQueue(u32 size);
private:
/**
* Create the sinks on startup.
*/
void CreateSinks();
/// Main audio manager for audio in/out
std::unique_ptr<AudioManager> audio_manager;
/// Sink used for audio renderer and audio out
std::unique_ptr<Sink::Sink> output_sink;
/// Sink used for audio input
std::unique_ptr<Sink::Sink> input_sink;
/// The ADSP in the sysmodule
std::unique_ptr<AudioRenderer::ADSP::ADSP> adsp;
/// Current size of the stream queue
std::atomic<u32> estimated_queue{0};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_event.h"
#include "common/assert.h"
namespace AudioCore {
size_t Event::GetManagerIndex(const Type type) const {
switch (type) {
case Type::AudioInManager:
return 0;
case Type::AudioOutManager:
return 1;
case Type::FinalOutputRecorderManager:
return 2;
case Type::Max:
return 3;
default:
UNREACHABLE();
}
return 3;
}
void Event::SetAudioEvent(const Type type, const bool signalled) {
events_signalled[GetManagerIndex(type)] = signalled;
if (signalled) {
manager_event.notify_one();
}
}
bool Event::CheckAudioEventSet(const Type type) const {
return events_signalled[GetManagerIndex(type)];
}
std::mutex& Event::GetAudioEventLock() {
return event_lock;
}
std::condition_variable_any& Event::GetAudioEvent() {
return manager_event;
}
bool Event::Wait(std::unique_lock<std::mutex>& l, const std::chrono::seconds timeout) {
bool timed_out{false};
if (!manager_event.wait_for(l, timeout, [&]() {
return std::ranges::any_of(events_signalled, [](bool x) { return x; });
})) {
timed_out = true;
}
return timed_out;
}
void Event::ClearEvents() {
events_signalled[0] = false;
events_signalled[1] = false;
events_signalled[2] = false;
events_signalled[3] = false;
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <atomic>
#include <chrono>
#include <condition_variable>
#include <mutex>
namespace AudioCore {
/**
* Responsible for the input/output events, set by the stream backend when buffers are consumed, and
* waited on by the audio manager. These callbacks signal the game's events to keep the audio buffer
* recycling going.
* In a real Switch this is not a seprate class, and exists entirely within the audio manager.
* On the Switch it's implemented more simply through a MultiWaitEventHolder, where it can
* wait on multiple events at once, and the events are not needed by the backend.
*/
class Event {
public:
enum class Type {
AudioInManager,
AudioOutManager,
FinalOutputRecorderManager,
Max,
};
/**
* Convert a manager type to an index.
*
* @param type - The manager type to convert
* @return The index of the type.
*/
size_t GetManagerIndex(Type type) const;
/**
* Set an audio event to true or false.
*
* @param type - The manager type to signal.
* @param signalled - Its signal state.
*/
void SetAudioEvent(Type type, bool signalled);
/**
* Check if the given manager type is signalled.
*
* @param type - The manager type to check.
* @return True if the event is signalled, otherwise false.
*/
bool CheckAudioEventSet(Type type) const;
/**
* Get the lock for audio events.
*
* @return Reference to the lock.
*/
std::mutex& GetAudioEventLock();
/**
* Get the manager event, this signals the audio manager to release buffers and signal the game
* for more.
*
* @return Reference to the condition variable.
*/
std::condition_variable_any& GetAudioEvent();
/**
* Wait on the manager_event.
*
* @param l - Lock held by the wait.
* @param timeout - Timeout for the wait. This is 2 seconds by default.
* @return True if the wait timed out, otherwise false if signalled.
*/
bool Wait(std::unique_lock<std::mutex>& l, std::chrono::seconds timeout);
/**
* Reset all manager events.
*/
void ClearEvents();
private:
/// Lock, used bythe audio manager
std::mutex event_lock;
/// Array of events, one per system type (see Type), last event is used to terminate
std::array<std::atomic<bool>, 4> events_signalled;
/// Event to signal the audio manager
std::condition_variable_any manager_event;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_core.h"
#include "audio_core/audio_in_manager.h"
#include "audio_core/audio_manager.h"
#include "audio_core/in/audio_in.h"
#include "audio_core/sink/sink_details.h"
#include "common/settings.h"
#include "core/core.h"
#include "core/hle/service/audio/errors.h"
namespace AudioCore::AudioIn {
Manager::Manager(Core::System& system_) : system{system_} {
std::iota(session_ids.begin(), session_ids.end(), 0);
num_free_sessions = MaxInSessions;
}
Result Manager::AcquireSessionId(size_t& session_id) {
if (num_free_sessions == 0) {
LOG_ERROR(Service_Audio, "All 4 AudioIn sessions are in use, cannot create any more");
return Service::Audio::ERR_MAXIMUM_SESSIONS_REACHED;
}
session_id = session_ids[next_session_id];
next_session_id = (next_session_id + 1) % MaxInSessions;
num_free_sessions--;
return ResultSuccess;
}
void Manager::ReleaseSessionId(const size_t session_id) {
std::scoped_lock l{mutex};
LOG_DEBUG(Service_Audio, "Freeing AudioIn session {}", session_id);
session_ids[free_session_id] = session_id;
num_free_sessions++;
free_session_id = (free_session_id + 1) % MaxInSessions;
sessions[session_id].reset();
applet_resource_user_ids[session_id] = 0;
}
Result Manager::LinkToManager() {
std::scoped_lock l{mutex};
if (!linked_to_manager) {
AudioManager& manager{system.AudioCore().GetAudioManager()};
manager.SetInManager(std::bind(&Manager::BufferReleaseAndRegister, this));
linked_to_manager = true;
}
return ResultSuccess;
}
void Manager::Start() {
if (sessions_started) {
return;
}
std::scoped_lock l{mutex};
for (auto& session : sessions) {
if (session) {
session->StartSession();
}
}
sessions_started = true;
}
void Manager::BufferReleaseAndRegister() {
std::scoped_lock l{mutex};
for (auto& session : sessions) {
if (session != nullptr) {
session->ReleaseAndRegisterBuffers();
}
}
}
u32 Manager::GetDeviceNames(std::vector<AudioRenderer::AudioDevice::AudioDeviceName>& names,
[[maybe_unused]] const u32 max_count,
[[maybe_unused]] const bool filter) {
std::scoped_lock l{mutex};
LinkToManager();
auto input_devices{Sink::GetDeviceListForSink(Settings::values.sink_id.GetValue(), true)};
if (input_devices.size() > 1) {
names.push_back(AudioRenderer::AudioDevice::AudioDeviceName("Uac"));
return 1;
}
return 0;
}
} // namespace AudioCore::AudioIn

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <mutex>
#include <vector>
#include "audio_core/renderer/audio_device.h"
namespace Core {
class System;
}
namespace AudioCore::AudioIn {
class In;
constexpr size_t MaxInSessions = 4;
/**
* Manages all audio in sessions.
*/
class Manager {
public:
explicit Manager(Core::System& system);
/**
* Acquire a free session id for opening a new audio in.
*
* @param session_id - Output session_id.
* @return Result code.
*/
Result AcquireSessionId(size_t& session_id);
/**
* Release a session id on close.
*
* @param session_id - Session id to free.
*/
void ReleaseSessionId(size_t session_id);
/**
* Link the audio in manager to the main audio manager.
*
* @return Result code.
*/
Result LinkToManager();
/**
* Start the audio in manager.
*/
void Start();
/**
* Callback function, called by the audio manager when the audio in event is signalled.
*/
void BufferReleaseAndRegister();
/**
* Get a list of audio in device names.
*
* @oaram names - Output container to write names to.
* @param max_count - Maximum numebr of deivce names to write. Unused
* @param filter - Should the list be filtered? Unused.
* @return Number of names written.
*/
u32 GetDeviceNames(std::vector<AudioRenderer::AudioDevice::AudioDeviceName>& names,
u32 max_count, bool filter);
/// Core system
Core::System& system;
/// Array of session ids
std::array<size_t, MaxInSessions> session_ids{};
/// Array of resource user ids
std::array<size_t, MaxInSessions> applet_resource_user_ids{};
/// Pointer to each open session
std::array<std::shared_ptr<In>, MaxInSessions> sessions{};
/// The number of free sessions
size_t num_free_sessions{};
/// The next session id to be taken
size_t next_session_id{};
/// The next session id to be freed
size_t free_session_id{};
/// Whether this is linked to the audio manager
bool linked_to_manager{};
/// Whether the sessions have been started
bool sessions_started{};
/// Protect state due to audio manager callback
std::recursive_mutex mutex{};
};
} // namespace AudioCore::AudioIn

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_in_manager.h"
#include "audio_core/audio_manager.h"
#include "audio_core/audio_out_manager.h"
#include "core/core.h"
namespace AudioCore {
AudioManager::AudioManager(Core::System& system_) : system{system_} {
thread = std::jthread([this]() { ThreadFunc(); });
}
void AudioManager::Shutdown() {
running = false;
events.SetAudioEvent(Event::Type::Max, true);
thread.join();
}
Result AudioManager::SetOutManager(BufferEventFunc buffer_func) {
if (!running) {
return Service::Audio::ERR_OPERATION_FAILED;
}
std::scoped_lock l{lock};
const auto index{events.GetManagerIndex(Event::Type::AudioOutManager)};
if (buffer_events[index] == nullptr) {
buffer_events[index] = buffer_func;
needs_update = true;
events.SetAudioEvent(Event::Type::AudioOutManager, true);
}
return ResultSuccess;
}
Result AudioManager::SetInManager(BufferEventFunc buffer_func) {
if (!running) {
return Service::Audio::ERR_OPERATION_FAILED;
}
std::scoped_lock l{lock};
const auto index{events.GetManagerIndex(Event::Type::AudioInManager)};
if (buffer_events[index] == nullptr) {
buffer_events[index] = buffer_func;
needs_update = true;
events.SetAudioEvent(Event::Type::AudioInManager, true);
}
return ResultSuccess;
}
void AudioManager::SetEvent(const Event::Type type, const bool signalled) {
events.SetAudioEvent(type, signalled);
}
void AudioManager::ThreadFunc() {
std::unique_lock l{events.GetAudioEventLock()};
events.ClearEvents();
running = true;
while (running) {
auto timed_out{events.Wait(l, std::chrono::seconds(2))};
if (events.CheckAudioEventSet(Event::Type::Max)) {
break;
}
for (size_t i = 0; i < buffer_events.size(); i++) {
if (events.CheckAudioEventSet(Event::Type(i)) || timed_out) {
if (buffer_events[i]) {
buffer_events[i]();
}
}
events.SetAudioEvent(Event::Type(i), false);
}
}
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <atomic>
#include <functional>
#include <mutex>
#include <thread>
#include "audio_core/audio_event.h"
#include "core/hle/service/audio/errors.h"
namespace Core {
class System;
}
namespace AudioCore {
namespace AudioOut {
class Manager;
}
namespace AudioIn {
class Manager;
}
/**
* The AudioManager's main purpose is to wait for buffer events for the audio in and out managers,
* and call an associated callback to release buffers.
*
* Execution pattern is:
* Buffers appended ->
* Buffers queued and played by the backend stream ->
* When consumed, set the corresponding manager event and signal the audio manager ->
* Consumed buffers are released, game is signalled ->
* Game appends more buffers.
*
* This is only used by audio in and audio out.
*/
class AudioManager {
using BufferEventFunc = std::function<void()>;
public:
explicit AudioManager(Core::System& system);
/**
* Shutdown the audio manager.
*/
void Shutdown();
/**
* Register the out manager, keeping a function to be called when the out event is signalled.
*
* @param buffer_func - Function to be called on signal.
* @return Result code.
*/
Result SetOutManager(BufferEventFunc buffer_func);
/**
* Register the in manager, keeping a function to be called when the in event is signalled.
*
* @param buffer_func - Function to be called on signal.
* @return Result code.
*/
Result SetInManager(BufferEventFunc buffer_func);
/**
* Set an event to signalled, and signal the thread.
*
* @param type - Manager type to set.
* @param signalled - Set the event to true or false?
*/
void SetEvent(Event::Type type, bool signalled);
private:
/**
* Main thread, waiting on a manager signal and calling the registered fucntion.
*/
void ThreadFunc();
/// Core system
Core::System& system;
/// Have sessions started palying?
bool sessions_started{};
/// Is the main thread running?
std::atomic<bool> running{};
/// Unused
bool needs_update{};
/// Events to be set and signalled
Event events{};
/// Callbacks for each manager
std::array<BufferEventFunc, 3> buffer_events{};
/// General lock
std::mutex lock{};
/// Main thread for waiting and callbacks
std::jthread thread;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_out.h"
#include "audio_core/sink.h"
#include "audio_core/sink_details.h"
#include "common/assert.h"
#include "common/logging/log.h"
#include "common/settings.h"
namespace AudioCore {
/// Returns the stream format from the specified number of channels
static Stream::Format ChannelsToStreamFormat(u32 num_channels) {
switch (num_channels) {
case 1:
return Stream::Format::Mono16;
case 2:
return Stream::Format::Stereo16;
case 6:
return Stream::Format::Multi51Channel16;
}
UNIMPLEMENTED_MSG("Unimplemented num_channels={}", num_channels);
return {};
}
StreamPtr AudioOut::OpenStream(Core::Timing::CoreTiming& core_timing, u32 sample_rate,
u32 num_channels, std::string&& name,
Stream::ReleaseCallback&& release_callback) {
if (!sink) {
sink = CreateSinkFromID(Settings::values.sink_id.GetValue(),
Settings::values.audio_device_id.GetValue());
}
return std::make_shared<Stream>(
core_timing, sample_rate, ChannelsToStreamFormat(num_channels), std::move(release_callback),
sink->AcquireSinkStream(sample_rate, num_channels, name), std::move(name));
}
std::vector<Buffer::Tag> AudioOut::GetTagsAndReleaseBuffers(StreamPtr stream,
std::size_t max_count) {
return stream->GetTagsAndReleaseBuffers(max_count);
}
std::vector<Buffer::Tag> AudioOut::GetTagsAndReleaseBuffers(StreamPtr stream) {
return stream->GetTagsAndReleaseBuffers();
}
void AudioOut::StartStream(StreamPtr stream) {
stream->Play();
}
void AudioOut::StopStream(StreamPtr stream) {
stream->Stop();
}
bool AudioOut::QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data) {
return stream->QueueBuffer(std::make_shared<Buffer>(tag, std::move(data)));
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <memory>
#include <string>
#include <vector>
#include "audio_core/buffer.h"
#include "audio_core/sink.h"
#include "audio_core/stream.h"
#include "common/common_types.h"
namespace Core::Timing {
class CoreTiming;
}
namespace AudioCore {
/**
* Represents an audio playback interface, used to open and play audio streams
*/
class AudioOut {
public:
/// Opens a new audio stream
StreamPtr OpenStream(Core::Timing::CoreTiming& core_timing, u32 sample_rate, u32 num_channels,
std::string&& name, Stream::ReleaseCallback&& release_callback);
/// Returns a vector of recently released buffers specified by tag for the specified stream
std::vector<Buffer::Tag> GetTagsAndReleaseBuffers(StreamPtr stream, std::size_t max_count);
/// Returns a vector of all recently released buffers specified by tag for the specified stream
std::vector<Buffer::Tag> GetTagsAndReleaseBuffers(StreamPtr stream);
/// Starts an audio stream for playback
void StartStream(StreamPtr stream);
/// Stops an audio stream that is currently playing
void StopStream(StreamPtr stream);
/// Queues a buffer into the specified audio stream, returns true on success
bool QueueBuffer(StreamPtr stream, Buffer::Tag tag, std::vector<s16>&& data);
private:
SinkPtr sink;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_core.h"
#include "audio_core/audio_manager.h"
#include "audio_core/audio_out_manager.h"
#include "audio_core/out/audio_out.h"
#include "core/core.h"
#include "core/hle/kernel/k_event.h"
#include "core/hle/service/audio/errors.h"
namespace AudioCore::AudioOut {
Manager::Manager(Core::System& system_) : system{system_} {
std::iota(session_ids.begin(), session_ids.end(), 0);
num_free_sessions = MaxOutSessions;
}
Result Manager::AcquireSessionId(size_t& session_id) {
if (num_free_sessions == 0) {
LOG_ERROR(Service_Audio, "All 12 Audio Out sessions are in use, cannot create any more");
return Service::Audio::ERR_MAXIMUM_SESSIONS_REACHED;
}
session_id = session_ids[next_session_id];
next_session_id = (next_session_id + 1) % MaxOutSessions;
num_free_sessions--;
return ResultSuccess;
}
void Manager::ReleaseSessionId(const size_t session_id) {
std::scoped_lock l{mutex};
LOG_DEBUG(Service_Audio, "Freeing AudioOut session {}", session_id);
session_ids[free_session_id] = session_id;
num_free_sessions++;
free_session_id = (free_session_id + 1) % MaxOutSessions;
sessions[session_id].reset();
applet_resource_user_ids[session_id] = 0;
}
Result Manager::LinkToManager() {
std::scoped_lock l{mutex};
if (!linked_to_manager) {
AudioManager& manager{system.AudioCore().GetAudioManager()};
manager.SetOutManager(std::bind(&Manager::BufferReleaseAndRegister, this));
linked_to_manager = true;
}
return ResultSuccess;
}
void Manager::Start() {
if (sessions_started) {
return;
}
std::scoped_lock l{mutex};
for (auto& session : sessions) {
if (session) {
session->StartSession();
}
}
sessions_started = true;
}
void Manager::BufferReleaseAndRegister() {
std::scoped_lock l{mutex};
for (auto& session : sessions) {
if (session != nullptr) {
session->ReleaseAndRegisterBuffers();
}
}
}
u32 Manager::GetAudioOutDeviceNames(
std::vector<AudioRenderer::AudioDevice::AudioDeviceName>& names) const {
names.push_back({"DeviceOut"});
return 1;
}
} // namespace AudioCore::AudioOut

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <mutex>
#include "audio_core/renderer/audio_device.h"
namespace Core {
class System;
}
namespace AudioCore::AudioOut {
class Out;
constexpr size_t MaxOutSessions = 12;
/**
* Manages all audio out sessions.
*/
class Manager {
public:
explicit Manager(Core::System& system);
/**
* Acquire a free session id for opening a new audio out.
*
* @param session_id - Output session_id.
* @return Result code.
*/
Result AcquireSessionId(size_t& session_id);
/**
* Release a session id on close.
*
* @param session_id - Session id to free.
*/
void ReleaseSessionId(size_t session_id);
/**
* Link this manager to the main audio manager.
*
* @return Result code.
*/
Result LinkToManager();
/**
* Start the audio out manager.
*/
void Start();
/**
* Callback function, called by the audio manager when the audio out event is signalled.
*/
void BufferReleaseAndRegister();
/**
* Get a list of audio out device names.
*
* @oaram names - Output container to write names to.
* @return Number of names written.
*/
u32 GetAudioOutDeviceNames(
std::vector<AudioRenderer::AudioDevice::AudioDeviceName>& names) const;
/// Core system
Core::System& system;
/// Array of session ids
std::array<size_t, MaxOutSessions> session_ids{};
/// Array of resource user ids
std::array<size_t, MaxOutSessions> applet_resource_user_ids{};
/// Pointer to each open session
std::array<std::shared_ptr<Out>, MaxOutSessions> sessions{};
/// The number of free sessions
size_t num_free_sessions{};
/// The next session id to be taken
size_t next_session_id{};
/// The next session id to be freed
size_t free_session_id{};
/// Whether this is linked to the audio manager
bool linked_to_manager{};
/// Whether the sessions have been started
bool sessions_started{};
/// Protect state due to audio manager callback
std::recursive_mutex mutex{};
};
} // namespace AudioCore::AudioOut

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_render_manager.h"
#include "audio_core/common/audio_renderer_parameter.h"
#include "audio_core/common/feature_support.h"
#include "core/core.h"
namespace AudioCore::AudioRenderer {
Manager::Manager(Core::System& system_)
: system{system_}, system_manager{std::make_unique<SystemManager>(system)} {
std::iota(session_ids.begin(), session_ids.end(), 0);
}
Manager::~Manager() {
Stop();
}
void Manager::Stop() {
system_manager->Stop();
}
SystemManager& Manager::GetSystemManager() {
return *system_manager;
}
auto Manager::GetWorkBufferSize(const AudioRendererParameterInternal& params, u64& out_count)
-> Result {
if (!CheckValidRevision(params.revision)) {
return Service::Audio::ERR_INVALID_REVISION;
}
out_count = System::GetWorkBufferSize(params);
return ResultSuccess;
}
s32 Manager::GetSessionId() {
std::scoped_lock l{session_lock};
auto session_id{session_ids[session_count]};
if (session_id == -1) {
return -1;
}
session_ids[session_count] = -1;
session_count++;
return session_id;
}
void Manager::ReleaseSessionId(const s32 session_id) {
std::scoped_lock l{session_lock};
session_ids[--session_count] = session_id;
}
u32 Manager::GetSessionCount() {
std::scoped_lock l{session_lock};
return session_count;
}
bool Manager::AddSystem(System& system_) {
return system_manager->Add(system_);
}
bool Manager::RemoveSystem(System& system_) {
return system_manager->Remove(system_);
}
} // 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 <array>
#include <memory>
#include <mutex>
#include "audio_core/common/common.h"
#include "audio_core/renderer/system_manager.h"
#include "core/hle/service/audio/errors.h"
namespace Core {
class System;
}
namespace AudioCore {
struct AudioRendererParameterInternal;
namespace AudioRenderer {
/**
* Wrapper for the audio system manager, handles service calls.
*/
class Manager {
public:
explicit Manager(Core::System& system);
~Manager();
/**
* Stop the manager.
*/
void Stop();
/**
* Get the system manager.
*
* @return The system manager.
*/
SystemManager& GetSystemManager();
/**
* Get required size for the audio renderer workbuffer.
*
* @param params - Input parameters with the numbers of voices/mixes/sinks etc.
* @param out_count - Output size of the required workbuffer.
* @return Result code.
*/
Result GetWorkBufferSize(const AudioRendererParameterInternal& params, u64& out_count);
/**
* Get a new session id.
*
* @return The new session id. -1 if invalid, otherwise 0-MaxRendererSessions.
*/
s32 GetSessionId();
/**
* Get the number of currently active sessions.
*
* @return The number of active sessions.
*/
u32 GetSessionCount();
/**
* Add a renderer system to the manager.
* The system will be reguarly called to generate commands for the AudioRenderer.
*
* @param system - The system to add.
* @return True if the system was sucessfully added, otherwise false.
*/
bool AddSystem(System& system);
/**
* Remove a renderer system from the manager.
*
* @param system - The system to remove.
* @return True if the system was sucessfully removed, otherwise false.
*/
bool RemoveSystem(System& system);
/**
* Free a session id when the system wants to shut down.
*
* @param session_id - The session id to free.
*/
void ReleaseSessionId(s32 session_id);
private:
/// Core system
Core::System& system;
/// Session ids, -1 when in use
std::array<s32, MaxRendererSessions> session_ids{};
/// Number of active renderers
u32 session_count{};
/// Lock for interacting with the sessions
std::mutex session_lock{};
/// Regularly generates commands from the registered systems for the AudioRenderer
std::unique_ptr<SystemManager> system_manager{};
};
} // namespace AudioRenderer
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <limits>
#include <optional>
#include <vector>
#include "audio_core/audio_out.h"
#include "audio_core/audio_renderer.h"
#include "audio_core/common.h"
#include "audio_core/info_updater.h"
#include "audio_core/voice_context.h"
#include "common/logging/log.h"
#include "common/settings.h"
#include "core/core_timing.h"
#include "core/memory.h"
namespace {
[[nodiscard]] static constexpr s16 ClampToS16(s32 value) {
return static_cast<s16>(std::clamp(value, s32{std::numeric_limits<s16>::min()},
s32{std::numeric_limits<s16>::max()}));
}
[[nodiscard]] static constexpr s16 Mix2To1(s16 l_channel, s16 r_channel) {
// Mix 50% from left and 50% from right channel
constexpr float l_mix_amount = 50.0f / 100.0f;
constexpr float r_mix_amount = 50.0f / 100.0f;
return ClampToS16(static_cast<s32>((static_cast<float>(l_channel) * l_mix_amount) +
(static_cast<float>(r_channel) * r_mix_amount)));
}
[[maybe_unused, nodiscard]] static constexpr std::tuple<s16, s16> Mix6To2(
s16 fl_channel, s16 fr_channel, s16 fc_channel, [[maybe_unused]] s16 lf_channel, s16 bl_channel,
s16 br_channel) {
// Front channels are mixed 36.94%, Center channels are mixed to be 26.12% & the back channels
// are mixed to be 36.94%
constexpr float front_mix_amount = 36.94f / 100.0f;
constexpr float center_mix_amount = 26.12f / 100.0f;
constexpr float back_mix_amount = 36.94f / 100.0f;
// Mix 50% from left and 50% from right channel
const auto left = front_mix_amount * static_cast<float>(fl_channel) +
center_mix_amount * static_cast<float>(fc_channel) +
back_mix_amount * static_cast<float>(bl_channel);
const auto right = front_mix_amount * static_cast<float>(fr_channel) +
center_mix_amount * static_cast<float>(fc_channel) +
back_mix_amount * static_cast<float>(br_channel);
return {ClampToS16(static_cast<s32>(left)), ClampToS16(static_cast<s32>(right))};
}
[[nodiscard]] static constexpr std::tuple<s16, s16> Mix6To2WithCoefficients(
s16 fl_channel, s16 fr_channel, s16 fc_channel, s16 lf_channel, s16 bl_channel, s16 br_channel,
const std::array<float_le, 4>& coeff) {
const auto left =
static_cast<float>(fl_channel) * coeff[0] + static_cast<float>(fc_channel) * coeff[1] +
static_cast<float>(lf_channel) * coeff[2] + static_cast<float>(bl_channel) * coeff[3];
const auto right =
static_cast<float>(fr_channel) * coeff[0] + static_cast<float>(fc_channel) * coeff[1] +
static_cast<float>(lf_channel) * coeff[2] + static_cast<float>(br_channel) * coeff[3];
return {ClampToS16(static_cast<s32>(left)), ClampToS16(static_cast<s32>(right))};
}
} // namespace
namespace AudioCore {
constexpr s32 NUM_BUFFERS = 2;
AudioRenderer::AudioRenderer(Core::Timing::CoreTiming& core_timing_, Core::Memory::Memory& memory_,
AudioCommon::AudioRendererParameter params,
Stream::ReleaseCallback&& release_callback,
std::size_t instance_number)
: worker_params{params}, memory_pool_info(params.effect_count + params.voice_count * 4),
voice_context(params.voice_count), effect_context(params.effect_count), mix_context(),
sink_context(params.sink_count), splitter_context(),
voices(params.voice_count), memory{memory_},
command_generator(worker_params, voice_context, mix_context, splitter_context, effect_context,
memory),
core_timing{core_timing_} {
behavior_info.SetUserRevision(params.revision);
splitter_context.Initialize(behavior_info, params.splitter_count,
params.num_splitter_send_channels);
mix_context.Initialize(behavior_info, params.submix_count + 1, params.effect_count);
audio_out = std::make_unique<AudioCore::AudioOut>();
stream = audio_out->OpenStream(
core_timing, params.sample_rate, AudioCommon::STREAM_NUM_CHANNELS,
fmt::format("AudioRenderer-Instance{}", instance_number), std::move(release_callback));
process_event =
Core::Timing::CreateEvent(fmt::format("AudioRenderer-Instance{}-Process", instance_number),
[this](std::uintptr_t, s64, std::chrono::nanoseconds) {
ReleaseAndQueueBuffers();
return std::nullopt;
});
for (s32 i = 0; i < NUM_BUFFERS; ++i) {
QueueMixedBuffer(i);
}
}
AudioRenderer::~AudioRenderer() = default;
Result AudioRenderer::Start() {
audio_out->StartStream(stream);
ReleaseAndQueueBuffers();
return ResultSuccess;
}
Result AudioRenderer::Stop() {
audio_out->StopStream(stream);
return ResultSuccess;
}
u32 AudioRenderer::GetSampleRate() const {
return worker_params.sample_rate;
}
u32 AudioRenderer::GetSampleCount() const {
return worker_params.sample_count;
}
u32 AudioRenderer::GetMixBufferCount() const {
return worker_params.mix_buffer_count;
}
Stream::State AudioRenderer::GetStreamState() const {
return stream->GetState();
}
Result AudioRenderer::UpdateAudioRenderer(const std::vector<u8>& input_params,
std::vector<u8>& output_params) {
std::scoped_lock lock{mutex};
InfoUpdater info_updater{input_params, output_params, behavior_info};
if (!info_updater.UpdateBehaviorInfo(behavior_info)) {
LOG_ERROR(Audio, "Failed to update behavior info input parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (!info_updater.UpdateMemoryPools(memory_pool_info)) {
LOG_ERROR(Audio, "Failed to update memory pool parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (!info_updater.UpdateVoiceChannelResources(voice_context)) {
LOG_ERROR(Audio, "Failed to update voice channel resource parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (!info_updater.UpdateVoices(voice_context, memory_pool_info, 0)) {
LOG_ERROR(Audio, "Failed to update voice parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
// TODO(ogniK): Deal with stopped audio renderer but updates still taking place
if (!info_updater.UpdateEffects(effect_context, true)) {
LOG_ERROR(Audio, "Failed to update effect parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (behavior_info.IsSplitterSupported()) {
if (!info_updater.UpdateSplitterInfo(splitter_context)) {
LOG_ERROR(Audio, "Failed to update splitter parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
}
const auto mix_result = info_updater.UpdateMixes(mix_context, worker_params.mix_buffer_count,
splitter_context, effect_context);
if (mix_result.IsError()) {
LOG_ERROR(Audio, "Failed to update mix parameters");
return mix_result;
}
// TODO(ogniK): Sinks
if (!info_updater.UpdateSinks(sink_context)) {
LOG_ERROR(Audio, "Failed to update sink parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
// TODO(ogniK): Performance buffer
if (!info_updater.UpdatePerformanceBuffer()) {
LOG_ERROR(Audio, "Failed to update performance buffer parameters");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (!info_updater.UpdateErrorInfo(behavior_info)) {
LOG_ERROR(Audio, "Failed to update error info");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (behavior_info.IsElapsedFrameCountSupported()) {
if (!info_updater.UpdateRendererInfo(elapsed_frame_count)) {
LOG_ERROR(Audio, "Failed to update renderer info");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
}
// TODO(ogniK): Statistics
if (!info_updater.WriteOutputHeader()) {
LOG_ERROR(Audio, "Failed to write output header");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
// TODO(ogniK): Check when all sections are implemented
if (!info_updater.CheckConsumedSize()) {
LOG_ERROR(Audio, "Audio buffers were not consumed!");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
return ResultSuccess;
}
void AudioRenderer::QueueMixedBuffer(Buffer::Tag tag) {
command_generator.PreCommand();
// Clear mix buffers before our next operation
command_generator.ClearMixBuffers();
// If the splitter is not in use, sort our mixes
if (!splitter_context.UsingSplitter()) {
mix_context.SortInfo();
}
// Sort our voices
voice_context.SortInfo();
// Handle samples
command_generator.GenerateVoiceCommands();
command_generator.GenerateSubMixCommands();
command_generator.GenerateFinalMixCommands();
command_generator.PostCommand();
// Base sample size
std::size_t BUFFER_SIZE{worker_params.sample_count};
// Samples, making sure to clear
std::vector<s16> buffer(BUFFER_SIZE * stream->GetNumChannels(), 0);
if (sink_context.InUse()) {
const auto stream_channel_count = stream->GetNumChannels();
const auto buffer_offsets = sink_context.OutputBuffers();
const auto channel_count = buffer_offsets.size();
const auto& final_mix = mix_context.GetFinalMixInfo();
const auto& in_params = final_mix.GetInParams();
std::vector<std::span<s32>> mix_buffers(channel_count);
for (std::size_t i = 0; i < channel_count; i++) {
mix_buffers[i] =
command_generator.GetMixBuffer(in_params.buffer_offset + buffer_offsets[i]);
}
for (std::size_t i = 0; i < BUFFER_SIZE; i++) {
if (channel_count == 1) {
const auto sample = ClampToS16(mix_buffers[0][i]);
// Place sample in all channels
for (u32 channel = 0; channel < stream_channel_count; channel++) {
buffer[i * stream_channel_count + channel] = sample;
}
if (stream_channel_count == 6) {
// Output stream has a LF channel, mute it!
buffer[i * stream_channel_count + 3] = 0;
}
} else if (channel_count == 2) {
const auto l_sample = ClampToS16(mix_buffers[0][i]);
const auto r_sample = ClampToS16(mix_buffers[1][i]);
if (stream_channel_count == 1) {
buffer[i * stream_channel_count + 0] = Mix2To1(l_sample, r_sample);
} else if (stream_channel_count == 2) {
buffer[i * stream_channel_count + 0] = l_sample;
buffer[i * stream_channel_count + 1] = r_sample;
} else if (stream_channel_count == 6) {
buffer[i * stream_channel_count + 0] = l_sample;
buffer[i * stream_channel_count + 1] = r_sample;
// Combine both left and right channels to the center channel
buffer[i * stream_channel_count + 2] = Mix2To1(l_sample, r_sample);
buffer[i * stream_channel_count + 4] = l_sample;
buffer[i * stream_channel_count + 5] = r_sample;
}
} else if (channel_count == 6) {
const auto fl_sample = ClampToS16(mix_buffers[0][i]);
const auto fr_sample = ClampToS16(mix_buffers[1][i]);
const auto fc_sample = ClampToS16(mix_buffers[2][i]);
const auto lf_sample = ClampToS16(mix_buffers[3][i]);
const auto bl_sample = ClampToS16(mix_buffers[4][i]);
const auto br_sample = ClampToS16(mix_buffers[5][i]);
if (stream_channel_count == 1) {
// Games seem to ignore the center channel half the time, we use the front left
// and right channel for mixing as that's where majority of the audio goes
buffer[i * stream_channel_count + 0] = Mix2To1(fl_sample, fr_sample);
} else if (stream_channel_count == 2) {
// Mix all channels into 2 channels
const auto [left, right] = Mix6To2WithCoefficients(
fl_sample, fr_sample, fc_sample, lf_sample, bl_sample, br_sample,
sink_context.GetDownmixCoefficients());
buffer[i * stream_channel_count + 0] = left;
buffer[i * stream_channel_count + 1] = right;
} else if (stream_channel_count == 6) {
// Pass through
buffer[i * stream_channel_count + 0] = fl_sample;
buffer[i * stream_channel_count + 1] = fr_sample;
buffer[i * stream_channel_count + 2] = fc_sample;
buffer[i * stream_channel_count + 3] = lf_sample;
buffer[i * stream_channel_count + 4] = bl_sample;
buffer[i * stream_channel_count + 5] = br_sample;
}
}
}
}
audio_out->QueueBuffer(stream, tag, std::move(buffer));
elapsed_frame_count++;
voice_context.UpdateStateByDspShared();
}
void AudioRenderer::ReleaseAndQueueBuffers() {
if (!stream->IsPlaying()) {
return;
}
{
std::scoped_lock lock{mutex};
const auto released_buffers{audio_out->GetTagsAndReleaseBuffers(stream)};
for (const auto& tag : released_buffers) {
QueueMixedBuffer(tag);
}
}
const f32 sample_rate = static_cast<f32>(GetSampleRate());
const f32 sample_count = static_cast<f32>(GetSampleCount());
const f32 consume_rate = sample_rate / (sample_count * (sample_count / 240));
const s32 ms = (1000 / static_cast<s32>(consume_rate)) - 1;
const std::chrono::milliseconds next_event_time(std::max(ms / NUM_BUFFERS, 1));
core_timing.ScheduleEvent(next_event_time, process_event, {});
}
} // namespace AudioCore

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src/audio_core/audio_renderer.h
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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <memory>
#include <mutex>
#include <vector>
#include "audio_core/behavior_info.h"
#include "audio_core/command_generator.h"
#include "audio_core/common.h"
#include "audio_core/effect_context.h"
#include "audio_core/memory_pool.h"
#include "audio_core/mix_context.h"
#include "audio_core/sink_context.h"
#include "audio_core/splitter_context.h"
#include "audio_core/stream.h"
#include "audio_core/voice_context.h"
#include "common/common_funcs.h"
#include "common/common_types.h"
#include "common/swap.h"
#include "core/hle/result.h"
namespace Core::Timing {
class CoreTiming;
}
namespace Core::Memory {
class Memory;
}
namespace AudioCore {
using DSPStateHolder = std::array<VoiceState*, AudioCommon::MAX_CHANNEL_COUNT>;
class AudioOut;
class AudioRenderer {
public:
AudioRenderer(Core::Timing::CoreTiming& core_timing, Core::Memory::Memory& memory_,
AudioCommon::AudioRendererParameter params,
Stream::ReleaseCallback&& release_callback, std::size_t instance_number);
~AudioRenderer();
[[nodiscard]] Result UpdateAudioRenderer(const std::vector<u8>& input_params,
std::vector<u8>& output_params);
[[nodiscard]] Result Start();
[[nodiscard]] Result Stop();
void QueueMixedBuffer(Buffer::Tag tag);
void ReleaseAndQueueBuffers();
[[nodiscard]] u32 GetSampleRate() const;
[[nodiscard]] u32 GetSampleCount() const;
[[nodiscard]] u32 GetMixBufferCount() const;
[[nodiscard]] Stream::State GetStreamState() const;
private:
BehaviorInfo behavior_info{};
AudioCommon::AudioRendererParameter worker_params;
std::vector<ServerMemoryPoolInfo> memory_pool_info;
VoiceContext voice_context;
EffectContext effect_context;
MixContext mix_context;
SinkContext sink_context;
SplitterContext splitter_context;
std::vector<VoiceState> voices;
std::unique_ptr<AudioOut> audio_out;
StreamPtr stream;
Core::Memory::Memory& memory;
CommandGenerator command_generator;
std::size_t elapsed_frame_count{};
Core::Timing::CoreTiming& core_timing;
std::shared_ptr<Core::Timing::EventType> process_event;
std::mutex mutex;
};
} // namespace AudioCore

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src/audio_core/behavior_info.cpp
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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <cstring>
#include "audio_core/behavior_info.h"
#include "audio_core/common.h"
#include "common/logging/log.h"
namespace AudioCore {
BehaviorInfo::BehaviorInfo() : process_revision(AudioCommon::CURRENT_PROCESS_REVISION) {}
BehaviorInfo::~BehaviorInfo() = default;
bool BehaviorInfo::UpdateOutput(std::vector<u8>& buffer, std::size_t offset) {
if (!AudioCommon::CanConsumeBuffer(buffer.size(), offset, sizeof(OutParams))) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
OutParams params{};
std::memcpy(params.errors.data(), errors.data(), sizeof(ErrorInfo) * errors.size());
params.error_count = static_cast<u32_le>(error_count);
std::memcpy(buffer.data() + offset, &params, sizeof(OutParams));
return true;
}
void BehaviorInfo::ClearError() {
error_count = 0;
}
void BehaviorInfo::UpdateFlags(u64_le dest_flags) {
flags = dest_flags;
}
void BehaviorInfo::SetUserRevision(u32_le revision) {
user_revision = revision;
}
u32_le BehaviorInfo::GetUserRevision() const {
return user_revision;
}
u32_le BehaviorInfo::GetProcessRevision() const {
return process_revision;
}
bool BehaviorInfo::IsAdpcmLoopContextBugFixed() const {
return AudioCommon::IsRevisionSupported(2, user_revision);
}
bool BehaviorInfo::IsSplitterSupported() const {
return AudioCommon::IsRevisionSupported(2, user_revision);
}
bool BehaviorInfo::IsLongSizePreDelaySupported() const {
return AudioCommon::IsRevisionSupported(3, user_revision);
}
bool BehaviorInfo::IsAudioRendererProcessingTimeLimit80PercentSupported() const {
return AudioCommon::IsRevisionSupported(5, user_revision);
}
bool BehaviorInfo::IsAudioRendererProcessingTimeLimit75PercentSupported() const {
return AudioCommon::IsRevisionSupported(4, user_revision);
}
bool BehaviorInfo::IsAudioRendererProcessingTimeLimit70PercentSupported() const {
return AudioCommon::IsRevisionSupported(1, user_revision);
}
bool BehaviorInfo::IsElapsedFrameCountSupported() const {
return AudioCommon::IsRevisionSupported(5, user_revision);
}
bool BehaviorInfo::IsMemoryPoolForceMappingEnabled() const {
return (flags & 1) != 0;
}
bool BehaviorInfo::IsFlushVoiceWaveBuffersSupported() const {
return AudioCommon::IsRevisionSupported(5, user_revision);
}
bool BehaviorInfo::IsVoicePlayedSampleCountResetAtLoopPointSupported() const {
return AudioCommon::IsRevisionSupported(5, user_revision);
}
bool BehaviorInfo::IsVoicePitchAndSrcSkippedSupported() const {
return AudioCommon::IsRevisionSupported(5, user_revision);
}
bool BehaviorInfo::IsMixInParameterDirtyOnlyUpdateSupported() const {
return AudioCommon::IsRevisionSupported(7, user_revision);
}
bool BehaviorInfo::IsSplitterBugFixed() const {
return AudioCommon::IsRevisionSupported(5, user_revision);
}
void BehaviorInfo::CopyErrorInfo(BehaviorInfo::OutParams& dst) {
dst.error_count = static_cast<u32>(error_count);
std::copy(errors.begin(), errors.begin() + error_count, dst.errors.begin());
}
} // namespace AudioCore

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src/audio_core/behavior_info.h
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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <vector>
#include "common/common_funcs.h"
#include "common/common_types.h"
#include "common/swap.h"
namespace AudioCore {
class BehaviorInfo {
public:
struct ErrorInfo {
u32_le result{};
INSERT_PADDING_WORDS(1);
u64_le result_info{};
};
static_assert(sizeof(ErrorInfo) == 0x10, "ErrorInfo is an invalid size");
struct InParams {
u32_le revision{};
u32_le padding{};
u64_le flags{};
};
static_assert(sizeof(InParams) == 0x10, "InParams is an invalid size");
struct OutParams {
std::array<ErrorInfo, 10> errors{};
u32_le error_count{};
INSERT_PADDING_BYTES(12);
};
static_assert(sizeof(OutParams) == 0xb0, "OutParams is an invalid size");
explicit BehaviorInfo();
~BehaviorInfo();
bool UpdateOutput(std::vector<u8>& buffer, std::size_t offset);
void ClearError();
void UpdateFlags(u64_le dest_flags);
void SetUserRevision(u32_le revision);
[[nodiscard]] u32_le GetUserRevision() const;
[[nodiscard]] u32_le GetProcessRevision() const;
[[nodiscard]] bool IsAdpcmLoopContextBugFixed() const;
[[nodiscard]] bool IsSplitterSupported() const;
[[nodiscard]] bool IsLongSizePreDelaySupported() const;
[[nodiscard]] bool IsAudioRendererProcessingTimeLimit80PercentSupported() const;
[[nodiscard]] bool IsAudioRendererProcessingTimeLimit75PercentSupported() const;
[[nodiscard]] bool IsAudioRendererProcessingTimeLimit70PercentSupported() const;
[[nodiscard]] bool IsElapsedFrameCountSupported() const;
[[nodiscard]] bool IsMemoryPoolForceMappingEnabled() const;
[[nodiscard]] bool IsFlushVoiceWaveBuffersSupported() const;
[[nodiscard]] bool IsVoicePlayedSampleCountResetAtLoopPointSupported() const;
[[nodiscard]] bool IsVoicePitchAndSrcSkippedSupported() const;
[[nodiscard]] bool IsMixInParameterDirtyOnlyUpdateSupported() const;
[[nodiscard]] bool IsSplitterBugFixed() const;
void CopyErrorInfo(OutParams& dst);
private:
u32_le process_revision{};
u32_le user_revision{};
u64_le flags{};
std::array<ErrorInfo, 10> errors{};
std::size_t error_count{};
};
} // namespace AudioCore

44
src/audio_core/buffer.h
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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <memory>
#include <vector>
#include "common/common_types.h"
namespace AudioCore {
/**
* Represents a buffer of audio samples to be played in an audio stream
*/
class Buffer {
public:
using Tag = u64;
Buffer(Tag tag_, std::vector<s16>&& samples_) : tag{tag_}, samples{std::move(samples_)} {}
/// Returns the raw audio data for the buffer
std::vector<s16>& GetSamples() {
return samples;
}
/// Returns the raw audio data for the buffer
const std::vector<s16>& GetSamples() const {
return samples;
}
/// Returns the buffer tag, this is provided by the game to the audout service
Tag GetTag() const {
return tag;
}
private:
Tag tag;
std::vector<s16> samples;
};
using BufferPtr = std::shared_ptr<Buffer>;
} // namespace AudioCore

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src/audio_core/codec.cpp
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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include "audio_core/codec.h"
namespace AudioCore::Codec {
std::vector<s16> DecodeADPCM(const u8* const data, std::size_t size, const ADPCM_Coeff& coeff,
ADPCMState& state) {
// GC-ADPCM with scale factor and variable coefficients.
// Frames are 8 bytes long containing 14 samples each.
// Samples are 4 bits (one nibble) long.
constexpr std::size_t FRAME_LEN = 8;
constexpr std::size_t SAMPLES_PER_FRAME = 14;
static constexpr std::array<int, 16> SIGNED_NIBBLES{
0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1,
};
const std::size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
const std::size_t ret_size =
sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
std::vector<s16> ret(ret_size);
int yn1 = state.yn1, yn2 = state.yn2;
const std::size_t NUM_FRAMES =
(sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
for (std::size_t framei = 0; framei < NUM_FRAMES; framei++) {
const int frame_header = data[framei * FRAME_LEN];
const int scale = 1 << (frame_header & 0xF);
const int idx = (frame_header >> 4) & 0x7;
// Coefficients are fixed point with 11 bits fractional part.
const int coef1 = coeff[idx * 2 + 0];
const int coef2 = coeff[idx * 2 + 1];
// Decodes an audio sample. One nibble produces one sample.
const auto decode_sample = [&](const int nibble) -> s16 {
const int xn = nibble * scale;
// We first transform everything into 11 bit fixed point, perform the second order
// digital filter, then transform back.
// 0x400 == 0.5 in 11 bit fixed point.
// Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
// Clamp to output range.
val = std::clamp<s32>(val, -32768, 32767);
// Advance output feedback.
yn2 = yn1;
yn1 = val;
return static_cast<s16>(val);
};
std::size_t outputi = framei * SAMPLES_PER_FRAME;
std::size_t datai = framei * FRAME_LEN + 1;
for (std::size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
ret[outputi] = sample1;
outputi++;
const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]);
ret[outputi] = sample2;
outputi++;
datai++;
}
}
state.yn1 = static_cast<s16>(yn1);
state.yn2 = static_cast<s16>(yn2);
return ret;
}
} // namespace AudioCore::Codec

43
src/audio_core/codec.h
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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <vector>
#include "common/common_types.h"
namespace AudioCore::Codec {
enum class PcmFormat : u32 {
Invalid = 0,
Int8 = 1,
Int16 = 2,
Int24 = 3,
Int32 = 4,
PcmFloat = 5,
Adpcm = 6,
};
/// See: Codec::DecodeADPCM
struct ADPCMState {
// Two historical samples from previous processed buffer,
// required for ADPCM decoding
s16 yn1; ///< y[n-1]
s16 yn2; ///< y[n-2]
};
using ADPCM_Coeff = std::array<s16, 16>;
/**
* @param data Pointer to buffer that contains ADPCM data to decode
* @param size Size of buffer in bytes
* @param coeff ADPCM coefficients
* @param state ADPCM state, this is updated with new state
* @return Decoded stereo signed PCM16 data, sample_count in length
*/
std::vector<s16> DecodeADPCM(const u8* data, std::size_t size, const ADPCM_Coeff& coeff,
ADPCMState& state);
}; // namespace AudioCore::Codec

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src/audio_core/command_generator.cpp
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src/audio_core/command_generator.h
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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <span>
#include "audio_core/common.h"
#include "audio_core/voice_context.h"
#include "common/common_types.h"
namespace Core::Memory {
class Memory;
}
namespace AudioCore {
class MixContext;
class SplitterContext;
class ServerSplitterDestinationData;
class ServerMixInfo;
class EffectContext;
class EffectBase;
struct AuxInfoDSP;
struct I3dl2ReverbParams;
struct I3dl2ReverbState;
using MixVolumeBuffer = std::array<float, AudioCommon::MAX_MIX_BUFFERS>;
class CommandGenerator {
public:
explicit CommandGenerator(AudioCommon::AudioRendererParameter& worker_params_,
VoiceContext& voice_context_, MixContext& mix_context_,
SplitterContext& splitter_context_, EffectContext& effect_context_,
Core::Memory::Memory& memory_);
~CommandGenerator();
void ClearMixBuffers();
void GenerateVoiceCommands();
void GenerateVoiceCommand(ServerVoiceInfo& voice_info);
void GenerateSubMixCommands();
void GenerateFinalMixCommands();
void PreCommand();
void PostCommand();
[[nodiscard]] std::span<s32> GetChannelMixBuffer(s32 channel);
[[nodiscard]] std::span<const s32> GetChannelMixBuffer(s32 channel) const;
[[nodiscard]] std::span<s32> GetMixBuffer(std::size_t index);
[[nodiscard]] std::span<const s32> GetMixBuffer(std::size_t index) const;
[[nodiscard]] std::size_t GetMixChannelBufferOffset(s32 channel) const;
[[nodiscard]] std::size_t GetTotalMixBufferCount() const;
private:
void GenerateDataSourceCommand(ServerVoiceInfo& voice_info, VoiceState& dsp_state, s32 channel);
void GenerateBiquadFilterCommandForVoice(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
s32 mix_buffer_count, s32 channel);
void GenerateVolumeRampCommand(float last_volume, float current_volume, s32 channel,
s32 node_id);
void GenerateVoiceMixCommand(const MixVolumeBuffer& mix_volumes,
const MixVolumeBuffer& last_mix_volumes, VoiceState& dsp_state,
s32 mix_buffer_offset, s32 mix_buffer_count, s32 voice_index,
s32 node_id);
void GenerateSubMixCommand(ServerMixInfo& mix_info);
void GenerateMixCommands(ServerMixInfo& mix_info);
void GenerateMixCommand(std::size_t output_offset, std::size_t input_offset, float volume,
s32 node_id);
void GenerateFinalMixCommand();
void GenerateBiquadFilterCommand(s32 mix_buffer, const BiquadFilterParameter& params,
std::array<s64, 2>& state, std::size_t input_offset,
std::size_t output_offset, s32 sample_count, s32 node_id);
void GenerateDepopPrepareCommand(VoiceState& dsp_state, std::size_t mix_buffer_count,
std::size_t mix_buffer_offset);
void GenerateDepopForMixBuffersCommand(std::size_t mix_buffer_count,
std::size_t mix_buffer_offset, s32 sample_rate);
void GenerateEffectCommand(ServerMixInfo& mix_info);
void GenerateI3dl2ReverbEffectCommand(s32 mix_buffer_offset, EffectBase* info, bool enabled);
void GenerateBiquadFilterEffectCommand(s32 mix_buffer_offset, EffectBase* info, bool enabled);
void GenerateAuxCommand(s32 mix_buffer_offset, EffectBase* info, bool enabled);
[[nodiscard]] ServerSplitterDestinationData* GetDestinationData(s32 splitter_id, s32 index);
s32 WriteAuxBuffer(AuxInfoDSP& dsp_info, VAddr send_buffer, u32 max_samples,
std::span<const s32> data, u32 sample_count, u32 write_offset,
u32 write_count);
s32 ReadAuxBuffer(AuxInfoDSP& recv_info, VAddr recv_buffer, u32 max_samples,
std::span<s32> out_data, u32 sample_count, u32 read_offset, u32 read_count);
void InitializeI3dl2Reverb(I3dl2ReverbParams& info, I3dl2ReverbState& state,
std::vector<u8>& work_buffer);
void UpdateI3dl2Reverb(I3dl2ReverbParams& info, I3dl2ReverbState& state, bool should_clear);
// DSP Code
template <typename T>
s32 DecodePcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state, s32 sample_start_offset,
s32 sample_end_offset, s32 sample_count, s32 channel, std::size_t mix_offset);
s32 DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state, s32 sample_start_offset,
s32 sample_end_offset, s32 sample_count, s32 channel, std::size_t mix_offset);
void DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, std::span<s32> output,
VoiceState& dsp_state, s32 channel, s32 target_sample_rate,
s32 sample_count, s32 node_id);
AudioCommon::AudioRendererParameter& worker_params;
VoiceContext& voice_context;
MixContext& mix_context;
SplitterContext& splitter_context;
EffectContext& effect_context;
Core::Memory::Memory& memory;
std::vector<s32> mix_buffer{};
std::vector<s32> sample_buffer{};
std::vector<s32> depop_buffer{};
bool dumping_frame{false};
};
} // namespace AudioCore

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src/audio_core/common.h
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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "common/common_funcs.h"
#include "common/common_types.h"
#include "common/swap.h"
#include "core/hle/result.h"
namespace AudioCommon {
namespace Audren {
constexpr Result ERR_INVALID_PARAMETERS{ErrorModule::Audio, 41};
constexpr Result ERR_SPLITTER_SORT_FAILED{ErrorModule::Audio, 43};
} // namespace Audren
constexpr u8 BASE_REVISION = '0';
constexpr u32_le CURRENT_PROCESS_REVISION =
Common::MakeMagic('R', 'E', 'V', static_cast<u8>(BASE_REVISION + 0xA));
constexpr std::size_t MAX_MIX_BUFFERS = 24;
constexpr std::size_t MAX_BIQUAD_FILTERS = 2;
constexpr std::size_t MAX_CHANNEL_COUNT = 6;
constexpr std::size_t MAX_WAVE_BUFFERS = 4;
constexpr std::size_t MAX_SAMPLE_HISTORY = 4;
constexpr u32 STREAM_SAMPLE_RATE = 48000;
constexpr u32 STREAM_NUM_CHANNELS = 2;
constexpr s32 NO_SPLITTER = -1;
constexpr s32 NO_MIX = 0x7fffffff;
constexpr s32 NO_FINAL_MIX = std::numeric_limits<s32>::min();
constexpr s32 FINAL_MIX = 0;
constexpr s32 NO_EFFECT_ORDER = -1;
constexpr std::size_t TEMP_MIX_BASE_SIZE = 0x3f00; // TODO(ogniK): Work out this constant
// Any size checks seem to take the sample history into account
// and our const ends up being 0x3f04, the 4 bytes are most
// likely the sample history
constexpr std::size_t TOTAL_TEMP_MIX_SIZE = TEMP_MIX_BASE_SIZE + AudioCommon::MAX_SAMPLE_HISTORY;
constexpr f32 I3DL2REVERB_MAX_LEVEL = 5000.0f;
constexpr f32 I3DL2REVERB_MIN_REFLECTION_DURATION = 0.02f;
constexpr std::size_t I3DL2REVERB_TAPS = 20;
constexpr std::size_t I3DL2REVERB_DELAY_LINE_COUNT = 4;
using Fractional = s32;
template <typename T>
constexpr Fractional ToFractional(T x) {
return static_cast<Fractional>(x * static_cast<T>(0x4000));
}
constexpr Fractional MultiplyFractional(Fractional lhs, Fractional rhs) {
return static_cast<Fractional>(static_cast<s64>(lhs) * rhs >> 14);
}
constexpr s32 FractionalToFixed(Fractional x) {
const auto s = x & (1 << 13);
return static_cast<s32>(x >> 14) + s;
}
constexpr s32 CalculateDelaySamples(s32 sample_rate_khz, float time) {
return FractionalToFixed(MultiplyFractional(ToFractional(sample_rate_khz), ToFractional(time)));
}
static constexpr u32 VersionFromRevision(u32_le rev) {
// "REV7" -> 7
return ((rev >> 24) & 0xff) - 0x30;
}
static constexpr bool IsRevisionSupported(u32 required, u32_le user_revision) {
const auto base = VersionFromRevision(user_revision);
return required <= base;
}
static constexpr bool IsValidRevision(u32_le revision) {
const auto base = VersionFromRevision(revision);
constexpr auto max_rev = VersionFromRevision(CURRENT_PROCESS_REVISION);
return base <= max_rev;
}
static constexpr bool CanConsumeBuffer(std::size_t size, std::size_t offset, std::size_t required) {
if (offset > size) {
return false;
}
if (size < required) {
return false;
}
if ((size - offset) < required) {
return false;
}
return true;
}
struct UpdateDataSizes {
u32_le behavior{};
u32_le memory_pool{};
u32_le voice{};
u32_le voice_channel_resource{};
u32_le effect{};
u32_le mixer{};
u32_le sink{};
u32_le performance{};
u32_le splitter{};
u32_le render_info{};
INSERT_PADDING_WORDS(4);
};
static_assert(sizeof(UpdateDataSizes) == 0x38, "UpdateDataSizes is an invalid size");
struct UpdateDataHeader {
u32_le revision{};
UpdateDataSizes size{};
u32_le total_size{};
};
static_assert(sizeof(UpdateDataHeader) == 0x40, "UpdateDataHeader is an invalid size");
struct AudioRendererParameter {
u32_le sample_rate;
u32_le sample_count;
u32_le mix_buffer_count;
u32_le submix_count;
u32_le voice_count;
u32_le sink_count;
u32_le effect_count;
u32_le performance_frame_count;
u8 is_voice_drop_enabled;
u8 unknown_21;
u8 unknown_22;
u8 execution_mode;
u32_le splitter_count;
u32_le num_splitter_send_channels;
u32_le unknown_30;
u32_le revision;
};
static_assert(sizeof(AudioRendererParameter) == 52, "AudioRendererParameter is an invalid size");
} // namespace AudioCommon

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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/behavior/behavior_info.h"
#include "audio_core/renderer/memory/memory_pool_info.h"
#include "audio_core/renderer/upsampler/upsampler_manager.h"
#include "common/common_types.h"
namespace AudioCore {
/**
* Execution mode of the audio renderer.
* Only Auto is currently supported.
*/
enum class ExecutionMode : u8 {
Auto,
Manual,
};
/**
* Parameters from the game, passed to the audio renderer for initialisation.
*/
struct AudioRendererParameterInternal {
/* 0x00 */ u32 sample_rate;
/* 0x04 */ u32 sample_count;
/* 0x08 */ u32 mixes;
/* 0x0C */ u32 sub_mixes;
/* 0x10 */ u32 voices;
/* 0x14 */ u32 sinks;
/* 0x18 */ u32 effects;
/* 0x1C */ u32 perf_frames;
/* 0x20 */ u16 voice_drop_enabled;
/* 0x22 */ u8 rendering_device;
/* 0x23 */ ExecutionMode execution_mode;
/* 0x24 */ u32 splitter_infos;
/* 0x28 */ s32 splitter_destinations;
/* 0x2C */ u32 external_context_size;
/* 0x30 */ u32 revision;
/* 0x34 */ char unk34[0x4];
};
static_assert(sizeof(AudioRendererParameterInternal) == 0x38,
"AudioRendererParameterInternal has the wrong size!");
/**
* Context for rendering, contains a bunch of useful fields for the command generator.
*/
struct AudioRendererSystemContext {
s32 session_id;
s8 channels;
s16 mix_buffer_count;
AudioRenderer::BehaviorInfo* behavior;
std::span<s32> depop_buffer;
AudioRenderer::UpsamplerManager* upsampler_manager;
AudioRenderer::MemoryPoolInfo* memory_pool_info;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <numeric>
#include <span>
#include "common/assert.h"
#include "common/common_funcs.h"
#include "common/common_types.h"
namespace AudioCore {
using CpuAddr = std::uintptr_t;
enum class PlayState : u8 {
Started,
Stopped,
Paused,
};
enum class SrcQuality : u8 {
Medium,
High,
Low,
};
enum class SampleFormat : u8 {
Invalid,
PcmInt8,
PcmInt16,
PcmInt24,
PcmInt32,
PcmFloat,
Adpcm,
};
enum class SessionTypes {
AudioIn,
AudioOut,
FinalOutputRecorder,
};
enum class Channels : u32 {
FrontLeft,
FrontRight,
Center,
LFE,
BackLeft,
BackRight,
};
// These are used by Delay, Reverb and I3dl2Reverb prior to Revision 11.
enum class OldChannels : u32 {
FrontLeft,
FrontRight,
BackLeft,
BackRight,
Center,
LFE,
};
constexpr u32 BufferCount = 32;
constexpr u32 MaxRendererSessions = 2;
constexpr u32 TargetSampleCount = 240;
constexpr u32 TargetSampleRate = 48'000;
constexpr u32 MaxChannels = 6;
constexpr u32 MaxMixBuffers = 24;
constexpr u32 MaxWaveBuffers = 4;
constexpr s32 LowestVoicePriority = 0xFF;
constexpr s32 HighestVoicePriority = 0;
constexpr u32 BufferAlignment = 0x40;
constexpr u32 WorkbufferAlignment = 0x1000;
constexpr s32 FinalMixId = 0;
constexpr s32 InvalidDistanceFromFinalMix = std::numeric_limits<s32>::min();
constexpr s32 UnusedSplitterId = -1;
constexpr s32 UnusedMixId = std::numeric_limits<s32>::max();
constexpr u32 InvalidNodeId = 0xF0000000;
constexpr s32 InvalidProcessOrder = -1;
constexpr u32 MaxBiquadFilters = 2;
constexpr u32 MaxEffects = 256;
constexpr bool IsChannelCountValid(u16 channel_count) {
return channel_count <= 6 &&
(channel_count == 1 || channel_count == 2 || channel_count == 4 || channel_count == 6);
}
constexpr void UseOldChannelMapping(std::span<s16> inputs, std::span<s16> outputs) {
constexpr auto old_center{static_cast<u32>(OldChannels::Center)};
constexpr auto new_center{static_cast<u32>(Channels::Center)};
constexpr auto old_lfe{static_cast<u32>(OldChannels::LFE)};
constexpr auto new_lfe{static_cast<u32>(Channels::LFE)};
auto center{inputs[old_center]};
auto lfe{inputs[old_lfe]};
inputs[old_center] = inputs[new_center];
inputs[old_lfe] = inputs[new_lfe];
inputs[new_center] = center;
inputs[new_lfe] = lfe;
center = outputs[old_center];
lfe = outputs[old_lfe];
outputs[old_center] = outputs[new_center];
outputs[old_lfe] = outputs[new_lfe];
outputs[new_center] = center;
outputs[new_lfe] = lfe;
}
constexpr u32 GetSplitterInParamHeaderMagic() {
return Common::MakeMagic('S', 'N', 'D', 'H');
}
constexpr u32 GetSplitterInfoMagic() {
return Common::MakeMagic('S', 'N', 'D', 'I');
}
constexpr u32 GetSplitterSendDataMagic() {
return Common::MakeMagic('S', 'N', 'D', 'D');
}
constexpr size_t GetSampleFormatByteSize(SampleFormat format) {
switch (format) {
case SampleFormat::PcmInt8:
return 1;
case SampleFormat::PcmInt16:
return 2;
case SampleFormat::PcmInt24:
return 3;
case SampleFormat::PcmInt32:
case SampleFormat::PcmFloat:
return 4;
default:
return 2;
}
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <map>
#include <ranges>
#include <tuple>
#include "common/assert.h"
#include "common/common_funcs.h"
#include "common/common_types.h"
namespace AudioCore {
constexpr u32 CurrentRevision = 11;
enum class SupportTags {
CommandProcessingTimeEstimatorVersion4,
CommandProcessingTimeEstimatorVersion3,
CommandProcessingTimeEstimatorVersion2,
MultiTapBiquadFilterProcessing,
EffectInfoVer2,
WaveBufferVer2,
BiquadFilterFloatProcessing,
VolumeMixParameterPrecisionQ23,
MixInParameterDirtyOnlyUpdate,
BiquadFilterEffectStateClearBugFix,
VoicePlayedSampleCountResetAtLoopPoint,
VoicePitchAndSrcSkipped,
SplitterBugFix,
FlushVoiceWaveBuffers,
ElapsedFrameCount,
AudioRendererVariadicCommandBufferSize,
PerformanceMetricsDataFormatVersion2,
AudioRendererProcessingTimeLimit80Percent,
AudioRendererProcessingTimeLimit75Percent,
AudioRendererProcessingTimeLimit70Percent,
AdpcmLoopContextBugFix,
Splitter,
LongSizePreDelay,
AudioUsbDeviceOutput,
DeviceApiVersion2,
DelayChannelMappingChange,
ReverbChannelMappingChange,
I3dl2ReverbChannelMappingChange,
// Not a real tag, just here to get the count.
Size
};
constexpr u32 GetRevisionNum(u32 user_revision) {
if (user_revision >= 0x100) {
user_revision -= Common::MakeMagic('R', 'E', 'V', '0');
user_revision >>= 24;
}
return user_revision;
};
constexpr bool CheckFeatureSupported(SupportTags tag, u32 user_revision) {
constexpr std::array<std::pair<SupportTags, u32>, static_cast<u32>(SupportTags::Size)> features{
{
{SupportTags::AudioRendererProcessingTimeLimit70Percent, 1},
{SupportTags::Splitter, 2},
{SupportTags::AdpcmLoopContextBugFix, 2},
{SupportTags::LongSizePreDelay, 3},
{SupportTags::AudioUsbDeviceOutput, 4},
{SupportTags::AudioRendererProcessingTimeLimit75Percent, 4},
{SupportTags::VoicePlayedSampleCountResetAtLoopPoint, 5},
{SupportTags::VoicePitchAndSrcSkipped, 5},
{SupportTags::SplitterBugFix, 5},
{SupportTags::FlushVoiceWaveBuffers, 5},
{SupportTags::ElapsedFrameCount, 5},
{SupportTags::AudioRendererProcessingTimeLimit80Percent, 5},
{SupportTags::AudioRendererVariadicCommandBufferSize, 5},
{SupportTags::PerformanceMetricsDataFormatVersion2, 5},
{SupportTags::CommandProcessingTimeEstimatorVersion2, 5},
{SupportTags::BiquadFilterEffectStateClearBugFix, 6},
{SupportTags::BiquadFilterFloatProcessing, 7},
{SupportTags::VolumeMixParameterPrecisionQ23, 7},
{SupportTags::MixInParameterDirtyOnlyUpdate, 7},
{SupportTags::WaveBufferVer2, 8},
{SupportTags::CommandProcessingTimeEstimatorVersion3, 8},
{SupportTags::EffectInfoVer2, 9},
{SupportTags::CommandProcessingTimeEstimatorVersion4, 10},
{SupportTags::MultiTapBiquadFilterProcessing, 10},
{SupportTags::DelayChannelMappingChange, 11},
{SupportTags::ReverbChannelMappingChange, 11},
{SupportTags::I3dl2ReverbChannelMappingChange, 11},
}};
const auto& feature =
std::ranges::find_if(features, [tag](const auto& entry) { return entry.first == tag; });
if (feature == features.cend()) {
LOG_ERROR(Service_Audio, "Invalid SupportTag {}!", static_cast<u32>(tag));
return false;
}
user_revision = GetRevisionNum(user_revision);
return (*feature).second <= user_revision;
}
constexpr bool CheckValidRevision(u32 user_revision) {
return GetRevisionNum(user_revision) <= CurrentRevision;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "common/common_types.h"
namespace AudioCore {
struct WaveBufferVersion1 {
CpuAddr buffer;
u64 buffer_size;
u32 start_offset;
u32 end_offset;
bool loop;
bool stream_ended;
CpuAddr context;
u64 context_size;
};
struct WaveBufferVersion2 {
CpuAddr buffer;
CpuAddr context;
u64 buffer_size;
u64 context_size;
u32 start_offset;
u32 end_offset;
u32 loop_start_offset;
u32 loop_end_offset;
s32 loop_count;
bool loop;
bool stream_ended;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <span>
#include "common/alignment.h"
#include "common/assert.h"
#include "common/common_types.h"
namespace AudioCore {
/**
* Responsible for allocating up a workbuffer into multiple pieces.
* Takes in a buffer and size (it does not own them), and allocates up the buffer via Allocate.
*/
class WorkbufferAllocator {
public:
explicit WorkbufferAllocator(std::span<u8> buffer_, u64 size_)
: buffer{reinterpret_cast<u64>(buffer_.data())}, size{size_} {}
/**
* Allocate the given count of T elements, aligned to alignment.
*
* @param count - The number of elements to allocate.
* @param alignment - The required starting alignment.
* @return Non-owning container of allocated elements.
*/
template <typename T>
std::span<T> Allocate(u64 count, u64 alignment) {
u64 out{0};
u64 byte_size{count * sizeof(T)};
if (byte_size > 0) {
auto current{buffer + offset};
auto aligned_buffer{Common::AlignUp(current, alignment)};
if (aligned_buffer + byte_size <= buffer + size) {
out = aligned_buffer;
offset = byte_size - buffer + aligned_buffer;
} else {
LOG_ERROR(
Service_Audio,
"Allocated buffer was too small to hold new alloc.\nAllocator size={:08X}, "
"offset={:08X}.\nAttempting to allocate {:08X} with alignment={:02X}",
size, offset, byte_size, alignment);
count = 0;
}
}
return std::span<T>(reinterpret_cast<T*>(out), count);
}
/**
* Align the current offset to the given alignment.
*
* @param alignment - The required starting alignment.
*/
void Align(u64 alignment) {
auto current{buffer + offset};
auto aligned_buffer{Common::AlignUp(current, alignment)};
offset = 0 - buffer + aligned_buffer;
}
/**
* Get the current buffer offset.
*
* @return The current allocating offset.
*/
u64 GetCurrentOffset() const {
return offset;
}
/**
* Get the current buffer size.
*
* @return The size of the current buffer.
*/
u64 GetSize() const {
return size;
}
/**
* Get the remaining size that can be allocated.
*
* @return The remaining size left in the buffer.
*/
u64 GetRemainingSize() const {
return size - offset;
}
private:
/// The buffer into which we are allocating.
u64 buffer;
/// Size of the buffer we're allocating to.
u64 size;
/// Current offset into the buffer, an error will be thrown if it exceeds size.
u64 offset{};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include <atomic>
#include <cstring>
#include "audio_core/cubeb_sink.h"
#include "audio_core/stream.h"
#include "common/assert.h"
#include "common/logging/log.h"
#include "common/ring_buffer.h"
#include "common/settings.h"
#ifdef _WIN32
#include <objbase.h>
#endif
namespace AudioCore {
class CubebSinkStream final : public SinkStream {
public:
CubebSinkStream(cubeb* ctx_, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
const std::string& name)
: ctx{ctx_}, num_channels{std::min(num_channels_, 6u)} {
cubeb_stream_params params{};
params.rate = sample_rate;
params.channels = num_channels;
params.format = CUBEB_SAMPLE_S16NE;
params.prefs = CUBEB_STREAM_PREF_PERSIST;
switch (num_channels) {
case 1:
params.layout = CUBEB_LAYOUT_MONO;
break;
case 2:
params.layout = CUBEB_LAYOUT_STEREO;
break;
case 6:
params.layout = CUBEB_LAYOUT_3F2_LFE;
break;
}
u32 minimum_latency{};
if (cubeb_get_min_latency(ctx, &params, &minimum_latency) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error getting minimum latency");
}
if (cubeb_stream_init(ctx, &stream_backend, name.c_str(), nullptr, nullptr, output_device,
&params, std::max(512u, minimum_latency),
&CubebSinkStream::DataCallback, &CubebSinkStream::StateCallback,
this) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream");
return;
}
if (cubeb_stream_start(stream_backend) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream");
return;
}
}
~CubebSinkStream() override {
if (!ctx) {
return;
}
if (cubeb_stream_stop(stream_backend) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error stopping cubeb stream");
}
cubeb_stream_destroy(stream_backend);
}
void EnqueueSamples(u32 source_num_channels, const std::vector<s16>& samples) override {
if (source_num_channels > num_channels) {
// Downsample 6 channels to 2
ASSERT_MSG(source_num_channels == 6, "Channel count must be 6");
std::vector<s16> buf;
buf.reserve(samples.size() * num_channels / source_num_channels);
for (std::size_t i = 0; i < samples.size(); i += source_num_channels) {
// Downmixing implementation taken from the ATSC standard
const s16 left{samples[i + 0]};
const s16 right{samples[i + 1]};
const s16 center{samples[i + 2]};
const s16 surround_left{samples[i + 4]};
const s16 surround_right{samples[i + 5]};
// Not used in the ATSC reference implementation
[[maybe_unused]] const s16 low_frequency_effects{samples[i + 3]};
constexpr s32 clev{707}; // center mixing level coefficient
constexpr s32 slev{707}; // surround mixing level coefficient
buf.push_back(static_cast<s16>(left + (clev * center / 1000) +
(slev * surround_left / 1000)));
buf.push_back(static_cast<s16>(right + (clev * center / 1000) +
(slev * surround_right / 1000)));
}
queue.Push(buf);
return;
}
queue.Push(samples);
}
std::size_t SamplesInQueue(u32 channel_count) const override {
if (!ctx)
return 0;
return queue.Size() / channel_count;
}
void Flush() override {
should_flush = true;
}
u32 GetNumChannels() const {
return num_channels;
}
private:
std::vector<std::string> device_list;
cubeb* ctx{};
cubeb_stream* stream_backend{};
u32 num_channels{};
Common::RingBuffer<s16, 0x10000> queue;
std::array<s16, 2> last_frame{};
std::atomic<bool> should_flush{};
static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
void* output_buffer, long num_frames);
static void StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state);
};
CubebSink::CubebSink(std::string_view target_device_name) {
// Cubeb requires COM to be initialized on the thread calling cubeb_init on Windows
#ifdef _WIN32
com_init_result = CoInitializeEx(nullptr, COINIT_MULTITHREADED);
#endif
if (cubeb_init(&ctx, "yuzu", nullptr) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
return;
}
if (target_device_name != auto_device_name && !target_device_name.empty()) {
cubeb_device_collection collection;
if (cubeb_enumerate_devices(ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) {
LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported");
} else {
const auto collection_end{collection.device + collection.count};
const auto device{
std::find_if(collection.device, collection_end, [&](const cubeb_device_info& info) {
return info.friendly_name != nullptr &&
target_device_name == info.friendly_name;
})};
if (device != collection_end) {
output_device = device->devid;
}
cubeb_device_collection_destroy(ctx, &collection);
}
}
}
CubebSink::~CubebSink() {
if (!ctx) {
return;
}
for (auto& sink_stream : sink_streams) {
sink_stream.reset();
}
cubeb_destroy(ctx);
#ifdef _WIN32
if (SUCCEEDED(com_init_result)) {
CoUninitialize();
}
#endif
}
SinkStream& CubebSink::AcquireSinkStream(u32 sample_rate, u32 num_channels,
const std::string& name) {
sink_streams.push_back(
std::make_unique<CubebSinkStream>(ctx, sample_rate, num_channels, output_device, name));
return *sink_streams.back();
}
long CubebSinkStream::DataCallback([[maybe_unused]] cubeb_stream* stream, void* user_data,
[[maybe_unused]] const void* input_buffer, void* output_buffer,
long num_frames) {
auto* impl = static_cast<CubebSinkStream*>(user_data);
auto* buffer = static_cast<u8*>(output_buffer);
if (!impl) {
return {};
}
const std::size_t num_channels = impl->GetNumChannels();
const std::size_t samples_to_write = num_channels * num_frames;
const std::size_t samples_written = impl->queue.Pop(buffer, samples_to_write);
if (samples_written >= num_channels) {
std::memcpy(&impl->last_frame[0], buffer + (samples_written - num_channels) * sizeof(s16),
num_channels * sizeof(s16));
}
// Fill the rest of the frames with last_frame
for (std::size_t i = samples_written; i < samples_to_write; i += num_channels) {
std::memcpy(buffer + i * sizeof(s16), &impl->last_frame[0], num_channels * sizeof(s16));
}
return num_frames;
}
void CubebSinkStream::StateCallback([[maybe_unused]] cubeb_stream* stream,
[[maybe_unused]] void* user_data,
[[maybe_unused]] cubeb_state state) {}
std::vector<std::string> ListCubebSinkDevices() {
std::vector<std::string> device_list;
cubeb* ctx;
if (cubeb_init(&ctx, "yuzu Device Enumerator", nullptr) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
return {};
}
cubeb_device_collection collection;
if (cubeb_enumerate_devices(ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) {
LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported");
} else {
for (std::size_t i = 0; i < collection.count; i++) {
const cubeb_device_info& device = collection.device[i];
if (device.friendly_name) {
device_list.emplace_back(device.friendly_name);
}
}
cubeb_device_collection_destroy(ctx, &collection);
}
cubeb_destroy(ctx);
return device_list;
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <string>
#include <vector>
#include <cubeb/cubeb.h>
#include "audio_core/sink.h"
namespace AudioCore {
class CubebSink final : public Sink {
public:
explicit CubebSink(std::string_view device_id);
~CubebSink() override;
SinkStream& AcquireSinkStream(u32 sample_rate, u32 num_channels,
const std::string& name) override;
private:
cubeb* ctx{};
cubeb_devid output_device{};
std::vector<SinkStreamPtr> sink_streams;
#ifdef _WIN32
u32 com_init_result = 0;
#endif
};
std::vector<std::string> ListCubebSinkDevices();
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2021 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <cstring>
#include "audio_core/delay_line.h"
namespace AudioCore {
DelayLineBase::DelayLineBase() = default;
DelayLineBase::~DelayLineBase() = default;
void DelayLineBase::Initialize(s32 max_delay_, float* src_buffer) {
buffer = src_buffer;
buffer_end = buffer + max_delay_;
max_delay = max_delay_;
output = buffer;
SetDelay(max_delay_);
Clear();
}
void DelayLineBase::SetDelay(s32 new_delay) {
if (max_delay < new_delay) {
return;
}
delay = new_delay;
input = (buffer + ((output - buffer) + new_delay) % (max_delay + 1));
}
s32 DelayLineBase::GetDelay() const {
return delay;
}
s32 DelayLineBase::GetMaxDelay() const {
return max_delay;
}
f32 DelayLineBase::TapOut(s32 last_sample) {
const float* ptr = input - (last_sample + 1);
if (ptr < buffer) {
ptr += (max_delay + 1);
}
return *ptr;
}
f32 DelayLineBase::Tick(f32 sample) {
*(input++) = sample;
const auto out_sample = *(output++);
if (buffer_end < input) {
input = buffer;
}
if (buffer_end < output) {
output = buffer;
}
return out_sample;
}
float* DelayLineBase::GetInput() {
return input;
}
const float* DelayLineBase::GetInput() const {
return input;
}
f32 DelayLineBase::GetOutputSample() const {
return *output;
}
void DelayLineBase::Clear() {
std::memset(buffer, 0, sizeof(float) * max_delay);
}
void DelayLineBase::Reset() {
buffer = nullptr;
buffer_end = nullptr;
max_delay = 0;
input = nullptr;
output = nullptr;
delay = 0;
}
DelayLineAllPass::DelayLineAllPass() = default;
DelayLineAllPass::~DelayLineAllPass() = default;
void DelayLineAllPass::Initialize(u32 delay_, float coeffcient_, f32* src_buffer) {
DelayLineBase::Initialize(delay_, src_buffer);
SetCoefficient(coeffcient_);
}
void DelayLineAllPass::SetCoefficient(float coeffcient_) {
coefficient = coeffcient_;
}
f32 DelayLineAllPass::Tick(f32 sample) {
const auto temp = sample - coefficient * *output;
return coefficient * temp + DelayLineBase::Tick(temp);
}
void DelayLineAllPass::Reset() {
coefficient = 0.0f;
DelayLineBase::Reset();
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2021 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "common/common_types.h"
namespace AudioCore {
class DelayLineBase {
public:
DelayLineBase();
~DelayLineBase();
void Initialize(s32 max_delay_, float* src_buffer);
void SetDelay(s32 new_delay);
s32 GetDelay() const;
s32 GetMaxDelay() const;
f32 TapOut(s32 last_sample);
f32 Tick(f32 sample);
float* GetInput();
const float* GetInput() const;
f32 GetOutputSample() const;
void Clear();
void Reset();
protected:
float* buffer{nullptr};
float* buffer_end{nullptr};
s32 max_delay{};
float* input{nullptr};
float* output{nullptr};
s32 delay{};
};
class DelayLineAllPass final : public DelayLineBase {
public:
DelayLineAllPass();
~DelayLineAllPass();
void Initialize(u32 delay, float coeffcient_, f32* src_buffer);
void SetCoefficient(float coeffcient_);
f32 Tick(f32 sample);
void Reset();
private:
float coefficient{};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "common/common_types.h"
namespace AudioCore {
struct AudioBuffer {
/// Timestamp this buffer completed playing.
s64 played_timestamp;
/// Game memory address for these samples.
VAddr samples;
/// Unqiue identifier for this buffer.
u64 tag;
/// Size of the samples buffer.
u64 size;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <mutex>
#include <span>
#include <vector>
#include "audio_buffer.h"
#include "audio_core/device/device_session.h"
#include "core/core_timing.h"
namespace AudioCore {
constexpr s32 BufferAppendLimit = 4;
/**
* A ringbuffer of N audio buffers.
* The buffer contains 3 sections:
* Appended - Buffers added to the ring, but have yet to be sent to the audio backend.
* Registered - Buffers sent to the backend and queued for playback.
* Released - Buffers which have been played, and can now be recycled.
* Any others are free/untracked.
*
* @tparam N - Maximum number of buffers in the ring.
*/
template <size_t N>
class AudioBuffers {
public:
explicit AudioBuffers(size_t limit) : append_limit{static_cast<u32>(limit)} {}
/**
* Append a new audio buffer to the ring.
*
* @param buffer - The new buffer.
*/
void AppendBuffer(AudioBuffer& buffer) {
std::scoped_lock l{lock};
buffers[appended_index] = buffer;
appended_count++;
appended_index = (appended_index + 1) % append_limit;
}
/**
* Register waiting buffers, up to a maximum of BufferAppendLimit.
*
* @param out_buffers - The buffers which were registered.
*/
void RegisterBuffers(std::vector<AudioBuffer>& out_buffers) {
std::scoped_lock l{lock};
const s32 to_register{std::min(std::min(appended_count, BufferAppendLimit),
BufferAppendLimit - registered_count)};
for (s32 i = 0; i < to_register; i++) {
s32 index{appended_index - appended_count};
if (index < 0) {
index += N;
}
out_buffers.push_back(buffers[index]);
registered_count++;
registered_index = (registered_index + 1) % append_limit;
appended_count--;
if (appended_count == 0) {
break;
}
}
}
/**
* Release a single buffer. Must be already registered.
*
* @param index - The buffer index to release.
* @param timestamp - The released timestamp for this buffer.
*/
void ReleaseBuffer(s32 index, s64 timestamp) {
std::scoped_lock l{lock};
buffers[index].played_timestamp = timestamp;
registered_count--;
released_count++;
released_index = (released_index + 1) % append_limit;
}
/**
* Release all registered buffers.
*
* @param timestamp - The released timestamp for this buffer.
* @return Is the buffer was released.
*/
bool ReleaseBuffers(Core::Timing::CoreTiming& core_timing, DeviceSession& session) {
std::scoped_lock l{lock};
bool buffer_released{false};
while (registered_count > 0) {
auto index{registered_index - registered_count};
if (index < 0) {
index += N;
}
// Check with the backend if this buffer can be released yet.
if (!session.IsBufferConsumed(buffers[index].tag)) {
break;
}
ReleaseBuffer(index, core_timing.GetGlobalTimeNs().count());
buffer_released = true;
}
return buffer_released || registered_count == 0;
}
/**
* Get all released buffers.
*
* @param tags - Container to be filled with the released buffers' tags.
* @return The number of buffers released.
*/
u32 GetReleasedBuffers(std::span<u64> tags) {
std::scoped_lock l{lock};
u32 released{0};
while (released_count > 0) {
auto index{released_index - released_count};
if (index < 0) {
index += N;
}
auto& buffer{buffers[index]};
released_count--;
auto tag{buffer.tag};
buffer.played_timestamp = 0;
buffer.samples = 0;
buffer.tag = 0;
buffer.size = 0;
if (tag == 0) {
break;
}
tags[released++] = tag;
if (released >= tags.size()) {
break;
}
}
return released;
}
/**
* Get all appended and registered buffers.
*
* @param buffers_flushed - Output vector for the buffers which are released.
* @param max_buffers - Maximum number of buffers to released.
* @return The number of buffers released.
*/
u32 GetRegisteredAppendedBuffers(std::vector<AudioBuffer>& buffers_flushed, u32 max_buffers) {
std::scoped_lock l{lock};
if (registered_count + appended_count == 0) {
return 0;
}
size_t buffers_to_flush{
std::min(static_cast<u32>(registered_count + appended_count), max_buffers)};
if (buffers_to_flush == 0) {
return 0;
}
while (registered_count > 0) {
auto index{registered_index - registered_count};
if (index < 0) {
index += N;
}
buffers_flushed.push_back(buffers[index]);
registered_count--;
released_count++;
released_index = (released_index + 1) % append_limit;
if (buffers_flushed.size() >= buffers_to_flush) {
break;
}
}
while (appended_count > 0) {
auto index{appended_index - appended_count};
if (index < 0) {
index += N;
}
buffers_flushed.push_back(buffers[index]);
appended_count--;
released_count++;
released_index = (released_index + 1) % append_limit;
if (buffers_flushed.size() >= buffers_to_flush) {
break;
}
}
return static_cast<u32>(buffers_flushed.size());
}
/**
* Check if the given tag is in the buffers.
*
* @param tag - Unique tag of the buffer to search for.
* @return True if the buffer is still in the ring, otherwise false.
*/
bool ContainsBuffer(const u64 tag) const {
std::scoped_lock l{lock};
const auto registered_buffers{appended_count + registered_count + released_count};
if (registered_buffers == 0) {
return false;
}
auto index{released_index - released_count};
if (index < 0) {
index += append_limit;
}
for (s32 i = 0; i < registered_buffers; i++) {
if (buffers[index].tag == tag) {
return true;
}
index = (index + 1) % append_limit;
}
return false;
}
/**
* Get the number of active buffers in the ring.
* That is, appended, registered and released buffers.
*
* @return Number of active buffers.
*/
u32 GetAppendedRegisteredCount() const {
std::scoped_lock l{lock};
return appended_count + registered_count;
}
/**
* Get the total number of active buffers in the ring.
* That is, appended, registered and released buffers.
*
* @return Number of active buffers.
*/
u32 GetTotalBufferCount() const {
std::scoped_lock l{lock};
return static_cast<u32>(appended_count + registered_count + released_count);
}
/**
* Flush all of the currently appended and registered buffers
*
* @param buffers_released - Output count for the number of buffers released.
* @return True if buffers were successfully flushed, otherwise false.
*/
bool FlushBuffers(u32& buffers_released) {
std::scoped_lock l{lock};
std::vector<AudioBuffer> buffers_flushed{};
buffers_released = GetRegisteredAppendedBuffers(buffers_flushed, append_limit);
if (registered_count > 0) {
return false;
}
if (static_cast<u32>(released_count + appended_count) > append_limit) {
return false;
}
return true;
}
private:
/// Buffer lock
mutable std::recursive_mutex lock{};
/// The audio buffers
std::array<AudioBuffer, N> buffers{};
/// Current released index
s32 released_index{};
/// Number of released buffers
s32 released_count{};
/// Current registered index
s32 registered_index{};
/// Number of registered buffers
s32 registered_count{};
/// Current appended index
s32 appended_index{};
/// Number of appended buffers
s32 appended_count{};
/// Maximum number of buffers (default 32)
u32 append_limit{};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_core.h"
#include "audio_core/audio_manager.h"
#include "audio_core/device/audio_buffer.h"
#include "audio_core/device/device_session.h"
#include "audio_core/sink/sink_stream.h"
#include "core/core.h"
#include "core/memory.h"
namespace AudioCore {
DeviceSession::DeviceSession(Core::System& system_) : system{system_} {}
DeviceSession::~DeviceSession() {
Finalize();
}
Result DeviceSession::Initialize(std::string_view name_, SampleFormat sample_format_,
u16 channel_count_, size_t session_id_, u32 handle_,
u64 applet_resource_user_id_, Sink::StreamType type_) {
if (stream) {
Finalize();
}
name = fmt::format("{}-{}", name_, session_id_);
type = type_;
sample_format = sample_format_;
channel_count = channel_count_;
session_id = session_id_;
handle = handle_;
applet_resource_user_id = applet_resource_user_id_;
if (type == Sink::StreamType::In) {
sink = &system.AudioCore().GetInputSink();
} else {
sink = &system.AudioCore().GetOutputSink();
}
stream = sink->AcquireSinkStream(system, channel_count, name, type);
initialized = true;
return ResultSuccess;
}
void DeviceSession::Finalize() {
if (initialized) {
Stop();
sink->CloseStream(stream);
stream = nullptr;
}
}
void DeviceSession::Start() {
stream->SetPlayedSampleCount(played_sample_count);
stream->Start();
}
void DeviceSession::Stop() {
if (stream) {
played_sample_count = stream->GetPlayedSampleCount();
stream->Stop();
}
}
void DeviceSession::AppendBuffers(std::span<AudioBuffer> buffers) const {
auto& memory{system.Memory()};
for (size_t i = 0; i < buffers.size(); i++) {
Sink::SinkBuffer new_buffer{
.frames = buffers[i].size / (channel_count * sizeof(s16)),
.frames_played = 0,
.tag = buffers[i].tag,
.consumed = false,
};
if (type == Sink::StreamType::In) {
std::vector<s16> samples{};
stream->AppendBuffer(new_buffer, samples);
} else {
std::vector<s16> samples(buffers[i].size / sizeof(s16));
memory.ReadBlockUnsafe(buffers[i].samples, samples.data(), buffers[i].size);
stream->AppendBuffer(new_buffer, samples);
}
}
}
void DeviceSession::ReleaseBuffer(AudioBuffer& buffer) const {
if (type == Sink::StreamType::In) {
auto& memory{system.Memory()};
auto samples{stream->ReleaseBuffer(buffer.size / sizeof(s16))};
memory.WriteBlockUnsafe(buffer.samples, samples.data(), buffer.size);
}
}
bool DeviceSession::IsBufferConsumed(u64 tag) const {
if (stream) {
return stream->IsBufferConsumed(tag);
}
return true;
}
void DeviceSession::SetVolume(f32 volume) const {
if (stream) {
stream->SetSystemVolume(volume);
}
}
u64 DeviceSession::GetPlayedSampleCount() const {
if (stream) {
return stream->GetPlayedSampleCount();
}
return 0;
}
} // namespace AudioCore

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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/common/common.h"
#include "audio_core/sink/sink.h"
#include "core/hle/service/audio/errors.h"
namespace Core {
class System;
}
namespace AudioCore {
namespace Sink {
class SinkStream;
struct SinkBuffer;
} // namespace Sink
struct AudioBuffer;
/**
* Represents an input or output device stream for audio in and audio out (not used for render).
**/
class DeviceSession {
public:
explicit DeviceSession(Core::System& system);
~DeviceSession();
/**
* Initialize this device session.
*
* @param name - Name of this device.
* @param sample_format - Sample format for this device's output.
* @param channel_count - Number of channels for this device (2 or 6).
* @param session_id - This session's id.
* @param handle - Handle for this device session (unused).
* @param applet_resource_user_id - Applet resource user id for this device session (unused).
* @param type - Type of this stream (Render, In, Out).
* @return Result code for this call.
*/
Result Initialize(std::string_view name, SampleFormat sample_format, u16 channel_count,
size_t session_id, u32 handle, u64 applet_resource_user_id,
Sink::StreamType type);
/**
* Finalize this device session.
*/
void Finalize();
/**
* Append audio buffers to this device session to be played back.
*
* @param buffers - The buffers to play.
*/
void AppendBuffers(std::span<AudioBuffer> buffers) const;
/**
* (Audio In only) Pop samples from the backend, and write them back to this buffer's address.
*
* @param buffer - The buffer to write to.
*/
void ReleaseBuffer(AudioBuffer& buffer) const;
/**
* Check if the buffer for the given tag has been consumed by the backend.
*
* @param tag - Unqiue tag of the buffer to check.
* @return true if the buffer has been consumed, otherwise false.
*/
bool IsBufferConsumed(u64 tag) const;
/**
* Start this device session, starting the backend stream.
*/
void Start();
/**
* Stop this device session, stopping the backend stream.
*/
void Stop();
/**
* Set this device session's volume.
*
* @param volume - New volume for this session.
*/
void SetVolume(f32 volume) const;
/**
* Get this device session's total played sample count.
*
* @return Samples played by this session.
*/
u64 GetPlayedSampleCount() const;
private:
/// System
Core::System& system;
/// Output sink this device will use
Sink::Sink* sink{};
/// The backend stream for this device session to send samples to
Sink::SinkStream* stream{};
/// Name of this device session
std::string name{};
/// Type of this device session (render/in/out)
Sink::StreamType type{};
/// Sample format for this device.
SampleFormat sample_format{SampleFormat::PcmInt16};
/// Channel count for this device session
u16 channel_count{};
/// Session id of this device session
size_t session_id{};
/// Handle of this device session
u32 handle{};
/// Applet resource user id of this device session
u64 applet_resource_user_id{};
/// Total number of samples played by this device session
u64 played_sample_count{};
/// Is this session initialised?
bool initialized{};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include "audio_core/effect_context.h"
namespace AudioCore {
namespace {
bool ValidChannelCountForEffect(s32 channel_count) {
return channel_count == 1 || channel_count == 2 || channel_count == 4 || channel_count == 6;
}
} // namespace
EffectContext::EffectContext(std::size_t effect_count_) : effect_count(effect_count_) {
effects.reserve(effect_count);
std::generate_n(std::back_inserter(effects), effect_count,
[] { return std::make_unique<EffectStubbed>(); });
}
EffectContext::~EffectContext() = default;
std::size_t EffectContext::GetCount() const {
return effect_count;
}
EffectBase* EffectContext::GetInfo(std::size_t i) {
return effects.at(i).get();
}
EffectBase* EffectContext::RetargetEffect(std::size_t i, EffectType effect) {
switch (effect) {
case EffectType::Invalid:
effects[i] = std::make_unique<EffectStubbed>();
break;
case EffectType::BufferMixer:
effects[i] = std::make_unique<EffectBufferMixer>();
break;
case EffectType::Aux:
effects[i] = std::make_unique<EffectAuxInfo>();
break;
case EffectType::Delay:
effects[i] = std::make_unique<EffectDelay>();
break;
case EffectType::Reverb:
effects[i] = std::make_unique<EffectReverb>();
break;
case EffectType::I3dl2Reverb:
effects[i] = std::make_unique<EffectI3dl2Reverb>();
break;
case EffectType::BiquadFilter:
effects[i] = std::make_unique<EffectBiquadFilter>();
break;
default:
ASSERT_MSG(false, "Unimplemented effect {}", effect);
effects[i] = std::make_unique<EffectStubbed>();
}
return GetInfo(i);
}
const EffectBase* EffectContext::GetInfo(std::size_t i) const {
return effects.at(i).get();
}
EffectStubbed::EffectStubbed() : EffectBase(EffectType::Invalid) {}
EffectStubbed::~EffectStubbed() = default;
void EffectStubbed::Update([[maybe_unused]] EffectInfo::InParams& in_params) {}
void EffectStubbed::UpdateForCommandGeneration() {}
EffectBase::EffectBase(EffectType effect_type_) : effect_type(effect_type_) {}
EffectBase::~EffectBase() = default;
UsageState EffectBase::GetUsage() const {
return usage;
}
EffectType EffectBase::GetType() const {
return effect_type;
}
bool EffectBase::IsEnabled() const {
return enabled;
}
s32 EffectBase::GetMixID() const {
return mix_id;
}
s32 EffectBase::GetProcessingOrder() const {
return processing_order;
}
std::vector<u8>& EffectBase::GetWorkBuffer() {
return work_buffer;
}
const std::vector<u8>& EffectBase::GetWorkBuffer() const {
return work_buffer;
}
EffectI3dl2Reverb::EffectI3dl2Reverb() : EffectGeneric(EffectType::I3dl2Reverb) {}
EffectI3dl2Reverb::~EffectI3dl2Reverb() = default;
void EffectI3dl2Reverb::Update(EffectInfo::InParams& in_params) {
auto& params = GetParams();
const auto* reverb_params = reinterpret_cast<I3dl2ReverbParams*>(in_params.raw.data());
if (!ValidChannelCountForEffect(reverb_params->max_channels)) {
ASSERT_MSG(false, "Invalid reverb max channel count {}", reverb_params->max_channels);
return;
}
const auto last_status = params.status;
mix_id = in_params.mix_id;
processing_order = in_params.processing_order;
params = *reverb_params;
if (!ValidChannelCountForEffect(reverb_params->channel_count)) {
params.channel_count = params.max_channels;
}
enabled = in_params.is_enabled;
if (last_status != ParameterStatus::Updated) {
params.status = last_status;
}
if (in_params.is_new || skipped) {
usage = UsageState::Initialized;
params.status = ParameterStatus::Initialized;
skipped = in_params.buffer_address == 0 || in_params.buffer_size == 0;
if (!skipped) {
auto& cur_work_buffer = GetWorkBuffer();
// Has two buffers internally
cur_work_buffer.resize(in_params.buffer_size * 2);
std::fill(cur_work_buffer.begin(), cur_work_buffer.end(), 0);
}
}
}
void EffectI3dl2Reverb::UpdateForCommandGeneration() {
if (enabled) {
usage = UsageState::Running;
} else {
usage = UsageState::Stopped;
}
GetParams().status = ParameterStatus::Updated;
}
I3dl2ReverbState& EffectI3dl2Reverb::GetState() {
return state;
}
const I3dl2ReverbState& EffectI3dl2Reverb::GetState() const {
return state;
}
EffectBiquadFilter::EffectBiquadFilter() : EffectGeneric(EffectType::BiquadFilter) {}
EffectBiquadFilter::~EffectBiquadFilter() = default;
void EffectBiquadFilter::Update(EffectInfo::InParams& in_params) {
auto& params = GetParams();
const auto* biquad_params = reinterpret_cast<BiquadFilterParams*>(in_params.raw.data());
mix_id = in_params.mix_id;
processing_order = in_params.processing_order;
params = *biquad_params;
enabled = in_params.is_enabled;
}
void EffectBiquadFilter::UpdateForCommandGeneration() {
if (enabled) {
usage = UsageState::Running;
} else {
usage = UsageState::Stopped;
}
GetParams().status = ParameterStatus::Updated;
}
EffectAuxInfo::EffectAuxInfo() : EffectGeneric(EffectType::Aux) {}
EffectAuxInfo::~EffectAuxInfo() = default;
void EffectAuxInfo::Update(EffectInfo::InParams& in_params) {
const auto* aux_params = reinterpret_cast<AuxInfo*>(in_params.raw.data());
mix_id = in_params.mix_id;
processing_order = in_params.processing_order;
GetParams() = *aux_params;
enabled = in_params.is_enabled;
if (in_params.is_new || skipped) {
skipped = aux_params->send_buffer_info == 0 || aux_params->return_buffer_info == 0;
if (skipped) {
return;
}
// There's two AuxInfos which are an identical size, the first one is managed by the cpu,
// the second is managed by the dsp. All we care about is managing the DSP one
send_info = aux_params->send_buffer_info + sizeof(AuxInfoDSP);
send_buffer = aux_params->send_buffer_info + (sizeof(AuxInfoDSP) * 2);
recv_info = aux_params->return_buffer_info + sizeof(AuxInfoDSP);
recv_buffer = aux_params->return_buffer_info + (sizeof(AuxInfoDSP) * 2);
}
}
void EffectAuxInfo::UpdateForCommandGeneration() {
if (enabled) {
usage = UsageState::Running;
} else {
usage = UsageState::Stopped;
}
}
VAddr EffectAuxInfo::GetSendInfo() const {
return send_info;
}
VAddr EffectAuxInfo::GetSendBuffer() const {
return send_buffer;
}
VAddr EffectAuxInfo::GetRecvInfo() const {
return recv_info;
}
VAddr EffectAuxInfo::GetRecvBuffer() const {
return recv_buffer;
}
EffectDelay::EffectDelay() : EffectGeneric(EffectType::Delay) {}
EffectDelay::~EffectDelay() = default;
void EffectDelay::Update(EffectInfo::InParams& in_params) {
const auto* delay_params = reinterpret_cast<DelayParams*>(in_params.raw.data());
auto& params = GetParams();
if (!ValidChannelCountForEffect(delay_params->max_channels)) {
return;
}
const auto last_status = params.status;
mix_id = in_params.mix_id;
processing_order = in_params.processing_order;
params = *delay_params;
if (!ValidChannelCountForEffect(delay_params->channels)) {
params.channels = params.max_channels;
}
enabled = in_params.is_enabled;
if (last_status != ParameterStatus::Updated) {
params.status = last_status;
}
if (in_params.is_new || skipped) {
usage = UsageState::Initialized;
params.status = ParameterStatus::Initialized;
skipped = in_params.buffer_address == 0 || in_params.buffer_size == 0;
}
}
void EffectDelay::UpdateForCommandGeneration() {
if (enabled) {
usage = UsageState::Running;
} else {
usage = UsageState::Stopped;
}
GetParams().status = ParameterStatus::Updated;
}
EffectBufferMixer::EffectBufferMixer() : EffectGeneric(EffectType::BufferMixer) {}
EffectBufferMixer::~EffectBufferMixer() = default;
void EffectBufferMixer::Update(EffectInfo::InParams& in_params) {
mix_id = in_params.mix_id;
processing_order = in_params.processing_order;
GetParams() = *reinterpret_cast<BufferMixerParams*>(in_params.raw.data());
enabled = in_params.is_enabled;
}
void EffectBufferMixer::UpdateForCommandGeneration() {
if (enabled) {
usage = UsageState::Running;
} else {
usage = UsageState::Stopped;
}
}
EffectReverb::EffectReverb() : EffectGeneric(EffectType::Reverb) {}
EffectReverb::~EffectReverb() = default;
void EffectReverb::Update(EffectInfo::InParams& in_params) {
const auto* reverb_params = reinterpret_cast<ReverbParams*>(in_params.raw.data());
auto& params = GetParams();
if (!ValidChannelCountForEffect(reverb_params->max_channels)) {
return;
}
const auto last_status = params.status;
mix_id = in_params.mix_id;
processing_order = in_params.processing_order;
params = *reverb_params;
if (!ValidChannelCountForEffect(reverb_params->channels)) {
params.channels = params.max_channels;
}
enabled = in_params.is_enabled;
if (last_status != ParameterStatus::Updated) {
params.status = last_status;
}
if (in_params.is_new || skipped) {
usage = UsageState::Initialized;
params.status = ParameterStatus::Initialized;
skipped = in_params.buffer_address == 0 || in_params.buffer_size == 0;
}
}
void EffectReverb::UpdateForCommandGeneration() {
if (enabled) {
usage = UsageState::Running;
} else {
usage = UsageState::Stopped;
}
GetParams().status = ParameterStatus::Updated;
}
} // namespace AudioCore

349
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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <memory>
#include <vector>
#include "audio_core/common.h"
#include "audio_core/delay_line.h"
#include "common/common_funcs.h"
#include "common/common_types.h"
#include "common/swap.h"
namespace AudioCore {
enum class EffectType : u8 {
Invalid = 0,
BufferMixer = 1,
Aux = 2,
Delay = 3,
Reverb = 4,
I3dl2Reverb = 5,
BiquadFilter = 6,
};
enum class UsageStatus : u8 {
Invalid = 0,
New = 1,
Initialized = 2,
Used = 3,
Removed = 4,
};
enum class UsageState {
Invalid = 0,
Initialized = 1,
Running = 2,
Stopped = 3,
};
enum class ParameterStatus : u8 {
Initialized = 0,
Updating = 1,
Updated = 2,
};
struct BufferMixerParams {
std::array<s8, AudioCommon::MAX_MIX_BUFFERS> input{};
std::array<s8, AudioCommon::MAX_MIX_BUFFERS> output{};
std::array<float_le, AudioCommon::MAX_MIX_BUFFERS> volume{};
s32_le count{};
};
static_assert(sizeof(BufferMixerParams) == 0x94, "BufferMixerParams is an invalid size");
struct AuxInfoDSP {
u32_le read_offset{};
u32_le write_offset{};
u32_le remaining{};
INSERT_PADDING_WORDS(13);
};
static_assert(sizeof(AuxInfoDSP) == 0x40, "AuxInfoDSP is an invalid size");
struct AuxInfo {
std::array<s8, AudioCommon::MAX_MIX_BUFFERS> input_mix_buffers{};
std::array<s8, AudioCommon::MAX_MIX_BUFFERS> output_mix_buffers{};
u32_le count{};
s32_le sample_rate{};
s32_le sample_count{};
s32_le mix_buffer_count{};
u64_le send_buffer_info{};
u64_le send_buffer_base{};
u64_le return_buffer_info{};
u64_le return_buffer_base{};
};
static_assert(sizeof(AuxInfo) == 0x60, "AuxInfo is an invalid size");
struct I3dl2ReverbParams {
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> input{};
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> output{};
u16_le max_channels{};
u16_le channel_count{};
INSERT_PADDING_BYTES(1);
u32_le sample_rate{};
f32 room_hf{};
f32 hf_reference{};
f32 decay_time{};
f32 hf_decay_ratio{};
f32 room{};
f32 reflection{};
f32 reverb{};
f32 diffusion{};
f32 reflection_delay{};
f32 reverb_delay{};
f32 density{};
f32 dry_gain{};
ParameterStatus status{};
INSERT_PADDING_BYTES(3);
};
static_assert(sizeof(I3dl2ReverbParams) == 0x4c, "I3dl2ReverbParams is an invalid size");
struct BiquadFilterParams {
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> input{};
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> output{};
std::array<s16_le, 3> numerator;
std::array<s16_le, 2> denominator;
s8 channel_count{};
ParameterStatus status{};
};
static_assert(sizeof(BiquadFilterParams) == 0x18, "BiquadFilterParams is an invalid size");
struct DelayParams {
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> input{};
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> output{};
u16_le max_channels{};
u16_le channels{};
s32_le max_delay{};
s32_le delay{};
s32_le sample_rate{};
s32_le gain{};
s32_le feedback_gain{};
s32_le out_gain{};
s32_le dry_gain{};
s32_le channel_spread{};
s32_le low_pass{};
ParameterStatus status{};
INSERT_PADDING_BYTES(3);
};
static_assert(sizeof(DelayParams) == 0x38, "DelayParams is an invalid size");
struct ReverbParams {
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> input{};
std::array<s8, AudioCommon::MAX_CHANNEL_COUNT> output{};
u16_le max_channels{};
u16_le channels{};
s32_le sample_rate{};
s32_le mode0{};
s32_le mode0_gain{};
s32_le pre_delay{};
s32_le mode1{};
s32_le mode1_gain{};
s32_le decay{};
s32_le hf_decay_ratio{};
s32_le coloration{};
s32_le reverb_gain{};
s32_le out_gain{};
s32_le dry_gain{};
ParameterStatus status{};
INSERT_PADDING_BYTES(3);
};
static_assert(sizeof(ReverbParams) == 0x44, "ReverbParams is an invalid size");
class EffectInfo {
public:
struct InParams {
EffectType type{};
u8 is_new{};
u8 is_enabled{};
INSERT_PADDING_BYTES(1);
s32_le mix_id{};
u64_le buffer_address{};
u64_le buffer_size{};
s32_le processing_order{};
INSERT_PADDING_BYTES(4);
union {
std::array<u8, 0xa0> raw;
};
};
static_assert(sizeof(InParams) == 0xc0, "InParams is an invalid size");
struct OutParams {
UsageStatus status{};
INSERT_PADDING_BYTES(15);
};
static_assert(sizeof(OutParams) == 0x10, "OutParams is an invalid size");
};
struct AuxAddress {
VAddr send_dsp_info{};
VAddr send_buffer_base{};
VAddr return_dsp_info{};
VAddr return_buffer_base{};
};
class EffectBase {
public:
explicit EffectBase(EffectType effect_type_);
virtual ~EffectBase();
virtual void Update(EffectInfo::InParams& in_params) = 0;
virtual void UpdateForCommandGeneration() = 0;
[[nodiscard]] UsageState GetUsage() const;
[[nodiscard]] EffectType GetType() const;
[[nodiscard]] bool IsEnabled() const;
[[nodiscard]] s32 GetMixID() const;
[[nodiscard]] s32 GetProcessingOrder() const;
[[nodiscard]] std::vector<u8>& GetWorkBuffer();
[[nodiscard]] const std::vector<u8>& GetWorkBuffer() const;
protected:
UsageState usage{UsageState::Invalid};
EffectType effect_type{};
s32 mix_id{};
s32 processing_order{};
bool enabled = false;
std::vector<u8> work_buffer{};
};
template <typename T>
class EffectGeneric : public EffectBase {
public:
explicit EffectGeneric(EffectType effect_type_) : EffectBase(effect_type_) {}
T& GetParams() {
return internal_params;
}
const T& GetParams() const {
return internal_params;
}
private:
T internal_params{};
};
class EffectStubbed : public EffectBase {
public:
explicit EffectStubbed();
~EffectStubbed() override;
void Update(EffectInfo::InParams& in_params) override;
void UpdateForCommandGeneration() override;
};
struct I3dl2ReverbState {
f32 lowpass_0{};
f32 lowpass_1{};
f32 lowpass_2{};
DelayLineBase early_delay_line{};
std::array<u32, AudioCommon::I3DL2REVERB_TAPS> early_tap_steps{};
f32 early_gain{};
f32 late_gain{};
u32 early_to_late_taps{};
std::array<DelayLineBase, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT> fdn_delay_line{};
std::array<DelayLineAllPass, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT> decay_delay_line0{};
std::array<DelayLineAllPass, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT> decay_delay_line1{};
f32 last_reverb_echo{};
DelayLineBase center_delay_line{};
std::array<std::array<f32, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT>, 3> lpf_coefficients{};
std::array<f32, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT> shelf_filter{};
f32 dry_gain{};
};
class EffectI3dl2Reverb : public EffectGeneric<I3dl2ReverbParams> {
public:
explicit EffectI3dl2Reverb();
~EffectI3dl2Reverb() override;
void Update(EffectInfo::InParams& in_params) override;
void UpdateForCommandGeneration() override;
I3dl2ReverbState& GetState();
const I3dl2ReverbState& GetState() const;
private:
bool skipped = false;
I3dl2ReverbState state{};
};
class EffectBiquadFilter : public EffectGeneric<BiquadFilterParams> {
public:
explicit EffectBiquadFilter();
~EffectBiquadFilter() override;
void Update(EffectInfo::InParams& in_params) override;
void UpdateForCommandGeneration() override;
};
class EffectAuxInfo : public EffectGeneric<AuxInfo> {
public:
explicit EffectAuxInfo();
~EffectAuxInfo() override;
void Update(EffectInfo::InParams& in_params) override;
void UpdateForCommandGeneration() override;
[[nodiscard]] VAddr GetSendInfo() const;
[[nodiscard]] VAddr GetSendBuffer() const;
[[nodiscard]] VAddr GetRecvInfo() const;
[[nodiscard]] VAddr GetRecvBuffer() const;
private:
VAddr send_info{};
VAddr send_buffer{};
VAddr recv_info{};
VAddr recv_buffer{};
bool skipped = false;
AuxAddress addresses{};
};
class EffectDelay : public EffectGeneric<DelayParams> {
public:
explicit EffectDelay();
~EffectDelay() override;
void Update(EffectInfo::InParams& in_params) override;
void UpdateForCommandGeneration() override;
private:
bool skipped = false;
};
class EffectBufferMixer : public EffectGeneric<BufferMixerParams> {
public:
explicit EffectBufferMixer();
~EffectBufferMixer() override;
void Update(EffectInfo::InParams& in_params) override;
void UpdateForCommandGeneration() override;
};
class EffectReverb : public EffectGeneric<ReverbParams> {
public:
explicit EffectReverb();
~EffectReverb() override;
void Update(EffectInfo::InParams& in_params) override;
void UpdateForCommandGeneration() override;
private:
bool skipped = false;
};
class EffectContext {
public:
explicit EffectContext(std::size_t effect_count_);
~EffectContext();
[[nodiscard]] std::size_t GetCount() const;
[[nodiscard]] EffectBase* GetInfo(std::size_t i);
[[nodiscard]] EffectBase* RetargetEffect(std::size_t i, EffectType effect);
[[nodiscard]] const EffectBase* GetInfo(std::size_t i) const;
private:
std::size_t effect_count{};
std::vector<std::unique_ptr<EffectBase>> effects;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_in_manager.h"
#include "audio_core/in/audio_in.h"
#include "core/hle/kernel/k_event.h"
namespace AudioCore::AudioIn {
In::In(Core::System& system_, Manager& manager_, Kernel::KEvent* event_, size_t session_id_)
: manager{manager_}, parent_mutex{manager.mutex}, event{event_}, system{system_, event,
session_id_} {}
void In::Free() {
std::scoped_lock l{parent_mutex};
manager.ReleaseSessionId(system.GetSessionId());
}
System& In::GetSystem() {
return system;
}
AudioIn::State In::GetState() {
std::scoped_lock l{parent_mutex};
return system.GetState();
}
Result In::StartSystem() {
std::scoped_lock l{parent_mutex};
return system.Start();
}
void In::StartSession() {
std::scoped_lock l{parent_mutex};
system.StartSession();
}
Result In::StopSystem() {
std::scoped_lock l{parent_mutex};
return system.Stop();
}
Result In::AppendBuffer(const AudioInBuffer& buffer, u64 tag) {
std::scoped_lock l{parent_mutex};
if (system.AppendBuffer(buffer, tag)) {
return ResultSuccess;
}
return Service::Audio::ERR_BUFFER_COUNT_EXCEEDED;
}
void In::ReleaseAndRegisterBuffers() {
std::scoped_lock l{parent_mutex};
if (system.GetState() == State::Started) {
system.ReleaseBuffers();
system.RegisterBuffers();
}
}
bool In::FlushAudioInBuffers() {
std::scoped_lock l{parent_mutex};
return system.FlushAudioInBuffers();
}
u32 In::GetReleasedBuffers(std::span<u64> tags) {
std::scoped_lock l{parent_mutex};
return system.GetReleasedBuffers(tags);
}
Kernel::KReadableEvent& In::GetBufferEvent() {
std::scoped_lock l{parent_mutex};
return event->GetReadableEvent();
}
f32 In::GetVolume() {
std::scoped_lock l{parent_mutex};
return system.GetVolume();
}
void In::SetVolume(f32 volume) {
std::scoped_lock l{parent_mutex};
system.SetVolume(volume);
}
bool In::ContainsAudioBuffer(u64 tag) {
std::scoped_lock l{parent_mutex};
return system.ContainsAudioBuffer(tag);
}
u32 In::GetBufferCount() {
std::scoped_lock l{parent_mutex};
return system.GetBufferCount();
}
u64 In::GetPlayedSampleCount() {
std::scoped_lock l{parent_mutex};
return system.GetPlayedSampleCount();
}
} // namespace AudioCore::AudioIn

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <mutex>
#include "audio_core/in/audio_in_system.h"
namespace Core {
class System;
}
namespace Kernel {
class KEvent;
class KReadableEvent;
} // namespace Kernel
namespace AudioCore::AudioIn {
class Manager;
/**
* Interface between the service and audio in system. Mainly responsible for forwarding service
* calls to the system.
*/
class In {
public:
explicit In(Core::System& system, Manager& manager, Kernel::KEvent* event, size_t session_id);
/**
* Free this audio in from the audio in manager.
*/
void Free();
/**
* Get this audio in's system.
*/
System& GetSystem();
/**
* Get the current state.
*
* @return Started or Stopped.
*/
AudioIn::State GetState();
/**
* Start the system
*
* @return Result code
*/
Result StartSystem();
/**
* Start the system's device session.
*/
void StartSession();
/**
* Stop the system.
*
* @return Result code
*/
Result StopSystem();
/**
* Append a new buffer to the system, the buffer event will be signalled when it is filled.
*
* @param buffer - The new buffer to append.
* @param tag - Unique tag for this buffer.
* @return Result code.
*/
Result AppendBuffer(const AudioInBuffer& buffer, u64 tag);
/**
* Release all completed buffers, and register any appended.
*/
void ReleaseAndRegisterBuffers();
/**
* Flush all buffers.
*/
bool FlushAudioInBuffers();
/**
* Get all of the currently released buffers.
*
* @param tags - Output container for the buffer tags which were released.
* @return The number of buffers released.
*/
u32 GetReleasedBuffers(std::span<u64> tags);
/**
* Get the buffer event for this audio in, this event will be signalled when a buffer is filled.
*
* @return The buffer event.
*/
Kernel::KReadableEvent& GetBufferEvent();
/**
* Get the current system volume.
*
* @return The current volume.
*/
f32 GetVolume();
/**
* Set the system volume.
*
* @param volume - The volume to set.
*/
void SetVolume(f32 volume);
/**
* Check if a buffer is in the system.
*
* @param tag - The tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
bool ContainsAudioBuffer(u64 tag);
/**
* Get the maximum number of buffers.
*
* @return The maximum number of buffers.
*/
u32 GetBufferCount();
/**
* Get the total played sample count for this audio in.
*
* @return The played sample count.
*/
u64 GetPlayedSampleCount();
private:
/// The AudioIn::Manager this audio in is registered with
Manager& manager;
/// Manager's mutex
std::recursive_mutex& parent_mutex;
/// Buffer event, signalled when buffers are ready to be released
Kernel::KEvent* event;
/// Main audio in system
System system;
};
} // namespace AudioCore::AudioIn

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <mutex>
#include "audio_core/audio_event.h"
#include "audio_core/audio_manager.h"
#include "audio_core/in/audio_in_system.h"
#include "common/logging/log.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/hle/kernel/k_event.h"
namespace AudioCore::AudioIn {
System::System(Core::System& system_, Kernel::KEvent* event_, const size_t session_id_)
: system{system_}, buffer_event{event_},
session_id{session_id_}, session{std::make_unique<DeviceSession>(system_)} {}
System::~System() {
Finalize();
}
void System::Finalize() {
Stop();
session->Finalize();
buffer_event->GetWritableEvent().Signal();
}
void System::StartSession() {
session->Start();
}
size_t System::GetSessionId() const {
return session_id;
}
std::string_view System::GetDefaultDeviceName() {
return "BuiltInHeadset";
}
std::string_view System::GetDefaultUacDeviceName() {
return "Uac";
}
Result System::IsConfigValid(const std::string_view device_name,
const AudioInParameter& in_params) {
if ((device_name.size() > 0) &&
(device_name != GetDefaultDeviceName() && device_name != GetDefaultUacDeviceName())) {
return Service::Audio::ERR_INVALID_DEVICE_NAME;
}
if (in_params.sample_rate != TargetSampleRate && in_params.sample_rate > 0) {
return Service::Audio::ERR_INVALID_SAMPLE_RATE;
}
return ResultSuccess;
}
Result System::Initialize(std::string& device_name, const AudioInParameter& in_params,
const u32 handle_, const u64 applet_resource_user_id_) {
auto result{IsConfigValid(device_name, in_params)};
if (result.IsError()) {
return result;
}
handle = handle_;
applet_resource_user_id = applet_resource_user_id_;
if (device_name.empty() || device_name[0] == '\0') {
name = std::string(GetDefaultDeviceName());
} else {
name = std::move(device_name);
}
sample_rate = TargetSampleRate;
sample_format = SampleFormat::PcmInt16;
channel_count = in_params.channel_count <= 2 ? 2 : 6;
volume = 1.0f;
is_uac = name == "Uac";
return ResultSuccess;
}
Result System::Start() {
if (state != State::Stopped) {
return Service::Audio::ERR_OPERATION_FAILED;
}
session->Initialize(name, sample_format, channel_count, session_id, handle,
applet_resource_user_id, Sink::StreamType::In);
session->SetVolume(volume);
session->Start();
state = State::Started;
std::vector<AudioBuffer> buffers_to_flush{};
buffers.RegisterBuffers(buffers_to_flush);
session->AppendBuffers(buffers_to_flush);
return ResultSuccess;
}
Result System::Stop() {
if (state == State::Started) {
session->Stop();
session->SetVolume(0.0f);
state = State::Stopped;
}
return ResultSuccess;
}
bool System::AppendBuffer(const AudioInBuffer& buffer, const u64 tag) {
if (buffers.GetTotalBufferCount() == BufferCount) {
return false;
}
AudioBuffer new_buffer{
.played_timestamp = 0, .samples = buffer.samples, .tag = tag, .size = buffer.size};
buffers.AppendBuffer(new_buffer);
RegisterBuffers();
return true;
}
void System::RegisterBuffers() {
if (state == State::Started) {
std::vector<AudioBuffer> registered_buffers{};
buffers.RegisterBuffers(registered_buffers);
session->AppendBuffers(registered_buffers);
}
}
void System::ReleaseBuffers() {
bool signal{buffers.ReleaseBuffers(system.CoreTiming(), *session)};
if (signal) {
// Signal if any buffer was released, or if none are registered, we need more.
buffer_event->GetWritableEvent().Signal();
}
}
u32 System::GetReleasedBuffers(std::span<u64> tags) {
return buffers.GetReleasedBuffers(tags);
}
bool System::FlushAudioInBuffers() {
if (state != State::Started) {
return false;
}
u32 buffers_released{};
buffers.FlushBuffers(buffers_released);
if (buffers_released > 0) {
buffer_event->GetWritableEvent().Signal();
}
return true;
}
u16 System::GetChannelCount() const {
return channel_count;
}
u32 System::GetSampleRate() const {
return sample_rate;
}
SampleFormat System::GetSampleFormat() const {
return sample_format;
}
State System::GetState() {
switch (state) {
case State::Started:
case State::Stopped:
return state;
default:
LOG_ERROR(Service_Audio, "AudioIn invalid state!");
state = State::Stopped;
break;
}
return state;
}
std::string System::GetName() const {
return name;
}
f32 System::GetVolume() const {
return volume;
}
void System::SetVolume(const f32 volume_) {
volume = volume_;
session->SetVolume(volume_);
}
bool System::ContainsAudioBuffer(const u64 tag) {
return buffers.ContainsBuffer(tag);
}
u32 System::GetBufferCount() {
return buffers.GetAppendedRegisteredCount();
}
u64 System::GetPlayedSampleCount() const {
return session->GetPlayedSampleCount();
}
bool System::IsUac() const {
return is_uac;
}
} // namespace AudioCore::AudioIn

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <atomic>
#include <memory>
#include <span>
#include <string>
#include "audio_core/common/common.h"
#include "audio_core/device/audio_buffers.h"
#include "audio_core/device/device_session.h"
#include "core/hle/service/audio/errors.h"
namespace Core {
class System;
}
namespace Kernel {
class KEvent;
}
namespace AudioCore::AudioIn {
constexpr SessionTypes SessionType = SessionTypes::AudioIn;
struct AudioInParameter {
/* 0x0 */ s32_le sample_rate;
/* 0x4 */ u16_le channel_count;
/* 0x6 */ u16_le reserved;
};
static_assert(sizeof(AudioInParameter) == 0x8, "AudioInParameter is an invalid size");
struct AudioInParameterInternal {
/* 0x0 */ u32_le sample_rate;
/* 0x4 */ u32_le channel_count;
/* 0x8 */ u32_le sample_format;
/* 0xC */ u32_le state;
};
static_assert(sizeof(AudioInParameterInternal) == 0x10,
"AudioInParameterInternal is an invalid size");
struct AudioInBuffer {
/* 0x00 */ AudioInBuffer* next;
/* 0x08 */ VAddr samples;
/* 0x10 */ u64 capacity;
/* 0x18 */ u64 size;
/* 0x20 */ u64 offset;
};
static_assert(sizeof(AudioInBuffer) == 0x28, "AudioInBuffer is an invalid size");
enum class State {
Started,
Stopped,
};
/**
* Controls and drives audio input.
*/
class System {
public:
explicit System(Core::System& system, Kernel::KEvent* event, size_t session_id);
~System();
/**
* Get the default audio input device name.
*
* @return The default audio input device name.
*/
std::string_view GetDefaultDeviceName();
/**
* Get the default USB audio input device name.
* This is preferred over non-USB as some games refuse to work with the BuiltInHeadset
* (e.g Let's Sing).
*
* @return The default USB audio input device name.
*/
std::string_view GetDefaultUacDeviceName();
/**
* Is the given initialize config valid?
*
* @param device_name - The name of the requested input device.
* @param in_params - Input parameters, see AudioInParameter.
* @return Result code.
*/
Result IsConfigValid(std::string_view device_name, const AudioInParameter& in_params);
/**
* Initialize this system.
*
* @param device_name - The name of the requested input device.
* @param in_params - Input parameters, see AudioInParameter.
* @param handle - Unused.
* @param applet_resource_user_id - Unused.
* @return Result code.
*/
Result Initialize(std::string& device_name, const AudioInParameter& in_params, u32 handle,
u64 applet_resource_user_id);
/**
* Start this system.
*
* @return Result code.
*/
Result Start();
/**
* Stop this system.
*
* @return Result code.
*/
Result Stop();
/**
* Finalize this system.
*/
void Finalize();
/**
* Start this system's device session.
*/
void StartSession();
/**
* Get this system's id.
*/
size_t GetSessionId() const;
/**
* Append a new buffer to the device.
*
* @param buffer - New buffer to append.
* @param tag - Unique tag of the buffer.
* @return True if the buffer was appended, otherwise false.
*/
bool AppendBuffer(const AudioInBuffer& buffer, u64 tag);
/**
* Register all appended buffers.
*/
void RegisterBuffers();
/**
* Release all registered buffers.
*/
void ReleaseBuffers();
/**
* Get all released buffers.
*
* @param tags - Container to be filled with the released buffers' tags.
* @return The number of buffers released.
*/
u32 GetReleasedBuffers(std::span<u64> tags);
/**
* Flush all appended and registered buffers.
*
* @return True if buffers were successfully flushed, otherwise false.
*/
bool FlushAudioInBuffers();
/**
* Get this system's current channel count.
*
* @return The channel count.
*/
u16 GetChannelCount() const;
/**
* Get this system's current sample rate.
*
* @return The sample rate.
*/
u32 GetSampleRate() const;
/**
* Get this system's current sample format.
*
* @return The sample format.
*/
SampleFormat GetSampleFormat() const;
/**
* Get this system's current state.
*
* @return The current state.
*/
State GetState();
/**
* Get this system's name.
*
* @return The system's name.
*/
std::string GetName() const;
/**
* Get this system's current volume.
*
* @return The system's current volume.
*/
f32 GetVolume() const;
/**
* Set this system's current volume.
*
* @param The new volume.
*/
void SetVolume(f32 volume);
/**
* Does the system contain this buffer?
*
* @param tag - Unique tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
bool ContainsAudioBuffer(u64 tag);
/**
* Get the maximum number of usable buffers (default 32).
*
* @return The number of buffers.
*/
u32 GetBufferCount();
/**
* Get the total number of samples played by this system.
*
* @return The number of samples.
*/
u64 GetPlayedSampleCount() const;
/**
* Is this system using a USB device?
*
* @return True if using a USB device, otherwise false.
*/
bool IsUac() const;
private:
/// Core system
Core::System& system;
/// (Unused)
u32 handle{};
/// (Unused)
u64 applet_resource_user_id{};
/// Buffer event, signalled when a buffer is ready
Kernel::KEvent* buffer_event;
/// Session id of this system
size_t session_id{};
/// Device session for this system
std::unique_ptr<DeviceSession> session;
/// Audio buffers in use by this system
AudioBuffers<BufferCount> buffers{BufferCount};
/// Sample rate of this system
u32 sample_rate{};
/// Sample format of this system
SampleFormat sample_format{SampleFormat::PcmInt16};
/// Channel count of this system
u16 channel_count{};
/// State of this system
std::atomic<State> state{State::Stopped};
/// Name of this system
std::string name{};
/// Volume of this system
f32 volume{1.0f};
/// Is this system's device USB?
bool is_uac{false};
};
} // namespace AudioCore::AudioIn

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@ -1,511 +0,0 @@
// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/behavior_info.h"
#include "audio_core/effect_context.h"
#include "audio_core/info_updater.h"
#include "audio_core/memory_pool.h"
#include "audio_core/mix_context.h"
#include "audio_core/sink_context.h"
#include "audio_core/splitter_context.h"
#include "audio_core/voice_context.h"
#include "common/logging/log.h"
namespace AudioCore {
InfoUpdater::InfoUpdater(const std::vector<u8>& in_params_, std::vector<u8>& out_params_,
BehaviorInfo& behavior_info_)
: in_params(in_params_), out_params(out_params_), behavior_info(behavior_info_) {
ASSERT(
AudioCommon::CanConsumeBuffer(in_params.size(), 0, sizeof(AudioCommon::UpdateDataHeader)));
std::memcpy(&input_header, in_params.data(), sizeof(AudioCommon::UpdateDataHeader));
output_header.total_size = sizeof(AudioCommon::UpdateDataHeader);
}
InfoUpdater::~InfoUpdater() = default;
bool InfoUpdater::UpdateBehaviorInfo(BehaviorInfo& in_behavior_info) {
if (input_header.size.behavior != sizeof(BehaviorInfo::InParams)) {
LOG_ERROR(Audio, "Behavior info is an invalid size, expecting 0x{:X} but got 0x{:X}",
sizeof(BehaviorInfo::InParams), input_header.size.behavior);
return false;
}
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset,
sizeof(BehaviorInfo::InParams))) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
BehaviorInfo::InParams behavior_in{};
std::memcpy(&behavior_in, in_params.data() + input_offset, sizeof(BehaviorInfo::InParams));
input_offset += sizeof(BehaviorInfo::InParams);
// Make sure it's an audio revision we can actually support
if (!AudioCommon::IsValidRevision(behavior_in.revision)) {
LOG_ERROR(Audio, "Invalid input revision, revision=0x{:08X}", behavior_in.revision);
return false;
}
// Make sure that our behavior info revision matches the input
if (in_behavior_info.GetUserRevision() != behavior_in.revision) {
LOG_ERROR(Audio,
"User revision differs from input revision, expecting 0x{:08X} but got 0x{:08X}",
in_behavior_info.GetUserRevision(), behavior_in.revision);
return false;
}
// Update behavior info flags
in_behavior_info.ClearError();
in_behavior_info.UpdateFlags(behavior_in.flags);
return true;
}
bool InfoUpdater::UpdateMemoryPools(std::vector<ServerMemoryPoolInfo>& memory_pool_info) {
const auto memory_pool_count = memory_pool_info.size();
const auto total_memory_pool_in = sizeof(ServerMemoryPoolInfo::InParams) * memory_pool_count;
const auto total_memory_pool_out = sizeof(ServerMemoryPoolInfo::OutParams) * memory_pool_count;
if (input_header.size.memory_pool != total_memory_pool_in) {
LOG_ERROR(Audio, "Memory pools are an invalid size, expecting 0x{:X} but got 0x{:X}",
total_memory_pool_in, input_header.size.memory_pool);
return false;
}
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset, total_memory_pool_in)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::vector<ServerMemoryPoolInfo::InParams> mempool_in(memory_pool_count);
std::vector<ServerMemoryPoolInfo::OutParams> mempool_out(memory_pool_count);
std::memcpy(mempool_in.data(), in_params.data() + input_offset, total_memory_pool_in);
input_offset += total_memory_pool_in;
// Update our memory pools
for (std::size_t i = 0; i < memory_pool_count; i++) {
if (!memory_pool_info[i].Update(mempool_in[i], mempool_out[i])) {
LOG_ERROR(Audio, "Failed to update memory pool {}!", i);
return false;
}
}
if (!AudioCommon::CanConsumeBuffer(out_params.size(), output_offset,
sizeof(BehaviorInfo::InParams))) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::memcpy(out_params.data() + output_offset, mempool_out.data(), total_memory_pool_out);
output_offset += total_memory_pool_out;
output_header.size.memory_pool = static_cast<u32>(total_memory_pool_out);
return true;
}
bool InfoUpdater::UpdateVoiceChannelResources(VoiceContext& voice_context) {
const auto voice_count = voice_context.GetVoiceCount();
const auto voice_size = voice_count * sizeof(VoiceChannelResource::InParams);
std::vector<VoiceChannelResource::InParams> resources_in(voice_count);
if (input_header.size.voice_channel_resource != voice_size) {
LOG_ERROR(Audio, "VoiceChannelResource is an invalid size, expecting 0x{:X} but got 0x{:X}",
voice_size, input_header.size.voice_channel_resource);
return false;
}
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset, voice_size)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::memcpy(resources_in.data(), in_params.data() + input_offset, voice_size);
input_offset += voice_size;
// Update our channel resources
for (std::size_t i = 0; i < voice_count; i++) {
// Grab our channel resource
auto& resource = voice_context.GetChannelResource(i);
resource.Update(resources_in[i]);
}
return true;
}
bool InfoUpdater::UpdateVoices(VoiceContext& voice_context,
[[maybe_unused]] std::vector<ServerMemoryPoolInfo>& memory_pool_info,
[[maybe_unused]] VAddr audio_codec_dsp_addr) {
const auto voice_count = voice_context.GetVoiceCount();
std::vector<VoiceInfo::InParams> voice_in(voice_count);
std::vector<VoiceInfo::OutParams> voice_out(voice_count);
const auto voice_in_size = voice_count * sizeof(VoiceInfo::InParams);
const auto voice_out_size = voice_count * sizeof(VoiceInfo::OutParams);
if (input_header.size.voice != voice_in_size) {
LOG_ERROR(Audio, "Voices are an invalid size, expecting 0x{:X} but got 0x{:X}",
voice_in_size, input_header.size.voice);
return false;
}
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset, voice_in_size)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::memcpy(voice_in.data(), in_params.data() + input_offset, voice_in_size);
input_offset += voice_in_size;
// Set all voices to not be in use
for (std::size_t i = 0; i < voice_count; i++) {
voice_context.GetInfo(i).GetInParams().in_use = false;
}
// Update our voices
for (std::size_t i = 0; i < voice_count; i++) {
auto& voice_in_params = voice_in[i];
const auto channel_count = static_cast<std::size_t>(voice_in_params.channel_count);
// Skip if it's not currently in use
if (!voice_in_params.is_in_use) {
continue;
}
// Voice states for each channel
std::array<VoiceState*, AudioCommon::MAX_CHANNEL_COUNT> voice_states{};
ASSERT(static_cast<std::size_t>(voice_in_params.id) < voice_count);
// Grab our current voice info
auto& voice_info = voice_context.GetInfo(static_cast<std::size_t>(voice_in_params.id));
ASSERT(channel_count <= AudioCommon::MAX_CHANNEL_COUNT);
// Get all our channel voice states
for (std::size_t channel = 0; channel < channel_count; channel++) {
voice_states[channel] =
&voice_context.GetState(voice_in_params.voice_channel_resource_ids[channel]);
}
if (voice_in_params.is_new) {
// Default our values for our voice
voice_info.Initialize();
// Zero out our voice states
for (std::size_t channel = 0; channel < channel_count; channel++) {
std::memset(voice_states[channel], 0, sizeof(VoiceState));
}
}
// Update our voice
voice_info.UpdateParameters(voice_in_params, behavior_info);
// TODO(ogniK): Handle mapping errors with behavior info based on in params response
// Update our wave buffers
voice_info.UpdateWaveBuffers(voice_in_params, voice_states, behavior_info);
voice_info.WriteOutStatus(voice_out[i], voice_in_params, voice_states);
}
if (!AudioCommon::CanConsumeBuffer(out_params.size(), output_offset, voice_out_size)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::memcpy(out_params.data() + output_offset, voice_out.data(), voice_out_size);
output_offset += voice_out_size;
output_header.size.voice = static_cast<u32>(voice_out_size);
return true;
}
bool InfoUpdater::UpdateEffects(EffectContext& effect_context, bool is_active) {
const auto effect_count = effect_context.GetCount();
std::vector<EffectInfo::InParams> effect_in(effect_count);
std::vector<EffectInfo::OutParams> effect_out(effect_count);
const auto total_effect_in = effect_count * sizeof(EffectInfo::InParams);
const auto total_effect_out = effect_count * sizeof(EffectInfo::OutParams);
if (input_header.size.effect != total_effect_in) {
LOG_ERROR(Audio, "Effects are an invalid size, expecting 0x{:X} but got 0x{:X}",
total_effect_in, input_header.size.effect);
return false;
}
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset, total_effect_in)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::memcpy(effect_in.data(), in_params.data() + input_offset, total_effect_in);
input_offset += total_effect_in;
// Update effects
for (std::size_t i = 0; i < effect_count; i++) {
auto* info = effect_context.GetInfo(i);
if (effect_in[i].type != info->GetType()) {
info = effect_context.RetargetEffect(i, effect_in[i].type);
}
info->Update(effect_in[i]);
if ((!is_active && info->GetUsage() != UsageState::Initialized) ||
info->GetUsage() == UsageState::Stopped) {
effect_out[i].status = UsageStatus::Removed;
} else {
effect_out[i].status = UsageStatus::Used;
}
}
if (!AudioCommon::CanConsumeBuffer(out_params.size(), output_offset, total_effect_out)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::memcpy(out_params.data() + output_offset, effect_out.data(), total_effect_out);
output_offset += total_effect_out;
output_header.size.effect = static_cast<u32>(total_effect_out);
return true;
}
bool InfoUpdater::UpdateSplitterInfo(SplitterContext& splitter_context) {
std::size_t start_offset = input_offset;
std::size_t bytes_read{};
// Update splitter context
if (!splitter_context.Update(in_params, input_offset, bytes_read)) {
LOG_ERROR(Audio, "Failed to update splitter context!");
return false;
}
const auto consumed = input_offset - start_offset;
if (input_header.size.splitter != consumed) {
LOG_ERROR(Audio, "Splitters is an invalid size, expecting 0x{:X} but got 0x{:X}",
bytes_read, input_header.size.splitter);
return false;
}
return true;
}
Result InfoUpdater::UpdateMixes(MixContext& mix_context, std::size_t mix_buffer_count,
SplitterContext& splitter_context, EffectContext& effect_context) {
std::vector<MixInfo::InParams> mix_in_params;
if (!behavior_info.IsMixInParameterDirtyOnlyUpdateSupported()) {
// If we're not dirty, get ALL mix in parameters
const auto context_mix_count = mix_context.GetCount();
const auto total_mix_in = context_mix_count * sizeof(MixInfo::InParams);
if (input_header.size.mixer != total_mix_in) {
LOG_ERROR(Audio, "Mixer is an invalid size, expecting 0x{:X} but got 0x{:X}",
total_mix_in, input_header.size.mixer);
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset, total_mix_in)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
mix_in_params.resize(context_mix_count);
std::memcpy(mix_in_params.data(), in_params.data() + input_offset, total_mix_in);
input_offset += total_mix_in;
} else {
// Only update the "dirty" mixes
MixInfo::DirtyHeader dirty_header{};
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset,
sizeof(MixInfo::DirtyHeader))) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
std::memcpy(&dirty_header, in_params.data() + input_offset, sizeof(MixInfo::DirtyHeader));
input_offset += sizeof(MixInfo::DirtyHeader);
const auto total_mix_in =
dirty_header.mixer_count * sizeof(MixInfo::InParams) + sizeof(MixInfo::DirtyHeader);
if (input_header.size.mixer != total_mix_in) {
LOG_ERROR(Audio, "Mixer is an invalid size, expecting 0x{:X} but got 0x{:X}",
total_mix_in, input_header.size.mixer);
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
if (dirty_header.mixer_count != 0) {
mix_in_params.resize(dirty_header.mixer_count);
std::memcpy(mix_in_params.data(), in_params.data() + input_offset,
mix_in_params.size() * sizeof(MixInfo::InParams));
input_offset += mix_in_params.size() * sizeof(MixInfo::InParams);
}
}
// Get our total input count
const auto mix_count = mix_in_params.size();
if (!behavior_info.IsMixInParameterDirtyOnlyUpdateSupported()) {
// Only verify our buffer count if we're not dirty
std::size_t total_buffer_count{};
for (std::size_t i = 0; i < mix_count; i++) {
const auto& in = mix_in_params[i];
total_buffer_count += in.buffer_count;
if (static_cast<std::size_t>(in.dest_mix_id) > mix_count &&
in.dest_mix_id != AudioCommon::NO_MIX && in.mix_id != AudioCommon::FINAL_MIX) {
LOG_ERROR(
Audio,
"Invalid mix destination, mix_id={:X}, dest_mix_id={:X}, mix_buffer_count={:X}",
in.mix_id, in.dest_mix_id, mix_buffer_count);
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
}
if (total_buffer_count > mix_buffer_count) {
LOG_ERROR(Audio,
"Too many mix buffers used! mix_buffer_count={:X}, requesting_buffers={:X}",
mix_buffer_count, total_buffer_count);
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
}
if (mix_buffer_count == 0) {
LOG_ERROR(Audio, "No mix buffers!");
return AudioCommon::Audren::ERR_INVALID_PARAMETERS;
}
bool should_sort = false;
for (std::size_t i = 0; i < mix_count; i++) {
const auto& mix_in = mix_in_params[i];
std::size_t target_mix{};
if (behavior_info.IsMixInParameterDirtyOnlyUpdateSupported()) {
target_mix = mix_in.mix_id;
} else {
// Non dirty supported games just use i instead of the actual mix_id
target_mix = i;
}
auto& mix_info = mix_context.GetInfo(target_mix);
auto& mix_info_params = mix_info.GetInParams();
if (mix_info_params.in_use != mix_in.in_use) {
mix_info_params.in_use = mix_in.in_use;
mix_info.ResetEffectProcessingOrder();
should_sort = true;
}
if (mix_in.in_use) {
should_sort |= mix_info.Update(mix_context.GetEdgeMatrix(), mix_in, behavior_info,
splitter_context, effect_context);
}
}
if (should_sort && behavior_info.IsSplitterSupported()) {
// Sort our splitter data
if (!mix_context.TsortInfo(splitter_context)) {
return AudioCommon::Audren::ERR_SPLITTER_SORT_FAILED;
}
}
// TODO(ogniK): Sort when splitter is suppoorted
return ResultSuccess;
}
bool InfoUpdater::UpdateSinks(SinkContext& sink_context) {
const auto sink_count = sink_context.GetCount();
std::vector<SinkInfo::InParams> sink_in_params(sink_count);
const auto total_sink_in = sink_count * sizeof(SinkInfo::InParams);
if (input_header.size.sink != total_sink_in) {
LOG_ERROR(Audio, "Sinks are an invalid size, expecting 0x{:X} but got 0x{:X}",
total_sink_in, input_header.size.effect);
return false;
}
if (!AudioCommon::CanConsumeBuffer(in_params.size(), input_offset, total_sink_in)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
std::memcpy(sink_in_params.data(), in_params.data() + input_offset, total_sink_in);
input_offset += total_sink_in;
// TODO(ogniK): Properly update sinks
if (!sink_in_params.empty()) {
sink_context.UpdateMainSink(sink_in_params[0]);
}
output_header.size.sink = static_cast<u32>(0x20 * sink_count);
output_offset += 0x20 * sink_count;
return true;
}
bool InfoUpdater::UpdatePerformanceBuffer() {
output_header.size.performance = 0x10;
output_offset += 0x10;
return true;
}
bool InfoUpdater::UpdateErrorInfo([[maybe_unused]] BehaviorInfo& in_behavior_info) {
const auto total_beahvior_info_out = sizeof(BehaviorInfo::OutParams);
if (!AudioCommon::CanConsumeBuffer(out_params.size(), output_offset, total_beahvior_info_out)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
BehaviorInfo::OutParams behavior_info_out{};
behavior_info.CopyErrorInfo(behavior_info_out);
std::memcpy(out_params.data() + output_offset, &behavior_info_out, total_beahvior_info_out);
output_offset += total_beahvior_info_out;
output_header.size.behavior = total_beahvior_info_out;
return true;
}
struct RendererInfo {
u64_le elasped_frame_count{};
INSERT_PADDING_WORDS(2);
};
static_assert(sizeof(RendererInfo) == 0x10, "RendererInfo is an invalid size");
bool InfoUpdater::UpdateRendererInfo(std::size_t elapsed_frame_count) {
const auto total_renderer_info_out = sizeof(RendererInfo);
if (!AudioCommon::CanConsumeBuffer(out_params.size(), output_offset, total_renderer_info_out)) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
RendererInfo out{};
out.elasped_frame_count = elapsed_frame_count;
std::memcpy(out_params.data() + output_offset, &out, total_renderer_info_out);
output_offset += total_renderer_info_out;
output_header.size.render_info = total_renderer_info_out;
return true;
}
bool InfoUpdater::CheckConsumedSize() const {
if (output_offset != out_params.size()) {
LOG_ERROR(Audio, "Output is not consumed! Consumed {}, but requires {}. {} bytes remaining",
output_offset, out_params.size(), out_params.size() - output_offset);
return false;
}
/*if (input_offset != in_params.size()) {
LOG_ERROR(Audio, "Input is not consumed!");
return false;
}*/
return true;
}
bool InfoUpdater::WriteOutputHeader() {
if (!AudioCommon::CanConsumeBuffer(out_params.size(), 0,
sizeof(AudioCommon::UpdateDataHeader))) {
LOG_ERROR(Audio, "Buffer is an invalid size!");
return false;
}
output_header.revision = AudioCommon::CURRENT_PROCESS_REVISION;
const auto& sz = output_header.size;
output_header.total_size += sz.behavior + sz.memory_pool + sz.voice +
sz.voice_channel_resource + sz.effect + sz.mixer + sz.sink +
sz.performance + sz.splitter + sz.render_info;
std::memcpy(out_params.data(), &output_header, sizeof(AudioCommon::UpdateDataHeader));
return true;
}
} // namespace AudioCore

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@ -1,57 +0,0 @@
// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <vector>
#include "audio_core/common.h"
#include "common/common_types.h"
namespace AudioCore {
class BehaviorInfo;
class ServerMemoryPoolInfo;
class VoiceContext;
class EffectContext;
class MixContext;
class SinkContext;
class SplitterContext;
class InfoUpdater {
public:
// TODO(ogniK): Pass process handle when we support it
InfoUpdater(const std::vector<u8>& in_params_, std::vector<u8>& out_params_,
BehaviorInfo& behavior_info_);
~InfoUpdater();
bool UpdateBehaviorInfo(BehaviorInfo& in_behavior_info);
bool UpdateMemoryPools(std::vector<ServerMemoryPoolInfo>& memory_pool_info);
bool UpdateVoiceChannelResources(VoiceContext& voice_context);
bool UpdateVoices(VoiceContext& voice_context,
std::vector<ServerMemoryPoolInfo>& memory_pool_info,
VAddr audio_codec_dsp_addr);
bool UpdateEffects(EffectContext& effect_context, bool is_active);
bool UpdateSplitterInfo(SplitterContext& splitter_context);
Result UpdateMixes(MixContext& mix_context, std::size_t mix_buffer_count,
SplitterContext& splitter_context, EffectContext& effect_context);
bool UpdateSinks(SinkContext& sink_context);
bool UpdatePerformanceBuffer();
bool UpdateErrorInfo(BehaviorInfo& in_behavior_info);
bool UpdateRendererInfo(std::size_t elapsed_frame_count);
bool CheckConsumedSize() const;
bool WriteOutputHeader();
private:
const std::vector<u8>& in_params;
std::vector<u8>& out_params;
BehaviorInfo& behavior_info;
AudioCommon::UpdateDataHeader input_header{};
AudioCommon::UpdateDataHeader output_header{};
std::size_t input_offset{sizeof(AudioCommon::UpdateDataHeader)};
std::size_t output_offset{sizeof(AudioCommon::UpdateDataHeader)};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/memory_pool.h"
#include "common/logging/log.h"
namespace AudioCore {
ServerMemoryPoolInfo::ServerMemoryPoolInfo() = default;
ServerMemoryPoolInfo::~ServerMemoryPoolInfo() = default;
bool ServerMemoryPoolInfo::Update(const InParams& in_params, OutParams& out_params) {
// Our state does not need to be changed
if (in_params.state != State::RequestAttach && in_params.state != State::RequestDetach) {
return true;
}
// Address or size is null
if (in_params.address == 0 || in_params.size == 0) {
LOG_ERROR(Audio, "Memory pool address or size is zero! address={:X}, size={:X}",
in_params.address, in_params.size);
return false;
}
// Address or size is not aligned
if ((in_params.address % 0x1000) != 0 || (in_params.size % 0x1000) != 0) {
LOG_ERROR(Audio, "Memory pool address or size is not aligned! address={:X}, size={:X}",
in_params.address, in_params.size);
return false;
}
if (in_params.state == State::RequestAttach) {
cpu_address = in_params.address;
size = in_params.size;
used = true;
out_params.state = State::Attached;
} else {
// Unexpected address
if (cpu_address != in_params.address) {
LOG_ERROR(Audio, "Memory pool address differs! Expecting {:X} but address is {:X}",
cpu_address, in_params.address);
return false;
}
if (size != in_params.size) {
LOG_ERROR(Audio, "Memory pool size differs! Expecting {:X} but size is {:X}", size,
in_params.size);
return false;
}
cpu_address = 0;
size = 0;
used = false;
out_params.state = State::Detached;
}
return true;
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "common/common_funcs.h"
#include "common/common_types.h"
#include "common/swap.h"
namespace AudioCore {
class ServerMemoryPoolInfo {
public:
ServerMemoryPoolInfo();
~ServerMemoryPoolInfo();
enum class State : u32_le {
Invalid = 0x0,
Aquired = 0x1,
RequestDetach = 0x2,
Detached = 0x3,
RequestAttach = 0x4,
Attached = 0x5,
Released = 0x6,
};
struct InParams {
u64_le address{};
u64_le size{};
State state{};
INSERT_PADDING_WORDS(3);
};
static_assert(sizeof(InParams) == 0x20, "InParams are an invalid size");
struct OutParams {
State state{};
INSERT_PADDING_WORDS(3);
};
static_assert(sizeof(OutParams) == 0x10, "OutParams are an invalid size");
bool Update(const InParams& in_params, OutParams& out_params);
private:
// There's another entry here which is the DSP address, however since we're not talking to the
// DSP we can just use the same address provided by the guest without needing to remap
u64_le cpu_address{};
u64_le size{};
bool used{};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <algorithm>
#include "audio_core/behavior_info.h"
#include "audio_core/common.h"
#include "audio_core/effect_context.h"
#include "audio_core/mix_context.h"
#include "audio_core/splitter_context.h"
namespace AudioCore {
MixContext::MixContext() = default;
MixContext::~MixContext() = default;
void MixContext::Initialize(const BehaviorInfo& behavior_info, std::size_t mix_count,
std::size_t effect_count) {
info_count = mix_count;
infos.resize(info_count);
auto& final_mix = GetInfo(AudioCommon::FINAL_MIX);
final_mix.GetInParams().mix_id = AudioCommon::FINAL_MIX;
sorted_info.reserve(infos.size());
for (auto& info : infos) {
sorted_info.push_back(&info);
}
for (auto& info : infos) {
info.SetEffectCount(effect_count);
}
// Only initialize our edge matrix and node states if splitters are supported
if (behavior_info.IsSplitterSupported()) {
node_states.Initialize(mix_count);
edge_matrix.Initialize(mix_count);
}
}
void MixContext::UpdateDistancesFromFinalMix() {
// Set all distances to be invalid
for (std::size_t i = 0; i < info_count; i++) {
GetInfo(i).GetInParams().final_mix_distance = AudioCommon::NO_FINAL_MIX;
}
for (std::size_t i = 0; i < info_count; i++) {
auto& info = GetInfo(i);
auto& in_params = info.GetInParams();
// Populate our sorted info
sorted_info[i] = &info;
if (!in_params.in_use) {
continue;
}
auto mix_id = in_params.mix_id;
// Needs to be referenced out of scope
s32 distance_to_final_mix{AudioCommon::FINAL_MIX};
for (; distance_to_final_mix < static_cast<s32>(info_count); distance_to_final_mix++) {
if (mix_id == AudioCommon::FINAL_MIX) {
// If we're at the final mix, we're done
break;
} else if (mix_id == AudioCommon::NO_MIX) {
// If we have no more mix ids, we're done
distance_to_final_mix = AudioCommon::NO_FINAL_MIX;
break;
} else {
const auto& dest_mix = GetInfo(mix_id);
const auto dest_mix_distance = dest_mix.GetInParams().final_mix_distance;
if (dest_mix_distance == AudioCommon::NO_FINAL_MIX) {
// If our current mix isn't pointing to a final mix, follow through
mix_id = dest_mix.GetInParams().dest_mix_id;
} else {
// Our current mix + 1 = final distance
distance_to_final_mix = dest_mix_distance + 1;
break;
}
}
}
// If we're out of range for our distance, mark it as no final mix
if (distance_to_final_mix >= static_cast<s32>(info_count)) {
distance_to_final_mix = AudioCommon::NO_FINAL_MIX;
}
in_params.final_mix_distance = distance_to_final_mix;
}
}
void MixContext::CalcMixBufferOffset() {
s32 offset{};
for (std::size_t i = 0; i < info_count; i++) {
auto& info = GetSortedInfo(i);
auto& in_params = info.GetInParams();
if (in_params.in_use) {
// Only update if in use
in_params.buffer_offset = offset;
offset += in_params.buffer_count;
}
}
}
void MixContext::SortInfo() {
// Get the distance to the final mix
UpdateDistancesFromFinalMix();
// Sort based on the distance to the final mix
std::sort(sorted_info.begin(), sorted_info.end(),
[](const ServerMixInfo* lhs, const ServerMixInfo* rhs) {
return lhs->GetInParams().final_mix_distance >
rhs->GetInParams().final_mix_distance;
});
// Calculate the mix buffer offset
CalcMixBufferOffset();
}
bool MixContext::TsortInfo(SplitterContext& splitter_context) {
// If we're not using mixes, just calculate the mix buffer offset
if (!splitter_context.UsingSplitter()) {
CalcMixBufferOffset();
return true;
}
// Sort our node states
if (!node_states.Tsort(edge_matrix)) {
return false;
}
// Get our sorted list
const auto sorted_list = node_states.GetIndexList();
std::size_t info_id{};
for (auto itr = sorted_list.rbegin(); itr != sorted_list.rend(); ++itr) {
// Set our sorted info
sorted_info[info_id++] = &GetInfo(*itr);
}
// Calculate the mix buffer offset
CalcMixBufferOffset();
return true;
}
std::size_t MixContext::GetCount() const {
return info_count;
}
ServerMixInfo& MixContext::GetInfo(std::size_t i) {
ASSERT(i < info_count);
return infos.at(i);
}
const ServerMixInfo& MixContext::GetInfo(std::size_t i) const {
ASSERT(i < info_count);
return infos.at(i);
}
ServerMixInfo& MixContext::GetSortedInfo(std::size_t i) {
ASSERT(i < info_count);
return *sorted_info.at(i);
}
const ServerMixInfo& MixContext::GetSortedInfo(std::size_t i) const {
ASSERT(i < info_count);
return *sorted_info.at(i);
}
ServerMixInfo& MixContext::GetFinalMixInfo() {
return infos.at(AudioCommon::FINAL_MIX);
}
const ServerMixInfo& MixContext::GetFinalMixInfo() const {
return infos.at(AudioCommon::FINAL_MIX);
}
EdgeMatrix& MixContext::GetEdgeMatrix() {
return edge_matrix;
}
const EdgeMatrix& MixContext::GetEdgeMatrix() const {
return edge_matrix;
}
ServerMixInfo::ServerMixInfo() {
Cleanup();
}
ServerMixInfo::~ServerMixInfo() = default;
const ServerMixInfo::InParams& ServerMixInfo::GetInParams() const {
return in_params;
}
ServerMixInfo::InParams& ServerMixInfo::GetInParams() {
return in_params;
}
bool ServerMixInfo::Update(EdgeMatrix& edge_matrix, const MixInfo::InParams& mix_in,
BehaviorInfo& behavior_info, SplitterContext& splitter_context,
EffectContext& effect_context) {
in_params.volume = mix_in.volume;
in_params.sample_rate = mix_in.sample_rate;
in_params.buffer_count = mix_in.buffer_count;
in_params.in_use = mix_in.in_use;
in_params.mix_id = mix_in.mix_id;
in_params.node_id = mix_in.node_id;
for (std::size_t i = 0; i < mix_in.mix_volume.size(); i++) {
std::copy(mix_in.mix_volume[i].begin(), mix_in.mix_volume[i].end(),
in_params.mix_volume[i].begin());
}
bool require_sort = false;
if (behavior_info.IsSplitterSupported()) {
require_sort = UpdateConnection(edge_matrix, mix_in, splitter_context);
} else {
in_params.dest_mix_id = mix_in.dest_mix_id;
in_params.splitter_id = AudioCommon::NO_SPLITTER;
}
ResetEffectProcessingOrder();
const auto effect_count = effect_context.GetCount();
for (std::size_t i = 0; i < effect_count; i++) {
auto* effect_info = effect_context.GetInfo(i);
if (effect_info->GetMixID() == in_params.mix_id) {
effect_processing_order[effect_info->GetProcessingOrder()] = static_cast<s32>(i);
}
}
// TODO(ogniK): Update effect processing order
return require_sort;
}
bool ServerMixInfo::HasAnyConnection() const {
return in_params.splitter_id != AudioCommon::NO_SPLITTER ||
in_params.mix_id != AudioCommon::NO_MIX;
}
void ServerMixInfo::Cleanup() {
in_params.volume = 0.0f;
in_params.sample_rate = 0;
in_params.buffer_count = 0;
in_params.in_use = false;
in_params.mix_id = AudioCommon::NO_MIX;
in_params.node_id = 0;
in_params.buffer_offset = 0;
in_params.dest_mix_id = AudioCommon::NO_MIX;
in_params.splitter_id = AudioCommon::NO_SPLITTER;
std::memset(in_params.mix_volume.data(), 0, sizeof(float) * in_params.mix_volume.size());
}
void ServerMixInfo::SetEffectCount(std::size_t count) {
effect_processing_order.resize(count);
ResetEffectProcessingOrder();
}
void ServerMixInfo::ResetEffectProcessingOrder() {
for (auto& order : effect_processing_order) {
order = AudioCommon::NO_EFFECT_ORDER;
}
}
s32 ServerMixInfo::GetEffectOrder(std::size_t i) const {
return effect_processing_order.at(i);
}
bool ServerMixInfo::UpdateConnection(EdgeMatrix& edge_matrix, const MixInfo::InParams& mix_in,
SplitterContext& splitter_context) {
// Mixes are identical
if (in_params.dest_mix_id == mix_in.dest_mix_id &&
in_params.splitter_id == mix_in.splitter_id &&
((in_params.splitter_id == AudioCommon::NO_SPLITTER) ||
!splitter_context.GetInfo(in_params.splitter_id).HasNewConnection())) {
return false;
}
// Remove current edges for mix id
edge_matrix.RemoveEdges(in_params.mix_id);
if (mix_in.dest_mix_id != AudioCommon::NO_MIX) {
// If we have a valid destination mix id, set our edge matrix
edge_matrix.Connect(in_params.mix_id, mix_in.dest_mix_id);
} else if (mix_in.splitter_id != AudioCommon::NO_SPLITTER) {
// Recurse our splitter linked and set our edges
auto& splitter_info = splitter_context.GetInfo(mix_in.splitter_id);
const auto length = splitter_info.GetLength();
for (s32 i = 0; i < length; i++) {
const auto* splitter_destination =
splitter_context.GetDestinationData(mix_in.splitter_id, i);
if (splitter_destination == nullptr) {
continue;
}
if (splitter_destination->ValidMixId()) {
edge_matrix.Connect(in_params.mix_id, splitter_destination->GetMixId());
}
}
}
in_params.dest_mix_id = mix_in.dest_mix_id;
in_params.splitter_id = mix_in.splitter_id;
return true;
}
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2020 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <vector>
#include "audio_core/common.h"
#include "audio_core/splitter_context.h"
#include "common/common_funcs.h"
#include "common/common_types.h"
namespace AudioCore {
class BehaviorInfo;
class EffectContext;
class MixInfo {
public:
struct DirtyHeader {
u32_le magic{};
u32_le mixer_count{};
INSERT_PADDING_BYTES(0x18);
};
static_assert(sizeof(DirtyHeader) == 0x20, "MixInfo::DirtyHeader is an invalid size");
struct InParams {
float_le volume{};
s32_le sample_rate{};
s32_le buffer_count{};
bool in_use{};
INSERT_PADDING_BYTES(3);
s32_le mix_id{};
s32_le effect_count{};
u32_le node_id{};
INSERT_PADDING_WORDS(2);
std::array<std::array<float_le, AudioCommon::MAX_MIX_BUFFERS>, AudioCommon::MAX_MIX_BUFFERS>
mix_volume{};
s32_le dest_mix_id{};
s32_le splitter_id{};
INSERT_PADDING_WORDS(1);
};
static_assert(sizeof(MixInfo::InParams) == 0x930, "MixInfo::InParams is an invalid size");
};
class ServerMixInfo {
public:
struct InParams {
float volume{};
s32 sample_rate{};
s32 buffer_count{};
bool in_use{};
s32 mix_id{};
u32 node_id{};
std::array<std::array<float_le, AudioCommon::MAX_MIX_BUFFERS>, AudioCommon::MAX_MIX_BUFFERS>
mix_volume{};
s32 dest_mix_id{};
s32 splitter_id{};
s32 buffer_offset{};
s32 final_mix_distance{};
};
ServerMixInfo();
~ServerMixInfo();
[[nodiscard]] const ServerMixInfo::InParams& GetInParams() const;
[[nodiscard]] ServerMixInfo::InParams& GetInParams();
bool Update(EdgeMatrix& edge_matrix, const MixInfo::InParams& mix_in,
BehaviorInfo& behavior_info, SplitterContext& splitter_context,
EffectContext& effect_context);
[[nodiscard]] bool HasAnyConnection() const;
void Cleanup();
void SetEffectCount(std::size_t count);
void ResetEffectProcessingOrder();
[[nodiscard]] s32 GetEffectOrder(std::size_t i) const;
private:
std::vector<s32> effect_processing_order;
InParams in_params{};
bool UpdateConnection(EdgeMatrix& edge_matrix, const MixInfo::InParams& mix_in,
SplitterContext& splitter_context);
};
class MixContext {
public:
MixContext();
~MixContext();
void Initialize(const BehaviorInfo& behavior_info, std::size_t mix_count,
std::size_t effect_count);
void SortInfo();
bool TsortInfo(SplitterContext& splitter_context);
[[nodiscard]] std::size_t GetCount() const;
[[nodiscard]] ServerMixInfo& GetInfo(std::size_t i);
[[nodiscard]] const ServerMixInfo& GetInfo(std::size_t i) const;
[[nodiscard]] ServerMixInfo& GetSortedInfo(std::size_t i);
[[nodiscard]] const ServerMixInfo& GetSortedInfo(std::size_t i) const;
[[nodiscard]] ServerMixInfo& GetFinalMixInfo();
[[nodiscard]] const ServerMixInfo& GetFinalMixInfo() const;
[[nodiscard]] EdgeMatrix& GetEdgeMatrix();
[[nodiscard]] const EdgeMatrix& GetEdgeMatrix() const;
private:
void CalcMixBufferOffset();
void UpdateDistancesFromFinalMix();
NodeStates node_states{};
EdgeMatrix edge_matrix{};
std::size_t info_count{};
std::vector<ServerMixInfo> infos{};
std::vector<ServerMixInfo*> sorted_info{};
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include "audio_core/sink.h"
namespace AudioCore {
class NullSink final : public Sink {
public:
explicit NullSink(std::string_view) {}
~NullSink() override = default;
SinkStream& AcquireSinkStream(u32 /*sample_rate*/, u32 /*num_channels*/,
const std::string& /*name*/) override {
return null_sink_stream;
}
private:
struct NullSinkStreamImpl final : SinkStream {
void EnqueueSamples(u32 /*num_channels*/, const std::vector<s16>& /*samples*/) override {}
std::size_t SamplesInQueue(u32 /*num_channels*/) const override {
return 0;
}
void Flush() override {}
} null_sink_stream;
};
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_out_manager.h"
#include "audio_core/out/audio_out.h"
#include "core/hle/kernel/k_event.h"
namespace AudioCore::AudioOut {
Out::Out(Core::System& system_, Manager& manager_, Kernel::KEvent* event_, size_t session_id_)
: manager{manager_}, parent_mutex{manager.mutex}, event{event_}, system{system_, event,
session_id_} {}
void Out::Free() {
std::scoped_lock l{parent_mutex};
manager.ReleaseSessionId(system.GetSessionId());
}
System& Out::GetSystem() {
return system;
}
AudioOut::State Out::GetState() {
std::scoped_lock l{parent_mutex};
return system.GetState();
}
Result Out::StartSystem() {
std::scoped_lock l{parent_mutex};
return system.Start();
}
void Out::StartSession() {
std::scoped_lock l{parent_mutex};
system.StartSession();
}
Result Out::StopSystem() {
std::scoped_lock l{parent_mutex};
return system.Stop();
}
Result Out::AppendBuffer(const AudioOutBuffer& buffer, const u64 tag) {
std::scoped_lock l{parent_mutex};
if (system.AppendBuffer(buffer, tag)) {
return ResultSuccess;
}
return Service::Audio::ERR_BUFFER_COUNT_EXCEEDED;
}
void Out::ReleaseAndRegisterBuffers() {
std::scoped_lock l{parent_mutex};
if (system.GetState() == State::Started) {
system.ReleaseBuffers();
system.RegisterBuffers();
}
}
bool Out::FlushAudioOutBuffers() {
std::scoped_lock l{parent_mutex};
return system.FlushAudioOutBuffers();
}
u32 Out::GetReleasedBuffers(std::span<u64> tags) {
std::scoped_lock l{parent_mutex};
return system.GetReleasedBuffers(tags);
}
Kernel::KReadableEvent& Out::GetBufferEvent() {
std::scoped_lock l{parent_mutex};
return event->GetReadableEvent();
}
f32 Out::GetVolume() {
std::scoped_lock l{parent_mutex};
return system.GetVolume();
}
void Out::SetVolume(const f32 volume) {
std::scoped_lock l{parent_mutex};
system.SetVolume(volume);
}
bool Out::ContainsAudioBuffer(const u64 tag) {
std::scoped_lock l{parent_mutex};
return system.ContainsAudioBuffer(tag);
}
u32 Out::GetBufferCount() {
std::scoped_lock l{parent_mutex};
return system.GetBufferCount();
}
u64 Out::GetPlayedSampleCount() {
std::scoped_lock l{parent_mutex};
return system.GetPlayedSampleCount();
}
} // namespace AudioCore::AudioOut

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <mutex>
#include "audio_core/out/audio_out_system.h"
namespace Core {
class System;
}
namespace Kernel {
class KEvent;
class KReadableEvent;
} // namespace Kernel
namespace AudioCore::AudioOut {
class Manager;
/**
* Interface between the service and audio out system. Mainly responsible for forwarding service
* calls to the system.
*/
class Out {
public:
explicit Out(Core::System& system, Manager& manager, Kernel::KEvent* event, size_t session_id);
/**
* Free this audio out from the audio out manager.
*/
void Free();
/**
* Get this audio out's system.
*/
System& GetSystem();
/**
* Get the current state.
*
* @return Started or Stopped.
*/
AudioOut::State GetState();
/**
* Start the system
*
* @return Result code
*/
Result StartSystem();
/**
* Start the system's device session.
*/
void StartSession();
/**
* Stop the system.
*
* @return Result code
*/
Result StopSystem();
/**
* Append a new buffer to the system, the buffer event will be signalled when it is filled.
*
* @param buffer - The new buffer to append.
* @param tag - Unique tag for this buffer.
* @return Result code.
*/
Result AppendBuffer(const AudioOutBuffer& buffer, u64 tag);
/**
* Release all completed buffers, and register any appended.
*/
void ReleaseAndRegisterBuffers();
/**
* Flush all buffers.
*/
bool FlushAudioOutBuffers();
/**
* Get all of the currently released buffers.
*
* @param tags - Output container for the buffer tags which were released.
* @return The number of buffers released.
*/
u32 GetReleasedBuffers(std::span<u64> tags);
/**
* Get the buffer event for this audio out, this event will be signalled when a buffer is
* filled.
* @return The buffer event.
*/
Kernel::KReadableEvent& GetBufferEvent();
/**
* Get the current system volume.
*
* @return The current volume.
*/
f32 GetVolume();
/**
* Set the system volume.
*
* @param volume - The volume to set.
*/
void SetVolume(f32 volume);
/**
* Check if a buffer is in the system.
*
* @param tag - The tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
bool ContainsAudioBuffer(u64 tag);
/**
* Get the maximum number of buffers.
*
* @return The maximum number of buffers.
*/
u32 GetBufferCount();
/**
* Get the total played sample count for this audio out.
*
* @return The played sample count.
*/
u64 GetPlayedSampleCount();
private:
/// The AudioOut::Manager this audio out is registered with
Manager& manager;
/// Manager's mutex
std::recursive_mutex& parent_mutex;
/// Buffer event, signalled when buffers are ready to be released
Kernel::KEvent* event;
/// Main audio out system
System system;
};
} // namespace AudioCore::AudioOut

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <mutex>
#include "audio_core/audio_event.h"
#include "audio_core/audio_manager.h"
#include "audio_core/out/audio_out_system.h"
#include "common/logging/log.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/hle/kernel/k_event.h"
namespace AudioCore::AudioOut {
System::System(Core::System& system_, Kernel::KEvent* event_, size_t session_id_)
: system{system_}, buffer_event{event_},
session_id{session_id_}, session{std::make_unique<DeviceSession>(system_)} {}
System::~System() {
Finalize();
}
void System::Finalize() {
Stop();
session->Finalize();
buffer_event->GetWritableEvent().Signal();
}
std::string_view System::GetDefaultOutputDeviceName() {
return "DeviceOut";
}
Result System::IsConfigValid(std::string_view device_name, const AudioOutParameter& in_params) {
if ((device_name.size() > 0) && (device_name != GetDefaultOutputDeviceName())) {
return Service::Audio::ERR_INVALID_DEVICE_NAME;
}
if (in_params.sample_rate != TargetSampleRate && in_params.sample_rate > 0) {
return Service::Audio::ERR_INVALID_SAMPLE_RATE;
}
if (in_params.channel_count == 0 || in_params.channel_count == 2 ||
in_params.channel_count == 6) {
return ResultSuccess;
}
return Service::Audio::ERR_INVALID_CHANNEL_COUNT;
}
Result System::Initialize(std::string& device_name, const AudioOutParameter& in_params, u32 handle_,
u64& applet_resource_user_id_) {
auto result = IsConfigValid(device_name, in_params);
if (result.IsError()) {
return result;
}
handle = handle_;
applet_resource_user_id = applet_resource_user_id_;
if (device_name.empty() || device_name[0] == '\0') {
name = std::string(GetDefaultOutputDeviceName());
} else {
name = std::move(device_name);
}
sample_rate = TargetSampleRate;
sample_format = SampleFormat::PcmInt16;
channel_count = in_params.channel_count <= 2 ? 2 : 6;
volume = 1.0f;
return ResultSuccess;
}
void System::StartSession() {
session->Start();
}
size_t System::GetSessionId() const {
return session_id;
}
Result System::Start() {
if (state != State::Stopped) {
return Service::Audio::ERR_OPERATION_FAILED;
}
session->Initialize(name, sample_format, channel_count, session_id, handle,
applet_resource_user_id, Sink::StreamType::Out);
session->SetVolume(volume);
session->Start();
state = State::Started;
std::vector<AudioBuffer> buffers_to_flush{};
buffers.RegisterBuffers(buffers_to_flush);
session->AppendBuffers(buffers_to_flush);
return ResultSuccess;
}
Result System::Stop() {
if (state == State::Started) {
session->Stop();
session->SetVolume(0.0f);
state = State::Stopped;
}
return ResultSuccess;
}
bool System::AppendBuffer(const AudioOutBuffer& buffer, u64 tag) {
if (buffers.GetTotalBufferCount() == BufferCount) {
return false;
}
AudioBuffer new_buffer{
.played_timestamp = 0, .samples = buffer.samples, .tag = tag, .size = buffer.size};
buffers.AppendBuffer(new_buffer);
RegisterBuffers();
return true;
}
void System::RegisterBuffers() {
if (state == State::Started) {
std::vector<AudioBuffer> registered_buffers{};
buffers.RegisterBuffers(registered_buffers);
session->AppendBuffers(registered_buffers);
}
}
void System::ReleaseBuffers() {
bool signal{buffers.ReleaseBuffers(system.CoreTiming(), *session)};
if (signal) {
// Signal if any buffer was released, or if none are registered, we need more.
buffer_event->GetWritableEvent().Signal();
}
}
u32 System::GetReleasedBuffers(std::span<u64> tags) {
return buffers.GetReleasedBuffers(tags);
}
bool System::FlushAudioOutBuffers() {
if (state != State::Started) {
return false;
}
u32 buffers_released{};
buffers.FlushBuffers(buffers_released);
if (buffers_released > 0) {
buffer_event->GetWritableEvent().Signal();
}
return true;
}
u16 System::GetChannelCount() const {
return channel_count;
}
u32 System::GetSampleRate() const {
return sample_rate;
}
SampleFormat System::GetSampleFormat() const {
return sample_format;
}
State System::GetState() {
switch (state) {
case State::Started:
case State::Stopped:
return state;
default:
LOG_ERROR(Service_Audio, "AudioOut invalid state!");
state = State::Stopped;
break;
}
return state;
}
std::string System::GetName() const {
return name;
}
f32 System::GetVolume() const {
return volume;
}
void System::SetVolume(const f32 volume_) {
volume = volume_;
session->SetVolume(volume_);
}
bool System::ContainsAudioBuffer(const u64 tag) {
return buffers.ContainsBuffer(tag);
}
u32 System::GetBufferCount() {
return buffers.GetAppendedRegisteredCount();
}
u64 System::GetPlayedSampleCount() const {
return session->GetPlayedSampleCount();
}
} // namespace AudioCore::AudioOut

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <atomic>
#include <memory>
#include <span>
#include <string>
#include "audio_core/common/common.h"
#include "audio_core/device/audio_buffers.h"
#include "audio_core/device/device_session.h"
#include "core/hle/service/audio/errors.h"
namespace Core {
class System;
}
namespace Kernel {
class KEvent;
}
namespace AudioCore::AudioOut {
constexpr SessionTypes SessionType = SessionTypes::AudioOut;
struct AudioOutParameter {
/* 0x0 */ s32_le sample_rate;
/* 0x4 */ u16_le channel_count;
/* 0x6 */ u16_le reserved;
};
static_assert(sizeof(AudioOutParameter) == 0x8, "AudioOutParameter is an invalid size");
struct AudioOutParameterInternal {
/* 0x0 */ u32_le sample_rate;
/* 0x4 */ u32_le channel_count;
/* 0x8 */ u32_le sample_format;
/* 0xC */ u32_le state;
};
static_assert(sizeof(AudioOutParameterInternal) == 0x10,
"AudioOutParameterInternal is an invalid size");
struct AudioOutBuffer {
/* 0x00 */ AudioOutBuffer* next;
/* 0x08 */ VAddr samples;
/* 0x10 */ u64 capacity;
/* 0x18 */ u64 size;
/* 0x20 */ u64 offset;
};
static_assert(sizeof(AudioOutBuffer) == 0x28, "AudioOutBuffer is an invalid size");
enum class State {
Started,
Stopped,
};
/**
* Controls and drives audio output.
*/
class System {
public:
explicit System(Core::System& system, Kernel::KEvent* event, size_t session_id);
~System();
/**
* Get the default audio output device name.
*
* @return The default audio output device name.
*/
std::string_view GetDefaultOutputDeviceName();
/**
* Is the given initialize config valid?
*
* @param device_name - The name of the requested output device.
* @param in_params - Input parameters, see AudioOutParameter.
* @return Result code.
*/
Result IsConfigValid(std::string_view device_name, const AudioOutParameter& in_params);
/**
* Initialize this system.
*
* @param device_name - The name of the requested output device.
* @param in_params - Input parameters, see AudioOutParameter.
* @param handle - Unused.
* @param applet_resource_user_id - Unused.
* @return Result code.
*/
Result Initialize(std::string& device_name, const AudioOutParameter& in_params, u32 handle,
u64& applet_resource_user_id);
/**
* Start this system.
*
* @return Result code.
*/
Result Start();
/**
* Stop this system.
*
* @return Result code.
*/
Result Stop();
/**
* Finalize this system.
*/
void Finalize();
/**
* Start this system's device session.
*/
void StartSession();
/**
* Get this system's id.
*/
size_t GetSessionId() const;
/**
* Append a new buffer to the device.
*
* @param buffer - New buffer to append.
* @param tag - Unique tag of the buffer.
* @return True if the buffer was appended, otherwise false.
*/
bool AppendBuffer(const AudioOutBuffer& buffer, u64 tag);
/**
* Register all appended buffers.
*/
void RegisterBuffers();
/**
* Release all registered buffers.
*/
void ReleaseBuffers();
/**
* Get all released buffers.
*
* @param tags - Container to be filled with the released buffers' tags.
* @return The number of buffers released.
*/
u32 GetReleasedBuffers(std::span<u64> tags);
/**
* Flush all appended and registered buffers.
*
* @return True if buffers were successfully flushed, otherwise false.
*/
bool FlushAudioOutBuffers();
/**
* Get this system's current channel count.
*
* @return The channel count.
*/
u16 GetChannelCount() const;
/**
* Get this system's current sample rate.
*
* @return The sample rate.
*/
u32 GetSampleRate() const;
/**
* Get this system's current sample format.
*
* @return The sample format.
*/
SampleFormat GetSampleFormat() const;
/**
* Get this system's current state.
*
* @return The current state.
*/
State GetState();
/**
* Get this system's name.
*
* @return The system's name.
*/
std::string GetName() const;
/**
* Get this system's current volume.
*
* @return The system's current volume.
*/
f32 GetVolume() const;
/**
* Set this system's current volume.
*
* @param The new volume.
*/
void SetVolume(f32 volume);
/**
* Does the system contain this buffer?
*
* @param tag - Unique tag to search for.
* @return True if the buffer is in the system, otherwise false.
*/
bool ContainsAudioBuffer(u64 tag);
/**
* Get the maximum number of usable buffers (default 32).
*
* @return The number of buffers.
*/
u32 GetBufferCount();
/**
* Get the total number of samples played by this system.
*
* @return The number of samples.
*/
u64 GetPlayedSampleCount() const;
private:
/// Core system
Core::System& system;
/// (Unused)
u32 handle{};
/// (Unused)
u64 applet_resource_user_id{};
/// Buffer event, signalled when a buffer is ready
Kernel::KEvent* buffer_event;
/// Session id of this system
size_t session_id{};
/// Device session for this system
std::unique_ptr<DeviceSession> session;
/// Audio buffers in use by this system
AudioBuffers<BufferCount> buffers{BufferCount};
/// Sample rate of this system
u32 sample_rate{};
/// Sample format of this system
SampleFormat sample_format{SampleFormat::PcmInt16};
/// Channel count of this system
u16 channel_count{};
/// State of this system
std::atomic<State> state{State::Stopped};
/// Name of this system
std::string name{};
/// Volume of this system
f32 volume{1.0f};
};
} // namespace AudioCore::AudioOut

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/renderer/adsp/adsp.h"
#include "audio_core/renderer/adsp/command_buffer.h"
#include "audio_core/sink/sink.h"
#include "common/logging/log.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/core_timing_util.h"
#include "core/memory.h"
namespace AudioCore::AudioRenderer::ADSP {
ADSP::ADSP(Core::System& system_, Sink::Sink& sink_)
: system{system_}, memory{system.Memory()}, sink{sink_} {}
ADSP::~ADSP() {
ClearCommandBuffers();
}
State ADSP::GetState() const {
if (running) {
return State::Started;
}
return State::Stopped;
}
AudioRenderer_Mailbox* ADSP::GetRenderMailbox() {
return &render_mailbox;
}
void ADSP::ClearRemainCount(const u32 session_id) {
render_mailbox.ClearRemainCount(session_id);
}
u64 ADSP::GetSignalledTick() const {
return render_mailbox.GetSignalledTick();
}
u64 ADSP::GetTimeTaken() const {
return render_mailbox.GetRenderTimeTaken();
}
u64 ADSP::GetRenderTimeTaken(const u32 session_id) {
return render_mailbox.GetCommandBuffer(session_id).render_time_taken;
}
u32 ADSP::GetRemainCommandCount(const u32 session_id) const {
return render_mailbox.GetRemainCommandCount(session_id);
}
void ADSP::SendCommandBuffer(const u32 session_id, CommandBuffer& command_buffer) {
render_mailbox.SetCommandBuffer(session_id, command_buffer);
}
u64 ADSP::GetRenderingStartTick(const u32 session_id) {
return render_mailbox.GetSignalledTick() +
render_mailbox.GetCommandBuffer(session_id).render_time_taken;
}
bool ADSP::Start() {
if (running) {
return running;
}
running = true;
systems_active++;
audio_renderer = std::make_unique<AudioRenderer>(system);
audio_renderer->Start(&render_mailbox);
render_mailbox.HostSendMessage(RenderMessage::AudioRenderer_InitializeOK);
if (render_mailbox.HostWaitMessage() != RenderMessage::AudioRenderer_InitializeOK) {
LOG_ERROR(
Service_Audio,
"Host Audio Renderer -- Failed to receive initialize message response from ADSP!");
}
return running;
}
void ADSP::Stop() {
systems_active--;
if (running && systems_active == 0) {
{
std::scoped_lock l{mailbox_lock};
render_mailbox.HostSendMessage(RenderMessage::AudioRenderer_Shutdown);
if (render_mailbox.HostWaitMessage() != RenderMessage::AudioRenderer_Shutdown) {
LOG_ERROR(Service_Audio, "Host Audio Renderer -- Failed to receive shutdown "
"message response from ADSP!");
}
}
audio_renderer->Stop();
running = false;
}
}
void ADSP::Signal() {
const auto signalled_tick{system.CoreTiming().GetClockTicks()};
render_mailbox.SetSignalledTick(signalled_tick);
render_mailbox.HostSendMessage(RenderMessage::AudioRenderer_Render);
}
void ADSP::Wait() {
std::scoped_lock l{mailbox_lock};
auto response{render_mailbox.HostWaitMessage()};
if (response != RenderMessage::AudioRenderer_RenderResponse) {
LOG_ERROR(Service_Audio, "Invalid ADSP response message, expected 0x{:02X}, got 0x{:02X}",
static_cast<u32>(RenderMessage::AudioRenderer_RenderResponse),
static_cast<u32>(response));
}
ClearCommandBuffers();
}
void ADSP::ClearCommandBuffers() {
render_mailbox.ClearCommandBuffers();
}
} // namespace AudioCore::AudioRenderer::ADSP

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <memory>
#include <mutex>
#include "audio_core/renderer/adsp/audio_renderer.h"
#include "common/common_types.h"
namespace Core {
namespace Memory {
class Memory;
}
class System;
} // namespace Core
namespace AudioCore {
namespace Sink {
class Sink;
}
namespace AudioRenderer::ADSP {
struct CommandBuffer;
enum class State {
Started,
Stopped,
};
/**
* Represents the ADSP embedded within the audio sysmodule.
* This is a 32-bit Linux4Tegra kernel from nVidia, which is launched with the sysmodule on boot.
*
* The kernel will run apps you program for it, Nintendo have the following:
*
* Gmix - Responsible for mixing final audio and sending it out to hardware. This is last place all
* audio samples end up, and we skip it entirely, since we have very different backends and
* mixing is implicitly handled by the OS (but also due to lack of research/simplicity).
*
* AudioRenderer - Receives command lists generated by the audio render
* system, processes them, and sends the samples to Gmix.
*
* OpusDecoder - Contains libopus, and controls processing Opus audio and sends it to Gmix.
* Not much research done here, TODO if needed.
*
* We only implement the AudioRenderer for now.
*
* Communication for the apps is done through mailboxes, and some shared memory.
*/
class ADSP {
public:
explicit ADSP(Core::System& system, Sink::Sink& sink);
~ADSP();
/**
* Start the ADSP.
*
* @return True if started or already running, otherwise false.
*/
bool Start();
/**
* Stop the ADSP.
*
* @return True if started or already running, otherwise false.
*/
void Stop();
/**
* Get the ADSP's state.
*
* @return Started or Stopped.
*/
State GetState() const;
/**
* Get the AudioRenderer mailbox to communicate with it.
*
* @return The AudioRenderer mailbox.
*/
AudioRenderer_Mailbox* GetRenderMailbox();
/**
* Get the tick the ADSP was signalled.
*
* @return The tick the ADSP was signalled.
*/
u64 GetSignalledTick() const;
/**
* Get the total time it took for the ADSP to run the last command lists (both command lists).
*
* @return The tick the ADSP was signalled.
*/
u64 GetTimeTaken() const;
/**
* Get the last time a given command list took to run.
*
* @param session_id - The session id to check (0 or 1).
* @return The time it took.
*/
u64 GetRenderTimeTaken(u32 session_id);
/**
* Clear the remaining command count for a given session.
*
* @param session_id - The session id to check (0 or 1).
*/
void ClearRemainCount(u32 session_id);
/**
* Get the remaining number of commands left to process for a command list.
*
* @param session_id - The session id to check (0 or 1).
* @return The number of commands remaining.
*/
u32 GetRemainCommandCount(u32 session_id) const;
/**
* Get the last tick a command list started processing.
*
* @param session_id - The session id to check (0 or 1).
* @return The last tick the given command list started.
*/
u64 GetRenderingStartTick(u32 session_id);
/**
* Set a command buffer to be processed.
*
* @param session_id - The session id to check (0 or 1).
* @param command_buffer - The command buffer to process.
*/
void SendCommandBuffer(u32 session_id, CommandBuffer& command_buffer);
/**
* Clear the command buffers (does not clear the time taken or the remaining command count)
*/
void ClearCommandBuffers();
/**
* Signal the AudioRenderer to begin processing.
*/
void Signal();
/**
* Wait for the AudioRenderer to finish processing.
*/
void Wait();
private:
/// Core system
Core::System& system;
/// Core memory
Core::Memory::Memory& memory;
/// Number of systems active, used to prevent accidental shutdowns
u8 systems_active{0};
/// ADSP running state
std::atomic<bool> running{false};
/// Output sink used by the ADSP
Sink::Sink& sink;
/// AudioRenderer app
std::unique_ptr<AudioRenderer> audio_renderer{};
/// Communication for the AudioRenderer
AudioRenderer_Mailbox render_mailbox{};
/// Mailbox lock ffor the render mailbox
std::mutex mailbox_lock;
};
} // namespace AudioRenderer::ADSP
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <array>
#include <chrono>
#include "audio_core/audio_core.h"
#include "audio_core/common/common.h"
#include "audio_core/renderer/adsp/audio_renderer.h"
#include "audio_core/sink/sink.h"
#include "common/logging/log.h"
#include "common/microprofile.h"
#include "common/thread.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/core_timing_util.h"
MICROPROFILE_DEFINE(Audio_Renderer, "Audio", "DSP", MP_RGB(60, 19, 97));
namespace AudioCore::AudioRenderer::ADSP {
void AudioRenderer_Mailbox::HostSendMessage(RenderMessage message_) {
adsp_messages.enqueue(message_);
adsp_event.Set();
}
RenderMessage AudioRenderer_Mailbox::HostWaitMessage() {
host_event.Wait();
RenderMessage msg{RenderMessage::Invalid};
if (!host_messages.try_dequeue(msg)) {
LOG_ERROR(Service_Audio, "Failed to dequeue host message!");
}
return msg;
}
void AudioRenderer_Mailbox::ADSPSendMessage(const RenderMessage message_) {
host_messages.enqueue(message_);
host_event.Set();
}
RenderMessage AudioRenderer_Mailbox::ADSPWaitMessage() {
adsp_event.Wait();
RenderMessage msg{RenderMessage::Invalid};
if (!adsp_messages.try_dequeue(msg)) {
LOG_ERROR(Service_Audio, "Failed to dequeue ADSP message!");
}
return msg;
}
CommandBuffer& AudioRenderer_Mailbox::GetCommandBuffer(const s32 session_id) {
return command_buffers[session_id];
}
void AudioRenderer_Mailbox::SetCommandBuffer(const u32 session_id, CommandBuffer& buffer) {
command_buffers[session_id] = buffer;
}
u64 AudioRenderer_Mailbox::GetRenderTimeTaken() const {
return command_buffers[0].render_time_taken + command_buffers[1].render_time_taken;
}
u64 AudioRenderer_Mailbox::GetSignalledTick() const {
return signalled_tick;
}
void AudioRenderer_Mailbox::SetSignalledTick(const u64 tick) {
signalled_tick = tick;
}
void AudioRenderer_Mailbox::ClearRemainCount(const u32 session_id) {
command_buffers[session_id].remaining_command_count = 0;
}
u32 AudioRenderer_Mailbox::GetRemainCommandCount(const u32 session_id) const {
return command_buffers[session_id].remaining_command_count;
}
void AudioRenderer_Mailbox::ClearCommandBuffers() {
command_buffers[0].buffer = 0;
command_buffers[0].size = 0;
command_buffers[0].reset_buffers = false;
command_buffers[1].buffer = 0;
command_buffers[1].size = 0;
command_buffers[1].reset_buffers = false;
}
AudioRenderer::AudioRenderer(Core::System& system_)
: system{system_}, sink{system.AudioCore().GetOutputSink()} {
CreateSinkStreams();
}
AudioRenderer::~AudioRenderer() {
Stop();
for (auto& stream : streams) {
if (stream) {
sink.CloseStream(stream);
}
stream = nullptr;
}
}
void AudioRenderer::Start(AudioRenderer_Mailbox* mailbox_) {
if (running) {
return;
}
mailbox = mailbox_;
thread = std::thread(&AudioRenderer::ThreadFunc, this);
for (auto& stream : streams) {
stream->Start();
}
running = true;
}
void AudioRenderer::Stop() {
if (!running) {
return;
}
for (auto& stream : streams) {
stream->Stop();
}
thread.join();
running = false;
}
void AudioRenderer::CreateSinkStreams() {
u32 channels{sink.GetDeviceChannels()};
for (u32 i = 0; i < MaxRendererSessions; i++) {
std::string name{fmt::format("ADSP_RenderStream-{}", i)};
streams[i] =
sink.AcquireSinkStream(system, channels, name, ::AudioCore::Sink::StreamType::Render);
}
}
void AudioRenderer::ThreadFunc() {
constexpr char name[]{"yuzu:AudioRenderer"};
MicroProfileOnThreadCreate(name);
Common::SetCurrentThreadName(name);
Common::SetCurrentThreadPriority(Common::ThreadPriority::Critical);
if (mailbox->ADSPWaitMessage() != RenderMessage::AudioRenderer_InitializeOK) {
LOG_ERROR(Service_Audio,
"ADSP Audio Renderer -- Failed to receive initialize message from host!");
return;
}
mailbox->ADSPSendMessage(RenderMessage::AudioRenderer_InitializeOK);
constexpr u64 max_process_time{2'304'000ULL};
while (true) {
auto message{mailbox->ADSPWaitMessage()};
switch (message) {
case RenderMessage::AudioRenderer_Shutdown:
mailbox->ADSPSendMessage(RenderMessage::AudioRenderer_Shutdown);
return;
case RenderMessage::AudioRenderer_Render: {
std::array<bool, MaxRendererSessions> buffers_reset{};
std::array<u64, MaxRendererSessions> render_times_taken{};
const auto start_time{system.CoreTiming().GetClockTicks()};
for (u32 index = 0; index < 2; index++) {
auto& command_buffer{mailbox->GetCommandBuffer(index)};
auto& command_list_processor{command_list_processors[index]};
// Check this buffer is valid, as it may not be used.
if (command_buffer.buffer != 0) {
// If there are no remaining commands (from the previous list),
// this is a new command list, initalize it.
if (command_buffer.remaining_command_count == 0) {
command_list_processor.Initialize(system, command_buffer.buffer,
command_buffer.size, streams[index]);
}
if (command_buffer.reset_buffers && !buffers_reset[index]) {
streams[index]->ClearQueue();
buffers_reset[index] = true;
}
u64 max_time{max_process_time};
if (index == 1 && command_buffer.applet_resource_user_id ==
mailbox->GetCommandBuffer(0).applet_resource_user_id) {
max_time = max_process_time -
Core::Timing::CyclesToNs(render_times_taken[0]).count();
if (render_times_taken[0] > max_process_time) {
max_time = 0;
}
}
max_time = std::min(command_buffer.time_limit, max_time);
command_list_processor.SetProcessTimeMax(max_time);
// Process the command list
{
MICROPROFILE_SCOPE(Audio_Renderer);
render_times_taken[index] =
command_list_processor.Process(index) - start_time;
}
if (index == 0) {
auto stream{command_list_processor.GetOutputSinkStream()};
system.AudioCore().SetStreamQueue(stream->GetQueueSize());
}
const auto end_time{system.CoreTiming().GetClockTicks()};
command_buffer.remaining_command_count =
command_list_processor.GetRemainingCommandCount();
command_buffer.render_time_taken = end_time - start_time;
}
}
mailbox->ADSPSendMessage(RenderMessage::AudioRenderer_RenderResponse);
} break;
default:
LOG_WARNING(Service_Audio,
"ADSP AudioRenderer received an invalid message, msg={:02X}!",
static_cast<u32>(message));
break;
}
}
}
} // namespace AudioCore::AudioRenderer::ADSP

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <memory>
#include <thread>
#include "audio_core/renderer/adsp/command_buffer.h"
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "common/common_types.h"
#include "common/reader_writer_queue.h"
#include "common/thread.h"
namespace Core {
namespace Timing {
struct EventType;
}
class System;
} // namespace Core
namespace AudioCore {
namespace Sink {
class Sink;
}
namespace AudioRenderer::ADSP {
enum class RenderMessage {
/* 0x00 */ Invalid,
/* 0x01 */ AudioRenderer_MapUnmap_Map,
/* 0x02 */ AudioRenderer_MapUnmap_MapResponse,
/* 0x03 */ AudioRenderer_MapUnmap_Unmap,
/* 0x04 */ AudioRenderer_MapUnmap_UnmapResponse,
/* 0x05 */ AudioRenderer_MapUnmap_InvalidateCache,
/* 0x06 */ AudioRenderer_MapUnmap_InvalidateCacheResponse,
/* 0x07 */ AudioRenderer_MapUnmap_Shutdown,
/* 0x08 */ AudioRenderer_MapUnmap_ShutdownResponse,
/* 0x16 */ AudioRenderer_InitializeOK = 0x16,
/* 0x20 */ AudioRenderer_RenderResponse = 0x20,
/* 0x2A */ AudioRenderer_Render = 0x2A,
/* 0x34 */ AudioRenderer_Shutdown = 0x34,
};
/**
* A mailbox for the AudioRenderer, allowing communication between the host and the AudioRenderer
* running on the ADSP.
*/
class AudioRenderer_Mailbox {
public:
/**
* Send a message from the host to the AudioRenderer.
*
* @param message_ - The message to send to the AudioRenderer.
*/
void HostSendMessage(RenderMessage message);
/**
* Host wait for a message from the AudioRenderer.
*
* @return The message returned from the AudioRenderer.
*/
RenderMessage HostWaitMessage();
/**
* Send a message from the AudioRenderer to the host.
*
* @param message_ - The message to send to the host.
*/
void ADSPSendMessage(RenderMessage message);
/**
* AudioRenderer wait for a message from the host.
*
* @return The message returned from the AudioRenderer.
*/
RenderMessage ADSPWaitMessage();
/**
* Get the command buffer with the given session id (0 or 1).
*
* @param session_id - The session id to get (0 or 1).
* @return The command buffer.
*/
CommandBuffer& GetCommandBuffer(s32 session_id);
/**
* Set the command buffer with the given session id (0 or 1).
*
* @param session_id - The session id to get (0 or 1).
* @param buffer - The command buffer to set.
*/
void SetCommandBuffer(u32 session_id, CommandBuffer& buffer);
/**
* Get the total render time taken for the last command lists sent.
*
* @return Total render time taken for the last command lists.
*/
u64 GetRenderTimeTaken() const;
/**
* Get the tick the AudioRenderer was signalled.
*
* @return The tick the AudioRenderer was signalled.
*/
u64 GetSignalledTick() const;
/**
* Set the tick the AudioRenderer was signalled.
*
* @param tick - The tick the AudioRenderer was signalled.
*/
void SetSignalledTick(u64 tick);
/**
* Clear the remaining command count.
*
* @param session_id - Index for which command list to clear (0 or 1).
*/
void ClearRemainCount(u32 session_id);
/**
* Get the remaining command count for a given command list.
*
* @param session_id - Index for which command list to clear (0 or 1).
* @return The remaining command count.
*/
u32 GetRemainCommandCount(u32 session_id) const;
/**
* Clear the command buffers (does not clear the time taken or the remaining command count).
*/
void ClearCommandBuffers();
private:
/// Host signalling event
Common::Event host_event{};
/// AudioRenderer signalling event
Common::Event adsp_event{};
/// Host message queue
Common::ReaderWriterQueue<RenderMessage> host_messages{};
/// AudioRenderer message queue
Common::ReaderWriterQueue<RenderMessage> adsp_messages{};
/// Command buffers
std::array<CommandBuffer, MaxRendererSessions> command_buffers{};
/// Tick the AudioRnederer was signalled
u64 signalled_tick{};
};
/**
* The AudioRenderer application running on the ADSP.
*/
class AudioRenderer {
public:
explicit AudioRenderer(Core::System& system);
~AudioRenderer();
/**
* Start the AudioRenderer.
*
* @param The mailbox to use for this session.
*/
void Start(AudioRenderer_Mailbox* mailbox);
/**
* Stop the AudioRenderer.
*/
void Stop();
private:
/**
* Main AudioRenderer thread, responsible for processing the command lists.
*/
void ThreadFunc();
/**
* Creates the streams which will receive the processed samples.
*/
void CreateSinkStreams();
/// Core system
Core::System& system;
/// Main thread
std::thread thread{};
/// The current state
std::atomic<bool> running{};
/// The active mailbox
AudioRenderer_Mailbox* mailbox{};
/// The command lists to process
std::array<CommandListProcessor, MaxRendererSessions> command_list_processors{};
/// The output sink the AudioRenderer will use
Sink::Sink& sink;
/// The streams which will receive the processed samples
std::array<Sink::SinkStream*, MaxRendererSessions> streams;
};
} // namespace AudioRenderer::ADSP
} // namespace AudioCore

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// 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::ADSP {
struct CommandBuffer {
CpuAddr buffer;
u64 size;
u64 time_limit;
u32 remaining_command_count;
bool reset_buffers;
u64 applet_resource_user_id;
u64 render_time_taken;
};
} // namespace AudioCore::AudioRenderer::ADSP

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// 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/command_list_header.h"
#include "audio_core/renderer/command/commands.h"
#include "common/settings.h"
#include "core/core.h"
#include "core/core_timing.h"
#include "core/core_timing_util.h"
#include "core/memory.h"
namespace AudioCore::AudioRenderer::ADSP {
void CommandListProcessor::Initialize(Core::System& system_, CpuAddr buffer, u64 size,
Sink::SinkStream* stream_) {
system = &system_;
memory = &system->Memory();
stream = stream_;
header = reinterpret_cast<CommandListHeader*>(buffer);
commands = reinterpret_cast<u8*>(buffer + sizeof(CommandListHeader));
commands_buffer_size = size;
command_count = header->command_count;
sample_count = header->sample_count;
target_sample_rate = header->sample_rate;
mix_buffers = header->samples_buffer;
buffer_count = header->buffer_count;
processed_command_count = 0;
}
void CommandListProcessor::SetProcessTimeMax(const u64 time) {
max_process_time = time;
}
u32 CommandListProcessor::GetRemainingCommandCount() const {
return command_count - processed_command_count;
}
void CommandListProcessor::SetBuffer(const CpuAddr buffer, const u64 size) {
commands = reinterpret_cast<u8*>(buffer + sizeof(CommandListHeader));
commands_buffer_size = size;
}
Sink::SinkStream* CommandListProcessor::GetOutputSinkStream() const {
return stream;
}
u64 CommandListProcessor::Process(u32 session_id) {
const auto start_time_{system->CoreTiming().GetClockTicks()};
const auto command_base{CpuAddr(commands)};
if (processed_command_count > 0) {
current_processing_time += start_time_ - end_time;
} else {
start_time = start_time_;
current_processing_time = 0;
}
std::string dump{fmt::format("\nSession {}\n", session_id)};
for (u32 index = 0; index < command_count; index++) {
auto& command{*reinterpret_cast<ICommand*>(commands)};
if (command.magic != 0xCAFEBABE) {
LOG_ERROR(Service_Audio, "Command has invalid magic! Expected 0xCAFEBABE, got {:08X}",
command.magic);
return system->CoreTiming().GetClockTicks() - start_time_;
}
auto current_offset{CpuAddr(commands) - command_base};
if (current_offset + command.size > commands_buffer_size) {
LOG_ERROR(Service_Audio,
"Command exceeded command buffer, buffer size {:08X}, command ends at {:08X}",
commands_buffer_size,
CpuAddr(commands) + command.size - sizeof(CommandListHeader));
return system->CoreTiming().GetClockTicks() - start_time_;
}
if (Settings::values.dump_audio_commands) {
command.Dump(*this, dump);
}
if (!command.Verify(*this)) {
break;
}
if (command.enabled) {
command.Process(*this);
} else {
dump += fmt::format("\tDisabled!\n");
}
processed_command_count++;
commands += command.size;
}
if (Settings::values.dump_audio_commands && dump != last_dump) {
LOG_WARNING(Service_Audio, "{}", dump);
last_dump = dump;
}
end_time = system->CoreTiming().GetClockTicks();
return end_time - start_time_;
}
} // namespace AudioCore::AudioRenderer::ADSP

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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/common/common.h"
#include "common/common_types.h"
namespace Core {
namespace Memory {
class Memory;
}
class System;
} // namespace Core
namespace AudioCore {
namespace Sink {
class SinkStream;
}
namespace AudioRenderer {
struct CommandListHeader;
namespace ADSP {
/**
* A processor for command lists given to the AudioRenderer.
*/
class CommandListProcessor {
public:
/**
* Initialize the processor.
*
* @param system_ - The core system.
* @param buffer - The command buffer to process.
* @param size - The size of the buffer.
* @param stream_ - The stream to be used for sending the samples.
*/
void Initialize(Core::System& system, CpuAddr buffer, u64 size, Sink::SinkStream* stream);
/**
* Set the maximum processing time for this command list.
*
* @param time - The maximum process time.
*/
void SetProcessTimeMax(u64 time);
/**
* Get the remaining command count for this list.
*
* @return The remaining command count.
*/
u32 GetRemainingCommandCount() const;
/**
* Set the command buffer.
*
* @param buffer - The buffer to use.
* @param size - The size of the buffer.
*/
void SetBuffer(CpuAddr buffer, u64 size);
/**
* Get the stream for this command list.
*
* @return The stream associated with this command list.
*/
Sink::SinkStream* GetOutputSinkStream() const;
/**
* Process the command list.
*
* @param index - Index of the current command list.
* @return The time taken to process.
*/
u64 Process(u32 session_id);
/// Core system
Core::System* system{};
/// Core memory
Core::Memory::Memory* memory{};
/// Stream for the processed samples
Sink::SinkStream* stream{};
/// Header info for this command list
CommandListHeader* header{};
/// The command buffer
u8* commands{};
/// The command buffer size
u64 commands_buffer_size{};
/// The maximum processing time alloted
u64 max_process_time{};
/// The number of commands in the buffer
u32 command_count{};
/// The target sample count for output
u32 sample_count{};
/// The target sample rate for output
u32 target_sample_rate{};
/// The mixing buffers used by the commands
std::span<s32> mix_buffers{};
/// The number of mix buffers
u32 buffer_count{};
/// The number of processed commands so far
u32 processed_command_count{};
/// The processing start time of this list
u64 start_time{};
/// The current processing time for this list
u64 current_processing_time{};
/// The end processing time for this list
u64 end_time{};
/// Last command list string generated, used for dumping audio commands to console
std::string last_dump{};
};
} // namespace ADSP
} // namespace AudioRenderer
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_core.h"
#include "audio_core/common/feature_support.h"
#include "audio_core/renderer/audio_device.h"
#include "audio_core/sink/sink.h"
#include "core/core.h"
namespace AudioCore::AudioRenderer {
AudioDevice::AudioDevice(Core::System& system, const u64 applet_resource_user_id_,
const u32 revision)
: output_sink{system.AudioCore().GetOutputSink()},
applet_resource_user_id{applet_resource_user_id_}, user_revision{revision} {}
u32 AudioDevice::ListAudioDeviceName(std::vector<AudioDeviceName>& out_buffer,
const size_t max_count) {
std::span<AudioDeviceName> names{};
if (CheckFeatureSupported(SupportTags::AudioUsbDeviceOutput, user_revision)) {
names = usb_device_names;
} else {
names = device_names;
}
u32 out_count{static_cast<u32>(std::min(max_count, names.size()))};
for (u32 i = 0; i < out_count; i++) {
out_buffer.push_back(names[i]);
}
return out_count;
}
u32 AudioDevice::ListAudioOutputDeviceName(std::vector<AudioDeviceName>& out_buffer,
const size_t max_count) {
u32 out_count{static_cast<u32>(std::min(max_count, output_device_names.size()))};
for (u32 i = 0; i < out_count; i++) {
out_buffer.push_back(output_device_names[i]);
}
return out_count;
}
void AudioDevice::SetDeviceVolumes(const f32 volume) {
output_sink.SetDeviceVolume(volume);
}
f32 AudioDevice::GetDeviceVolume([[maybe_unused]] std::string_view name) {
return output_sink.GetDeviceVolume();
}
} // 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/audio_render_manager.h"
namespace Core {
class System;
}
namespace AudioCore {
namespace Sink {
class Sink;
}
namespace AudioRenderer {
/**
* An interface to an output audio device available to the Switch.
*/
class AudioDevice {
public:
struct AudioDeviceName {
std::array<char, 0x100> name;
AudioDeviceName(const char* name_) {
std::strncpy(name.data(), name_, name.size());
}
};
std::array<AudioDeviceName, 4> usb_device_names{"AudioStereoJackOutput",
"AudioBuiltInSpeakerOutput", "AudioTvOutput",
"AudioUsbDeviceOutput"};
std::array<AudioDeviceName, 3> device_names{"AudioStereoJackOutput",
"AudioBuiltInSpeakerOutput", "AudioTvOutput"};
std::array<AudioDeviceName, 3> output_device_names{"AudioBuiltInSpeakerOutput", "AudioTvOutput",
"AudioExternalOutput"};
explicit AudioDevice(Core::System& system, u64 applet_resource_user_id, u32 revision);
/**
* Get a list of the available output devices.
*
* @param out_buffer - Output buffer to write the available device names.
* @param max_count - Maximum number of devices to write (count of out_buffer).
* @return Number of device names written.
*/
u32 ListAudioDeviceName(std::vector<AudioDeviceName>& out_buffer, size_t max_count);
/**
* Get a list of the available output devices.
* Different to above somehow...
*
* @param out_buffer - Output buffer to write the available device names.
* @param max_count - Maximum number of devices to write (count of out_buffer).
* @return Number of device names written.
*/
u32 ListAudioOutputDeviceName(std::vector<AudioDeviceName>& out_buffer, size_t max_count);
/**
* Set the volume of all streams in the backend sink.
*
* @param volume - Volume to set.
*/
void SetDeviceVolumes(f32 volume);
/**
* Get the volume for a given device name.
* Note: This is not fully implemented, we only assume 1 device for all streams.
*
* @param name - Name of the device to check. Unused.
* @return Volume of the device.
*/
f32 GetDeviceVolume(std::string_view name);
private:
/// Backend output sink for the device
Sink::Sink& output_sink;
/// Resource id this device is used for
const u64 applet_resource_user_id;
/// User audio renderer revision
const u32 user_revision;
};
} // namespace AudioRenderer
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/audio_render_manager.h"
#include "audio_core/common/audio_renderer_parameter.h"
#include "audio_core/renderer/audio_renderer.h"
#include "audio_core/renderer/system_manager.h"
#include "core/core.h"
#include "core/hle/kernel/k_transfer_memory.h"
#include "core/hle/service/audio/errors.h"
namespace AudioCore::AudioRenderer {
Renderer::Renderer(Core::System& system_, Manager& manager_, Kernel::KEvent* rendered_event)
: core{system_}, manager{manager_}, system{system_, rendered_event} {}
Result Renderer::Initialize(const AudioRendererParameterInternal& params,
Kernel::KTransferMemory* transfer_memory,
const u64 transfer_memory_size, const u32 process_handle,
const u64 applet_resource_user_id, const s32 session_id) {
if (params.execution_mode == ExecutionMode::Auto) {
if (!manager.AddSystem(system)) {
LOG_ERROR(Service_Audio,
"Both Audio Render sessions are in use, cannot create any more");
return Service::Audio::ERR_MAXIMUM_SESSIONS_REACHED;
}
system_registered = true;
}
initialized = true;
system.Initialize(params, transfer_memory, transfer_memory_size, process_handle,
applet_resource_user_id, session_id);
return ResultSuccess;
}
void Renderer::Finalize() {
auto session_id{system.GetSessionId()};
system.Finalize();
if (system_registered) {
manager.RemoveSystem(system);
system_registered = false;
}
manager.ReleaseSessionId(session_id);
}
System& Renderer::GetSystem() {
return system;
}
void Renderer::Start() {
system.Start();
}
void Renderer::Stop() {
system.Stop();
}
Result Renderer::RequestUpdate(std::span<const u8> input, std::span<u8> performance,
std::span<u8> output) {
return system.Update(input, performance, output);
}
} // 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/system.h"
#include "core/hle/service/audio/errors.h"
namespace Core {
class System;
}
namespace Kernel {
class KTransferMemory;
}
namespace AudioCore {
struct AudioRendererParameterInternal;
namespace AudioRenderer {
class Manager;
/**
* Audio Renderer, wraps the main audio system and is mainly responsible for handling service calls.
*/
class Renderer {
public:
explicit Renderer(Core::System& system, Manager& manager, Kernel::KEvent* rendered_event);
/**
* Initialize the renderer.
* Registers the system with the AudioRenderer::Manager, allocates workbuffers and initializes
* everything to a default state.
*
* @param params - Input parameters to initialize the system with.
* @param transfer_memory - Game-supplied memory for all workbuffers. Unused.
* @param transfer_memory_size - Size of the transfer memory. Unused.
* @param process_handle - Process handle, also used for memory. Unused.
* @param applet_resource_user_id - Applet id for this renderer. Unused.
* @param session_id - Session id of this renderer.
* @return Result code.
*/
Result Initialize(const AudioRendererParameterInternal& params,
Kernel::KTransferMemory* transfer_memory, u64 transfer_memory_size,
u32 process_handle, u64 applet_resource_user_id, s32 session_id);
/**
* Finalize the renderer for shutdown.
*/
void Finalize();
/**
* Get the renderer's system.
*
* @return Reference to the system.
*/
System& GetSystem();
/**
* Start the renderer.
*/
void Start();
/**
* Stop the renderer.
*/
void Stop();
/**
* Update the audio renderer with new information.
* Called via RequestUpdate from the AudRen:U service.
*
* @param input - Input buffer containing the new data.
* @param performance - Optional performance buffer for outputting performance metrics.
* @param output - Output data from the renderer.
* @return Result code.
*/
Result RequestUpdate(std::span<const u8> input, std::span<u8> performance,
std::span<u8> output);
private:
/// System core
Core::System& core;
/// Manager this renderer is registered with
Manager& manager;
/// Is the audio renderer initialized?
bool initialized{};
/// Is the system registered with the manager?
bool system_registered{};
/// Audio render system, main driver of audio rendering
System system;
};
} // namespace AudioRenderer
} // namespace AudioCore

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/common/feature_support.h"
#include "audio_core/renderer/behavior/behavior_info.h"
namespace AudioCore::AudioRenderer {
BehaviorInfo::BehaviorInfo() : process_revision{CurrentRevision} {}
u32 BehaviorInfo::GetProcessRevisionNum() const {
return process_revision;
}
u32 BehaviorInfo::GetProcessRevision() const {
return Common::MakeMagic('R', 'E', 'V',
static_cast<char>(static_cast<u8>('0') + process_revision));
}
u32 BehaviorInfo::GetUserRevisionNum() const {
return user_revision;
}
u32 BehaviorInfo::GetUserRevision() const {
return Common::MakeMagic('R', 'E', 'V',
static_cast<char>(static_cast<u8>('0') + user_revision));
}
void BehaviorInfo::SetUserLibRevision(const u32 user_revision_) {
user_revision = GetRevisionNum(user_revision_);
}
void BehaviorInfo::ClearError() {
error_count = 0;
}
void BehaviorInfo::AppendError(ErrorInfo& error) {
LOG_ERROR(Service_Audio, "Error during RequestUpdate, reporting code {:04X} address {:08X}",
error.error_code.raw, error.address);
if (error_count < MaxErrors) {
errors[error_count++] = error;
}
}
void BehaviorInfo::CopyErrorInfo(std::span<ErrorInfo> out_errors, u32& out_count) {
auto error_count_{std::min(error_count, MaxErrors)};
std::memset(out_errors.data(), 0, MaxErrors * sizeof(ErrorInfo));
for (size_t i = 0; i < error_count_; i++) {
out_errors[i] = errors[i];
}
out_count = error_count_;
}
void BehaviorInfo::UpdateFlags(const Flags flags_) {
flags = flags_;
}
bool BehaviorInfo::IsMemoryForceMappingEnabled() const {
return flags.IsMemoryForceMappingEnabled;
}
bool BehaviorInfo::IsAdpcmLoopContextBugFixed() const {
return CheckFeatureSupported(SupportTags::AdpcmLoopContextBugFix, user_revision);
}
bool BehaviorInfo::IsSplitterSupported() const {
return CheckFeatureSupported(SupportTags::Splitter, user_revision);
}
bool BehaviorInfo::IsSplitterBugFixed() const {
return CheckFeatureSupported(SupportTags::SplitterBugFix, user_revision);
}
bool BehaviorInfo::IsEffectInfoVersion2Supported() const {
return CheckFeatureSupported(SupportTags::EffectInfoVer2, user_revision);
}
bool BehaviorInfo::IsVariadicCommandBufferSizeSupported() const {
return CheckFeatureSupported(SupportTags::AudioRendererVariadicCommandBufferSize,
user_revision);
}
bool BehaviorInfo::IsWaveBufferVer2Supported() const {
return CheckFeatureSupported(SupportTags::WaveBufferVer2, user_revision);
}
bool BehaviorInfo::IsLongSizePreDelaySupported() const {
return CheckFeatureSupported(SupportTags::LongSizePreDelay, user_revision);
}
bool BehaviorInfo::IsCommandProcessingTimeEstimatorVersion2Supported() const {
return CheckFeatureSupported(SupportTags::CommandProcessingTimeEstimatorVersion2,
user_revision);
}
bool BehaviorInfo::IsCommandProcessingTimeEstimatorVersion3Supported() const {
return CheckFeatureSupported(SupportTags::CommandProcessingTimeEstimatorVersion3,
user_revision);
}
bool BehaviorInfo::IsCommandProcessingTimeEstimatorVersion4Supported() const {
return CheckFeatureSupported(SupportTags::CommandProcessingTimeEstimatorVersion4,
user_revision);
}
bool BehaviorInfo::IsCommandProcessingTimeEstimatorVersion5Supported() const {
return CheckFeatureSupported(SupportTags::CommandProcessingTimeEstimatorVersion4,
user_revision);
}
bool BehaviorInfo::IsAudioRendererProcessingTimeLimit70PercentSupported() const {
return CheckFeatureSupported(SupportTags::AudioRendererProcessingTimeLimit70Percent,
user_revision);
}
bool BehaviorInfo::IsAudioRendererProcessingTimeLimit75PercentSupported() const {
return CheckFeatureSupported(SupportTags::AudioRendererProcessingTimeLimit75Percent,
user_revision);
}
bool BehaviorInfo::IsAudioRendererProcessingTimeLimit80PercentSupported() const {
return CheckFeatureSupported(SupportTags::AudioRendererProcessingTimeLimit80Percent,
user_revision);
}
bool BehaviorInfo::IsFlushVoiceWaveBuffersSupported() const {
return CheckFeatureSupported(SupportTags::FlushVoiceWaveBuffers, user_revision);
}
bool BehaviorInfo::IsElapsedFrameCountSupported() const {
return CheckFeatureSupported(SupportTags::ElapsedFrameCount, user_revision);
}
bool BehaviorInfo::IsPerformanceMetricsDataFormatVersion2Supported() const {
return CheckFeatureSupported(SupportTags::PerformanceMetricsDataFormatVersion2, user_revision);
}
size_t BehaviorInfo::GetPerformanceMetricsDataFormat() const {
if (CheckFeatureSupported(SupportTags::PerformanceMetricsDataFormatVersion2, user_revision)) {
return 2;
}
return 1;
}
bool BehaviorInfo::IsVoicePitchAndSrcSkippedSupported() const {
return CheckFeatureSupported(SupportTags::VoicePitchAndSrcSkipped, user_revision);
}
bool BehaviorInfo::IsVoicePlayedSampleCountResetAtLoopPointSupported() const {
return CheckFeatureSupported(SupportTags::VoicePlayedSampleCountResetAtLoopPoint,
user_revision);
}
bool BehaviorInfo::IsBiquadFilterEffectStateClearBugFixed() const {
return CheckFeatureSupported(SupportTags::BiquadFilterEffectStateClearBugFix, user_revision);
}
bool BehaviorInfo::IsVolumeMixParameterPrecisionQ23Supported() const {
return CheckFeatureSupported(SupportTags::VolumeMixParameterPrecisionQ23, user_revision);
}
bool BehaviorInfo::UseBiquadFilterFloatProcessing() const {
return CheckFeatureSupported(SupportTags::BiquadFilterFloatProcessing, user_revision);
}
bool BehaviorInfo::IsMixInParameterDirtyOnlyUpdateSupported() const {
return CheckFeatureSupported(SupportTags::MixInParameterDirtyOnlyUpdate, user_revision);
}
bool BehaviorInfo::UseMultiTapBiquadFilterProcessing() const {
return CheckFeatureSupported(SupportTags::MultiTapBiquadFilterProcessing, user_revision);
}
bool BehaviorInfo::IsDeviceApiVersion2Supported() const {
return CheckFeatureSupported(SupportTags::DeviceApiVersion2, user_revision);
}
bool BehaviorInfo::IsDelayChannelMappingChanged() const {
return CheckFeatureSupported(SupportTags::DelayChannelMappingChange, user_revision);
}
bool BehaviorInfo::IsReverbChannelMappingChanged() const {
return CheckFeatureSupported(SupportTags::ReverbChannelMappingChange, user_revision);
}
bool BehaviorInfo::IsI3dl2ReverbChannelMappingChanged() const {
return CheckFeatureSupported(SupportTags::I3dl2ReverbChannelMappingChange, user_revision);
}
} // 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 <array>
#include <span>
#include "audio_core/common/common.h"
#include "common/common_types.h"
#include "core/hle/service/audio/errors.h"
namespace AudioCore::AudioRenderer {
/**
* Holds host and user revisions, checks whether render features can be enabled, and reports errors.
*/
class BehaviorInfo {
static constexpr u32 MaxErrors = 10;
public:
struct ErrorInfo {
/* 0x00 */ Result error_code{0};
/* 0x04 */ u32 unk_04;
/* 0x08 */ CpuAddr address;
};
static_assert(sizeof(ErrorInfo) == 0x10, "BehaviorInfo::ErrorInfo has the wrong size!");
struct Flags {
u64 IsMemoryForceMappingEnabled : 1;
};
struct InParameter {
/* 0x00 */ u32 revision;
/* 0x08 */ Flags flags;
};
static_assert(sizeof(InParameter) == 0x10, "BehaviorInfo::InParameter has the wrong size!");
struct OutStatus {
/* 0x00 */ std::array<ErrorInfo, MaxErrors> errors;
/* 0xA0 */ u32 error_count;
/* 0xA4 */ char unkA4[0xC];
};
static_assert(sizeof(OutStatus) == 0xB0, "BehaviorInfo::OutStatus has the wrong size!");
BehaviorInfo();
/**
* Get the host revision as a number.
*
* @return The host revision.
*/
u32 GetProcessRevisionNum() const;
/**
* Get the host revision in chars, e.g REV8.
* Rev 10 and higher use the ascii characters above 9.
* E.g:
* Rev 10 = REV:
* Rev 11 = REV;
*
* @return The host revision.
*/
u32 GetProcessRevision() const;
/**
* Get the user revision as a number.
*
* @return The user revision.
*/
u32 GetUserRevisionNum() const;
/**
* Get the user revision in chars, e.g REV8.
* Rev 10 and higher use the ascii characters above 9. REV: REV; etc.
*
* @return The user revision.
*/
u32 GetUserRevision() const;
/**
* Set the user revision.
*
* @param user_revision - The user's revision.
*/
void SetUserLibRevision(u32 user_revision);
/**
* Clear the current error count.
*/
void ClearError();
/**
* Append an error to the error list.
*
* @param error - The new error.
*/
void AppendError(ErrorInfo& error);
/**
* Copy errors to the given output container.
*
* @param out_errors - Output container to receive the errors.
* @param out_count - The number of errors written.
*/
void CopyErrorInfo(std::span<ErrorInfo> out_errors, u32& out_count);
/**
* Update the behaviour flags.
*
* @param flags - New flags to use.
*/
void UpdateFlags(Flags flags);
/**
* Check if memory pools can be forcibly mapped.
*
* @return True if enabled, otherwise false.
*/
bool IsMemoryForceMappingEnabled() const;
/**
* Check if the ADPCM context bug is fixed.
* The ADPCM context was not being sent to the AudioRenderer, leading to incorrect scaling being
* used.
*
* @return True if fixed, otherwise false.
*/
bool IsAdpcmLoopContextBugFixed() const;
/**
* Check if the splitter is supported.
*
* @return True if supported, otherwise false.
*/
bool IsSplitterSupported() const;
/**
* Check if the splitter bug is fixed.
* Update is given the wrong number of splitter destinations, leading to invalid data
* being processed.
*
* @return True if supported, otherwise false.
*/
bool IsSplitterBugFixed() const;
/**
* Check if effects version 2 are supported.
* This gives support for returning effect states from the AudioRenderer, currently only used
* for Limiter statistics.
*
* @return True if supported, otherwise false.
*/
bool IsEffectInfoVersion2Supported() const;
/**
* Check if a variadic command buffer is supported.
* As of Rev 5 with the added optional performance metric logging, the command
* buffer can be a variable size, so take that into account for calcualting its size.
*
* @return True if supported, otherwise false.
*/
bool IsVariadicCommandBufferSizeSupported() const;
/**
* Check if wave buffers version 2 are supported.
* See WaveBufferVersion1 and WaveBufferVersion2.
*
* @return True if supported, otherwise false.
*/
bool IsWaveBufferVer2Supported() const;
/**
* Check if long size pre delay is supported.
* This allows a longer initial delay time for the Reverb command.
*
* @return True if supported, otherwise false.
*/
bool IsLongSizePreDelaySupported() const;
/**
* Check if the command time estimator version 2 is supported.
*
* @return True if supported, otherwise false.
*/
bool IsCommandProcessingTimeEstimatorVersion2Supported() const;
/**
* Check if the command time estimator version 3 is supported.
*
* @return True if supported, otherwise false.
*/
bool IsCommandProcessingTimeEstimatorVersion3Supported() const;
/**
* Check if the command time estimator version 4 is supported.
*
* @return True if supported, otherwise false.
*/
bool IsCommandProcessingTimeEstimatorVersion4Supported() const;
/**
* Check if the command time estimator version 5 is supported.
*
* @return True if supported, otherwise false.
*/
bool IsCommandProcessingTimeEstimatorVersion5Supported() const;
/**
* Check if the AudioRenderer can use up to 70% of the allocated processing timeslice.
*
* @return True if supported, otherwise false.
*/
bool IsAudioRendererProcessingTimeLimit70PercentSupported() const;
/**
* Check if the AudioRenderer can use up to 75% of the allocated processing timeslice.
*
* @return True if supported, otherwise false.
*/
bool IsAudioRendererProcessingTimeLimit75PercentSupported() const;
/**
* Check if the AudioRenderer can use up to 80% of the allocated processing timeslice.
*
* @return True if supported, otherwise false.
*/
bool IsAudioRendererProcessingTimeLimit80PercentSupported() const;
/**
* Check if voice flushing is supported
* This allowws low-priority voices to be dropped if the AudioRenderer is running behind.
*
* @return True if supported, otherwise false.
*/
bool IsFlushVoiceWaveBuffersSupported() const;
/**
* Check if counting the number of elapsed frames is supported.
* This adds extra output to RequestUpdate, returning the number of times the AudioRenderer
* processed a command list.
*
* @return True if supported, otherwise false.
*/
bool IsElapsedFrameCountSupported() const;
/**
* Check if performance metrics version 2 are supported.
* This adds extra output to RequestUpdate, returning the number of times the AudioRenderer
* (Unused?).
*
* @return True if supported, otherwise false.
*/
bool IsPerformanceMetricsDataFormatVersion2Supported() const;
/**
* Get the supported performance metrics version.
* Version 2 logs some extra fields in output, such as number of voices dropped,
* processing start time, if the AudioRenderer exceeded its time, etc.
*
* @return Version supported, either 1 or 2.
*/
size_t GetPerformanceMetricsDataFormat() const;
/**
* Check if skipping voice pitch and sample rate conversion is supported.
* This speeds up the data source commands by skipping resampling if unwanted.
* See AudioCore::AudioRenderer::DecodeFromWaveBuffers
*
* @return True if supported, otherwise false.
*/
bool IsVoicePitchAndSrcSkippedSupported() const;
/**
* Check if resetting played sample count at loop points is supported.
* This resets the number of samples played in a voice state when a loop point is reached.
* See AudioCore::AudioRenderer::DecodeFromWaveBuffers
*
* @return True if supported, otherwise false.
*/
bool IsVoicePlayedSampleCountResetAtLoopPointSupported() const;
/**
* Check if the clear state bug for biquad filters is fixed.
* The biquad state was not marked as needing re-initialisation when the effect was updated, it
* was only initialized once with a new effect.
*
* @return True if fixed, otherwise false.
*/
bool IsBiquadFilterEffectStateClearBugFixed() const;
/**
* Check if Q23 precision is supported for fixed point.
*
* @return True if supported, otherwise false.
*/
bool IsVolumeMixParameterPrecisionQ23Supported() const;
/**
* Check if float processing for biuad filters is supported.
*
* @return True if supported, otherwise false.
*/
bool UseBiquadFilterFloatProcessing() const;
/**
* Check if dirty-only mix updates are supported.
* This saves a lot of buffer size as mixes can be large and not change much.
*
* @return True if supported, otherwise false.
*/
bool IsMixInParameterDirtyOnlyUpdateSupported() const;
/**
* Check if multi-tap biquad filters are supported.
*
* @return True if supported, otherwise false.
*/
bool UseMultiTapBiquadFilterProcessing() const;
/**
* Check if device api version 2 is supported.
* In the SDK but not in any sysmodule? Not sure, left here for completeness anyway.
*
* @return True if supported, otherwise false.
*/
bool IsDeviceApiVersion2Supported() const;
/**
* Check if new channel mappings are used for Delay commands.
* Older commands used:
* front left/front right/back left/back right/center/lfe
* Whereas everywhere else in the code uses:
* front left/front right/center/lfe/back left/back right
* This corrects that and makes everything standardised.
*
* @return True if supported, otherwise false.
*/
bool IsDelayChannelMappingChanged() const;
/**
* Check if new channel mappings are used for Reverb commands.
* Older commands used:
* front left/front right/back left/back right/center/lfe
* Whereas everywhere else in the code uses:
* front left/front right/center/lfe/back left/back right
* This corrects that and makes everything standardised.
*
* @return True if supported, otherwise false.
*/
bool IsReverbChannelMappingChanged() const;
/**
* Check if new channel mappings are used for I3dl2Reverb commands.
* Older commands used:
* front left/front right/back left/back right/center/lfe
* Whereas everywhere else in the code uses:
* front left/front right/center/lfe/back left/back right
* This corrects that and makes everything standardised.
*
* @return True if supported, otherwise false.
*/
bool IsI3dl2ReverbChannelMappingChanged() const;
/// Host version
u32 process_revision;
/// User version
u32 user_revision{};
/// Behaviour flags
Flags flags{};
/// Errors generated and reported during Update
std::array<ErrorInfo, MaxErrors> errors{};
/// Error count
u32 error_count{};
};
} // namespace AudioCore::AudioRenderer

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/common/feature_support.h"
#include "audio_core/renderer/behavior/behavior_info.h"
#include "audio_core/renderer/behavior/info_updater.h"
#include "audio_core/renderer/effect/effect_context.h"
#include "audio_core/renderer/effect/effect_reset.h"
#include "audio_core/renderer/memory/memory_pool_info.h"
#include "audio_core/renderer/mix/mix_context.h"
#include "audio_core/renderer/performance/performance_manager.h"
#include "audio_core/renderer/sink/circular_buffer_sink_info.h"
#include "audio_core/renderer/sink/device_sink_info.h"
#include "audio_core/renderer/sink/sink_context.h"
#include "audio_core/renderer/splitter/splitter_context.h"
#include "audio_core/renderer/voice/voice_context.h"
namespace AudioCore::AudioRenderer {
InfoUpdater::InfoUpdater(std::span<const u8> input_, std::span<u8> output_,
const u32 process_handle_, BehaviorInfo& behaviour_)
: input{input_.data() + sizeof(UpdateDataHeader)},
input_origin{input_}, output{output_.data() + sizeof(UpdateDataHeader)},
output_origin{output_}, in_header{reinterpret_cast<const UpdateDataHeader*>(
input_origin.data())},
out_header{reinterpret_cast<UpdateDataHeader*>(output_origin.data())},
expected_input_size{input_.size()}, expected_output_size{output_.size()},
process_handle{process_handle_}, behaviour{behaviour_} {
std::construct_at<UpdateDataHeader>(out_header, behaviour.GetProcessRevision());
}
Result InfoUpdater::UpdateVoiceChannelResources(VoiceContext& voice_context) {
const auto voice_count{voice_context.GetCount()};
std::span<const VoiceChannelResource::InParameter> in_params{
reinterpret_cast<const VoiceChannelResource::InParameter*>(input), voice_count};
for (u32 i = 0; i < voice_count; i++) {
auto& resource{voice_context.GetChannelResource(i)};
resource.in_use = in_params[i].in_use;
if (in_params[i].in_use) {
resource.mix_volumes = in_params[i].mix_volumes;
}
}
const auto consumed_input_size{voice_count *
static_cast<u32>(sizeof(VoiceChannelResource::InParameter))};
if (consumed_input_size != in_header->voice_resources_size) {
LOG_ERROR(Service_Audio,
"Consumed an incorrect voice resource size, header size={}, consumed={}",
in_header->voice_resources_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
input += consumed_input_size;
return ResultSuccess;
}
Result InfoUpdater::UpdateVoices(VoiceContext& voice_context,
std::span<MemoryPoolInfo> memory_pools,
const u32 memory_pool_count) {
const PoolMapper pool_mapper(process_handle, memory_pools, memory_pool_count,
behaviour.IsMemoryForceMappingEnabled());
const auto voice_count{voice_context.GetCount()};
std::span<const VoiceInfo::InParameter> in_params{
reinterpret_cast<const VoiceInfo::InParameter*>(input), voice_count};
std::span<VoiceInfo::OutStatus> out_params{reinterpret_cast<VoiceInfo::OutStatus*>(output),
voice_count};
for (u32 i = 0; i < voice_count; i++) {
auto& voice_info{voice_context.GetInfo(i)};
voice_info.in_use = false;
}
u32 new_voice_count{0};
for (u32 i = 0; i < voice_count; i++) {
const auto& in_param{in_params[i]};
std::array<VoiceState*, MaxChannels> voice_states{};
if (!in_param.in_use) {
continue;
}
auto& voice_info{voice_context.GetInfo(in_param.id)};
for (u32 channel = 0; channel < in_param.channel_count; channel++) {
voice_states[channel] = &voice_context.GetState(in_param.channel_resource_ids[channel]);
}
if (in_param.is_new) {
voice_info.Initialize();
for (u32 channel = 0; channel < in_param.channel_count; channel++) {
std::memset(voice_states[channel], 0, sizeof(VoiceState));
}
}
BehaviorInfo::ErrorInfo update_error{};
voice_info.UpdateParameters(update_error, in_param, pool_mapper, behaviour);
if (!update_error.error_code.IsSuccess()) {
behaviour.AppendError(update_error);
}
std::array<std::array<BehaviorInfo::ErrorInfo, 2>, MaxWaveBuffers> wavebuffer_errors{};
voice_info.UpdateWaveBuffers(wavebuffer_errors, MaxWaveBuffers * 2, in_param, voice_states,
pool_mapper, behaviour);
for (auto& wavebuffer_error : wavebuffer_errors) {
for (auto& error : wavebuffer_error) {
if (error.error_code.IsError()) {
behaviour.AppendError(error);
}
}
}
voice_info.WriteOutStatus(out_params[i], in_param, voice_states);
new_voice_count += in_param.channel_count;
}
auto consumed_input_size{voice_count * static_cast<u32>(sizeof(VoiceInfo::InParameter))};
auto consumed_output_size{voice_count * static_cast<u32>(sizeof(VoiceInfo::OutStatus))};
if (consumed_input_size != in_header->voices_size) {
LOG_ERROR(Service_Audio, "Consumed an incorrect voices size, header size={}, consumed={}",
in_header->voices_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
out_header->voices_size = consumed_output_size;
out_header->size += consumed_output_size;
input += consumed_input_size;
output += consumed_output_size;
voice_context.SetActiveCount(new_voice_count);
return ResultSuccess;
}
Result InfoUpdater::UpdateEffects(EffectContext& effect_context, const bool renderer_active,
std::span<MemoryPoolInfo> memory_pools,
const u32 memory_pool_count) {
if (behaviour.IsEffectInfoVersion2Supported()) {
return UpdateEffectsVersion2(effect_context, renderer_active, memory_pools,
memory_pool_count);
} else {
return UpdateEffectsVersion1(effect_context, renderer_active, memory_pools,
memory_pool_count);
}
}
Result InfoUpdater::UpdateEffectsVersion1(EffectContext& effect_context, const bool renderer_active,
std::span<MemoryPoolInfo> memory_pools,
const u32 memory_pool_count) {
PoolMapper pool_mapper(process_handle, memory_pools, memory_pool_count,
behaviour.IsMemoryForceMappingEnabled());
const auto effect_count{effect_context.GetCount()};
std::span<const EffectInfoBase::InParameterVersion1> in_params{
reinterpret_cast<const EffectInfoBase::InParameterVersion1*>(input), effect_count};
std::span<EffectInfoBase::OutStatusVersion1> out_params{
reinterpret_cast<EffectInfoBase::OutStatusVersion1*>(output), effect_count};
for (u32 i = 0; i < effect_count; i++) {
auto effect_info{&effect_context.GetInfo(i)};
if (effect_info->GetType() != in_params[i].type) {
effect_info->ForceUnmapBuffers(pool_mapper);
ResetEffect(effect_info, in_params[i].type);
}
BehaviorInfo::ErrorInfo error_info{};
effect_info->Update(error_info, in_params[i], pool_mapper);
if (error_info.error_code.IsError()) {
behaviour.AppendError(error_info);
}
effect_info->StoreStatus(out_params[i], renderer_active);
}
auto consumed_input_size{effect_count *
static_cast<u32>(sizeof(EffectInfoBase::InParameterVersion1))};
auto consumed_output_size{effect_count *
static_cast<u32>(sizeof(EffectInfoBase::OutStatusVersion1))};
if (consumed_input_size != in_header->effects_size) {
LOG_ERROR(Service_Audio, "Consumed an incorrect effects size, header size={}, consumed={}",
in_header->effects_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
out_header->effects_size = consumed_output_size;
out_header->size += consumed_output_size;
input += consumed_input_size;
output += consumed_output_size;
return ResultSuccess;
}
Result InfoUpdater::UpdateEffectsVersion2(EffectContext& effect_context, const bool renderer_active,
std::span<MemoryPoolInfo> memory_pools,
const u32 memory_pool_count) {
PoolMapper pool_mapper(process_handle, memory_pools, memory_pool_count,
behaviour.IsMemoryForceMappingEnabled());
const auto effect_count{effect_context.GetCount()};
std::span<const EffectInfoBase::InParameterVersion2> in_params{
reinterpret_cast<const EffectInfoBase::InParameterVersion2*>(input), effect_count};
std::span<EffectInfoBase::OutStatusVersion2> out_params{
reinterpret_cast<EffectInfoBase::OutStatusVersion2*>(output), effect_count};
for (u32 i = 0; i < effect_count; i++) {
auto effect_info{&effect_context.GetInfo(i)};
if (effect_info->GetType() != in_params[i].type) {
effect_info->ForceUnmapBuffers(pool_mapper);
ResetEffect(effect_info, in_params[i].type);
}
BehaviorInfo::ErrorInfo error_info{};
effect_info->Update(error_info, in_params[i], pool_mapper);
if (error_info.error_code.IsError()) {
behaviour.AppendError(error_info);
}
effect_info->StoreStatus(out_params[i], renderer_active);
if (in_params[i].is_new) {
effect_info->InitializeResultState(effect_context.GetDspSharedResultState(i));
effect_info->InitializeResultState(effect_context.GetResultState(i));
}
effect_info->UpdateResultState(out_params[i].result_state,
effect_context.GetResultState(i));
}
auto consumed_input_size{effect_count *
static_cast<u32>(sizeof(EffectInfoBase::InParameterVersion2))};
auto consumed_output_size{effect_count *
static_cast<u32>(sizeof(EffectInfoBase::OutStatusVersion2))};
if (consumed_input_size != in_header->effects_size) {
LOG_ERROR(Service_Audio, "Consumed an incorrect effects size, header size={}, consumed={}",
in_header->effects_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
out_header->effects_size = consumed_output_size;
out_header->size += consumed_output_size;
input += consumed_input_size;
output += consumed_output_size;
return ResultSuccess;
}
Result InfoUpdater::UpdateMixes(MixContext& mix_context, const u32 mix_buffer_count,
EffectContext& effect_context, SplitterContext& splitter_context) {
s32 mix_count{0};
u32 consumed_input_size{0};
if (behaviour.IsMixInParameterDirtyOnlyUpdateSupported()) {
auto in_dirty_params{reinterpret_cast<const MixInfo::InDirtyParameter*>(input)};
mix_count = in_dirty_params->count;
input += sizeof(MixInfo::InDirtyParameter);
consumed_input_size = static_cast<u32>(sizeof(MixInfo::InDirtyParameter) +
mix_count * sizeof(MixInfo::InParameter));
} else {
mix_count = mix_context.GetCount();
consumed_input_size = static_cast<u32>(mix_count * sizeof(MixInfo::InParameter));
}
if (mix_buffer_count == 0) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
std::span<const MixInfo::InParameter> in_params{
reinterpret_cast<const MixInfo::InParameter*>(input), static_cast<size_t>(mix_count)};
u32 total_buffer_count{0};
for (s32 i = 0; i < mix_count; i++) {
const auto& params{in_params[i]};
if (params.in_use) {
total_buffer_count += params.buffer_count;
if (params.dest_mix_id > static_cast<s32>(mix_context.GetCount()) &&
params.dest_mix_id != UnusedMixId && params.mix_id != FinalMixId) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
}
}
if (total_buffer_count > mix_buffer_count) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
bool mix_dirty{false};
for (s32 i = 0; i < mix_count; i++) {
const auto& params{in_params[i]};
s32 mix_id{i};
if (behaviour.IsMixInParameterDirtyOnlyUpdateSupported()) {
mix_id = params.mix_id;
}
auto mix_info{mix_context.GetInfo(mix_id)};
if (mix_info->in_use != params.in_use) {
mix_info->in_use = params.in_use;
if (!params.in_use) {
mix_info->ClearEffectProcessingOrder();
}
mix_dirty = true;
}
if (params.in_use) {
mix_dirty |= mix_info->Update(mix_context.GetEdgeMatrix(), params, effect_context,
splitter_context, behaviour);
}
}
if (mix_dirty) {
if (behaviour.IsSplitterSupported() && splitter_context.UsingSplitter()) {
if (!mix_context.TSortInfo(splitter_context)) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
} else {
mix_context.SortInfo();
}
}
if (consumed_input_size != in_header->mix_size) {
LOG_ERROR(Service_Audio, "Consumed an incorrect mixes size, header size={}, consumed={}",
in_header->mix_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
input += mix_count * sizeof(MixInfo::InParameter);
return ResultSuccess;
}
Result InfoUpdater::UpdateSinks(SinkContext& sink_context, std::span<MemoryPoolInfo> memory_pools,
const u32 memory_pool_count) {
PoolMapper pool_mapper(process_handle, memory_pools, memory_pool_count,
behaviour.IsMemoryForceMappingEnabled());
std::span<const SinkInfoBase::InParameter> in_params{
reinterpret_cast<const SinkInfoBase::InParameter*>(input), memory_pool_count};
std::span<SinkInfoBase::OutStatus> out_params{
reinterpret_cast<SinkInfoBase::OutStatus*>(output), memory_pool_count};
const auto sink_count{sink_context.GetCount()};
for (u32 i = 0; i < sink_count; i++) {
const auto& params{in_params[i]};
auto sink_info{sink_context.GetInfo(i)};
if (sink_info->GetType() != params.type) {
sink_info->CleanUp();
switch (params.type) {
case SinkInfoBase::Type::Invalid:
std::construct_at<SinkInfoBase>(reinterpret_cast<SinkInfoBase*>(sink_info));
break;
case SinkInfoBase::Type::DeviceSink:
std::construct_at<DeviceSinkInfo>(reinterpret_cast<DeviceSinkInfo*>(sink_info));
break;
case SinkInfoBase::Type::CircularBufferSink:
std::construct_at<CircularBufferSinkInfo>(
reinterpret_cast<CircularBufferSinkInfo*>(sink_info));
break;
default:
LOG_ERROR(Service_Audio, "Invalid sink type {}", static_cast<u32>(params.type));
break;
}
}
BehaviorInfo::ErrorInfo error_info{};
sink_info->Update(error_info, out_params[i], params, pool_mapper);
if (error_info.error_code.IsError()) {
behaviour.AppendError(error_info);
}
}
const auto consumed_input_size{sink_count *
static_cast<u32>(sizeof(SinkInfoBase::InParameter))};
const auto consumed_output_size{sink_count * static_cast<u32>(sizeof(SinkInfoBase::OutStatus))};
if (consumed_input_size != in_header->sinks_size) {
LOG_ERROR(Service_Audio, "Consumed an incorrect sinks size, header size={}, consumed={}",
in_header->sinks_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
input += consumed_input_size;
output += consumed_output_size;
out_header->sinks_size = consumed_output_size;
out_header->size += consumed_output_size;
return ResultSuccess;
}
Result InfoUpdater::UpdateMemoryPools(std::span<MemoryPoolInfo> memory_pools,
const u32 memory_pool_count) {
PoolMapper pool_mapper(process_handle, memory_pools, memory_pool_count,
behaviour.IsMemoryForceMappingEnabled());
std::span<const MemoryPoolInfo::InParameter> in_params{
reinterpret_cast<const MemoryPoolInfo::InParameter*>(input), memory_pool_count};
std::span<MemoryPoolInfo::OutStatus> out_params{
reinterpret_cast<MemoryPoolInfo::OutStatus*>(output), memory_pool_count};
for (size_t i = 0; i < memory_pool_count; i++) {
auto state{pool_mapper.Update(memory_pools[i], in_params[i], out_params[i])};
if (state != MemoryPoolInfo::ResultState::Success &&
state != MemoryPoolInfo::ResultState::BadParam &&
state != MemoryPoolInfo::ResultState::MapFailed &&
state != MemoryPoolInfo::ResultState::InUse) {
LOG_WARNING(Service_Audio, "Invalid ResultState from updating memory pools");
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
}
const auto consumed_input_size{memory_pool_count *
static_cast<u32>(sizeof(MemoryPoolInfo::InParameter))};
const auto consumed_output_size{memory_pool_count *
static_cast<u32>(sizeof(MemoryPoolInfo::OutStatus))};
if (consumed_input_size != in_header->memory_pool_size) {
LOG_ERROR(Service_Audio,
"Consumed an incorrect memory pool size, header size={}, consumed={}",
in_header->memory_pool_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
input += consumed_input_size;
output += consumed_output_size;
out_header->memory_pool_size = consumed_output_size;
out_header->size += consumed_output_size;
return ResultSuccess;
}
Result InfoUpdater::UpdatePerformanceBuffer(std::span<u8> performance_output,
const u64 performance_output_size,
PerformanceManager* performance_manager) {
auto in_params{reinterpret_cast<const PerformanceManager::InParameter*>(input)};
auto out_params{reinterpret_cast<PerformanceManager::OutStatus*>(output)};
if (performance_manager != nullptr) {
out_params->history_size =
performance_manager->CopyHistories(performance_output.data(), performance_output_size);
performance_manager->SetDetailTarget(in_params->target_node_id);
} else {
out_params->history_size = 0;
}
const auto consumed_input_size{static_cast<u32>(sizeof(PerformanceManager::InParameter))};
const auto consumed_output_size{static_cast<u32>(sizeof(PerformanceManager::OutStatus))};
if (consumed_input_size != in_header->performance_buffer_size) {
LOG_ERROR(Service_Audio,
"Consumed an incorrect performance size, header size={}, consumed={}",
in_header->performance_buffer_size, consumed_input_size);
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
input += consumed_input_size;
output += consumed_output_size;
out_header->performance_buffer_size = consumed_output_size;
out_header->size += consumed_output_size;
return ResultSuccess;
}
Result InfoUpdater::UpdateBehaviorInfo(BehaviorInfo& behaviour_) {
const auto in_params{reinterpret_cast<const BehaviorInfo::InParameter*>(input)};
if (!CheckValidRevision(in_params->revision)) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
if (in_params->revision != behaviour_.GetUserRevision()) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
behaviour_.ClearError();
behaviour_.UpdateFlags(in_params->flags);
if (in_header->behaviour_size != sizeof(BehaviorInfo::InParameter)) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
input += sizeof(BehaviorInfo::InParameter);
return ResultSuccess;
}
Result InfoUpdater::UpdateErrorInfo(BehaviorInfo& behaviour_) {
auto out_params{reinterpret_cast<BehaviorInfo::OutStatus*>(output)};
behaviour_.CopyErrorInfo(out_params->errors, out_params->error_count);
const auto consumed_output_size{static_cast<u32>(sizeof(BehaviorInfo::OutStatus))};
output += consumed_output_size;
out_header->behaviour_size = consumed_output_size;
out_header->size += consumed_output_size;
return ResultSuccess;
}
Result InfoUpdater::UpdateSplitterInfo(SplitterContext& splitter_context) {
u32 consumed_size{0};
if (!splitter_context.Update(input, consumed_size)) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
input += consumed_size;
return ResultSuccess;
}
Result InfoUpdater::UpdateRendererInfo(const u64 elapsed_frames) {
struct RenderInfo {
/* 0x00 */ u64 frames_elapsed;
/* 0x08 */ char unk08[0x8];
};
static_assert(sizeof(RenderInfo) == 0x10, "RenderInfo has the wrong size!");
auto out_params{reinterpret_cast<RenderInfo*>(output)};
out_params->frames_elapsed = elapsed_frames;
const auto consumed_output_size{static_cast<u32>(sizeof(RenderInfo))};
output += consumed_output_size;
out_header->render_info_size = consumed_output_size;
out_header->size += consumed_output_size;
return ResultSuccess;
}
Result InfoUpdater::CheckConsumedSize() {
if (CpuAddr(input) - CpuAddr(input_origin.data()) != expected_input_size) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
} else if (CpuAddr(output) - CpuAddr(output_origin.data()) != expected_output_size) {
return Service::Audio::ERR_INVALID_UPDATE_DATA;
}
return ResultSuccess;
}
} // 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 "common/common_types.h"
#include "core/hle/service/audio/errors.h"
namespace AudioCore::AudioRenderer {
class BehaviorInfo;
class VoiceContext;
class MixContext;
class SinkContext;
class SplitterContext;
class EffectContext;
class MemoryPoolInfo;
class PerformanceManager;
class InfoUpdater {
struct UpdateDataHeader {
explicit UpdateDataHeader(u32 revision_) : revision{revision_} {}
/* 0x00 */ u32 revision;
/* 0x04 */ u32 behaviour_size{};
/* 0x08 */ u32 memory_pool_size{};
/* 0x0C */ u32 voices_size{};
/* 0x10 */ u32 voice_resources_size{};
/* 0x14 */ u32 effects_size{};
/* 0x18 */ u32 mix_size{};
/* 0x1C */ u32 sinks_size{};
/* 0x20 */ u32 performance_buffer_size{};
/* 0x24 */ char unk24[4];
/* 0x28 */ u32 render_info_size{};
/* 0x2C */ char unk2C[0x10];
/* 0x3C */ u32 size{sizeof(UpdateDataHeader)};
};
static_assert(sizeof(UpdateDataHeader) == 0x40, "UpdateDataHeader has the wrong size!");
public:
explicit InfoUpdater(std::span<const u8> input, std::span<u8> output, u32 process_handle,
BehaviorInfo& behaviour);
/**
* Update the voice channel resources.
*
* @param voice_context - Voice context to update.
* @return Result code.
*/
Result UpdateVoiceChannelResources(VoiceContext& voice_context);
/**
* Update voices.
*
* @param voice_context - Voice context to update.
* @param memory_pools - Memory pools to use for these voices.
* @param memory_pool_count - Number of memory pools.
* @return Result code.
*/
Result UpdateVoices(VoiceContext& voice_context, std::span<MemoryPoolInfo> memory_pools,
u32 memory_pool_count);
/**
* Update effects.
*
* @param effect_context - Effect context to update.
* @param renderer_active - Whether the AudioRenderer is active.
* @param memory_pools - Memory pools to use for these voices.
* @param memory_pool_count - Number of memory pools.
* @return Result code.
*/
Result UpdateEffects(EffectContext& effect_context, bool renderer_active,
std::span<MemoryPoolInfo> memory_pools, u32 memory_pool_count);
/**
* Update mixes.
*
* @param mix_context - Mix context to update.
* @param mix_buffer_count - Number of mix buffers.
* @param effect_context - Effect context to update effort order.
* @param splitter_context - Splitter context for the mixes.
* @return Result code.
*/
Result UpdateMixes(MixContext& mix_context, u32 mix_buffer_count, EffectContext& effect_context,
SplitterContext& splitter_context);
/**
* Update sinks.
*
* @param sink_context - Sink context to update.
* @param memory_pools - Memory pools to use for these voices.
* @param memory_pool_count - Number of memory pools.
* @return Result code.
*/
Result UpdateSinks(SinkContext& sink_context, std::span<MemoryPoolInfo> memory_pools,
u32 memory_pool_count);
/**
* Update memory pools.
*
* @param memory_pools - Memory pools to use for these voices.
* @param memory_pool_count - Number of memory pools.
* @return Result code.
*/
Result UpdateMemoryPools(std::span<MemoryPoolInfo> memory_pools, u32 memory_pool_count);
/**
* Update the performance buffer.
*
* @param output - Output buffer for performance metrics.
* @param output_size - Output buffer size.
* @param performance_manager - Performance manager..
* @return Result code.
*/
Result UpdatePerformanceBuffer(std::span<u8> output, u64 output_size,
PerformanceManager* performance_manager);
/**
* Update behaviour.
*
* @param behaviour - Behaviour to update.
* @return Result code.
*/
Result UpdateBehaviorInfo(BehaviorInfo& behaviour);
/**
* Update errors.
*
* @param behaviour - Behaviour to update.
* @return Result code.
*/
Result UpdateErrorInfo(BehaviorInfo& behaviour);
/**
* Update splitter.
*
* @param splitter_context - Splitter context to update.
* @return Result code.
*/
Result UpdateSplitterInfo(SplitterContext& splitter_context);
/**
* Update renderer info.
*
* @param elapsed_frames - Number of elapsed frames.
* @return Result code.
*/
Result UpdateRendererInfo(u64 elapsed_frames);
/**
* Check that the input.output sizes match their expected values.
*
* @return Result code.
*/
Result CheckConsumedSize();
private:
/**
* Update effects version 1.
*
* @param effect_context - Effect context to update.
* @param renderer_active - Is the AudioRenderer active?
* @param memory_pools - Memory pools to use for these voices.
* @param memory_pool_count - Number of memory pools.
* @return Result code.
*/
Result UpdateEffectsVersion1(EffectContext& effect_context, bool renderer_active,
std::span<MemoryPoolInfo> memory_pools, u32 memory_pool_count);
/**
* Update effects version 2.
*
* @param effect_context - Effect context to update.
* @param renderer_active - Is the AudioRenderer active?
* @param memory_pools - Memory pools to use for these voices.
* @param memory_pool_count - Number of memory pools.
* @return Result code.
*/
Result UpdateEffectsVersion2(EffectContext& effect_context, bool renderer_active,
std::span<MemoryPoolInfo> memory_pools, u32 memory_pool_count);
/// Input buffer
u8 const* input;
/// Input buffer start
std::span<const u8> input_origin;
/// Output buffer start
u8* output;
/// Output buffer start
std::span<u8> output_origin;
/// Input header
const UpdateDataHeader* in_header;
/// Output header
UpdateDataHeader* out_header;
/// Expected input size, see CheckConsumedSize
u64 expected_input_size;
/// Expected output size, see CheckConsumedSize
u64 expected_output_size;
/// Unused
u32 process_handle;
/// Behaviour
BehaviorInfo& behaviour;
};
} // namespace AudioCore::AudioRenderer

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/renderer/behavior/behavior_info.h"
#include "audio_core/renderer/command/command_buffer.h"
#include "audio_core/renderer/command/command_list_header.h"
#include "audio_core/renderer/command/command_processing_time_estimator.h"
#include "audio_core/renderer/effect/biquad_filter.h"
#include "audio_core/renderer/effect/delay.h"
#include "audio_core/renderer/effect/reverb.h"
#include "audio_core/renderer/memory/memory_pool_info.h"
#include "audio_core/renderer/mix/mix_info.h"
#include "audio_core/renderer/sink/circular_buffer_sink_info.h"
#include "audio_core/renderer/sink/device_sink_info.h"
#include "audio_core/renderer/sink/sink_info_base.h"
#include "audio_core/renderer/voice/voice_info.h"
#include "audio_core/renderer/voice/voice_state.h"
namespace AudioCore::AudioRenderer {
template <typename T, CommandId Id>
T& CommandBuffer::GenerateStart(const s32 node_id) {
if (size + sizeof(T) >= command_list.size_bytes()) {
LOG_ERROR(
Service_Audio,
"Attempting to write commands beyond the end of allocated command buffer memory!");
UNREACHABLE();
}
auto& cmd{*std::construct_at<T>(reinterpret_cast<T*>(&command_list[size]))};
cmd.magic = CommandMagic;
cmd.enabled = true;
cmd.type = Id;
cmd.size = sizeof(T);
cmd.node_id = node_id;
return cmd;
}
template <typename T>
void CommandBuffer::GenerateEnd(T& cmd) {
cmd.estimated_process_time = time_estimator->Estimate(cmd);
estimated_process_time += cmd.estimated_process_time;
size += sizeof(T);
count++;
}
void CommandBuffer::GeneratePcmInt16Version1Command(const s32 node_id,
const MemoryPoolInfo& memory_pool_,
VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel) {
auto& cmd{
GenerateStart<PcmInt16DataSourceVersion1Command, CommandId::DataSourcePcmInt16Version1>(
node_id)};
cmd.src_quality = voice_info.src_quality;
cmd.output_index = buffer_count + channel;
cmd.flags = voice_info.flags & 3;
cmd.sample_rate = voice_info.sample_rate;
cmd.pitch = voice_info.pitch;
cmd.channel_index = channel;
cmd.channel_count = voice_info.channel_count;
for (u32 i = 0; i < MaxWaveBuffers; i++) {
voice_info.wavebuffers[i].Copy(cmd.wave_buffers[i]);
}
cmd.voice_state = memory_pool_.Translate(CpuAddr(&voice_state), sizeof(VoiceState));
GenerateEnd<PcmInt16DataSourceVersion1Command>(cmd);
}
void CommandBuffer::GeneratePcmInt16Version2Command(const s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel) {
auto& cmd{
GenerateStart<PcmInt16DataSourceVersion2Command, CommandId::DataSourcePcmInt16Version2>(
node_id)};
cmd.src_quality = voice_info.src_quality;
cmd.output_index = buffer_count + channel;
cmd.flags = voice_info.flags & 3;
cmd.sample_rate = voice_info.sample_rate;
cmd.pitch = voice_info.pitch;
cmd.channel_index = channel;
cmd.channel_count = voice_info.channel_count;
for (u32 i = 0; i < MaxWaveBuffers; i++) {
voice_info.wavebuffers[i].Copy(cmd.wave_buffers[i]);
}
cmd.voice_state = memory_pool->Translate(CpuAddr(&voice_state), sizeof(VoiceState));
GenerateEnd<PcmInt16DataSourceVersion2Command>(cmd);
}
void CommandBuffer::GeneratePcmFloatVersion1Command(const s32 node_id,
const MemoryPoolInfo& memory_pool_,
VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel) {
auto& cmd{
GenerateStart<PcmFloatDataSourceVersion1Command, CommandId::DataSourcePcmFloatVersion1>(
node_id)};
cmd.src_quality = voice_info.src_quality;
cmd.output_index = buffer_count + channel;
cmd.flags = voice_info.flags & 3;
cmd.sample_rate = voice_info.sample_rate;
cmd.pitch = voice_info.pitch;
cmd.channel_index = channel;
cmd.channel_count = voice_info.channel_count;
for (u32 i = 0; i < MaxWaveBuffers; i++) {
voice_info.wavebuffers[i].Copy(cmd.wave_buffers[i]);
}
cmd.voice_state = memory_pool_.Translate(CpuAddr(&voice_state), sizeof(VoiceState));
GenerateEnd<PcmFloatDataSourceVersion1Command>(cmd);
}
void CommandBuffer::GeneratePcmFloatVersion2Command(const s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel) {
auto& cmd{
GenerateStart<PcmFloatDataSourceVersion2Command, CommandId::DataSourcePcmFloatVersion2>(
node_id)};
cmd.src_quality = voice_info.src_quality;
cmd.output_index = buffer_count + channel;
cmd.flags = voice_info.flags & 3;
cmd.sample_rate = voice_info.sample_rate;
cmd.pitch = voice_info.pitch;
cmd.channel_index = channel;
cmd.channel_count = voice_info.channel_count;
for (u32 i = 0; i < MaxWaveBuffers; i++) {
voice_info.wavebuffers[i].Copy(cmd.wave_buffers[i]);
}
cmd.voice_state = memory_pool->Translate(CpuAddr(&voice_state), sizeof(VoiceState));
GenerateEnd<PcmFloatDataSourceVersion2Command>(cmd);
}
void CommandBuffer::GenerateAdpcmVersion1Command(const s32 node_id,
const MemoryPoolInfo& memory_pool_,
VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel) {
auto& cmd{
GenerateStart<AdpcmDataSourceVersion1Command, CommandId::DataSourceAdpcmVersion1>(node_id)};
cmd.src_quality = voice_info.src_quality;
cmd.output_index = buffer_count + channel;
cmd.flags = voice_info.flags & 3;
cmd.sample_rate = voice_info.sample_rate;
cmd.pitch = voice_info.pitch;
for (u32 i = 0; i < MaxWaveBuffers; i++) {
voice_info.wavebuffers[i].Copy(cmd.wave_buffers[i]);
}
cmd.voice_state = memory_pool_.Translate(CpuAddr(&voice_state), sizeof(VoiceState));
cmd.data_address = voice_info.data_address.GetReference(true);
cmd.data_size = voice_info.data_address.GetSize();
GenerateEnd<AdpcmDataSourceVersion1Command>(cmd);
}
void CommandBuffer::GenerateAdpcmVersion2Command(const s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel) {
auto& cmd{
GenerateStart<AdpcmDataSourceVersion2Command, CommandId::DataSourceAdpcmVersion2>(node_id)};
cmd.src_quality = voice_info.src_quality;
cmd.output_index = buffer_count + channel;
cmd.flags = voice_info.flags & 3;
cmd.sample_rate = voice_info.sample_rate;
cmd.pitch = voice_info.pitch;
cmd.channel_index = channel;
cmd.channel_count = voice_info.channel_count;
for (u32 i = 0; i < MaxWaveBuffers; i++) {
voice_info.wavebuffers[i].Copy(cmd.wave_buffers[i]);
}
cmd.voice_state = memory_pool->Translate(CpuAddr(&voice_state), sizeof(VoiceState));
cmd.data_address = voice_info.data_address.GetReference(true);
cmd.data_size = voice_info.data_address.GetSize();
GenerateEnd<AdpcmDataSourceVersion2Command>(cmd);
}
void CommandBuffer::GenerateVolumeCommand(const s32 node_id, const s16 buffer_offset,
const s16 input_index, const f32 volume,
const u8 precision) {
auto& cmd{GenerateStart<VolumeCommand, CommandId::Volume>(node_id)};
cmd.precision = precision;
cmd.input_index = buffer_offset + input_index;
cmd.output_index = buffer_offset + input_index;
cmd.volume = volume;
GenerateEnd<VolumeCommand>(cmd);
}
void CommandBuffer::GenerateVolumeRampCommand(const s32 node_id, VoiceInfo& voice_info,
const s16 buffer_count, const u8 precision) {
auto& cmd{GenerateStart<VolumeRampCommand, CommandId::VolumeRamp>(node_id)};
cmd.input_index = buffer_count;
cmd.output_index = buffer_count;
cmd.prev_volume = voice_info.prev_volume;
cmd.volume = voice_info.volume;
cmd.precision = precision;
GenerateEnd<VolumeRampCommand>(cmd);
}
void CommandBuffer::GenerateBiquadFilterCommand(const s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel,
const u32 biquad_index,
const bool use_float_processing) {
auto& cmd{GenerateStart<BiquadFilterCommand, CommandId::BiquadFilter>(node_id)};
cmd.input = buffer_count + channel;
cmd.output = buffer_count + channel;
cmd.biquad = voice_info.biquads[biquad_index];
cmd.state = memory_pool->Translate(CpuAddr(voice_state.biquad_states[biquad_index].data()),
MaxBiquadFilters * sizeof(VoiceState::BiquadFilterState));
cmd.needs_init = !voice_info.biquad_initialized[biquad_index];
cmd.use_float_processing = use_float_processing;
GenerateEnd<BiquadFilterCommand>(cmd);
}
void CommandBuffer::GenerateBiquadFilterCommand(const s32 node_id, EffectInfoBase& effect_info,
const s16 buffer_offset, const s8 channel,
const bool needs_init,
const bool use_float_processing) {
auto& cmd{GenerateStart<BiquadFilterCommand, CommandId::BiquadFilter>(node_id)};
const auto& parameter{
*reinterpret_cast<BiquadFilterInfo::ParameterVersion1*>(effect_info.GetParameter())};
const auto state{
reinterpret_cast<VoiceState::BiquadFilterState*>(effect_info.GetStateBuffer())};
cmd.input = buffer_offset + parameter.inputs[channel];
cmd.output = buffer_offset + parameter.outputs[channel];
cmd.biquad.b = parameter.b;
cmd.biquad.a = parameter.a;
cmd.state = memory_pool->Translate(CpuAddr(state),
MaxBiquadFilters * sizeof(VoiceState::BiquadFilterState));
cmd.needs_init = needs_init;
cmd.use_float_processing = use_float_processing;
GenerateEnd<BiquadFilterCommand>(cmd);
}
void CommandBuffer::GenerateMixCommand(const s32 node_id, const s16 input_index,
const s16 output_index, const s16 buffer_offset,
const f32 volume, const u8 precision) {
auto& cmd{GenerateStart<MixCommand, CommandId::Mix>(node_id)};
cmd.input_index = input_index;
cmd.output_index = output_index;
cmd.volume = volume;
cmd.precision = precision;
GenerateEnd<MixCommand>(cmd);
}
void CommandBuffer::GenerateMixRampCommand(const s32 node_id,
[[maybe_unused]] const s16 buffer_count,
const s16 input_index, const s16 output_index,
const f32 volume, const f32 prev_volume,
const CpuAddr prev_samples, const u8 precision) {
if (volume == 0.0f && prev_volume == 0.0f) {
return;
}
auto& cmd{GenerateStart<MixRampCommand, CommandId::MixRamp>(node_id)};
cmd.input_index = input_index;
cmd.output_index = output_index;
cmd.prev_volume = prev_volume;
cmd.volume = volume;
cmd.previous_sample = prev_samples;
cmd.precision = precision;
GenerateEnd<MixRampCommand>(cmd);
}
void CommandBuffer::GenerateMixRampGroupedCommand(const s32 node_id, const s16 buffer_count,
const s16 input_index, s16 output_index,
std::span<const f32> volumes,
std::span<const f32> prev_volumes,
const CpuAddr prev_samples, const u8 precision) {
auto& cmd{GenerateStart<MixRampGroupedCommand, CommandId::MixRampGrouped>(node_id)};
cmd.buffer_count = buffer_count;
for (s32 i = 0; i < buffer_count; i++) {
cmd.inputs[i] = input_index;
cmd.outputs[i] = output_index++;
cmd.prev_volumes[i] = prev_volumes[i];
cmd.volumes[i] = volumes[i];
}
cmd.previous_samples = prev_samples;
cmd.precision = precision;
GenerateEnd<MixRampGroupedCommand>(cmd);
}
void CommandBuffer::GenerateDepopPrepareCommand(const s32 node_id, const VoiceState& voice_state,
std::span<const s32> buffer, const s16 buffer_count,
s16 buffer_offset, const bool was_playing) {
auto& cmd{GenerateStart<DepopPrepareCommand, CommandId::DepopPrepare>(node_id)};
cmd.enabled = was_playing;
for (u32 i = 0; i < MaxMixBuffers; i++) {
cmd.inputs[i] = buffer_offset++;
}
cmd.previous_samples = memory_pool->Translate(CpuAddr(voice_state.previous_samples.data()),
MaxMixBuffers * sizeof(s32));
cmd.buffer_count = buffer_count;
cmd.depop_buffer = memory_pool->Translate(CpuAddr(buffer.data()), buffer_count * sizeof(s32));
GenerateEnd<DepopPrepareCommand>(cmd);
}
void CommandBuffer::GenerateDepopForMixBuffersCommand(const s32 node_id, const MixInfo& mix_info,
std::span<const s32> depop_buffer) {
auto& cmd{GenerateStart<DepopForMixBuffersCommand, CommandId::DepopForMixBuffers>(node_id)};
cmd.input = mix_info.buffer_offset;
cmd.count = mix_info.buffer_count;
cmd.decay = mix_info.sample_rate == TargetSampleRate ? 0.96218872f : 0.94369507f;
cmd.depop_buffer =
memory_pool->Translate(CpuAddr(depop_buffer.data()), mix_info.buffer_count * sizeof(s32));
GenerateEnd<DepopForMixBuffersCommand>(cmd);
}
void CommandBuffer::GenerateDelayCommand(const s32 node_id, EffectInfoBase& effect_info,
const s16 buffer_offset) {
auto& cmd{GenerateStart<DelayCommand, CommandId::Delay>(node_id)};
const auto& parameter{
*reinterpret_cast<DelayInfo::ParameterVersion1*>(effect_info.GetParameter())};
const auto state{effect_info.GetStateBuffer()};
if (IsChannelCountValid(parameter.channel_count)) {
const auto state_buffer{memory_pool->Translate(CpuAddr(state), sizeof(DelayInfo::State))};
if (state_buffer) {
for (s16 channel = 0; channel < parameter.channel_count; channel++) {
cmd.inputs[channel] = buffer_offset + parameter.inputs[channel];
cmd.outputs[channel] = buffer_offset + parameter.outputs[channel];
}
if (!behavior->IsDelayChannelMappingChanged() && parameter.channel_count == 6) {
UseOldChannelMapping(cmd.inputs, cmd.outputs);
}
cmd.parameter = parameter;
cmd.effect_enabled = effect_info.IsEnabled();
cmd.state = state_buffer;
cmd.workbuffer = effect_info.GetWorkbuffer(-1);
}
}
GenerateEnd<DelayCommand>(cmd);
}
void CommandBuffer::GenerateUpsampleCommand(const s32 node_id, const s16 buffer_offset,
UpsamplerInfo& upsampler_info, const u32 input_count,
std::span<const s8> inputs, const s16 buffer_count,
const u32 sample_count_, const u32 sample_rate_) {
auto& cmd{GenerateStart<UpsampleCommand, CommandId::Upsample>(node_id)};
cmd.samples_buffer = memory_pool->Translate(upsampler_info.samples_pos,
upsampler_info.sample_count * sizeof(s32));
cmd.inputs = memory_pool->Translate(CpuAddr(upsampler_info.inputs.data()), MaxChannels);
cmd.buffer_count = buffer_count;
cmd.unk_20 = 0;
cmd.source_sample_count = sample_count_;
cmd.source_sample_rate = sample_rate_;
upsampler_info.input_count = input_count;
for (u32 i = 0; i < input_count; i++) {
upsampler_info.inputs[i] = buffer_offset + inputs[i];
}
cmd.upsampler_info = memory_pool->Translate(CpuAddr(&upsampler_info), sizeof(UpsamplerInfo));
GenerateEnd<UpsampleCommand>(cmd);
}
void CommandBuffer::GenerateDownMix6chTo2chCommand(const s32 node_id, std::span<const s8> inputs,
const s16 buffer_offset,
std::span<const f32> downmix_coeff) {
auto& cmd{GenerateStart<DownMix6chTo2chCommand, CommandId::DownMix6chTo2ch>(node_id)};
for (u32 i = 0; i < MaxChannels; i++) {
cmd.inputs[i] = buffer_offset + inputs[i];
cmd.outputs[i] = buffer_offset + inputs[i];
}
for (u32 i = 0; i < 4; i++) {
cmd.down_mix_coeff[i] = downmix_coeff[i];
}
GenerateEnd<DownMix6chTo2chCommand>(cmd);
}
void CommandBuffer::GenerateAuxCommand(const s32 node_id, EffectInfoBase& effect_info,
const s16 input_index, const s16 output_index,
const s16 buffer_offset, const u32 update_count,
const u32 count_max, const u32 write_offset) {
auto& cmd{GenerateStart<AuxCommand, CommandId::Aux>(node_id)};
if (effect_info.GetSendBuffer() != 0 && effect_info.GetReturnBuffer() != 0) {
cmd.input = buffer_offset + input_index;
cmd.output = buffer_offset + output_index;
cmd.send_buffer_info = effect_info.GetSendBufferInfo();
cmd.send_buffer = effect_info.GetSendBuffer();
cmd.return_buffer_info = effect_info.GetReturnBufferInfo();
cmd.return_buffer = effect_info.GetReturnBuffer();
cmd.count_max = count_max;
cmd.write_offset = write_offset;
cmd.update_count = update_count;
cmd.effect_enabled = effect_info.IsEnabled();
}
GenerateEnd<AuxCommand>(cmd);
}
void CommandBuffer::GenerateDeviceSinkCommand(const s32 node_id, const s16 buffer_offset,
SinkInfoBase& sink_info, const u32 session_id,
std::span<s32> samples_buffer) {
auto& cmd{GenerateStart<DeviceSinkCommand, CommandId::DeviceSink>(node_id)};
const auto& parameter{
*reinterpret_cast<DeviceSinkInfo::DeviceInParameter*>(sink_info.GetParameter())};
auto state{*reinterpret_cast<DeviceSinkInfo::DeviceState*>(sink_info.GetState())};
cmd.session_id = session_id;
if (state.upsampler_info != nullptr) {
const auto size_{state.upsampler_info->sample_count * parameter.input_count};
const auto size_bytes{size_ * sizeof(s32)};
const auto addr{memory_pool->Translate(state.upsampler_info->samples_pos, size_bytes)};
cmd.sample_buffer = {reinterpret_cast<s32*>(addr),
parameter.input_count * state.upsampler_info->sample_count};
} else {
cmd.sample_buffer = samples_buffer;
}
cmd.input_count = parameter.input_count;
for (u32 i = 0; i < parameter.input_count; i++) {
cmd.inputs[i] = buffer_offset + parameter.inputs[i];
}
GenerateEnd<DeviceSinkCommand>(cmd);
}
void CommandBuffer::GenerateCircularBufferSinkCommand(const s32 node_id, SinkInfoBase& sink_info,
const s16 buffer_offset) {
auto& cmd{GenerateStart<CircularBufferSinkCommand, CommandId::CircularBufferSink>(node_id)};
const auto& parameter{*reinterpret_cast<CircularBufferSinkInfo::CircularBufferInParameter*>(
sink_info.GetParameter())};
auto state{
*reinterpret_cast<CircularBufferSinkInfo::CircularBufferState*>(sink_info.GetState())};
cmd.input_count = parameter.input_count;
for (u32 i = 0; i < parameter.input_count; i++) {
cmd.inputs[i] = buffer_offset + parameter.inputs[i];
}
cmd.address = state.address_info.GetReference(true);
cmd.size = parameter.size;
cmd.pos = state.current_pos;
GenerateEnd<CircularBufferSinkCommand>(cmd);
}
void CommandBuffer::GenerateReverbCommand(const s32 node_id, EffectInfoBase& effect_info,
const s16 buffer_offset,
const bool long_size_pre_delay_supported) {
auto& cmd{GenerateStart<ReverbCommand, CommandId::Reverb>(node_id)};
const auto& parameter{
*reinterpret_cast<ReverbInfo::ParameterVersion2*>(effect_info.GetParameter())};
const auto state{effect_info.GetStateBuffer()};
if (IsChannelCountValid(parameter.channel_count)) {
const auto state_buffer{memory_pool->Translate(CpuAddr(state), sizeof(ReverbInfo::State))};
if (state_buffer) {
for (s16 channel = 0; channel < parameter.channel_count; channel++) {
cmd.inputs[channel] = buffer_offset + parameter.inputs[channel];
cmd.outputs[channel] = buffer_offset + parameter.outputs[channel];
}
if (!behavior->IsReverbChannelMappingChanged() && parameter.channel_count == 6) {
UseOldChannelMapping(cmd.inputs, cmd.outputs);
}
cmd.parameter = parameter;
cmd.effect_enabled = effect_info.IsEnabled();
cmd.state = state_buffer;
cmd.workbuffer = effect_info.GetWorkbuffer(-1);
cmd.long_size_pre_delay_supported = long_size_pre_delay_supported;
}
}
GenerateEnd<ReverbCommand>(cmd);
}
void CommandBuffer::GenerateI3dl2ReverbCommand(const s32 node_id, EffectInfoBase& effect_info,
const s16 buffer_offset) {
auto& cmd{GenerateStart<I3dl2ReverbCommand, CommandId::I3dl2Reverb>(node_id)};
const auto& parameter{
*reinterpret_cast<I3dl2ReverbInfo::ParameterVersion1*>(effect_info.GetParameter())};
const auto state{effect_info.GetStateBuffer()};
if (IsChannelCountValid(parameter.channel_count)) {
const auto state_buffer{
memory_pool->Translate(CpuAddr(state), sizeof(I3dl2ReverbInfo::State))};
if (state_buffer) {
for (s16 channel = 0; channel < parameter.channel_count; channel++) {
cmd.inputs[channel] = buffer_offset + parameter.inputs[channel];
cmd.outputs[channel] = buffer_offset + parameter.outputs[channel];
}
if (!behavior->IsI3dl2ReverbChannelMappingChanged() && parameter.channel_count == 6) {
UseOldChannelMapping(cmd.inputs, cmd.outputs);
}
cmd.parameter = parameter;
cmd.effect_enabled = effect_info.IsEnabled();
cmd.state = state_buffer;
cmd.workbuffer = effect_info.GetWorkbuffer(-1);
}
}
GenerateEnd<I3dl2ReverbCommand>(cmd);
}
void CommandBuffer::GeneratePerformanceCommand(const s32 node_id, const PerformanceState state,
const PerformanceEntryAddresses& entry_addresses) {
auto& cmd{GenerateStart<PerformanceCommand, CommandId::Performance>(node_id)};
cmd.state = state;
cmd.entry_address = entry_addresses;
GenerateEnd<PerformanceCommand>(cmd);
}
void CommandBuffer::GenerateClearMixCommand(const s32 node_id) {
auto& cmd{GenerateStart<ClearMixBufferCommand, CommandId::ClearMixBuffer>(node_id)};
GenerateEnd<ClearMixBufferCommand>(cmd);
}
void CommandBuffer::GenerateCopyMixBufferCommand(const s32 node_id, EffectInfoBase& effect_info,
const s16 buffer_offset, const s8 channel) {
auto& cmd{GenerateStart<CopyMixBufferCommand, CommandId::CopyMixBuffer>(node_id)};
const auto& parameter{
*reinterpret_cast<BiquadFilterInfo::ParameterVersion1*>(effect_info.GetParameter())};
cmd.input_index = buffer_offset + parameter.inputs[channel];
cmd.output_index = buffer_offset + parameter.outputs[channel];
GenerateEnd<CopyMixBufferCommand>(cmd);
}
void CommandBuffer::GenerateLightLimiterCommand(
const s32 node_id, const s16 buffer_offset,
const LightLimiterInfo::ParameterVersion1& parameter, const LightLimiterInfo::State& state,
const bool enabled, const CpuAddr workbuffer) {
auto& cmd{GenerateStart<LightLimiterVersion1Command, CommandId::LightLimiterVersion1>(node_id)};
if (IsChannelCountValid(parameter.channel_count)) {
const auto state_buffer{
memory_pool->Translate(CpuAddr(&state), sizeof(LightLimiterInfo::State))};
if (state_buffer) {
for (s8 channel = 0; channel < parameter.channel_count; channel++) {
cmd.inputs[channel] = buffer_offset + parameter.inputs[channel];
cmd.outputs[channel] = buffer_offset + parameter.outputs[channel];
}
std::memcpy(&cmd.parameter, &parameter, sizeof(LightLimiterInfo::ParameterVersion1));
cmd.effect_enabled = enabled;
cmd.state = state_buffer;
cmd.workbuffer = workbuffer;
}
}
GenerateEnd<LightLimiterVersion1Command>(cmd);
}
void CommandBuffer::GenerateLightLimiterCommand(
const s32 node_id, const s16 buffer_offset,
const LightLimiterInfo::ParameterVersion2& parameter,
const LightLimiterInfo::StatisticsInternal& statistics, const LightLimiterInfo::State& state,
const bool enabled, const CpuAddr workbuffer) {
auto& cmd{GenerateStart<LightLimiterVersion2Command, CommandId::LightLimiterVersion2>(node_id)};
if (IsChannelCountValid(parameter.channel_count)) {
const auto state_buffer{
memory_pool->Translate(CpuAddr(&state), sizeof(LightLimiterInfo::State))};
if (state_buffer) {
for (s8 channel = 0; channel < parameter.channel_count; channel++) {
cmd.inputs[channel] = buffer_offset + parameter.inputs[channel];
cmd.outputs[channel] = buffer_offset + parameter.outputs[channel];
}
cmd.parameter = parameter;
cmd.effect_enabled = enabled;
cmd.state = state_buffer;
if (cmd.parameter.statistics_enabled) {
cmd.result_state = memory_pool->Translate(
CpuAddr(&statistics), sizeof(LightLimiterInfo::StatisticsInternal));
} else {
cmd.result_state = 0;
}
cmd.workbuffer = workbuffer;
}
}
GenerateEnd<LightLimiterVersion2Command>(cmd);
}
void CommandBuffer::GenerateMultitapBiquadFilterCommand(const s32 node_id, VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel) {
auto& cmd{GenerateStart<MultiTapBiquadFilterCommand, CommandId::MultiTapBiquadFilter>(node_id)};
cmd.input = buffer_count + channel;
cmd.output = buffer_count + channel;
cmd.biquads = voice_info.biquads;
cmd.states[0] =
memory_pool->Translate(CpuAddr(voice_state.biquad_states[0].data()),
MaxBiquadFilters * sizeof(VoiceState::BiquadFilterState));
cmd.states[1] =
memory_pool->Translate(CpuAddr(voice_state.biquad_states[1].data()),
MaxBiquadFilters * sizeof(VoiceState::BiquadFilterState));
cmd.needs_init[0] = !voice_info.biquad_initialized[0];
cmd.needs_init[1] = !voice_info.biquad_initialized[1];
cmd.filter_tap_count = MaxBiquadFilters;
GenerateEnd<MultiTapBiquadFilterCommand>(cmd);
}
void CommandBuffer::GenerateCaptureCommand(const s32 node_id, EffectInfoBase& effect_info,
const s16 input_index, const s16 output_index,
const s16 buffer_offset, const u32 update_count,
const u32 count_max, const u32 write_offset) {
auto& cmd{GenerateStart<CaptureCommand, CommandId::Capture>(node_id)};
if (effect_info.GetSendBuffer()) {
cmd.input = buffer_offset + input_index;
cmd.output = buffer_offset + output_index;
cmd.send_buffer_info = effect_info.GetSendBufferInfo();
cmd.send_buffer = effect_info.GetSendBuffer();
cmd.count_max = count_max;
cmd.write_offset = write_offset;
cmd.update_count = update_count;
cmd.effect_enabled = effect_info.IsEnabled();
}
GenerateEnd<CaptureCommand>(cmd);
}
void CommandBuffer::GenerateCompressorCommand(s16 buffer_offset, EffectInfoBase& effect_info,
s32 node_id) {
auto& cmd{GenerateStart<CompressorCommand, CommandId::Compressor>(node_id)};
auto& parameter{
*reinterpret_cast<CompressorInfo::ParameterVersion2*>(effect_info.GetParameter())};
auto state{reinterpret_cast<CompressorInfo::State*>(effect_info.GetStateBuffer())};
if (IsChannelCountValid(parameter.channel_count)) {
auto state_buffer{memory_pool->Translate(CpuAddr(state), sizeof(CompressorInfo::State))};
if (state_buffer) {
for (u16 channel = 0; channel < parameter.channel_count; channel++) {
cmd.inputs[channel] = buffer_offset + parameter.inputs[channel];
cmd.outputs[channel] = buffer_offset + parameter.outputs[channel];
}
cmd.parameter = parameter;
cmd.workbuffer = state_buffer;
cmd.enabled = effect_info.IsEnabled();
}
}
GenerateEnd<CompressorCommand>(cmd);
}
} // namespace AudioCore::AudioRenderer

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src/audio_core/renderer/command/command_buffer.h Normal file
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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 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 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.
* @session_id - System session id this command is generated from.
* @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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src/audio_core/renderer/command/command_generator.cpp Normal file
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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/common/audio_renderer_parameter.h"
#include "audio_core/renderer/behavior/behavior_info.h"
#include "audio_core/renderer/command/command_buffer.h"
#include "audio_core/renderer/command/command_generator.h"
#include "audio_core/renderer/command/command_list_header.h"
#include "audio_core/renderer/effect/aux_.h"
#include "audio_core/renderer/effect/biquad_filter.h"
#include "audio_core/renderer/effect/buffer_mixer.h"
#include "audio_core/renderer/effect/capture.h"
#include "audio_core/renderer/effect/effect_context.h"
#include "audio_core/renderer/effect/light_limiter.h"
#include "audio_core/renderer/mix/mix_context.h"
#include "audio_core/renderer/performance/detail_aspect.h"
#include "audio_core/renderer/performance/entry_aspect.h"
#include "audio_core/renderer/sink/device_sink_info.h"
#include "audio_core/renderer/sink/sink_context.h"
#include "audio_core/renderer/splitter/splitter_context.h"
#include "audio_core/renderer/voice/voice_context.h"
#include "common/alignment.h"
namespace AudioCore::AudioRenderer {
CommandGenerator::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_)
: command_buffer{command_buffer_}, command_header{command_list_header_},
render_context{render_context_}, voice_context{voice_context_}, mix_context{mix_context_},
effect_context{effect_context_}, sink_context{sink_context_},
splitter_context{splitter_context_}, performance_manager{performance_manager_} {
command_buffer.GenerateClearMixCommand(InvalidNodeId);
}
void CommandGenerator::GenerateDataSourceCommand(VoiceInfo& voice_info,
const VoiceState& voice_state, const s8 channel) {
if (voice_info.mix_id == UnusedMixId) {
if (voice_info.splitter_id != UnusedSplitterId) {
auto destination{splitter_context.GetDesintationData(voice_info.splitter_id, 0)};
u32 dest_id{0};
while (destination != nullptr) {
if (destination->IsConfigured()) {
auto mix_id{destination->GetMixId()};
if (mix_id < mix_context.GetCount()) {
auto mix_info{mix_context.GetInfo(mix_id)};
command_buffer.GenerateDepopPrepareCommand(
voice_info.node_id, voice_state, render_context.depop_buffer,
mix_info->buffer_count, mix_info->buffer_offset,
voice_info.was_playing);
}
}
dest_id++;
destination = splitter_context.GetDesintationData(voice_info.splitter_id, dest_id);
}
}
} else {
auto mix_info{mix_context.GetInfo(voice_info.mix_id)};
command_buffer.GenerateDepopPrepareCommand(
voice_info.node_id, voice_state, render_context.depop_buffer, mix_info->buffer_count,
mix_info->buffer_offset, voice_info.was_playing);
}
if (voice_info.was_playing) {
return;
}
if (render_context.behavior->IsWaveBufferVer2Supported()) {
switch (voice_info.sample_format) {
case SampleFormat::PcmInt16:
command_buffer.GeneratePcmInt16Version2Command(
voice_info.node_id, voice_info, voice_state, render_context.mix_buffer_count,
channel);
break;
case SampleFormat::PcmFloat:
command_buffer.GeneratePcmFloatVersion2Command(
voice_info.node_id, voice_info, voice_state, render_context.mix_buffer_count,
channel);
break;
case SampleFormat::Adpcm:
command_buffer.GenerateAdpcmVersion2Command(voice_info.node_id, voice_info, voice_state,
render_context.mix_buffer_count, channel);
break;
default:
LOG_ERROR(Service_Audio, "Invalid SampleFormat {}",
static_cast<u32>(voice_info.sample_format));
break;
}
} else {
switch (voice_info.sample_format) {
case SampleFormat::PcmInt16:
command_buffer.GeneratePcmInt16Version1Command(
voice_info.node_id, *command_buffer.memory_pool, voice_info, voice_state,
render_context.mix_buffer_count, channel);
break;
case SampleFormat::PcmFloat:
command_buffer.GeneratePcmFloatVersion1Command(
voice_info.node_id, *command_buffer.memory_pool, voice_info, voice_state,
render_context.mix_buffer_count, channel);
break;
case SampleFormat::Adpcm:
command_buffer.GenerateAdpcmVersion1Command(
voice_info.node_id, *command_buffer.memory_pool, voice_info, voice_state,
render_context.mix_buffer_count, channel);
break;
default:
LOG_ERROR(Service_Audio, "Invalid SampleFormat {}",
static_cast<u32>(voice_info.sample_format));
break;
}
}
}
void CommandGenerator::GenerateVoiceMixCommand(std::span<const f32> mix_volumes,
std::span<const f32> prev_mix_volumes,
const VoiceState& voice_state, s16 output_index,
const s16 buffer_count, const s16 input_index,
const s32 node_id) {
u8 precision{15};
if (render_context.behavior->IsVolumeMixParameterPrecisionQ23Supported()) {
precision = 23;
}
if (buffer_count > 8) {
const auto prev_samples{render_context.memory_pool_info->Translate(
CpuAddr(voice_state.previous_samples.data()), buffer_count * sizeof(s32))};
command_buffer.GenerateMixRampGroupedCommand(node_id, buffer_count, input_index,
output_index, mix_volumes, prev_mix_volumes,
prev_samples, precision);
} else {
for (s16 i = 0; i < buffer_count; i++, output_index++) {
const auto prev_samples{render_context.memory_pool_info->Translate(
CpuAddr(&voice_state.previous_samples[i]), sizeof(s32))};
command_buffer.GenerateMixRampCommand(node_id, buffer_count, input_index, output_index,
mix_volumes[i], prev_mix_volumes[i], prev_samples,
precision);
}
}
}
void CommandGenerator::GenerateBiquadFilterCommandForVoice(VoiceInfo& voice_info,
const VoiceState& voice_state,
const s16 buffer_count, const s8 channel,
const s32 node_id) {
const bool both_biquads_enabled{voice_info.biquads[0].enabled && voice_info.biquads[1].enabled};
const auto use_float_processing{render_context.behavior->UseBiquadFilterFloatProcessing()};
if (both_biquads_enabled && render_context.behavior->UseMultiTapBiquadFilterProcessing() &&
use_float_processing) {
command_buffer.GenerateMultitapBiquadFilterCommand(node_id, voice_info, voice_state,
buffer_count, channel);
} else {
for (u32 i = 0; i < MaxBiquadFilters; i++) {
if (voice_info.biquads[i].enabled) {
command_buffer.GenerateBiquadFilterCommand(node_id, voice_info, voice_state,
buffer_count, channel, i,
use_float_processing);
}
}
}
}
void CommandGenerator::GenerateVoiceCommand(VoiceInfo& voice_info) {
u8 precision{15};
if (render_context.behavior->IsVolumeMixParameterPrecisionQ23Supported()) {
precision = 23;
}
for (s8 channel = 0; channel < voice_info.channel_count; channel++) {
const auto resource_id{voice_info.channel_resource_ids[channel]};
auto& voice_state{voice_context.GetDspSharedState(resource_id)};
auto& channel_resource{voice_context.GetChannelResource(resource_id)};
PerformanceDetailType detail_type{PerformanceDetailType::Invalid};
switch (voice_info.sample_format) {
case SampleFormat::PcmInt16:
detail_type = PerformanceDetailType::Unk1;
break;
case SampleFormat::PcmFloat:
detail_type = PerformanceDetailType::Unk10;
break;
default:
detail_type = PerformanceDetailType::Unk2;
break;
}
DetailAspect data_source_detail(*this, PerformanceEntryType::Voice, voice_info.node_id,
detail_type);
GenerateDataSourceCommand(voice_info, voice_state, channel);
if (data_source_detail.initialized) {
command_buffer.GeneratePerformanceCommand(data_source_detail.node_id,
PerformanceState::Stop,
data_source_detail.performance_entry_address);
}
if (voice_info.was_playing) {
voice_info.prev_volume = 0.0f;
continue;
}
if (!voice_info.HasAnyConnection()) {
continue;
}
DetailAspect biquad_detail_aspect(*this, PerformanceEntryType::Voice, voice_info.node_id,
PerformanceDetailType::Unk4);
GenerateBiquadFilterCommandForVoice(
voice_info, voice_state, render_context.mix_buffer_count, channel, voice_info.node_id);
if (biquad_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
biquad_detail_aspect.node_id, PerformanceState::Stop,
biquad_detail_aspect.performance_entry_address);
}
DetailAspect volume_ramp_detail_aspect(*this, PerformanceEntryType::Voice,
voice_info.node_id, PerformanceDetailType::Unk3);
command_buffer.GenerateVolumeRampCommand(
voice_info.node_id, voice_info, render_context.mix_buffer_count + channel, precision);
if (volume_ramp_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
volume_ramp_detail_aspect.node_id, PerformanceState::Stop,
volume_ramp_detail_aspect.performance_entry_address);
}
voice_info.prev_volume = voice_info.volume;
if (voice_info.mix_id == UnusedMixId) {
if (voice_info.splitter_id != UnusedSplitterId) {
auto i{channel};
auto destination{splitter_context.GetDesintationData(voice_info.splitter_id, i)};
while (destination != nullptr) {
if (destination->IsConfigured()) {
const auto mix_id{destination->GetMixId()};
if (mix_id < mix_context.GetCount() &&
static_cast<s32>(mix_id) != UnusedSplitterId) {
auto mix_info{mix_context.GetInfo(mix_id)};
GenerateVoiceMixCommand(
destination->GetMixVolume(), destination->GetMixVolumePrev(),
voice_state, mix_info->buffer_offset, mix_info->buffer_count,
render_context.mix_buffer_count + channel, voice_info.node_id);
destination->MarkAsNeedToUpdateInternalState();
}
}
i += voice_info.channel_count;
destination = splitter_context.GetDesintationData(voice_info.splitter_id, i);
}
}
} else {
DetailAspect volume_mix_detail_aspect(*this, PerformanceEntryType::Voice,
voice_info.node_id, PerformanceDetailType::Unk3);
auto mix_info{mix_context.GetInfo(voice_info.mix_id)};
GenerateVoiceMixCommand(channel_resource.mix_volumes, channel_resource.prev_mix_volumes,
voice_state, mix_info->buffer_offset, mix_info->buffer_count,
render_context.mix_buffer_count + channel, voice_info.node_id);
if (volume_mix_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
volume_mix_detail_aspect.node_id, PerformanceState::Stop,
volume_mix_detail_aspect.performance_entry_address);
}
channel_resource.prev_mix_volumes = channel_resource.mix_volumes;
}
voice_info.biquad_initialized[0] = voice_info.biquads[0].enabled;
voice_info.biquad_initialized[1] = voice_info.biquads[1].enabled;
}
}
void CommandGenerator::GenerateVoiceCommands() {
const auto voice_count{voice_context.GetCount()};
for (u32 i = 0; i < voice_count; i++) {
auto sorted_info{voice_context.GetSortedInfo(i)};
if (sorted_info->ShouldSkip() || !sorted_info->UpdateForCommandGeneration(voice_context)) {
continue;
}
EntryAspect voice_entry_aspect(*this, PerformanceEntryType::Voice, sorted_info->node_id);
GenerateVoiceCommand(*sorted_info);
if (voice_entry_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(voice_entry_aspect.node_id,
PerformanceState::Stop,
voice_entry_aspect.performance_entry_address);
}
}
splitter_context.UpdateInternalState();
}
void CommandGenerator::GenerateBufferMixerCommand(const s16 buffer_offset,
EffectInfoBase& effect_info, const s32 node_id) {
u8 precision{15};
if (render_context.behavior->IsVolumeMixParameterPrecisionQ23Supported()) {
precision = 23;
}
if (effect_info.IsEnabled()) {
const auto& parameter{
*reinterpret_cast<BufferMixerInfo::ParameterVersion1*>(effect_info.GetParameter())};
for (u32 i = 0; i < parameter.mix_count; i++) {
if (parameter.volumes[i] != 0.0f) {
command_buffer.GenerateMixCommand(node_id, buffer_offset + parameter.inputs[i],
buffer_offset + parameter.outputs[i],
buffer_offset, parameter.volumes[i], precision);
}
}
}
}
void CommandGenerator::GenerateDelayCommand(const s16 buffer_offset, EffectInfoBase& effect_info,
const s32 node_id) {
command_buffer.GenerateDelayCommand(node_id, effect_info, buffer_offset);
}
void CommandGenerator::GenerateReverbCommand(const s16 buffer_offset, EffectInfoBase& effect_info,
const s32 node_id,
const bool long_size_pre_delay_supported) {
command_buffer.GenerateReverbCommand(node_id, effect_info, buffer_offset,
long_size_pre_delay_supported);
}
void CommandGenerator::GenerateI3dl2ReverbEffectCommand(const s16 buffer_offset,
EffectInfoBase& effect_info,
const s32 node_id) {
command_buffer.GenerateI3dl2ReverbCommand(node_id, effect_info, buffer_offset);
}
void CommandGenerator::GenerateAuxCommand(const s16 buffer_offset, EffectInfoBase& effect_info,
const s32 node_id) {
if (effect_info.IsEnabled()) {
effect_info.GetWorkbuffer(0);
effect_info.GetWorkbuffer(1);
}
if (effect_info.GetSendBuffer() != 0 && effect_info.GetReturnBuffer() != 0) {
const auto& parameter{
*reinterpret_cast<AuxInfo::ParameterVersion1*>(effect_info.GetParameter())};
auto channel_index{parameter.mix_buffer_count - 1};
u32 write_offset{0};
for (u32 i = 0; i < parameter.mix_buffer_count; i++, channel_index--) {
auto new_update_count{command_header.sample_count + write_offset};
const auto update_count{channel_index > 0 ? 0 : new_update_count};
command_buffer.GenerateAuxCommand(node_id, effect_info, parameter.inputs[i],
parameter.outputs[i], buffer_offset, update_count,
parameter.count_max, write_offset);
write_offset = new_update_count;
}
}
}
void CommandGenerator::GenerateBiquadFilterEffectCommand(const s16 buffer_offset,
EffectInfoBase& effect_info,
const s32 node_id) {
const auto& parameter{
*reinterpret_cast<BiquadFilterInfo::ParameterVersion1*>(effect_info.GetParameter())};
if (effect_info.IsEnabled()) {
bool needs_init{false};
switch (parameter.state) {
case EffectInfoBase::ParameterState::Initialized:
needs_init = true;
break;
case EffectInfoBase::ParameterState::Updating:
case EffectInfoBase::ParameterState::Updated:
if (render_context.behavior->IsBiquadFilterEffectStateClearBugFixed()) {
needs_init = false;
} else {
needs_init = parameter.state == EffectInfoBase::ParameterState::Updating;
}
break;
default:
LOG_ERROR(Service_Audio, "Invalid biquad parameter state {}",
static_cast<u32>(parameter.state));
break;
}
for (s8 channel = 0; channel < parameter.channel_count; channel++) {
command_buffer.GenerateBiquadFilterCommand(
node_id, effect_info, buffer_offset, channel, needs_init,
render_context.behavior->UseBiquadFilterFloatProcessing());
}
} else {
for (s8 channel = 0; channel < parameter.channel_count; channel++) {
command_buffer.GenerateCopyMixBufferCommand(node_id, effect_info, buffer_offset,
channel);
}
}
}
void CommandGenerator::GenerateLightLimiterEffectCommand(const s16 buffer_offset,
EffectInfoBase& effect_info,
const s32 node_id,
const u32 effect_index) {
const auto& state{*reinterpret_cast<LightLimiterInfo::State*>(effect_info.GetStateBuffer())};
if (render_context.behavior->IsEffectInfoVersion2Supported()) {
const auto& parameter{
*reinterpret_cast<LightLimiterInfo::ParameterVersion2*>(effect_info.GetParameter())};
const auto& result_state{*reinterpret_cast<LightLimiterInfo::StatisticsInternal*>(
&effect_context.GetDspSharedResultState(effect_index))};
command_buffer.GenerateLightLimiterCommand(node_id, buffer_offset, parameter, result_state,
state, effect_info.IsEnabled(),
effect_info.GetWorkbuffer(-1));
} else {
const auto& parameter{
*reinterpret_cast<LightLimiterInfo::ParameterVersion1*>(effect_info.GetParameter())};
command_buffer.GenerateLightLimiterCommand(node_id, buffer_offset, parameter, state,
effect_info.IsEnabled(),
effect_info.GetWorkbuffer(-1));
}
}
void CommandGenerator::GenerateCaptureCommand(const s16 buffer_offset, EffectInfoBase& effect_info,
const s32 node_id) {
if (effect_info.IsEnabled()) {
effect_info.GetWorkbuffer(0);
}
if (effect_info.GetSendBuffer()) {
const auto& parameter{
*reinterpret_cast<AuxInfo::ParameterVersion1*>(effect_info.GetParameter())};
auto channel_index{parameter.mix_buffer_count - 1};
u32 write_offset{0};
for (u32 i = 0; i < parameter.mix_buffer_count; i++, channel_index--) {
auto new_update_count{command_header.sample_count + write_offset};
const auto update_count{channel_index > 0 ? 0 : new_update_count};
command_buffer.GenerateCaptureCommand(node_id, effect_info, parameter.inputs[i],
parameter.outputs[i], buffer_offset, update_count,
parameter.count_max, write_offset);
write_offset = new_update_count;
}
}
}
void CommandGenerator::GenerateCompressorCommand(const s16 buffer_offset,
EffectInfoBase& effect_info, const s32 node_id) {
command_buffer.GenerateCompressorCommand(buffer_offset, effect_info, node_id);
}
void CommandGenerator::GenerateEffectCommand(MixInfo& mix_info) {
const auto effect_count{effect_context.GetCount()};
for (u32 i = 0; i < effect_count; i++) {
const auto effect_index{mix_info.effect_order_buffer[i]};
if (effect_index == -1) {
break;
}
auto& effect_info = effect_context.GetInfo(effect_index);
if (effect_info.ShouldSkip()) {
continue;
}
const auto entry_type{mix_info.mix_id == FinalMixId ? PerformanceEntryType::FinalMix
: PerformanceEntryType::SubMix};
switch (effect_info.GetType()) {
case EffectInfoBase::Type::Mix: {
DetailAspect mix_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk5);
GenerateBufferMixerCommand(mix_info.buffer_offset, effect_info, mix_info.node_id);
if (mix_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
mix_detail_aspect.node_id, PerformanceState::Stop,
mix_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::Aux: {
DetailAspect aux_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk7);
GenerateAuxCommand(mix_info.buffer_offset, effect_info, mix_info.node_id);
if (aux_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
aux_detail_aspect.node_id, PerformanceState::Stop,
aux_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::Delay: {
DetailAspect delay_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk6);
GenerateDelayCommand(mix_info.buffer_offset, effect_info, mix_info.node_id);
if (delay_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
delay_detail_aspect.node_id, PerformanceState::Stop,
delay_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::Reverb: {
DetailAspect reverb_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk8);
GenerateReverbCommand(mix_info.buffer_offset, effect_info, mix_info.node_id,
render_context.behavior->IsLongSizePreDelaySupported());
if (reverb_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
reverb_detail_aspect.node_id, PerformanceState::Stop,
reverb_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::I3dl2Reverb: {
DetailAspect i3dl2_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk9);
GenerateI3dl2ReverbEffectCommand(mix_info.buffer_offset, effect_info, mix_info.node_id);
if (i3dl2_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
i3dl2_detail_aspect.node_id, PerformanceState::Stop,
i3dl2_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::BiquadFilter: {
DetailAspect biquad_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk4);
GenerateBiquadFilterEffectCommand(mix_info.buffer_offset, effect_info,
mix_info.node_id);
if (biquad_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
biquad_detail_aspect.node_id, PerformanceState::Stop,
biquad_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::LightLimiter: {
DetailAspect light_limiter_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk11);
GenerateLightLimiterEffectCommand(mix_info.buffer_offset, effect_info, mix_info.node_id,
effect_index);
if (light_limiter_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
light_limiter_detail_aspect.node_id, PerformanceState::Stop,
light_limiter_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::Capture: {
DetailAspect capture_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk12);
GenerateCaptureCommand(mix_info.buffer_offset, effect_info, mix_info.node_id);
if (capture_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
capture_detail_aspect.node_id, PerformanceState::Stop,
capture_detail_aspect.performance_entry_address);
}
} break;
case EffectInfoBase::Type::Compressor: {
DetailAspect capture_detail_aspect(*this, entry_type, mix_info.node_id,
PerformanceDetailType::Unk13);
GenerateCompressorCommand(mix_info.buffer_offset, effect_info, mix_info.node_id);
if (capture_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
capture_detail_aspect.node_id, PerformanceState::Stop,
capture_detail_aspect.performance_entry_address);
}
} break;
default:
LOG_ERROR(Service_Audio, "Invalid effect type {}",
static_cast<u32>(effect_info.GetType()));
break;
}
effect_info.UpdateForCommandGeneration();
}
}
void CommandGenerator::GenerateMixCommands(MixInfo& mix_info) {
u8 precision{15};
if (render_context.behavior->IsVolumeMixParameterPrecisionQ23Supported()) {
precision = 23;
}
if (!mix_info.HasAnyConnection()) {
return;
}
if (mix_info.dst_mix_id == UnusedMixId) {
if (mix_info.dst_splitter_id != UnusedSplitterId) {
s16 dest_id{0};
auto destination{
splitter_context.GetDesintationData(mix_info.dst_splitter_id, dest_id)};
while (destination != nullptr) {
if (destination->IsConfigured()) {
auto splitter_mix_id{destination->GetMixId()};
if (splitter_mix_id < mix_context.GetCount()) {
auto splitter_mix_info{mix_context.GetInfo(splitter_mix_id)};
const s16 input_index{static_cast<s16>(mix_info.buffer_offset +
(dest_id % mix_info.buffer_count))};
for (s16 i = 0; i < splitter_mix_info->buffer_count; i++) {
auto volume{mix_info.volume * destination->GetMixVolume(i)};
if (volume != 0.0f) {
command_buffer.GenerateMixCommand(
mix_info.node_id, input_index,
splitter_mix_info->buffer_offset + i, mix_info.buffer_offset,
volume, precision);
}
}
}
}
dest_id++;
destination =
splitter_context.GetDesintationData(mix_info.dst_splitter_id, dest_id);
}
}
} else {
auto dest_mix_info{mix_context.GetInfo(mix_info.dst_mix_id)};
for (s16 i = 0; i < mix_info.buffer_count; i++) {
for (s16 j = 0; j < dest_mix_info->buffer_count; j++) {
auto volume{mix_info.volume * mix_info.mix_volumes[i][j]};
if (volume != 0.0f) {
command_buffer.GenerateMixCommand(mix_info.node_id, mix_info.buffer_offset + i,
dest_mix_info->buffer_offset + j,
mix_info.buffer_offset, volume, precision);
}
}
}
}
}
void CommandGenerator::GenerateSubMixCommand(MixInfo& mix_info) {
command_buffer.GenerateDepopForMixBuffersCommand(mix_info.node_id, mix_info,
render_context.depop_buffer);
GenerateEffectCommand(mix_info);
DetailAspect mix_detail_aspect(*this, PerformanceEntryType::SubMix, mix_info.node_id,
PerformanceDetailType::Unk5);
GenerateMixCommands(mix_info);
if (mix_detail_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(mix_detail_aspect.node_id, PerformanceState::Stop,
mix_detail_aspect.performance_entry_address);
}
}
void CommandGenerator::GenerateSubMixCommands() {
const auto submix_count{mix_context.GetCount()};
for (s32 i = 0; i < submix_count; i++) {
auto sorted_info{mix_context.GetSortedInfo(i)};
if (!sorted_info->in_use || sorted_info->mix_id == FinalMixId) {
continue;
}
EntryAspect submix_entry_aspect(*this, PerformanceEntryType::SubMix, sorted_info->node_id);
GenerateSubMixCommand(*sorted_info);
if (submix_entry_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(
submix_entry_aspect.node_id, PerformanceState::Stop,
submix_entry_aspect.performance_entry_address);
}
}
}
void CommandGenerator::GenerateFinalMixCommand() {
auto& final_mix_info{*mix_context.GetFinalMixInfo()};
command_buffer.GenerateDepopForMixBuffersCommand(final_mix_info.node_id, final_mix_info,
render_context.depop_buffer);
GenerateEffectCommand(final_mix_info);
u8 precision{15};
if (render_context.behavior->IsVolumeMixParameterPrecisionQ23Supported()) {
precision = 23;
}
for (s16 i = 0; i < final_mix_info.buffer_count; i++) {
DetailAspect volume_aspect(*this, PerformanceEntryType::FinalMix, final_mix_info.node_id,
PerformanceDetailType::Unk3);
command_buffer.GenerateVolumeCommand(final_mix_info.node_id, final_mix_info.buffer_offset,
i, final_mix_info.volume, precision);
if (volume_aspect.initialized) {
command_buffer.GeneratePerformanceCommand(volume_aspect.node_id, PerformanceState::Stop,
volume_aspect.performance_entry_address);
}
}
}
void CommandGenerator::GenerateFinalMixCommands() {
auto final_mix_info{mix_context.GetFinalMixInfo()};
EntryAspect final_mix_entry(*this, PerformanceEntryType::FinalMix, final_mix_info->node_id);
GenerateFinalMixCommand();
if (final_mix_entry.initialized) {
command_buffer.GeneratePerformanceCommand(final_mix_entry.node_id, PerformanceState::Stop,
final_mix_entry.performance_entry_address);
}
}
void CommandGenerator::GenerateSinkCommands() {
const auto sink_count{sink_context.GetCount()};
for (u32 i = 0; i < sink_count; i++) {
auto sink_info{sink_context.GetInfo(i)};
if (sink_info->IsUsed() && sink_info->GetType() == SinkInfoBase::Type::DeviceSink) {
auto state{reinterpret_cast<DeviceSinkInfo::DeviceState*>(sink_info->GetState())};
if (command_header.sample_rate != TargetSampleRate &&
state->upsampler_info == nullptr) {
auto device_state{sink_info->GetDeviceState()};
device_state->upsampler_info = render_context.upsampler_manager->Allocate();
}
EntryAspect device_sink_entry(*this, PerformanceEntryType::Sink,
sink_info->GetNodeId());
auto final_mix{mix_context.GetFinalMixInfo()};
GenerateSinkCommand(final_mix->buffer_offset, *sink_info);
if (device_sink_entry.initialized) {
command_buffer.GeneratePerformanceCommand(
device_sink_entry.node_id, PerformanceState::Stop,
device_sink_entry.performance_entry_address);
}
}
}
for (u32 i = 0; i < sink_count; i++) {
auto sink_info{sink_context.GetInfo(i)};
if (sink_info->IsUsed() && sink_info->GetType() == SinkInfoBase::Type::CircularBufferSink) {
EntryAspect circular_buffer_entry(*this, PerformanceEntryType::Sink,
sink_info->GetNodeId());
auto final_mix{mix_context.GetFinalMixInfo()};
GenerateSinkCommand(final_mix->buffer_offset, *sink_info);
if (circular_buffer_entry.initialized) {
command_buffer.GeneratePerformanceCommand(
circular_buffer_entry.node_id, PerformanceState::Stop,
circular_buffer_entry.performance_entry_address);
}
}
}
}
void CommandGenerator::GenerateSinkCommand(const s16 buffer_offset, SinkInfoBase& sink_info) {
if (sink_info.ShouldSkip()) {
return;
}
switch (sink_info.GetType()) {
case SinkInfoBase::Type::DeviceSink:
GenerateDeviceSinkCommand(buffer_offset, sink_info);
break;
case SinkInfoBase::Type::CircularBufferSink:
command_buffer.GenerateCircularBufferSinkCommand(sink_info.GetNodeId(), sink_info,
buffer_offset);
break;
default:
LOG_ERROR(Service_Audio, "Invalid sink type {}", static_cast<u32>(sink_info.GetType()));
break;
}
sink_info.UpdateForCommandGeneration();
}
void CommandGenerator::GenerateDeviceSinkCommand(const s16 buffer_offset, SinkInfoBase& sink_info) {
auto& parameter{
*reinterpret_cast<DeviceSinkInfo::DeviceInParameter*>(sink_info.GetParameter())};
auto state{*reinterpret_cast<DeviceSinkInfo::DeviceState*>(sink_info.GetState())};
if (render_context.channels == 2 && parameter.downmix_enabled) {
command_buffer.GenerateDownMix6chTo2chCommand(InvalidNodeId, parameter.inputs,
buffer_offset, parameter.downmix_coeff);
}
if (state.upsampler_info != nullptr) {
command_buffer.GenerateUpsampleCommand(
InvalidNodeId, buffer_offset, *state.upsampler_info, parameter.input_count,
parameter.inputs, command_header.buffer_count, command_header.sample_count,
command_header.sample_rate);
}
command_buffer.GenerateDeviceSinkCommand(InvalidNodeId, buffer_offset, sink_info,
render_context.session_id,
command_header.samples_buffer);
}
void CommandGenerator::GeneratePerformanceCommand(
s32 node_id, PerformanceState state, const PerformanceEntryAddresses& entry_addresses) {
command_buffer.GeneratePerformanceCommand(node_id, state, entry_addresses);
}
} // 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_base - 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_base - 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_base - 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_base - 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_base - 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_base - Compressor effect info.
* @param node_id - Node id of the mix this command is generated for.
*/
void GenerateCompressorCommand(const s16 buffer_offset, EffectInfoBase& effect_info,
const 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 buffer_offset - Base mix buffer offset to use.
* @param sink_info - Sink info to generate the commands from.
*/
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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// 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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// 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

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// 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"

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// 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

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// 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

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// 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

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// 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

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// 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

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// 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

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src/audio_core/renderer/command/data_source/pcm_int16.cpp Normal file
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// 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

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// 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

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src/audio_core/renderer/command/effect/aux_.cpp Normal file
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// 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

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// 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

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/effect/biquad_filter.h"
#include "audio_core/renderer/voice/voice_state.h"
namespace AudioCore::AudioRenderer {
/**
* Biquad filter float implementation.
*
* @param output - Output container for filtered samples.
* @param input - Input container for samples to be filtered.
* @param b - Feedforward coefficients.
* @param a - Feedback coefficients.
* @param state - State to track previous samples between calls.
* @param sample_count - Number of samples to process.
*/
void ApplyBiquadFilterFloat(std::span<s32> output, std::span<const s32> input,
std::array<s16, 3>& b_, std::array<s16, 2>& a_,
VoiceState::BiquadFilterState& state, const u32 sample_count) {
constexpr s64 min{std::numeric_limits<s32>::min()};
constexpr s64 max{std::numeric_limits<s32>::max()};
std::array<f64, 3> b{Common::FixedPoint<50, 14>::from_base(b_[0]).to_double(),
Common::FixedPoint<50, 14>::from_base(b_[1]).to_double(),
Common::FixedPoint<50, 14>::from_base(b_[2]).to_double()};
std::array<f64, 2> a{Common::FixedPoint<50, 14>::from_base(a_[0]).to_double(),
Common::FixedPoint<50, 14>::from_base(a_[1]).to_double()};
std::array<f64, 4> s{state.s0.to_double(), state.s1.to_double(), state.s2.to_double(),
state.s3.to_double()};
for (u32 i = 0; i < sample_count; i++) {
f64 in_sample{static_cast<f64>(input[i])};
auto sample{in_sample * b[0] + s[0] * b[1] + s[1] * b[2] + s[2] * a[0] + s[3] * a[1]};
output[i] = static_cast<s32>(std::clamp(static_cast<s64>(sample), min, max));
s[1] = s[0];
s[0] = in_sample;
s[3] = s[2];
s[2] = sample;
}
state.s0 = s[0];
state.s1 = s[1];
state.s2 = s[2];
state.s3 = s[3];
}
/**
* Biquad filter s32 implementation.
*
* @param output - Output container for filtered samples.
* @param input - Input container for samples to be filtered.
* @param b - Feedforward coefficients.
* @param a - Feedback coefficients.
* @param state - State to track previous samples between calls.
* @param sample_count - Number of samples to process.
*/
static void ApplyBiquadFilterInt(std::span<s32> output, std::span<const s32> input,
std::array<s16, 3>& b_, std::array<s16, 2>& a_,
VoiceState::BiquadFilterState& state, const u32 sample_count) {
constexpr s64 min{std::numeric_limits<s32>::min()};
constexpr s64 max{std::numeric_limits<s32>::max()};
std::array<Common::FixedPoint<50, 14>, 3> b{
Common::FixedPoint<50, 14>::from_base(b_[0]),
Common::FixedPoint<50, 14>::from_base(b_[1]),
Common::FixedPoint<50, 14>::from_base(b_[2]),
};
std::array<Common::FixedPoint<50, 14>, 3> a{
Common::FixedPoint<50, 14>::from_base(a_[0]),
Common::FixedPoint<50, 14>::from_base(a_[1]),
};
for (u32 i = 0; i < sample_count; i++) {
s64 in_sample{input[i]};
auto sample{in_sample * b[0] + state.s0};
const auto out_sample{std::clamp(sample.to_long(), min, max)};
output[i] = static_cast<s32>(out_sample);
state.s0 = state.s1 + b[1] * in_sample + a[0] * out_sample;
state.s1 = 0 + b[2] * in_sample + a[1] * out_sample;
}
}
void BiquadFilterCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format(
"BiquadFilterCommand\n\tinput {:02X} output {:02X} needs_init {} use_float_processing {}\n",
input, output, needs_init, use_float_processing);
}
void BiquadFilterCommand::Process(const ADSP::CommandListProcessor& processor) {
auto state_{reinterpret_cast<VoiceState::BiquadFilterState*>(state)};
if (needs_init) {
std::memset(state_, 0, sizeof(VoiceState::BiquadFilterState));
}
auto input_buffer{
processor.mix_buffers.subspan(input * processor.sample_count, processor.sample_count)};
auto output_buffer{
processor.mix_buffers.subspan(output * processor.sample_count, processor.sample_count)};
if (use_float_processing) {
ApplyBiquadFilterFloat(output_buffer, input_buffer, biquad.b, biquad.a, *state_,
processor.sample_count);
} else {
ApplyBiquadFilterInt(output_buffer, input_buffer, biquad.b, biquad.a, *state_,
processor.sample_count);
}
}
bool BiquadFilterCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

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// 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

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// 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

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// 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

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// SPDX-FileCopyrightText: Copyright 2022 yuzu Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include <cmath>
#include <span>
#include <vector>
#include "audio_core/renderer/adsp/command_list_processor.h"
#include "audio_core/renderer/command/effect/compressor.h"
#include "audio_core/renderer/effect/compressor.h"
namespace AudioCore::AudioRenderer {
static void SetCompressorEffectParameter(CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state) {
const auto ratio{1.0f / params.compressor_ratio};
auto makeup_gain{0.0f};
if (params.makeup_gain_enabled) {
makeup_gain = (params.threshold * 0.5f) * (ratio - 1.0f) - 3.0f;
}
state.makeup_gain = makeup_gain;
state.unk_18 = params.unk_28;
const auto a{(params.out_gain + makeup_gain) / 20.0f * 3.3219f};
const auto b{(a - std::trunc(a)) * 0.69315f};
const auto c{std::pow(2.0f, b)};
state.unk_0C = (1.0f - ratio) / 6.0f;
state.unk_14 = params.threshold + 1.5f;
state.unk_10 = params.threshold - 1.5f;
state.unk_20 = c;
}
static void InitializeCompressorEffect(CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state) {
std::memset(&state, 0, sizeof(CompressorInfo::State));
state.unk_00 = 0;
state.unk_04 = 1.0f;
state.unk_08 = 1.0f;
SetCompressorEffectParameter(params, state);
}
static void ApplyCompressorEffect(CompressorInfo::ParameterVersion2& params,
CompressorInfo::State& state, bool enabled,
std::vector<std::span<const s32>> input_buffers,
std::vector<std::span<s32>> output_buffers, u32 sample_count) {
if (enabled) {
auto state_00{state.unk_00};
auto state_04{state.unk_04};
auto state_08{state.unk_08};
auto state_18{state.unk_18};
for (u32 i = 0; i < sample_count; i++) {
auto a{0.0f};
for (s16 channel = 0; channel < params.channel_count; channel++) {
const auto input_sample{Common::FixedPoint<49, 15>(input_buffers[channel][i])};
a += (input_sample * input_sample).to_float();
}
state_00 += params.unk_24 * ((a / params.channel_count) - state.unk_00);
auto b{-100.0f};
auto c{0.0f};
if (state_00 >= 1.0e-10) {
b = std::log10(state_00) * 10.0f;
c = 1.0f;
}
if (b >= state.unk_10) {
const auto d{b >= state.unk_14
? ((1.0f / params.compressor_ratio) - 1.0f) *
(b - params.threshold)
: (b - state.unk_10) * (b - state.unk_10) * -state.unk_0C};
const auto e{d / 20.0f * 3.3219f};
const auto f{(e - std::trunc(e)) * 0.69315f};
c = std::pow(2.0f, f);
}
state_18 = params.unk_28;
auto tmp{c};
if ((state_04 - c) <= 0.08f) {
state_18 = params.unk_2C;
if (((state_04 - c) >= -0.08f) && (std::abs(state_08 - c) >= 0.001f)) {
tmp = state_04;
}
}
state_04 = tmp;
state_08 += (c - state_08) * state_18;
for (s16 channel = 0; channel < params.channel_count; channel++) {
output_buffers[channel][i] = static_cast<s32>(
static_cast<f32>(input_buffers[channel][i]) * state_08 * state.unk_20);
}
}
state.unk_00 = state_00;
state.unk_04 = state_04;
state.unk_08 = state_08;
state.unk_18 = state_18;
} else {
for (s16 channel = 0; channel < params.channel_count; channel++) {
if (params.inputs[channel] != params.outputs[channel]) {
std::memcpy((char*)output_buffers[channel].data(),
(char*)input_buffers[channel].data(),
output_buffers[channel].size_bytes());
}
}
}
}
void CompressorCommand::Dump([[maybe_unused]] const ADSP::CommandListProcessor& processor,
std::string& string) {
string += fmt::format("CompressorCommand\n\tenabled {} \n\tinputs: ", effect_enabled);
for (s16 i = 0; i < parameter.channel_count; i++) {
string += fmt::format("{:02X}, ", inputs[i]);
}
string += "\n\toutputs: ";
for (s16 i = 0; i < parameter.channel_count; i++) {
string += fmt::format("{:02X}, ", outputs[i]);
}
string += "\n";
}
void CompressorCommand::Process(const ADSP::CommandListProcessor& processor) {
std::vector<std::span<const s32>> input_buffers(parameter.channel_count);
std::vector<std::span<s32>> output_buffers(parameter.channel_count);
for (s16 i = 0; i < parameter.channel_count; i++) {
input_buffers[i] = processor.mix_buffers.subspan(inputs[i] * processor.sample_count,
processor.sample_count);
output_buffers[i] = processor.mix_buffers.subspan(outputs[i] * processor.sample_count,
processor.sample_count);
}
auto state_{reinterpret_cast<CompressorInfo::State*>(state)};
if (effect_enabled) {
if (parameter.state == CompressorInfo::ParameterState::Updating) {
SetCompressorEffectParameter(parameter, *state_);
} else if (parameter.state == CompressorInfo::ParameterState::Initialized) {
InitializeCompressorEffect(parameter, *state_);
}
}
ApplyCompressorEffect(parameter, *state_, effect_enabled, input_buffers, output_buffers,
processor.sample_count);
}
bool CompressorCommand::Verify(const ADSP::CommandListProcessor& processor) {
return true;
}
} // namespace AudioCore::AudioRenderer

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// 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

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// 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

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