5daf3abe65
This reduces the load of requiring to include std::chrono in all files which include log.h
1371 lines
58 KiB
C++
1371 lines
58 KiB
C++
// Copyright 2020 yuzu Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <algorithm>
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#include <cmath>
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#include <numbers>
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#include "audio_core/algorithm/interpolate.h"
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#include "audio_core/command_generator.h"
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#include "audio_core/effect_context.h"
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#include "audio_core/mix_context.h"
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#include "audio_core/voice_context.h"
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#include "common/common_types.h"
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#include "core/memory.h"
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namespace AudioCore {
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namespace {
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constexpr std::size_t MIX_BUFFER_SIZE = 0x3f00;
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constexpr std::size_t SCALED_MIX_BUFFER_SIZE = MIX_BUFFER_SIZE << 15ULL;
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using DelayLineTimes = std::array<f32, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT>;
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constexpr DelayLineTimes FDN_MIN_DELAY_LINE_TIMES{5.0f, 6.0f, 13.0f, 14.0f};
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constexpr DelayLineTimes FDN_MAX_DELAY_LINE_TIMES{45.704f, 82.782f, 149.94f, 271.58f};
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constexpr DelayLineTimes DECAY0_MAX_DELAY_LINE_TIMES{17.0f, 13.0f, 9.0f, 7.0f};
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constexpr DelayLineTimes DECAY1_MAX_DELAY_LINE_TIMES{19.0f, 11.0f, 10.0f, 6.0f};
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constexpr std::array<f32, AudioCommon::I3DL2REVERB_TAPS> EARLY_TAP_TIMES{
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0.017136f, 0.059154f, 0.161733f, 0.390186f, 0.425262f, 0.455411f, 0.689737f,
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0.745910f, 0.833844f, 0.859502f, 0.000000f, 0.075024f, 0.168788f, 0.299901f,
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0.337443f, 0.371903f, 0.599011f, 0.716741f, 0.817859f, 0.851664f};
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constexpr std::array<f32, AudioCommon::I3DL2REVERB_TAPS> EARLY_GAIN{
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0.67096f, 0.61027f, 1.0f, 0.35680f, 0.68361f, 0.65978f, 0.51939f,
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0.24712f, 0.45945f, 0.45021f, 0.64196f, 0.54879f, 0.92925f, 0.38270f,
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0.72867f, 0.69794f, 0.5464f, 0.24563f, 0.45214f, 0.44042f};
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template <std::size_t N>
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void ApplyMix(std::span<s32> output, std::span<const s32> input, s32 gain, s32 sample_count) {
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for (std::size_t i = 0; i < static_cast<std::size_t>(sample_count); i += N) {
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for (std::size_t j = 0; j < N; j++) {
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output[i + j] +=
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static_cast<s32>((static_cast<s64>(input[i + j]) * gain + 0x4000) >> 15);
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}
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}
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}
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s32 ApplyMixRamp(std::span<s32> output, std::span<const s32> input, float gain, float delta,
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s32 sample_count) {
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// XC2 passes in NaN mix volumes, causing further issues as we handle everything as s32 rather
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// than float, so the NaN propogation is lost. As the samples get further modified for
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// volume etc, they can get out of NaN range, so a later heuristic for catching this is
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// more difficult. Handle it here by setting these samples to silence.
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if (std::isnan(gain)) {
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gain = 0.0f;
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delta = 0.0f;
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}
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s32 x = 0;
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for (s32 i = 0; i < sample_count; i++) {
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x = static_cast<s32>(static_cast<float>(input[i]) * gain);
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output[i] += x;
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gain += delta;
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}
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return x;
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}
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void ApplyGain(std::span<s32> output, std::span<const s32> input, s32 gain, s32 delta,
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s32 sample_count) {
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for (s32 i = 0; i < sample_count; i++) {
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output[i] = static_cast<s32>((static_cast<s64>(input[i]) * gain + 0x4000) >> 15);
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gain += delta;
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}
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}
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void ApplyGainWithoutDelta(std::span<s32> output, std::span<const s32> input, s32 gain,
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s32 sample_count) {
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for (s32 i = 0; i < sample_count; i++) {
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output[i] = static_cast<s32>((static_cast<s64>(input[i]) * gain + 0x4000) >> 15);
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}
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}
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s32 ApplyMixDepop(std::span<s32> output, s32 first_sample, s32 delta, s32 sample_count) {
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const bool positive = first_sample > 0;
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auto final_sample = std::abs(first_sample);
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for (s32 i = 0; i < sample_count; i++) {
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final_sample = static_cast<s32>((static_cast<s64>(final_sample) * delta) >> 15);
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if (positive) {
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output[i] += final_sample;
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} else {
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output[i] -= final_sample;
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}
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}
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if (positive) {
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return final_sample;
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} else {
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return -final_sample;
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}
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}
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float Pow10(float x) {
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if (x >= 0.0f) {
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return 1.0f;
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} else if (x <= -5.3f) {
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return 0.0f;
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}
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return std::pow(10.0f, x);
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}
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float SinD(float degrees) {
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return std::sin(degrees * std::numbers::pi_v<float> / 180.0f);
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}
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float CosD(float degrees) {
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return std::cos(degrees * std::numbers::pi_v<float> / 180.0f);
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}
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float ToFloat(s32 sample) {
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return static_cast<float>(sample) / 65536.f;
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}
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s32 ToS32(float sample) {
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constexpr auto min = -8388608.0f;
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constexpr auto max = 8388607.f;
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float rescaled_sample = sample * 65536.0f;
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if (rescaled_sample < min) {
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rescaled_sample = min;
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}
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if (rescaled_sample > max) {
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rescaled_sample = max;
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}
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return static_cast<s32>(rescaled_sample);
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}
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constexpr std::array<std::size_t, 20> REVERB_TAP_INDEX_1CH{0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
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constexpr std::array<std::size_t, 20> REVERB_TAP_INDEX_2CH{0, 0, 0, 1, 1, 1, 1, 0, 0, 0,
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1, 1, 1, 0, 0, 0, 0, 1, 1, 1};
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constexpr std::array<std::size_t, 20> REVERB_TAP_INDEX_4CH{0, 0, 0, 1, 1, 1, 1, 2, 2, 2,
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1, 1, 1, 0, 0, 0, 0, 3, 3, 3};
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constexpr std::array<std::size_t, 20> REVERB_TAP_INDEX_6CH{4, 0, 0, 1, 1, 1, 1, 2, 2, 2,
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1, 1, 1, 0, 0, 0, 0, 3, 3, 3};
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template <std::size_t CHANNEL_COUNT>
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void ApplyReverbGeneric(
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I3dl2ReverbState& state,
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const std::array<std::span<const s32>, AudioCommon::MAX_CHANNEL_COUNT>& input,
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const std::array<std::span<s32>, AudioCommon::MAX_CHANNEL_COUNT>& output, s32 sample_count) {
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auto GetTapLookup = []() {
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if constexpr (CHANNEL_COUNT == 1) {
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return REVERB_TAP_INDEX_1CH;
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} else if constexpr (CHANNEL_COUNT == 2) {
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return REVERB_TAP_INDEX_2CH;
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} else if constexpr (CHANNEL_COUNT == 4) {
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return REVERB_TAP_INDEX_4CH;
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} else if constexpr (CHANNEL_COUNT == 6) {
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return REVERB_TAP_INDEX_6CH;
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}
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};
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const auto& tap_index_lut = GetTapLookup();
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for (s32 sample = 0; sample < sample_count; sample++) {
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std::array<f32, CHANNEL_COUNT> out_samples{};
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std::array<f32, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT> fsamp{};
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std::array<f32, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT> mixed{};
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std::array<f32, AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT> osamp{};
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// Mix everything into a single sample
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s32 temp_mixed_sample = 0;
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for (std::size_t i = 0; i < CHANNEL_COUNT; i++) {
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temp_mixed_sample += input[i][sample];
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}
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const auto current_sample = ToFloat(temp_mixed_sample);
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const auto early_tap = state.early_delay_line.TapOut(state.early_to_late_taps);
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for (std::size_t i = 0; i < AudioCommon::I3DL2REVERB_TAPS; i++) {
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const auto tapped_samp =
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state.early_delay_line.TapOut(state.early_tap_steps[i]) * EARLY_GAIN[i];
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out_samples[tap_index_lut[i]] += tapped_samp;
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if constexpr (CHANNEL_COUNT == 6) {
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// handle lfe
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out_samples[5] += tapped_samp;
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}
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}
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state.lowpass_0 = current_sample * state.lowpass_2 + state.lowpass_0 * state.lowpass_1;
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state.early_delay_line.Tick(state.lowpass_0);
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for (std::size_t i = 0; i < CHANNEL_COUNT; i++) {
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out_samples[i] *= state.early_gain;
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}
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// Two channel seems to apply a latet gain, we require to save this
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f32 filter{};
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for (std::size_t i = 0; i < AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT; i++) {
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filter = state.fdn_delay_line[i].GetOutputSample();
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const auto computed = filter * state.lpf_coefficients[0][i] + state.shelf_filter[i];
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state.shelf_filter[i] =
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filter * state.lpf_coefficients[1][i] + computed * state.lpf_coefficients[2][i];
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fsamp[i] = computed;
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}
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// Mixing matrix
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mixed[0] = fsamp[1] + fsamp[2];
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mixed[1] = -fsamp[0] - fsamp[3];
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mixed[2] = fsamp[0] - fsamp[3];
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mixed[3] = fsamp[1] - fsamp[2];
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if constexpr (CHANNEL_COUNT == 2) {
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for (auto& mix : mixed) {
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mix *= (filter * state.late_gain);
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}
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}
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for (std::size_t i = 0; i < AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT; i++) {
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const auto late = early_tap * state.late_gain;
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osamp[i] = state.decay_delay_line0[i].Tick(late + mixed[i]);
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osamp[i] = state.decay_delay_line1[i].Tick(osamp[i]);
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state.fdn_delay_line[i].Tick(osamp[i]);
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}
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if constexpr (CHANNEL_COUNT == 1) {
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output[0][sample] = ToS32(state.dry_gain * ToFloat(input[0][sample]) +
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(out_samples[0] + osamp[0] + osamp[1]));
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} else if constexpr (CHANNEL_COUNT == 2 || CHANNEL_COUNT == 4) {
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for (std::size_t i = 0; i < CHANNEL_COUNT; i++) {
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output[i][sample] =
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ToS32(state.dry_gain * ToFloat(input[i][sample]) + (out_samples[i] + osamp[i]));
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}
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} else if constexpr (CHANNEL_COUNT == 6) {
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const auto temp_center = state.center_delay_line.Tick(0.5f * (osamp[2] - osamp[3]));
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for (std::size_t i = 0; i < 4; i++) {
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output[i][sample] =
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ToS32(state.dry_gain * ToFloat(input[i][sample]) + (out_samples[i] + osamp[i]));
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}
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output[4][sample] =
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ToS32(state.dry_gain * ToFloat(input[4][sample]) + (out_samples[4] + temp_center));
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output[5][sample] =
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ToS32(state.dry_gain * ToFloat(input[5][sample]) + (out_samples[5] + osamp[3]));
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}
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}
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}
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} // namespace
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CommandGenerator::CommandGenerator(AudioCommon::AudioRendererParameter& worker_params_,
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VoiceContext& voice_context_, MixContext& mix_context_,
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SplitterContext& splitter_context_,
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EffectContext& effect_context_, Core::Memory::Memory& memory_)
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: worker_params(worker_params_), voice_context(voice_context_), mix_context(mix_context_),
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splitter_context(splitter_context_), effect_context(effect_context_), memory(memory_),
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mix_buffer((worker_params.mix_buffer_count + AudioCommon::MAX_CHANNEL_COUNT) *
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worker_params.sample_count),
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sample_buffer(MIX_BUFFER_SIZE),
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depop_buffer((worker_params.mix_buffer_count + AudioCommon::MAX_CHANNEL_COUNT) *
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worker_params.sample_count) {}
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CommandGenerator::~CommandGenerator() = default;
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void CommandGenerator::ClearMixBuffers() {
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std::fill(mix_buffer.begin(), mix_buffer.end(), 0);
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std::fill(sample_buffer.begin(), sample_buffer.end(), 0);
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// std::fill(depop_buffer.begin(), depop_buffer.end(), 0);
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}
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void CommandGenerator::GenerateVoiceCommands() {
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if (dumping_frame) {
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LOG_DEBUG(Audio, "(DSP_TRACE) GenerateVoiceCommands");
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}
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// Grab all our voices
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const auto voice_count = voice_context.GetVoiceCount();
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for (std::size_t i = 0; i < voice_count; i++) {
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auto& voice_info = voice_context.GetSortedInfo(i);
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// Update voices and check if we should queue them
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if (voice_info.ShouldSkip() || !voice_info.UpdateForCommandGeneration(voice_context)) {
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continue;
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}
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// Queue our voice
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GenerateVoiceCommand(voice_info);
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}
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// Update our splitters
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splitter_context.UpdateInternalState();
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}
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void CommandGenerator::GenerateVoiceCommand(ServerVoiceInfo& voice_info) {
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auto& in_params = voice_info.GetInParams();
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const auto channel_count = in_params.channel_count;
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for (s32 channel = 0; channel < channel_count; channel++) {
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const auto resource_id = in_params.voice_channel_resource_id[channel];
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auto& dsp_state = voice_context.GetDspSharedState(resource_id);
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auto& channel_resource = voice_context.GetChannelResource(resource_id);
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// Decode our samples for our channel
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GenerateDataSourceCommand(voice_info, dsp_state, channel);
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if (in_params.should_depop) {
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in_params.last_volume = 0.0f;
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} else if (in_params.splitter_info_id != AudioCommon::NO_SPLITTER ||
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in_params.mix_id != AudioCommon::NO_MIX) {
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// Apply a biquad filter if needed
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GenerateBiquadFilterCommandForVoice(voice_info, dsp_state,
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worker_params.mix_buffer_count, channel);
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// Base voice volume ramping
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GenerateVolumeRampCommand(in_params.last_volume, in_params.volume, channel,
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in_params.node_id);
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in_params.last_volume = in_params.volume;
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if (in_params.mix_id != AudioCommon::NO_MIX) {
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// If we're using a mix id
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auto& mix_info = mix_context.GetInfo(in_params.mix_id);
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const auto& dest_mix_params = mix_info.GetInParams();
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// Voice Mixing
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GenerateVoiceMixCommand(
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channel_resource.GetCurrentMixVolume(), channel_resource.GetLastMixVolume(),
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dsp_state, dest_mix_params.buffer_offset, dest_mix_params.buffer_count,
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worker_params.mix_buffer_count + channel, in_params.node_id);
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// Update last mix volumes
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channel_resource.UpdateLastMixVolumes();
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} else if (in_params.splitter_info_id != AudioCommon::NO_SPLITTER) {
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s32 base = channel;
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while (auto* destination_data =
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GetDestinationData(in_params.splitter_info_id, base)) {
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base += channel_count;
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if (!destination_data->IsConfigured()) {
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continue;
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}
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if (destination_data->GetMixId() >= static_cast<int>(mix_context.GetCount())) {
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continue;
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}
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const auto& mix_info = mix_context.GetInfo(destination_data->GetMixId());
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const auto& dest_mix_params = mix_info.GetInParams();
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GenerateVoiceMixCommand(
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destination_data->CurrentMixVolumes(), destination_data->LastMixVolumes(),
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dsp_state, dest_mix_params.buffer_offset, dest_mix_params.buffer_count,
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worker_params.mix_buffer_count + channel, in_params.node_id);
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destination_data->MarkDirty();
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}
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}
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// Update biquad filter enabled states
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for (std::size_t i = 0; i < AudioCommon::MAX_BIQUAD_FILTERS; i++) {
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in_params.was_biquad_filter_enabled[i] = in_params.biquad_filter[i].enabled;
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}
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}
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}
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}
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void CommandGenerator::GenerateSubMixCommands() {
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const auto mix_count = mix_context.GetCount();
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for (std::size_t i = 0; i < mix_count; i++) {
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auto& mix_info = mix_context.GetSortedInfo(i);
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const auto& in_params = mix_info.GetInParams();
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if (!in_params.in_use || in_params.mix_id == AudioCommon::FINAL_MIX) {
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continue;
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}
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GenerateSubMixCommand(mix_info);
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}
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}
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void CommandGenerator::GenerateFinalMixCommands() {
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GenerateFinalMixCommand();
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}
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void CommandGenerator::PreCommand() {
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if (!dumping_frame) {
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return;
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}
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for (std::size_t i = 0; i < splitter_context.GetInfoCount(); i++) {
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const auto& base = splitter_context.GetInfo(i);
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std::string graph = fmt::format("b[{}]", i);
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const auto* head = base.GetHead();
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while (head != nullptr) {
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graph += fmt::format("->{}", head->GetMixId());
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head = head->GetNextDestination();
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}
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LOG_DEBUG(Audio, "(DSP_TRACE) SplitterGraph splitter_info={}, {}", i, graph);
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}
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}
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void CommandGenerator::PostCommand() {
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if (!dumping_frame) {
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return;
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}
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dumping_frame = false;
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}
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void CommandGenerator::GenerateDataSourceCommand(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
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s32 channel) {
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const auto& in_params = voice_info.GetInParams();
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const auto depop = in_params.should_depop;
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if (depop) {
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if (in_params.mix_id != AudioCommon::NO_MIX) {
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auto& mix_info = mix_context.GetInfo(in_params.mix_id);
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const auto& mix_in = mix_info.GetInParams();
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GenerateDepopPrepareCommand(dsp_state, mix_in.buffer_count, mix_in.buffer_offset);
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} else if (in_params.splitter_info_id != AudioCommon::NO_SPLITTER) {
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s32 index{};
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while (const auto* destination =
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GetDestinationData(in_params.splitter_info_id, index++)) {
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if (!destination->IsConfigured()) {
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continue;
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}
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auto& mix_info = mix_context.GetInfo(destination->GetMixId());
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const auto& mix_in = mix_info.GetInParams();
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GenerateDepopPrepareCommand(dsp_state, mix_in.buffer_count, mix_in.buffer_offset);
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}
|
|
}
|
|
} else {
|
|
switch (in_params.sample_format) {
|
|
case SampleFormat::Pcm8:
|
|
case SampleFormat::Pcm16:
|
|
case SampleFormat::Pcm32:
|
|
case SampleFormat::PcmFloat:
|
|
DecodeFromWaveBuffers(voice_info, GetChannelMixBuffer(channel), dsp_state, channel,
|
|
worker_params.sample_rate, worker_params.sample_count,
|
|
in_params.node_id);
|
|
break;
|
|
case SampleFormat::Adpcm:
|
|
ASSERT(channel == 0 && in_params.channel_count == 1);
|
|
DecodeFromWaveBuffers(voice_info, GetChannelMixBuffer(0), dsp_state, 0,
|
|
worker_params.sample_rate, worker_params.sample_count,
|
|
in_params.node_id);
|
|
break;
|
|
default:
|
|
UNREACHABLE_MSG("Unimplemented sample format={}", in_params.sample_format);
|
|
}
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateBiquadFilterCommandForVoice(ServerVoiceInfo& voice_info,
|
|
VoiceState& dsp_state,
|
|
[[maybe_unused]] s32 mix_buffer_count,
|
|
[[maybe_unused]] s32 channel) {
|
|
for (std::size_t i = 0; i < AudioCommon::MAX_BIQUAD_FILTERS; i++) {
|
|
const auto& in_params = voice_info.GetInParams();
|
|
auto& biquad_filter = in_params.biquad_filter[i];
|
|
// Check if biquad filter is actually used
|
|
if (!biquad_filter.enabled) {
|
|
continue;
|
|
}
|
|
|
|
// Reinitialize our biquad filter state if it was enabled previously
|
|
if (!in_params.was_biquad_filter_enabled[i]) {
|
|
dsp_state.biquad_filter_state.fill(0);
|
|
}
|
|
|
|
// Generate biquad filter
|
|
// GenerateBiquadFilterCommand(mix_buffer_count, biquad_filter,
|
|
// dsp_state.biquad_filter_state,
|
|
// mix_buffer_count + channel, mix_buffer_count + channel,
|
|
// worker_params.sample_count, voice_info.GetInParams().node_id);
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateBiquadFilterCommand([[maybe_unused]] s32 mix_buffer_id,
|
|
const BiquadFilterParameter& params,
|
|
std::array<s64, 2>& state,
|
|
std::size_t input_offset,
|
|
std::size_t output_offset, s32 sample_count,
|
|
s32 node_id) {
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(Audio,
|
|
"(DSP_TRACE) GenerateBiquadFilterCommand node_id={}, "
|
|
"input_mix_buffer={}, output_mix_buffer={}",
|
|
node_id, input_offset, output_offset);
|
|
}
|
|
std::span<const s32> input = GetMixBuffer(input_offset);
|
|
std::span<s32> output = GetMixBuffer(output_offset);
|
|
|
|
// Biquad filter parameters
|
|
const auto [n0, n1, n2] = params.numerator;
|
|
const auto [d0, d1] = params.denominator;
|
|
|
|
// Biquad filter states
|
|
auto [s0, s1] = state;
|
|
|
|
constexpr s64 int32_min = std::numeric_limits<s32>::min();
|
|
constexpr s64 int32_max = std::numeric_limits<s32>::max();
|
|
|
|
for (int i = 0; i < sample_count; ++i) {
|
|
const auto sample = static_cast<s64>(input[i]);
|
|
const auto f = (sample * n0 + s0 + 0x4000) >> 15;
|
|
const auto y = std::clamp(f, int32_min, int32_max);
|
|
s0 = sample * n1 + y * d0 + s1;
|
|
s1 = sample * n2 + y * d1;
|
|
output[i] = static_cast<s32>(y);
|
|
}
|
|
|
|
state = {s0, s1};
|
|
}
|
|
|
|
void CommandGenerator::GenerateDepopPrepareCommand(VoiceState& dsp_state,
|
|
std::size_t mix_buffer_count,
|
|
std::size_t mix_buffer_offset) {
|
|
for (std::size_t i = 0; i < mix_buffer_count; i++) {
|
|
auto& sample = dsp_state.previous_samples[i];
|
|
if (sample != 0) {
|
|
depop_buffer[mix_buffer_offset + i] += sample;
|
|
sample = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateDepopForMixBuffersCommand(std::size_t mix_buffer_count,
|
|
std::size_t mix_buffer_offset,
|
|
s32 sample_rate) {
|
|
const std::size_t end_offset =
|
|
std::min(mix_buffer_offset + mix_buffer_count, GetTotalMixBufferCount());
|
|
const s32 delta = sample_rate == 48000 ? 0x7B29 : 0x78CB;
|
|
for (std::size_t i = mix_buffer_offset; i < end_offset; i++) {
|
|
if (depop_buffer[i] == 0) {
|
|
continue;
|
|
}
|
|
|
|
depop_buffer[i] =
|
|
ApplyMixDepop(GetMixBuffer(i), depop_buffer[i], delta, worker_params.sample_count);
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateEffectCommand(ServerMixInfo& mix_info) {
|
|
const std::size_t effect_count = effect_context.GetCount();
|
|
const auto buffer_offset = mix_info.GetInParams().buffer_offset;
|
|
for (std::size_t i = 0; i < effect_count; i++) {
|
|
const auto index = mix_info.GetEffectOrder(i);
|
|
if (index == AudioCommon::NO_EFFECT_ORDER) {
|
|
break;
|
|
}
|
|
auto* info = effect_context.GetInfo(index);
|
|
const auto type = info->GetType();
|
|
|
|
// TODO(ogniK): Finish remaining effects
|
|
switch (type) {
|
|
case EffectType::Aux:
|
|
GenerateAuxCommand(buffer_offset, info, info->IsEnabled());
|
|
break;
|
|
case EffectType::I3dl2Reverb:
|
|
GenerateI3dl2ReverbEffectCommand(buffer_offset, info, info->IsEnabled());
|
|
break;
|
|
case EffectType::BiquadFilter:
|
|
GenerateBiquadFilterEffectCommand(buffer_offset, info, info->IsEnabled());
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
info->UpdateForCommandGeneration();
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateI3dl2ReverbEffectCommand(s32 mix_buffer_offset, EffectBase* info,
|
|
bool enabled) {
|
|
auto* reverb = dynamic_cast<EffectI3dl2Reverb*>(info);
|
|
const auto& params = reverb->GetParams();
|
|
auto& state = reverb->GetState();
|
|
const auto channel_count = params.channel_count;
|
|
|
|
if (channel_count != 1 && channel_count != 2 && channel_count != 4 && channel_count != 6) {
|
|
return;
|
|
}
|
|
|
|
std::array<std::span<const s32>, AudioCommon::MAX_CHANNEL_COUNT> input{};
|
|
std::array<std::span<s32>, AudioCommon::MAX_CHANNEL_COUNT> output{};
|
|
|
|
const auto status = params.status;
|
|
for (s32 i = 0; i < channel_count; i++) {
|
|
input[i] = GetMixBuffer(mix_buffer_offset + params.input[i]);
|
|
output[i] = GetMixBuffer(mix_buffer_offset + params.output[i]);
|
|
}
|
|
|
|
if (enabled) {
|
|
if (status == ParameterStatus::Initialized) {
|
|
InitializeI3dl2Reverb(reverb->GetParams(), state, info->GetWorkBuffer());
|
|
} else if (status == ParameterStatus::Updating) {
|
|
UpdateI3dl2Reverb(reverb->GetParams(), state, false);
|
|
}
|
|
}
|
|
|
|
if (enabled) {
|
|
switch (channel_count) {
|
|
case 1:
|
|
ApplyReverbGeneric<1>(state, input, output, worker_params.sample_count);
|
|
break;
|
|
case 2:
|
|
ApplyReverbGeneric<2>(state, input, output, worker_params.sample_count);
|
|
break;
|
|
case 4:
|
|
ApplyReverbGeneric<4>(state, input, output, worker_params.sample_count);
|
|
break;
|
|
case 6:
|
|
ApplyReverbGeneric<6>(state, input, output, worker_params.sample_count);
|
|
break;
|
|
}
|
|
} else {
|
|
for (s32 i = 0; i < channel_count; i++) {
|
|
// Only copy if the buffer input and output do not match!
|
|
if ((mix_buffer_offset + params.input[i]) != (mix_buffer_offset + params.output[i])) {
|
|
std::memcpy(output[i].data(), input[i].data(),
|
|
worker_params.sample_count * sizeof(s32));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateBiquadFilterEffectCommand(s32 mix_buffer_offset, EffectBase* info,
|
|
bool enabled) {
|
|
if (!enabled) {
|
|
return;
|
|
}
|
|
const auto& params = dynamic_cast<EffectBiquadFilter*>(info)->GetParams();
|
|
const auto channel_count = params.channel_count;
|
|
for (s32 i = 0; i < channel_count; i++) {
|
|
// TODO(ogniK): Actually implement biquad filter
|
|
if (params.input[i] != params.output[i]) {
|
|
std::span<const s32> input = GetMixBuffer(mix_buffer_offset + params.input[i]);
|
|
std::span<s32> output = GetMixBuffer(mix_buffer_offset + params.output[i]);
|
|
ApplyMix<1>(output, input, 32768, worker_params.sample_count);
|
|
}
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateAuxCommand(s32 mix_buffer_offset, EffectBase* info, bool enabled) {
|
|
auto* aux = dynamic_cast<EffectAuxInfo*>(info);
|
|
const auto& params = aux->GetParams();
|
|
if (aux->GetSendBuffer() != 0 && aux->GetRecvBuffer() != 0) {
|
|
const auto max_channels = params.count;
|
|
u32 offset{};
|
|
for (u32 channel = 0; channel < max_channels; channel++) {
|
|
u32 write_count = 0;
|
|
if (channel == (max_channels - 1)) {
|
|
write_count = offset + worker_params.sample_count;
|
|
}
|
|
|
|
const auto input_index = params.input_mix_buffers[channel] + mix_buffer_offset;
|
|
const auto output_index = params.output_mix_buffers[channel] + mix_buffer_offset;
|
|
|
|
if (enabled) {
|
|
AuxInfoDSP send_info{};
|
|
AuxInfoDSP recv_info{};
|
|
memory.ReadBlock(aux->GetSendInfo(), &send_info, sizeof(AuxInfoDSP));
|
|
memory.ReadBlock(aux->GetRecvInfo(), &recv_info, sizeof(AuxInfoDSP));
|
|
|
|
WriteAuxBuffer(send_info, aux->GetSendBuffer(), params.sample_count,
|
|
GetMixBuffer(input_index), worker_params.sample_count, offset,
|
|
write_count);
|
|
memory.WriteBlock(aux->GetSendInfo(), &send_info, sizeof(AuxInfoDSP));
|
|
|
|
const auto samples_read = ReadAuxBuffer(
|
|
recv_info, aux->GetRecvBuffer(), params.sample_count,
|
|
GetMixBuffer(output_index), worker_params.sample_count, offset, write_count);
|
|
memory.WriteBlock(aux->GetRecvInfo(), &recv_info, sizeof(AuxInfoDSP));
|
|
|
|
if (samples_read != static_cast<int>(worker_params.sample_count) &&
|
|
samples_read <= params.sample_count) {
|
|
std::memset(GetMixBuffer(output_index).data(), 0,
|
|
params.sample_count - samples_read);
|
|
}
|
|
} else {
|
|
AuxInfoDSP empty{};
|
|
memory.WriteBlock(aux->GetSendInfo(), &empty, sizeof(AuxInfoDSP));
|
|
memory.WriteBlock(aux->GetRecvInfo(), &empty, sizeof(AuxInfoDSP));
|
|
if (output_index != input_index) {
|
|
std::memcpy(GetMixBuffer(output_index).data(), GetMixBuffer(input_index).data(),
|
|
worker_params.sample_count * sizeof(s32));
|
|
}
|
|
}
|
|
|
|
offset += worker_params.sample_count;
|
|
}
|
|
}
|
|
}
|
|
|
|
ServerSplitterDestinationData* CommandGenerator::GetDestinationData(s32 splitter_id, s32 index) {
|
|
if (splitter_id == AudioCommon::NO_SPLITTER) {
|
|
return nullptr;
|
|
}
|
|
return splitter_context.GetDestinationData(splitter_id, index);
|
|
}
|
|
|
|
s32 CommandGenerator::WriteAuxBuffer(AuxInfoDSP& dsp_info, VAddr send_buffer, u32 max_samples,
|
|
std::span<const s32> data, u32 sample_count, u32 write_offset,
|
|
u32 write_count) {
|
|
if (max_samples == 0) {
|
|
return 0;
|
|
}
|
|
u32 offset = dsp_info.write_offset + write_offset;
|
|
if (send_buffer == 0 || offset > max_samples) {
|
|
return 0;
|
|
}
|
|
|
|
s32 data_offset{};
|
|
u32 remaining = sample_count;
|
|
while (remaining > 0) {
|
|
// Get position in buffer
|
|
const auto base = send_buffer + (offset * sizeof(u32));
|
|
const auto samples_to_grab = std::min(max_samples - offset, remaining);
|
|
// Write to output
|
|
memory.WriteBlock(base, (data.data() + data_offset), samples_to_grab * sizeof(u32));
|
|
offset = (offset + samples_to_grab) % max_samples;
|
|
remaining -= samples_to_grab;
|
|
data_offset += samples_to_grab;
|
|
}
|
|
|
|
if (write_count != 0) {
|
|
dsp_info.write_offset = (dsp_info.write_offset + write_count) % max_samples;
|
|
}
|
|
return sample_count;
|
|
}
|
|
|
|
s32 CommandGenerator::ReadAuxBuffer(AuxInfoDSP& recv_info, VAddr recv_buffer, u32 max_samples,
|
|
std::span<s32> out_data, u32 sample_count, u32 read_offset,
|
|
u32 read_count) {
|
|
if (max_samples == 0) {
|
|
return 0;
|
|
}
|
|
|
|
u32 offset = recv_info.read_offset + read_offset;
|
|
if (recv_buffer == 0 || offset > max_samples) {
|
|
return 0;
|
|
}
|
|
|
|
u32 remaining = sample_count;
|
|
s32 data_offset{};
|
|
while (remaining > 0) {
|
|
const auto base = recv_buffer + (offset * sizeof(u32));
|
|
const auto samples_to_grab = std::min(max_samples - offset, remaining);
|
|
std::vector<s32> buffer(samples_to_grab);
|
|
memory.ReadBlock(base, buffer.data(), buffer.size() * sizeof(u32));
|
|
std::memcpy(out_data.data() + data_offset, buffer.data(), buffer.size() * sizeof(u32));
|
|
offset = (offset + samples_to_grab) % max_samples;
|
|
remaining -= samples_to_grab;
|
|
data_offset += samples_to_grab;
|
|
}
|
|
|
|
if (read_count != 0) {
|
|
recv_info.read_offset = (recv_info.read_offset + read_count) % max_samples;
|
|
}
|
|
return sample_count;
|
|
}
|
|
|
|
void CommandGenerator::InitializeI3dl2Reverb(I3dl2ReverbParams& info, I3dl2ReverbState& state,
|
|
std::vector<u8>& work_buffer) {
|
|
// Reset state
|
|
state.lowpass_0 = 0.0f;
|
|
state.lowpass_1 = 0.0f;
|
|
state.lowpass_2 = 0.0f;
|
|
|
|
state.early_delay_line.Reset();
|
|
state.early_tap_steps.fill(0);
|
|
state.early_gain = 0.0f;
|
|
state.late_gain = 0.0f;
|
|
state.early_to_late_taps = 0;
|
|
for (std::size_t i = 0; i < AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT; i++) {
|
|
state.fdn_delay_line[i].Reset();
|
|
state.decay_delay_line0[i].Reset();
|
|
state.decay_delay_line1[i].Reset();
|
|
}
|
|
state.last_reverb_echo = 0.0f;
|
|
state.center_delay_line.Reset();
|
|
for (auto& coef : state.lpf_coefficients) {
|
|
coef.fill(0.0f);
|
|
}
|
|
state.shelf_filter.fill(0.0f);
|
|
state.dry_gain = 0.0f;
|
|
|
|
const auto sample_rate = info.sample_rate / 1000;
|
|
f32* work_buffer_ptr = reinterpret_cast<f32*>(work_buffer.data());
|
|
|
|
s32 delay_samples{};
|
|
for (std::size_t i = 0; i < AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT; i++) {
|
|
delay_samples =
|
|
AudioCommon::CalculateDelaySamples(sample_rate, FDN_MAX_DELAY_LINE_TIMES[i]);
|
|
state.fdn_delay_line[i].Initialize(delay_samples, work_buffer_ptr);
|
|
work_buffer_ptr += delay_samples + 1;
|
|
|
|
delay_samples =
|
|
AudioCommon::CalculateDelaySamples(sample_rate, DECAY0_MAX_DELAY_LINE_TIMES[i]);
|
|
state.decay_delay_line0[i].Initialize(delay_samples, 0.0f, work_buffer_ptr);
|
|
work_buffer_ptr += delay_samples + 1;
|
|
|
|
delay_samples =
|
|
AudioCommon::CalculateDelaySamples(sample_rate, DECAY1_MAX_DELAY_LINE_TIMES[i]);
|
|
state.decay_delay_line1[i].Initialize(delay_samples, 0.0f, work_buffer_ptr);
|
|
work_buffer_ptr += delay_samples + 1;
|
|
}
|
|
delay_samples = AudioCommon::CalculateDelaySamples(sample_rate, 5.0f);
|
|
state.center_delay_line.Initialize(delay_samples, work_buffer_ptr);
|
|
work_buffer_ptr += delay_samples + 1;
|
|
|
|
delay_samples = AudioCommon::CalculateDelaySamples(sample_rate, 400.0f);
|
|
state.early_delay_line.Initialize(delay_samples, work_buffer_ptr);
|
|
|
|
UpdateI3dl2Reverb(info, state, true);
|
|
}
|
|
|
|
void CommandGenerator::UpdateI3dl2Reverb(I3dl2ReverbParams& info, I3dl2ReverbState& state,
|
|
bool should_clear) {
|
|
|
|
state.dry_gain = info.dry_gain;
|
|
state.shelf_filter.fill(0.0f);
|
|
state.lowpass_0 = 0.0f;
|
|
state.early_gain = Pow10(std::min(info.room + info.reflection, 5000.0f) / 2000.0f);
|
|
state.late_gain = Pow10(std::min(info.room + info.reverb, 5000.0f) / 2000.0f);
|
|
|
|
const auto sample_rate = info.sample_rate / 1000;
|
|
const f32 hf_gain = Pow10(info.room_hf / 2000.0f);
|
|
if (hf_gain >= 1.0f) {
|
|
state.lowpass_2 = 1.0f;
|
|
state.lowpass_1 = 0.0f;
|
|
} else {
|
|
const auto a = 1.0f - hf_gain;
|
|
const auto b = 2.0f * (2.0f - hf_gain * CosD(256.0f * info.hf_reference /
|
|
static_cast<f32>(info.sample_rate)));
|
|
const auto c = std::sqrt(b * b - 4.0f * a * a);
|
|
|
|
state.lowpass_1 = (b - c) / (2.0f * a);
|
|
state.lowpass_2 = 1.0f - state.lowpass_1;
|
|
}
|
|
state.early_to_late_taps = AudioCommon::CalculateDelaySamples(
|
|
sample_rate, 1000.0f * (info.reflection_delay + info.reverb_delay));
|
|
|
|
state.last_reverb_echo = 0.6f * info.diffusion * 0.01f;
|
|
for (std::size_t i = 0; i < AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT; i++) {
|
|
const auto length =
|
|
FDN_MIN_DELAY_LINE_TIMES[i] +
|
|
(info.density / 100.0f) * (FDN_MAX_DELAY_LINE_TIMES[i] - FDN_MIN_DELAY_LINE_TIMES[i]);
|
|
state.fdn_delay_line[i].SetDelay(AudioCommon::CalculateDelaySamples(sample_rate, length));
|
|
|
|
const auto delay_sample_counts = state.fdn_delay_line[i].GetDelay() +
|
|
state.decay_delay_line0[i].GetDelay() +
|
|
state.decay_delay_line1[i].GetDelay();
|
|
|
|
float a = (-60.0f * static_cast<f32>(delay_sample_counts)) /
|
|
(info.decay_time * static_cast<f32>(info.sample_rate));
|
|
float b = a / info.hf_decay_ratio;
|
|
float c = CosD(128.0f * 0.5f * info.hf_reference / static_cast<f32>(info.sample_rate)) /
|
|
SinD(128.0f * 0.5f * info.hf_reference / static_cast<f32>(info.sample_rate));
|
|
float d = Pow10((b - a) / 40.0f);
|
|
float e = Pow10((b + a) / 40.0f) * 0.7071f;
|
|
|
|
state.lpf_coefficients[0][i] = e * ((d * c) + 1.0f) / (c + d);
|
|
state.lpf_coefficients[1][i] = e * (1.0f - (d * c)) / (c + d);
|
|
state.lpf_coefficients[2][i] = (c - d) / (c + d);
|
|
|
|
state.decay_delay_line0[i].SetCoefficient(state.last_reverb_echo);
|
|
state.decay_delay_line1[i].SetCoefficient(-0.9f * state.last_reverb_echo);
|
|
}
|
|
|
|
if (should_clear) {
|
|
for (std::size_t i = 0; i < AudioCommon::I3DL2REVERB_DELAY_LINE_COUNT; i++) {
|
|
state.fdn_delay_line[i].Clear();
|
|
state.decay_delay_line0[i].Clear();
|
|
state.decay_delay_line1[i].Clear();
|
|
}
|
|
state.early_delay_line.Clear();
|
|
state.center_delay_line.Clear();
|
|
}
|
|
|
|
const auto max_early_delay = state.early_delay_line.GetMaxDelay();
|
|
const auto reflection_time = 1000.0f * (0.9998f * info.reverb_delay + 0.02f);
|
|
for (std::size_t tap = 0; tap < AudioCommon::I3DL2REVERB_TAPS; tap++) {
|
|
const auto length = AudioCommon::CalculateDelaySamples(
|
|
sample_rate, 1000.0f * info.reflection_delay + reflection_time * EARLY_TAP_TIMES[tap]);
|
|
state.early_tap_steps[tap] = std::min(length, max_early_delay);
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateVolumeRampCommand(float last_volume, float current_volume,
|
|
s32 channel, s32 node_id) {
|
|
const auto last = static_cast<s32>(last_volume * 32768.0f);
|
|
const auto current = static_cast<s32>(current_volume * 32768.0f);
|
|
const auto delta = static_cast<s32>((static_cast<float>(current) - static_cast<float>(last)) /
|
|
static_cast<float>(worker_params.sample_count));
|
|
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(Audio,
|
|
"(DSP_TRACE) GenerateVolumeRampCommand node_id={}, input={}, output={}, "
|
|
"last_volume={}, current_volume={}",
|
|
node_id, GetMixChannelBufferOffset(channel), GetMixChannelBufferOffset(channel),
|
|
last_volume, current_volume);
|
|
}
|
|
// Apply generic gain on samples
|
|
ApplyGain(GetChannelMixBuffer(channel), GetChannelMixBuffer(channel), last, delta,
|
|
worker_params.sample_count);
|
|
}
|
|
|
|
void CommandGenerator::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) {
|
|
// Loop all our mix buffers
|
|
for (s32 i = 0; i < mix_buffer_count; i++) {
|
|
if (last_mix_volumes[i] != 0.0f || mix_volumes[i] != 0.0f) {
|
|
const auto delta = static_cast<float>((mix_volumes[i] - last_mix_volumes[i])) /
|
|
static_cast<float>(worker_params.sample_count);
|
|
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(Audio,
|
|
"(DSP_TRACE) GenerateVoiceMixCommand node_id={}, input={}, "
|
|
"output={}, last_volume={}, current_volume={}",
|
|
node_id, voice_index, mix_buffer_offset + i, last_mix_volumes[i],
|
|
mix_volumes[i]);
|
|
}
|
|
|
|
dsp_state.previous_samples[i] =
|
|
ApplyMixRamp(GetMixBuffer(mix_buffer_offset + i), GetMixBuffer(voice_index),
|
|
last_mix_volumes[i], delta, worker_params.sample_count);
|
|
} else {
|
|
dsp_state.previous_samples[i] = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateSubMixCommand(ServerMixInfo& mix_info) {
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(Audio, "(DSP_TRACE) GenerateSubMixCommand");
|
|
}
|
|
const auto& in_params = mix_info.GetInParams();
|
|
GenerateDepopForMixBuffersCommand(in_params.buffer_count, in_params.buffer_offset,
|
|
in_params.sample_rate);
|
|
|
|
GenerateEffectCommand(mix_info);
|
|
|
|
GenerateMixCommands(mix_info);
|
|
}
|
|
|
|
void CommandGenerator::GenerateMixCommands(ServerMixInfo& mix_info) {
|
|
if (!mix_info.HasAnyConnection()) {
|
|
return;
|
|
}
|
|
const auto& in_params = mix_info.GetInParams();
|
|
if (in_params.dest_mix_id != AudioCommon::NO_MIX) {
|
|
const auto& dest_mix = mix_context.GetInfo(in_params.dest_mix_id);
|
|
const auto& dest_in_params = dest_mix.GetInParams();
|
|
|
|
const auto buffer_count = in_params.buffer_count;
|
|
|
|
for (s32 i = 0; i < buffer_count; i++) {
|
|
for (s32 j = 0; j < dest_in_params.buffer_count; j++) {
|
|
const auto mixed_volume = in_params.volume * in_params.mix_volume[i][j];
|
|
if (mixed_volume != 0.0f) {
|
|
GenerateMixCommand(dest_in_params.buffer_offset + j,
|
|
in_params.buffer_offset + i, mixed_volume,
|
|
in_params.node_id);
|
|
}
|
|
}
|
|
}
|
|
} else if (in_params.splitter_id != AudioCommon::NO_SPLITTER) {
|
|
s32 base{};
|
|
while (const auto* destination_data = GetDestinationData(in_params.splitter_id, base++)) {
|
|
if (!destination_data->IsConfigured()) {
|
|
continue;
|
|
}
|
|
|
|
const auto& dest_mix = mix_context.GetInfo(destination_data->GetMixId());
|
|
const auto& dest_in_params = dest_mix.GetInParams();
|
|
const auto mix_index = (base - 1) % in_params.buffer_count + in_params.buffer_offset;
|
|
for (std::size_t i = 0; i < static_cast<std::size_t>(dest_in_params.buffer_count);
|
|
i++) {
|
|
const auto mixed_volume = in_params.volume * destination_data->GetMixVolume(i);
|
|
if (mixed_volume != 0.0f) {
|
|
GenerateMixCommand(dest_in_params.buffer_offset + i, mix_index, mixed_volume,
|
|
in_params.node_id);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateMixCommand(std::size_t output_offset, std::size_t input_offset,
|
|
float volume, s32 node_id) {
|
|
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(Audio,
|
|
"(DSP_TRACE) GenerateMixCommand node_id={}, input={}, output={}, volume={}",
|
|
node_id, input_offset, output_offset, volume);
|
|
}
|
|
|
|
std::span<s32> output = GetMixBuffer(output_offset);
|
|
std::span<const s32> input = GetMixBuffer(input_offset);
|
|
|
|
const s32 gain = static_cast<s32>(volume * 32768.0f);
|
|
// Mix with loop unrolling
|
|
if (worker_params.sample_count % 4 == 0) {
|
|
ApplyMix<4>(output, input, gain, worker_params.sample_count);
|
|
} else if (worker_params.sample_count % 2 == 0) {
|
|
ApplyMix<2>(output, input, gain, worker_params.sample_count);
|
|
} else {
|
|
ApplyMix<1>(output, input, gain, worker_params.sample_count);
|
|
}
|
|
}
|
|
|
|
void CommandGenerator::GenerateFinalMixCommand() {
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(Audio, "(DSP_TRACE) GenerateFinalMixCommand");
|
|
}
|
|
auto& mix_info = mix_context.GetFinalMixInfo();
|
|
const auto& in_params = mix_info.GetInParams();
|
|
|
|
GenerateDepopForMixBuffersCommand(in_params.buffer_count, in_params.buffer_offset,
|
|
in_params.sample_rate);
|
|
|
|
GenerateEffectCommand(mix_info);
|
|
|
|
for (s32 i = 0; i < in_params.buffer_count; i++) {
|
|
const s32 gain = static_cast<s32>(in_params.volume * 32768.0f);
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(
|
|
Audio,
|
|
"(DSP_TRACE) ApplyGainWithoutDelta node_id={}, input={}, output={}, volume={}",
|
|
in_params.node_id, in_params.buffer_offset + i, in_params.buffer_offset + i,
|
|
in_params.volume);
|
|
}
|
|
ApplyGainWithoutDelta(GetMixBuffer(in_params.buffer_offset + i),
|
|
GetMixBuffer(in_params.buffer_offset + i), gain,
|
|
worker_params.sample_count);
|
|
}
|
|
}
|
|
|
|
template <typename T>
|
|
s32 CommandGenerator::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) {
|
|
const auto& in_params = voice_info.GetInParams();
|
|
const auto& wave_buffer = in_params.wave_buffer[dsp_state.wave_buffer_index];
|
|
if (wave_buffer.buffer_address == 0) {
|
|
return 0;
|
|
}
|
|
if (wave_buffer.buffer_size == 0) {
|
|
return 0;
|
|
}
|
|
if (sample_end_offset < sample_start_offset) {
|
|
return 0;
|
|
}
|
|
const auto samples_remaining = (sample_end_offset - sample_start_offset) - dsp_state.offset;
|
|
const auto start_offset =
|
|
((dsp_state.offset + sample_start_offset) * in_params.channel_count) * sizeof(T);
|
|
const auto buffer_pos = wave_buffer.buffer_address + start_offset;
|
|
const auto samples_processed = std::min(sample_count, samples_remaining);
|
|
|
|
const auto channel_count = in_params.channel_count;
|
|
std::vector<T> buffer(samples_processed * channel_count);
|
|
memory.ReadBlock(buffer_pos, buffer.data(), buffer.size() * sizeof(T));
|
|
|
|
if constexpr (std::is_floating_point_v<T>) {
|
|
for (std::size_t i = 0; i < static_cast<std::size_t>(samples_processed); i++) {
|
|
sample_buffer[mix_offset + i] = static_cast<s32>(buffer[i * channel_count + channel] *
|
|
std::numeric_limits<s16>::max());
|
|
}
|
|
} else if constexpr (sizeof(T) == 1) {
|
|
for (std::size_t i = 0; i < static_cast<std::size_t>(samples_processed); i++) {
|
|
sample_buffer[mix_offset + i] =
|
|
static_cast<s32>(static_cast<f32>(buffer[i * channel_count + channel] /
|
|
std::numeric_limits<s8>::max()) *
|
|
std::numeric_limits<s16>::max());
|
|
}
|
|
} else if constexpr (sizeof(T) == 2) {
|
|
for (std::size_t i = 0; i < static_cast<std::size_t>(samples_processed); i++) {
|
|
sample_buffer[mix_offset + i] = buffer[i * channel_count + channel];
|
|
}
|
|
} else {
|
|
for (std::size_t i = 0; i < static_cast<std::size_t>(samples_processed); i++) {
|
|
sample_buffer[mix_offset + i] =
|
|
static_cast<s32>(static_cast<f32>(buffer[i * channel_count + channel] /
|
|
std::numeric_limits<s32>::max()) *
|
|
std::numeric_limits<s16>::max());
|
|
}
|
|
}
|
|
|
|
return samples_processed;
|
|
}
|
|
|
|
s32 CommandGenerator::DecodeAdpcm(ServerVoiceInfo& voice_info, VoiceState& dsp_state,
|
|
s32 sample_start_offset, s32 sample_end_offset, s32 sample_count,
|
|
[[maybe_unused]] s32 channel, std::size_t mix_offset) {
|
|
const auto& in_params = voice_info.GetInParams();
|
|
const auto& wave_buffer = in_params.wave_buffer[dsp_state.wave_buffer_index];
|
|
if (wave_buffer.buffer_address == 0) {
|
|
return 0;
|
|
}
|
|
if (wave_buffer.buffer_size == 0) {
|
|
return 0;
|
|
}
|
|
if (sample_end_offset < sample_start_offset) {
|
|
return 0;
|
|
}
|
|
|
|
static constexpr std::array<int, 16> SIGNED_NIBBLES{
|
|
0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1,
|
|
};
|
|
|
|
constexpr std::size_t FRAME_LEN = 8;
|
|
constexpr std::size_t NIBBLES_PER_SAMPLE = 16;
|
|
constexpr std::size_t SAMPLES_PER_FRAME = 14;
|
|
|
|
auto frame_header = dsp_state.context.header;
|
|
s32 idx = (frame_header >> 4) & 0xf;
|
|
s32 scale = frame_header & 0xf;
|
|
s16 yn1 = dsp_state.context.yn1;
|
|
s16 yn2 = dsp_state.context.yn2;
|
|
|
|
Codec::ADPCM_Coeff coeffs;
|
|
memory.ReadBlock(in_params.additional_params_address, coeffs.data(),
|
|
sizeof(Codec::ADPCM_Coeff));
|
|
|
|
s32 coef1 = coeffs[idx * 2];
|
|
s32 coef2 = coeffs[idx * 2 + 1];
|
|
|
|
const auto samples_remaining = (sample_end_offset - sample_start_offset) - dsp_state.offset;
|
|
const auto samples_processed = std::min(sample_count, samples_remaining);
|
|
const auto sample_pos = dsp_state.offset + sample_start_offset;
|
|
|
|
const auto samples_remaining_in_frame = sample_pos % SAMPLES_PER_FRAME;
|
|
auto position_in_frame = ((sample_pos / SAMPLES_PER_FRAME) * NIBBLES_PER_SAMPLE) +
|
|
samples_remaining_in_frame + (samples_remaining_in_frame != 0 ? 2 : 0);
|
|
|
|
const auto decode_sample = [&](const int nibble) -> s16 {
|
|
const int xn = nibble * (1 << 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 = static_cast<s16>(val);
|
|
return yn1;
|
|
};
|
|
|
|
std::size_t buffer_offset{};
|
|
std::vector<u8> buffer(
|
|
std::max((samples_processed / FRAME_LEN) * SAMPLES_PER_FRAME, FRAME_LEN));
|
|
memory.ReadBlock(wave_buffer.buffer_address + (position_in_frame / 2), buffer.data(),
|
|
buffer.size());
|
|
std::size_t cur_mix_offset = mix_offset;
|
|
|
|
auto remaining_samples = samples_processed;
|
|
while (remaining_samples > 0) {
|
|
if (position_in_frame % NIBBLES_PER_SAMPLE == 0) {
|
|
// Read header
|
|
frame_header = buffer[buffer_offset++];
|
|
idx = (frame_header >> 4) & 0xf;
|
|
scale = frame_header & 0xf;
|
|
coef1 = coeffs[idx * 2];
|
|
coef2 = coeffs[idx * 2 + 1];
|
|
position_in_frame += 2;
|
|
|
|
// Decode entire frame
|
|
if (remaining_samples >= static_cast<int>(SAMPLES_PER_FRAME)) {
|
|
for (std::size_t i = 0; i < SAMPLES_PER_FRAME / 2; i++) {
|
|
// Sample 1
|
|
const s32 s0 = SIGNED_NIBBLES[buffer[buffer_offset] >> 4];
|
|
const s32 s1 = SIGNED_NIBBLES[buffer[buffer_offset++] & 0xf];
|
|
const s16 sample_1 = decode_sample(s0);
|
|
const s16 sample_2 = decode_sample(s1);
|
|
sample_buffer[cur_mix_offset++] = sample_1;
|
|
sample_buffer[cur_mix_offset++] = sample_2;
|
|
}
|
|
remaining_samples -= static_cast<int>(SAMPLES_PER_FRAME);
|
|
position_in_frame += SAMPLES_PER_FRAME;
|
|
continue;
|
|
}
|
|
}
|
|
// Decode mid frame
|
|
s32 current_nibble = buffer[buffer_offset];
|
|
if (position_in_frame++ & 0x1) {
|
|
current_nibble &= 0xf;
|
|
buffer_offset++;
|
|
} else {
|
|
current_nibble >>= 4;
|
|
}
|
|
const s16 sample = decode_sample(SIGNED_NIBBLES[current_nibble]);
|
|
sample_buffer[cur_mix_offset++] = sample;
|
|
remaining_samples--;
|
|
}
|
|
|
|
dsp_state.context.header = frame_header;
|
|
dsp_state.context.yn1 = yn1;
|
|
dsp_state.context.yn2 = yn2;
|
|
|
|
return samples_processed;
|
|
}
|
|
|
|
std::span<s32> CommandGenerator::GetMixBuffer(std::size_t index) {
|
|
return std::span<s32>(mix_buffer.data() + (index * worker_params.sample_count),
|
|
worker_params.sample_count);
|
|
}
|
|
|
|
std::span<const s32> CommandGenerator::GetMixBuffer(std::size_t index) const {
|
|
return std::span<const s32>(mix_buffer.data() + (index * worker_params.sample_count),
|
|
worker_params.sample_count);
|
|
}
|
|
|
|
std::size_t CommandGenerator::GetMixChannelBufferOffset(s32 channel) const {
|
|
return worker_params.mix_buffer_count + channel;
|
|
}
|
|
|
|
std::size_t CommandGenerator::GetTotalMixBufferCount() const {
|
|
return worker_params.mix_buffer_count + AudioCommon::MAX_CHANNEL_COUNT;
|
|
}
|
|
|
|
std::span<s32> CommandGenerator::GetChannelMixBuffer(s32 channel) {
|
|
return GetMixBuffer(worker_params.mix_buffer_count + channel);
|
|
}
|
|
|
|
std::span<const s32> CommandGenerator::GetChannelMixBuffer(s32 channel) const {
|
|
return GetMixBuffer(worker_params.mix_buffer_count + channel);
|
|
}
|
|
|
|
void CommandGenerator::DecodeFromWaveBuffers(ServerVoiceInfo& voice_info, std::span<s32> output,
|
|
VoiceState& dsp_state, s32 channel,
|
|
s32 target_sample_rate, s32 sample_count,
|
|
s32 node_id) {
|
|
const auto& in_params = voice_info.GetInParams();
|
|
if (dumping_frame) {
|
|
LOG_DEBUG(Audio,
|
|
"(DSP_TRACE) DecodeFromWaveBuffers, node_id={}, channel={}, "
|
|
"format={}, sample_count={}, sample_rate={}, mix_id={}, splitter_id={}",
|
|
node_id, channel, in_params.sample_format, sample_count, in_params.sample_rate,
|
|
in_params.mix_id, in_params.splitter_info_id);
|
|
}
|
|
ASSERT_OR_EXECUTE(output.data() != nullptr, { return; });
|
|
|
|
const auto resample_rate = static_cast<s32>(
|
|
static_cast<float>(in_params.sample_rate) / static_cast<float>(target_sample_rate) *
|
|
static_cast<float>(static_cast<s32>(in_params.pitch * 32768.0f)));
|
|
if (dsp_state.fraction + sample_count * resample_rate >
|
|
static_cast<s32>(SCALED_MIX_BUFFER_SIZE - 4ULL)) {
|
|
return;
|
|
}
|
|
|
|
auto min_required_samples =
|
|
std::min(static_cast<s32>(SCALED_MIX_BUFFER_SIZE) - dsp_state.fraction, resample_rate);
|
|
if (min_required_samples >= sample_count) {
|
|
min_required_samples = sample_count;
|
|
}
|
|
|
|
std::size_t temp_mix_offset{};
|
|
s32 samples_output{};
|
|
auto samples_remaining = sample_count;
|
|
while (samples_remaining > 0) {
|
|
const auto samples_to_output = std::min(samples_remaining, min_required_samples);
|
|
const auto samples_to_read = (samples_to_output * resample_rate + dsp_state.fraction) >> 15;
|
|
|
|
if (!in_params.behavior_flags.is_pitch_and_src_skipped) {
|
|
// Append sample histtory for resampler
|
|
for (std::size_t i = 0; i < AudioCommon::MAX_SAMPLE_HISTORY; i++) {
|
|
sample_buffer[temp_mix_offset + i] = dsp_state.sample_history[i];
|
|
}
|
|
temp_mix_offset += 4;
|
|
}
|
|
|
|
s32 samples_read{};
|
|
while (samples_read < samples_to_read) {
|
|
const auto& wave_buffer = in_params.wave_buffer[dsp_state.wave_buffer_index];
|
|
// No more data can be read
|
|
if (!dsp_state.is_wave_buffer_valid[dsp_state.wave_buffer_index]) {
|
|
break;
|
|
}
|
|
|
|
if (in_params.sample_format == SampleFormat::Adpcm && dsp_state.offset == 0 &&
|
|
wave_buffer.context_address != 0 && wave_buffer.context_size != 0) {
|
|
memory.ReadBlock(wave_buffer.context_address, &dsp_state.context,
|
|
sizeof(ADPCMContext));
|
|
}
|
|
|
|
s32 samples_offset_start;
|
|
s32 samples_offset_end;
|
|
if (dsp_state.loop_count > 0 && wave_buffer.loop_start_sample != 0 &&
|
|
wave_buffer.loop_end_sample != 0 &&
|
|
wave_buffer.loop_start_sample <= wave_buffer.loop_end_sample) {
|
|
samples_offset_start = wave_buffer.loop_start_sample;
|
|
samples_offset_end = wave_buffer.loop_end_sample;
|
|
} else {
|
|
samples_offset_start = wave_buffer.start_sample_offset;
|
|
samples_offset_end = wave_buffer.end_sample_offset;
|
|
}
|
|
|
|
s32 samples_decoded{0};
|
|
switch (in_params.sample_format) {
|
|
case SampleFormat::Pcm8:
|
|
samples_decoded =
|
|
DecodePcm<s8>(voice_info, dsp_state, samples_offset_start, samples_offset_end,
|
|
samples_to_read - samples_read, channel, temp_mix_offset);
|
|
break;
|
|
case SampleFormat::Pcm16:
|
|
samples_decoded =
|
|
DecodePcm<s16>(voice_info, dsp_state, samples_offset_start, samples_offset_end,
|
|
samples_to_read - samples_read, channel, temp_mix_offset);
|
|
break;
|
|
case SampleFormat::Pcm32:
|
|
samples_decoded =
|
|
DecodePcm<s32>(voice_info, dsp_state, samples_offset_start, samples_offset_end,
|
|
samples_to_read - samples_read, channel, temp_mix_offset);
|
|
break;
|
|
case SampleFormat::PcmFloat:
|
|
samples_decoded =
|
|
DecodePcm<f32>(voice_info, dsp_state, samples_offset_start, samples_offset_end,
|
|
samples_to_read - samples_read, channel, temp_mix_offset);
|
|
break;
|
|
case SampleFormat::Adpcm:
|
|
samples_decoded =
|
|
DecodeAdpcm(voice_info, dsp_state, samples_offset_start, samples_offset_end,
|
|
samples_to_read - samples_read, channel, temp_mix_offset);
|
|
break;
|
|
default:
|
|
UNREACHABLE_MSG("Unimplemented sample format={}", in_params.sample_format);
|
|
}
|
|
|
|
temp_mix_offset += samples_decoded;
|
|
samples_read += samples_decoded;
|
|
dsp_state.offset += samples_decoded;
|
|
dsp_state.played_sample_count += samples_decoded;
|
|
|
|
if (dsp_state.offset >= (samples_offset_end - samples_offset_start) ||
|
|
samples_decoded == 0) {
|
|
// Reset our sample offset
|
|
dsp_state.offset = 0;
|
|
if (wave_buffer.is_looping) {
|
|
dsp_state.loop_count++;
|
|
if (wave_buffer.loop_count > 0 &&
|
|
(dsp_state.loop_count > wave_buffer.loop_count || samples_decoded == 0)) {
|
|
// End of our buffer
|
|
voice_info.SetWaveBufferCompleted(dsp_state, wave_buffer);
|
|
}
|
|
|
|
if (samples_decoded == 0) {
|
|
break;
|
|
}
|
|
|
|
if (in_params.behavior_flags.is_played_samples_reset_at_loop_point.Value()) {
|
|
dsp_state.played_sample_count = 0;
|
|
}
|
|
} else {
|
|
// Update our wave buffer states
|
|
voice_info.SetWaveBufferCompleted(dsp_state, wave_buffer);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (in_params.behavior_flags.is_pitch_and_src_skipped.Value()) {
|
|
// No need to resample
|
|
std::memcpy(output.data() + samples_output, sample_buffer.data(),
|
|
samples_read * sizeof(s32));
|
|
} else {
|
|
std::fill(sample_buffer.begin() + temp_mix_offset,
|
|
sample_buffer.begin() + temp_mix_offset + (samples_to_read - samples_read),
|
|
0);
|
|
AudioCore::Resample(output.data() + samples_output, sample_buffer.data(), resample_rate,
|
|
dsp_state.fraction, samples_to_output);
|
|
// Resample
|
|
for (std::size_t i = 0; i < AudioCommon::MAX_SAMPLE_HISTORY; i++) {
|
|
dsp_state.sample_history[i] = sample_buffer[samples_to_read + i];
|
|
}
|
|
}
|
|
samples_remaining -= samples_to_output;
|
|
samples_output += samples_to_output;
|
|
}
|
|
}
|
|
|
|
} // namespace AudioCore
|