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AudioCore: Implement interpolation
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15c907317c
commit
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@ -4,6 +4,7 @@ set(SRCS
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hle/dsp.cpp
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hle/dsp.cpp
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hle/filter.cpp
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hle/filter.cpp
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hle/pipe.cpp
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hle/pipe.cpp
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interpolate.cpp
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)
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set(HEADERS
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set(HEADERS
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@ -13,6 +14,7 @@ set(HEADERS
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hle/dsp.h
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hle/dsp.h
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hle/filter.h
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hle/filter.h
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hle/pipe.h
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hle/pipe.h
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interpolate.h
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sink.h
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sink.h
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)
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)
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85
src/audio_core/interpolate.cpp
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85
src/audio_core/interpolate.cpp
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@ -0,0 +1,85 @@
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// Copyright 2016 Citra 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 "audio_core/interpolate.h"
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#include "common/assert.h"
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#include "common/math_util.h"
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namespace AudioInterp {
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// Calculations are done in fixed point with 24 fractional bits.
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// (This is not verified. This was chosen for minimal error.)
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constexpr u64 scale_factor = 1 << 24;
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constexpr u64 scale_mask = scale_factor - 1;
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/// Here we step over the input in steps of rate_multiplier, until we consume all of the input.
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/// Three adjacent samples are passed to fn each step.
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template <typename Function>
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static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input, float rate_multiplier, Function fn) {
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ASSERT(rate_multiplier > 0);
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if (input.size() < 2)
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return {};
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StereoBuffer16 output;
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output.reserve(static_cast<size_t>(input.size() / rate_multiplier));
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u64 step_size = static_cast<u64>(rate_multiplier * scale_factor);
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u64 fposition = 0;
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const u64 max_fposition = input.size() * scale_factor;
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while (fposition < 1 * scale_factor) {
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u64 fraction = fposition & scale_mask;
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output.push_back(fn(fraction, state.xn2, state.xn1, input[0]));
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fposition += step_size;
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}
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while (fposition < 2 * scale_factor) {
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u64 fraction = fposition & scale_mask;
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output.push_back(fn(fraction, state.xn1, input[0], input[1]));
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fposition += step_size;
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}
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while (fposition < max_fposition) {
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u64 fraction = fposition & scale_mask;
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size_t index = static_cast<size_t>(fposition / scale_factor);
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output.push_back(fn(fraction, input[index - 2], input[index - 1], input[index]));
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fposition += step_size;
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}
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state.xn2 = input[input.size() - 2];
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state.xn1 = input[input.size() - 1];
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return output;
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}
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StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
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return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
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return x0;
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});
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}
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StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
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// Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
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return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
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// This is a saturated subtraction. (Verified by black-box fuzzing.)
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s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
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s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
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return std::array<s16, 2> {
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static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
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static_cast<s16>(x0[1] + fraction * delta1 / scale_factor)
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};
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});
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}
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} // namespace AudioInterp
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41
src/audio_core/interpolate.h
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41
src/audio_core/interpolate.h
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@ -0,0 +1,41 @@
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// Copyright 2016 Citra 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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#pragma once
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#include <array>
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#include <vector>
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#include "common/common_types.h"
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namespace AudioInterp {
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/// A variable length buffer of signed PCM16 stereo samples.
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using StereoBuffer16 = std::vector<std::array<s16, 2>>;
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struct State {
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// Two historical samples.
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std::array<s16, 2> xn1 = {}; ///< x[n-1]
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std::array<s16, 2> xn2 = {}; ///< x[n-2]
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};
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/**
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* No interpolation. This is equivalent to a zero-order hold. There is a two-sample predelay.
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* @param input Input buffer.
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* @param rate_multiplier Stretch factor. Must be a positive non-zero value.
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* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
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* @return The resampled audio buffer.
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*/
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StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
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/**
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* Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
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* @param input Input buffer.
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* @param rate_multiplier Stretch factor. Must be a positive non-zero value.
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* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
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* @return The resampled audio buffer.
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*/
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StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);
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} // namespace AudioInterp
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