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https://github.com/Lime3DS/Lime3DS.git
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d7459354f5
The current code inserts and deletes elements from the beginning of the audio buffer, which is very inefficient in an std::vector. Profiling was done using VisualStudio2017's Performance Analyzer in Super Mario 3D Land. Before this change: AudioInterp::Linear had 14.14% of the runtime (inclusive) and most of that time was spent in std::vector's insert implementation. After this change: AudioInterp::Linear has 0.36% of the runtime (inclusive)
77 lines
2.8 KiB
C++
77 lines
2.8 KiB
C++
// 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, 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 void StepOverSamples(State& state, StereoBuffer16& input, float rate,
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DSP::HLE::StereoFrame16& output, size_t& outputi, Function fn) {
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ASSERT(rate > 0);
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if (input.empty())
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return;
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input.insert(input.begin(), {state.xn2, state.xn1});
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const u64 step_size = static_cast<u64>(rate * scale_factor);
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u64 fposition = state.fposition;
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size_t inputi = 0;
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while (outputi < output.size()) {
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inputi = static_cast<size_t>(fposition / scale_factor);
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if (inputi + 2 >= input.size()) {
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inputi = input.size() - 2;
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break;
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}
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u64 fraction = fposition & scale_mask;
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output[outputi++] = fn(fraction, input[inputi], input[inputi + 1], input[inputi + 2]);
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fposition += step_size;
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}
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state.xn2 = input[inputi];
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state.xn1 = input[inputi + 1];
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state.fposition = fposition - inputi * scale_factor;
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input.erase(input.begin(), std::next(input.begin(), inputi + 2));
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}
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void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
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size_t& outputi) {
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StepOverSamples(
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state, input, rate, output, outputi,
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[](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
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}
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void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
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size_t& outputi) {
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// Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
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StepOverSamples(state, input, rate, output, outputi,
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[](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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