// Copyright 2016 Citra Emulator Project // Licensed under GPLv2 or any later version // Refer to the license.txt file included. #include #include "audio_core/interpolate.h" #include "common/assert.h" namespace AudioCore { namespace AudioInterp { // Calculations are done in fixed point with 24 fractional bits. // (This is not verified. This was chosen for minimal error.) constexpr u64 scale_factor = 1 << 24; constexpr u64 scale_mask = scale_factor - 1; /// Here we step over the input in steps of rate, until we consume all of the input. /// Three adjacent samples are passed to fn each step. template static void StepOverSamples(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, std::size_t& outputi, Function fn) { ASSERT(rate > 0); if (input.empty()) return; input.insert(input.begin(), {state.xn2, state.xn1}); const u64 step_size = static_cast(rate * scale_factor); u64 fposition = state.fposition; std::size_t inputi = 0; while (outputi < output.size()) { inputi = static_cast(fposition / scale_factor); if (inputi + 2 >= input.size()) { inputi = input.size() - 2; break; } u64 fraction = fposition & scale_mask; output[outputi++] = fn(fraction, input[inputi], input[inputi + 1], input[inputi + 2]); fposition += step_size; } state.xn2 = input[inputi]; state.xn1 = input[inputi + 1]; state.fposition = fposition - inputi * scale_factor; input.erase(input.begin(), std::next(input.begin(), inputi + 2)); } void None(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, std::size_t& outputi) { StepOverSamples( state, input, rate, output, outputi, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; }); } void Linear(State& state, StereoBuffer16& input, float rate, StereoFrame16& output, std::size_t& outputi) { // Note on accuracy: Some values that this produces are +/- 1 from the actual firmware. StepOverSamples(state, input, rate, output, outputi, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { // This is a saturated subtraction. (Verified by black-box fuzzing.) s64 delta0 = std::clamp(x1[0] - x0[0], -32768, 32767); s64 delta1 = std::clamp(x1[1] - x0[1], -32768, 32767); return std::array{ static_cast(x0[0] + fraction * delta0 / scale_factor), static_cast(x0[1] + fraction * delta1 / scale_factor), }; }); } } // namespace AudioInterp } // namespace AudioCore