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-rw-r--r--src/audio_core/interpolate.cpp86
1 files changed, 38 insertions, 48 deletions
diff --git a/src/audio_core/interpolate.cpp b/src/audio_core/interpolate.cpp
index 8a5d4181a..16e68bc5c 100644
--- a/src/audio_core/interpolate.cpp
+++ b/src/audio_core/interpolate.cpp
@@ -13,74 +13,64 @@ namespace AudioInterp {
constexpr u64 scale_factor = 1 << 24;
constexpr u64 scale_mask = scale_factor - 1;
-/// Here we step over the input in steps of rate_multiplier, until we consume all of the input.
+/// 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 <typename Function>
-static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input,
- float rate_multiplier, Function fn) {
- ASSERT(rate_multiplier > 0);
+static void StepOverSamples(State& state, StereoBuffer16& input, float rate,
+ DSP::HLE::StereoFrame16& output, size_t& outputi, Function fn) {
+ ASSERT(rate > 0);
- if (input.size() < 2)
- return {};
+ if (input.empty())
+ return;
- StereoBuffer16 output;
- output.reserve(static_cast<size_t>(input.size() / rate_multiplier));
+ input.insert(input.begin(), {state.xn2, state.xn1});
- u64 step_size = static_cast<u64>(rate_multiplier * scale_factor);
+ const u64 step_size = static_cast<u64>(rate * scale_factor);
+ u64 fposition = state.fposition;
+ size_t inputi = 0;
- u64 fposition = 0;
- const u64 max_fposition = input.size() * scale_factor;
+ while (outputi < output.size()) {
+ inputi = static_cast<size_t>(fposition / scale_factor);
- while (fposition < 1 * scale_factor) {
- u64 fraction = fposition & scale_mask;
-
- output.push_back(fn(fraction, state.xn2, state.xn1, input[0]));
-
- fposition += step_size;
- }
-
- while (fposition < 2 * scale_factor) {
- u64 fraction = fposition & scale_mask;
-
- output.push_back(fn(fraction, state.xn1, input[0], input[1]));
-
- fposition += step_size;
- }
+ if (inputi + 2 >= input.size()) {
+ inputi = input.size() - 2;
+ break;
+ }
- while (fposition < max_fposition) {
u64 fraction = fposition & scale_mask;
-
- size_t index = static_cast<size_t>(fposition / scale_factor);
- output.push_back(fn(fraction, input[index - 2], input[index - 1], input[index]));
+ output[outputi++] = fn(fraction, input[inputi], input[inputi + 1], input[inputi + 2]);
fposition += step_size;
}
- state.xn2 = input[input.size() - 2];
- state.xn1 = input[input.size() - 1];
+ state.xn2 = input[inputi];
+ state.xn1 = input[inputi + 1];
+ state.fposition = fposition - inputi * scale_factor;
- return output;
+ input.erase(input.begin(), input.begin() + inputi + 2);
}
-StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
- return StepOverSamples(
- state, input, rate_multiplier,
+void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+ size_t& outputi) {
+ StepOverSamples(
+ state, input, rate, output, outputi,
[](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
}
-StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
+void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
+ size_t& outputi) {
// Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
- return StepOverSamples(state, input, rate_multiplier,
- [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
- // This is a saturated subtraction. (Verified by black-box fuzzing.)
- s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
- s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
-
- return std::array<s16, 2>{
- static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
- static_cast<s16>(x0[1] + fraction * delta1 / scale_factor),
- };
- });
+ 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 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
+ s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
+
+ return std::array<s16, 2>{
+ static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
+ static_cast<s16>(x0[1] + fraction * delta1 / scale_factor),
+ };
+ });
}
} // namespace AudioInterp