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interpolate: Interpolate on a frame-by-frame basis

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MerryMage
2017-08-03 12:22:51 +01:00
parent 035716d57b
commit 933508e2a2
3 changed files with 74 additions and 88 deletions
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86
src/audio_core/interpolate.cpp
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@ -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);
if (inputi + 2 >= input.size()) {
inputi = input.size() - 2;
break;
}
while (fposition < 1 * scale_factor) {
u64 fraction = fposition & scale_mask;
output.push_back(fn(fraction, state.xn2, state.xn1, input[0]));
output[outputi++] = fn(fraction, input[inputi], input[inputi + 1], input[inputi + 2]);
fposition += step_size;
}
while (fposition < 2 * scale_factor) {
u64 fraction = fposition & scale_mask;
state.xn2 = input[inputi];
state.xn1 = input[inputi + 1];
state.fposition = fposition - inputi * scale_factor;
output.push_back(fn(fraction, state.xn1, input[0], input[1]));
fposition += step_size;
}
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]));
fposition += step_size;
}
state.xn2 = input[input.size() - 2];
state.xn1 = input[input.size() - 1];
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);
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),
};
});
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