Merge pull request #1723 from MerryMage/audio-interp

AudioCore: Implement interpolation
This commit is contained in:
bunnei 2016-04-29 10:43:20 -04:00
commit 4c235955cf
3 changed files with 128 additions and 0 deletions

View File

@ -4,6 +4,7 @@ set(SRCS
hle/dsp.cpp hle/dsp.cpp
hle/filter.cpp hle/filter.cpp
hle/pipe.cpp hle/pipe.cpp
interpolate.cpp
) )
set(HEADERS set(HEADERS
@ -13,6 +14,7 @@ set(HEADERS
hle/dsp.h hle/dsp.h
hle/filter.h hle/filter.h
hle/pipe.h hle/pipe.h
interpolate.h
sink.h sink.h
) )

View File

@ -0,0 +1,85 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include "audio_core/interpolate.h"
#include "common/assert.h"
#include "common/math_util.h"
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_multiplier, 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);
if (input.size() < 2)
return {};
StereoBuffer16 output;
output.reserve(static_cast<size_t>(input.size() / rate_multiplier));
u64 step_size = static_cast<u64>(rate_multiplier * scale_factor);
u64 fposition = 0;
const u64 max_fposition = input.size() * 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;
}
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;
}
StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
return StepOverSamples(state, input, rate_multiplier, [](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
return x0;
});
}
StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
// 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)
};
});
}
} // namespace AudioInterp

View File

@ -0,0 +1,41 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <array>
#include <vector>
#include "common/common_types.h"
namespace AudioInterp {
/// A variable length buffer of signed PCM16 stereo samples.
using StereoBuffer16 = std::vector<std::array<s16, 2>>;
struct State {
// Two historical samples.
std::array<s16, 2> xn1 = {}; ///< x[n-1]
std::array<s16, 2> xn2 = {}; ///< x[n-2]
};
/**
* No interpolation. This is equivalent to a zero-order hold. There is a two-sample predelay.
* @param input Input buffer.
* @param rate_multiplier Stretch factor. Must be a positive non-zero value.
* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
* @return The resampled audio buffer.
*/
StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
/**
* Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
* @param input Input buffer.
* @param rate_multiplier Stretch factor. Must be a positive non-zero value.
* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0 performs upsampling.
* @return The resampled audio buffer.
*/
StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);
} // namespace AudioInterp