diff options
author | bunnei <bunneidev@gmail.com> | 2016-04-29 16:43:20 +0200 |
---|---|---|
committer | bunnei <bunneidev@gmail.com> | 2016-04-29 16:43:20 +0200 |
commit | 4c235955cf09ad38ca4723f6c5861298f7e845cc (patch) | |
tree | b933a3f130e5e5f4e57ca90ce2ecdfa57e0ea1d8 /src | |
parent | Merge pull request #1730 from hrydgard/vertex-loader (diff) | |
parent | AudioCore: Implement interpolation (diff) | |
download | yuzu-4c235955cf09ad38ca4723f6c5861298f7e845cc.tar yuzu-4c235955cf09ad38ca4723f6c5861298f7e845cc.tar.gz yuzu-4c235955cf09ad38ca4723f6c5861298f7e845cc.tar.bz2 yuzu-4c235955cf09ad38ca4723f6c5861298f7e845cc.tar.lz yuzu-4c235955cf09ad38ca4723f6c5861298f7e845cc.tar.xz yuzu-4c235955cf09ad38ca4723f6c5861298f7e845cc.tar.zst yuzu-4c235955cf09ad38ca4723f6c5861298f7e845cc.zip |
Diffstat (limited to '')
-rw-r--r-- | src/audio_core/CMakeLists.txt | 2 | ||||
-rw-r--r-- | src/audio_core/interpolate.cpp | 85 | ||||
-rw-r--r-- | src/audio_core/interpolate.h | 41 |
3 files changed, 128 insertions, 0 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt index c08ce69e0..a965af291 100644 --- a/src/audio_core/CMakeLists.txt +++ b/src/audio_core/CMakeLists.txt @@ -4,6 +4,7 @@ set(SRCS hle/dsp.cpp hle/filter.cpp hle/pipe.cpp + interpolate.cpp ) set(HEADERS @@ -13,6 +14,7 @@ set(HEADERS hle/dsp.h hle/filter.h hle/pipe.h + interpolate.h sink.h ) diff --git a/src/audio_core/interpolate.cpp b/src/audio_core/interpolate.cpp new file mode 100644 index 000000000..fcd3aa066 --- /dev/null +++ b/src/audio_core/interpolate.cpp @@ -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 diff --git a/src/audio_core/interpolate.h b/src/audio_core/interpolate.h new file mode 100644 index 000000000..a4c0a453d --- /dev/null +++ b/src/audio_core/interpolate.h @@ -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 |