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author | James Rowe <jroweboy@gmail.com> | 2018-01-12 03:21:20 +0100 |
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committer | James Rowe <jroweboy@gmail.com> | 2018-01-13 03:11:03 +0100 |
commit | ebf9a784a9f7f4148a669dbb39e7cd50df779a14 (patch) | |
tree | d585685a1c0a34b903af1d086d62560bf56bb29f /src/audio_core/interpolate.cpp | |
parent | config: Default CPU core to Unicorn. (diff) | |
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Diffstat (limited to '')
-rw-r--r-- | src/audio_core/interpolate.cpp | 76 |
1 files changed, 0 insertions, 76 deletions
diff --git a/src/audio_core/interpolate.cpp b/src/audio_core/interpolate.cpp deleted file mode 100644 index 83573d772..000000000 --- a/src/audio_core/interpolate.cpp +++ /dev/null @@ -1,76 +0,0 @@ -// 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, until we consume all of the input. -/// Three adjacent samples are passed to fn each step. -template <typename Function> -static void StepOverSamples(State& state, StereoBuffer16& input, float rate, - DSP::HLE::StereoFrame16& output, 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<u64>(rate * scale_factor); - u64 fposition = state.fposition; - size_t inputi = 0; - - while (outputi < output.size()) { - inputi = static_cast<size_t>(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, 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; }); -} - -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. - 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 |