From ebf9a784a9f7f4148a669dbb39e7cd50df779a14 Mon Sep 17 00:00:00 2001
From: James Rowe <jroweboy@gmail.com>
Date: Thu, 11 Jan 2018 19:21:20 -0700
Subject: Massive removal of unused modules

---
 src/audio_core/interpolate.cpp | 76 ------------------------------------------
 1 file changed, 76 deletions(-)
 delete mode 100644 src/audio_core/interpolate.cpp

(limited to 'src/audio_core/interpolate.cpp')

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
-- 
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