1 files changed, 0 insertions, 77 deletions

diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp
deleted file mode 100644
index 868b7a173..000000000
--- a/src/audio_core/codec.cpp
+++ /dev/null
@@ -1,77 +0,0 @@
-// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project
-// SPDX-License-Identifier: GPL-2.0-or-later
-
-#include <algorithm>
-
-#include "audio_core/codec.h"
-
-namespace AudioCore::Codec {
-
-std::vector<s16> DecodeADPCM(const u8* const data, std::size_t size, const ADPCM_Coeff& coeff,
-                             ADPCMState& state) {
-    // GC-ADPCM with scale factor and variable coefficients.
-    // Frames are 8 bytes long containing 14 samples each.
-    // Samples are 4 bits (one nibble) long.
-
-    constexpr std::size_t FRAME_LEN = 8;
-    constexpr std::size_t SAMPLES_PER_FRAME = 14;
-    static constexpr std::array<int, 16> SIGNED_NIBBLES{
-        0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1,
-    };
-
-    const std::size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
-    const std::size_t ret_size =
-        sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
-    std::vector<s16> ret(ret_size);
-
-    int yn1 = state.yn1, yn2 = state.yn2;
-
-    const std::size_t NUM_FRAMES =
-        (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
-    for (std::size_t framei = 0; framei < NUM_FRAMES; framei++) {
-        const int frame_header = data[framei * FRAME_LEN];
-        const int scale = 1 << (frame_header & 0xF);
-        const int idx = (frame_header >> 4) & 0x7;
-
-        // Coefficients are fixed point with 11 bits fractional part.
-        const int coef1 = coeff[idx * 2 + 0];
-        const int coef2 = coeff[idx * 2 + 1];
-
-        // Decodes an audio sample. One nibble produces one sample.
-        const auto decode_sample = [&](const int nibble) -> s16 {
-            const int xn = nibble * scale;
-            // We first transform everything into 11 bit fixed point, perform the second order
-            // digital filter, then transform back.
-            // 0x400 == 0.5 in 11 bit fixed point.
-            // Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
-            int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
-            // Clamp to output range.
-            val = std::clamp<s32>(val, -32768, 32767);
-            // Advance output feedback.
-            yn2 = yn1;
-            yn1 = val;
-            return static_cast<s16>(val);
-        };
-
-        std::size_t outputi = framei * SAMPLES_PER_FRAME;
-        std::size_t datai = framei * FRAME_LEN + 1;
-        for (std::size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
-            const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
-            ret[outputi] = sample1;
-            outputi++;
-
-            const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]);
-            ret[outputi] = sample2;
-            outputi++;
-
-            datai++;
-        }
-    }
-
-    state.yn1 = static_cast<s16>(yn1);
-    state.yn2 = static_cast<s16>(yn2);
-
-    return ret;
-}
-
-} // namespace AudioCore::Codec