f0e3637c7a
Allows compilers to elide needing to push these values on the stack every time the function is called. Co-authored-by: Lioncash <mathew1800@gmail.com>
127 lines
4.3 KiB
C++
127 lines
4.3 KiB
C++
// Copyright 2016 Citra Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <algorithm>
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#include <array>
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#include <cstddef>
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#include <cstring>
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#include "audio_core/audio_types.h"
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#include "audio_core/codec.h"
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#include "common/assert.h"
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#include "common/common_types.h"
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namespace AudioCore::Codec {
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StereoBuffer16 DecodeADPCM(const u8* const data, const std::size_t sample_count,
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const std::array<s16, 16>& adpcm_coeff, ADPCMState& state) {
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// GC-ADPCM with scale factor and variable coefficients.
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// Frames are 8 bytes long containing 14 samples each.
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// Samples are 4 bits (one nibble) long.
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constexpr std::size_t FRAME_LEN = 8;
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constexpr std::size_t SAMPLES_PER_FRAME = 14;
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static constexpr std::array<int, 16> SIGNED_NIBBLES{
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0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1,
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};
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const std::size_t ret_size =
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sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two.
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StereoBuffer16 ret(ret_size);
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int yn1 = state.yn1, yn2 = state.yn2;
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const std::size_t NUM_FRAMES =
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(sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
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for (std::size_t framei = 0; framei < NUM_FRAMES; framei++) {
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const int frame_header = data[framei * FRAME_LEN];
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const int scale = 1 << (frame_header & 0xF);
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const int idx = (frame_header >> 4) & 0x7;
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// Coefficients are fixed point with 11 bits fractional part.
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const int coef1 = adpcm_coeff[idx * 2 + 0];
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const int coef2 = adpcm_coeff[idx * 2 + 1];
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// Decodes an audio sample. One nibble produces one sample.
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const auto decode_sample = [&](const int nibble) -> s16 {
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const int xn = nibble * scale;
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// We first transform everything into 11 bit fixed point, perform the second order
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// digital filter, then transform back.
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// 0x400 == 0.5 in 11 bit fixed point.
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// Filter: y[n] = x[n] + 0.5 + c1 * y[n-1] + c2 * y[n-2]
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int val = ((xn << 11) + 0x400 + coef1 * yn1 + coef2 * yn2) >> 11;
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// Clamp to output range.
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val = std::clamp(val, -32768, 32767);
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// Advance output feedback.
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yn2 = yn1;
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yn1 = val;
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return (s16)val;
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};
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std::size_t outputi = framei * SAMPLES_PER_FRAME;
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std::size_t datai = framei * FRAME_LEN + 1;
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for (std::size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
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const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]);
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ret[outputi].fill(sample1);
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outputi++;
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const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]);
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ret[outputi].fill(sample2);
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outputi++;
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datai++;
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}
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}
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state.yn1 = static_cast<s16>(yn1);
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state.yn2 = static_cast<s16>(yn2);
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return ret;
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}
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StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data,
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const std::size_t sample_count) {
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ASSERT(num_channels == 1 || num_channels == 2);
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const auto decode_sample = [](u8 sample) {
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return static_cast<s16>(static_cast<u16>(sample) << 8);
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};
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StereoBuffer16 ret(sample_count);
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if (num_channels == 1) {
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for (std::size_t i = 0; i < sample_count; i++) {
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ret[i].fill(decode_sample(data[i]));
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}
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} else {
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for (std::size_t i = 0; i < sample_count; i++) {
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ret[i][0] = decode_sample(data[i * 2 + 0]);
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ret[i][1] = decode_sample(data[i * 2 + 1]);
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}
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}
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return ret;
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}
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StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data,
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const std::size_t sample_count) {
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ASSERT(num_channels == 1 || num_channels == 2);
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StereoBuffer16 ret(sample_count);
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if (num_channels == 1) {
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for (std::size_t i = 0; i < sample_count; i++) {
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s16 sample;
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std::memcpy(&sample, data + i * sizeof(s16), sizeof(s16));
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ret[i].fill(sample);
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}
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} else {
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for (std::size_t i = 0; i < sample_count; ++i) {
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std::memcpy(&ret[i], data + i * sizeof(s16) * 2, 2 * sizeof(s16));
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}
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}
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return ret;
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}
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} // namespace AudioCore::Codec
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