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DSP: Implement audio codecs (PCM8, PCM16, ADPCM)
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parent
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commit
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@ -1,11 +1,13 @@
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set(SRCS
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audio_core.cpp
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codec.cpp
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hle/dsp.cpp
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hle/pipe.cpp
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)
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set(HEADERS
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audio_core.h
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codec.h
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hle/dsp.h
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hle/pipe.h
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sink.h
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122
src/audio_core/codec.cpp
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122
src/audio_core/codec.cpp
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// 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 <array>
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#include <cstddef>
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#include <cstring>
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#include <vector>
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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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#include "common/math_util.h"
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namespace Codec {
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StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, 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 size_t FRAME_LEN = 8;
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constexpr size_t SAMPLES_PER_FRAME = 14;
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constexpr std::array<int, 16> SIGNED_NIBBLES {{ 0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1 }};
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const size_t ret_size = 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,
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yn2 = state.yn2;
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const size_t NUM_FRAMES = (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
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for (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 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 = MathUtil::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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size_t outputi = framei * SAMPLES_PER_FRAME;
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size_t datai = framei * FRAME_LEN + 1;
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for (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] & 0xF]);
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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] >> 4]);
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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 = yn1;
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state.yn2 = yn2;
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return ret;
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}
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static s16 SignExtendS8(u8 x) {
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// The data is actually signed PCM8.
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// We sign extend this to signed PCM16.
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return static_cast<s16>(static_cast<s8>(x));
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}
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StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const 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 (size_t i = 0; i < sample_count; i++) {
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ret[i].fill(SignExtendS8(data[i]));
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}
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} else {
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for (size_t i = 0; i < sample_count; i++) {
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ret[i][0] = SignExtendS8(data[i * 2 + 0]);
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ret[i][1] = SignExtendS8(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, const 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 (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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std::memcpy(ret.data(), data, sample_count * 2 * sizeof(u16));
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}
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return ret;
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}
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};
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50
src/audio_core/codec.h
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50
src/audio_core/codec.h
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// 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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#pragma once
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#include <array>
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#include <vector>
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#include "common/common_types.h"
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namespace Codec {
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/// A variable length buffer of signed PCM16 stereo samples.
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using StereoBuffer16 = std::vector<std::array<s16, 2>>;
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/// See: Codec::DecodeADPCM
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struct ADPCMState {
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// Two historical samples from previous processed buffer,
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// required for ADPCM decoding
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s16 yn1; ///< y[n-1]
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s16 yn2; ///< y[n-2]
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};
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/**
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* @param data Pointer to buffer that contains ADPCM data to decode
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* @param sample_count Length of buffer in terms of number of samples
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* @param adpcm_coeff ADPCM coefficients
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* @param state ADPCM state, this is updated with new state
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* @return Decoded stereo signed PCM16 data, sample_count in length
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*/
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StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, const std::array<s16, 16>& adpcm_coeff, ADPCMState& state);
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/**
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* @param num_channels Number of channels
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* @param data Pointer to buffer that contains PCM8 data to decode
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* @param sample_count Length of buffer in terms of number of samples
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* @return Decoded stereo signed PCM16 data, sample_count in length
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*/
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StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count);
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/**
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* @param num_channels Number of channels
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* @param data Pointer to buffer that contains PCM16 data to decode
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* @param sample_count Length of buffer in terms of number of samples
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* @return Decoded stereo signed PCM16 data, sample_count in length
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*/
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StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, const size_t sample_count);
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};
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