From efd1c3f8c3f649b0fa3fec0b236e9a748fc34e98 Mon Sep 17 00:00:00 2001 From: MerryMage Date: Thu, 24 Mar 2016 00:12:54 +0000 Subject: [PATCH] DSP: Implement audio codecs (PCM8, PCM16, ADPCM) --- src/audio_core/CMakeLists.txt | 2 + src/audio_core/codec.cpp | 122 ++++++++++++++++++++++++++++++++++ src/audio_core/codec.h | 50 ++++++++++++++ 3 files changed, 174 insertions(+) create mode 100644 src/audio_core/codec.cpp create mode 100644 src/audio_core/codec.h diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt index b0d1c7eb66..c4bad8cb06 100644 --- a/src/audio_core/CMakeLists.txt +++ b/src/audio_core/CMakeLists.txt @@ -1,11 +1,13 @@ set(SRCS audio_core.cpp + codec.cpp hle/dsp.cpp hle/pipe.cpp ) set(HEADERS audio_core.h + codec.h hle/dsp.h hle/pipe.h sink.h diff --git a/src/audio_core/codec.cpp b/src/audio_core/codec.cpp new file mode 100644 index 0000000000..ab65514b73 --- /dev/null +++ b/src/audio_core/codec.cpp @@ -0,0 +1,122 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#include +#include +#include +#include + +#include "audio_core/codec.h" + +#include "common/assert.h" +#include "common/common_types.h" +#include "common/math_util.h" + +namespace Codec { + +StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, const std::array& adpcm_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 size_t FRAME_LEN = 8; + constexpr size_t SAMPLES_PER_FRAME = 14; + constexpr std::array SIGNED_NIBBLES {{ 0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1 }}; + + const size_t ret_size = sample_count % 2 == 0 ? sample_count : sample_count + 1; // Ensure multiple of two. + StereoBuffer16 ret(ret_size); + + int yn1 = state.yn1, + yn2 = state.yn2; + + const size_t NUM_FRAMES = (sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up. + for (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 = adpcm_coeff[idx * 2 + 0]; + const int coef2 = adpcm_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 = MathUtil::Clamp(val, -32768, 32767); + // Advance output feedback. + yn2 = yn1; + yn1 = val; + return (s16)val; + }; + + size_t outputi = framei * SAMPLES_PER_FRAME; + size_t datai = framei * FRAME_LEN + 1; + for (size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) { + const s16 sample1 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]); + ret[outputi].fill(sample1); + outputi++; + + const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] >> 4]); + ret[outputi].fill(sample2); + outputi++; + + datai++; + } + } + + state.yn1 = yn1; + state.yn2 = yn2; + + return ret; +} + +static s16 SignExtendS8(u8 x) { + // The data is actually signed PCM8. + // We sign extend this to signed PCM16. + return static_cast(static_cast(x)); +} + +StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count) { + ASSERT(num_channels == 1 || num_channels == 2); + + StereoBuffer16 ret(sample_count); + + if (num_channels == 1) { + for (size_t i = 0; i < sample_count; i++) { + ret[i].fill(SignExtendS8(data[i])); + } + } else { + for (size_t i = 0; i < sample_count; i++) { + ret[i][0] = SignExtendS8(data[i * 2 + 0]); + ret[i][1] = SignExtendS8(data[i * 2 + 1]); + } + } + + return ret; +} + +StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, const size_t sample_count) { + ASSERT(num_channels == 1 || num_channels == 2); + + StereoBuffer16 ret(sample_count); + + if (num_channels == 1) { + for (size_t i = 0; i < sample_count; i++) { + s16 sample; + std::memcpy(&sample, data + i * sizeof(s16), sizeof(s16)); + ret[i].fill(sample); + } + } else { + std::memcpy(ret.data(), data, sample_count * 2 * sizeof(u16)); + } + + return ret; +} + +}; diff --git a/src/audio_core/codec.h b/src/audio_core/codec.h new file mode 100644 index 0000000000..e695f2edcd --- /dev/null +++ b/src/audio_core/codec.h @@ -0,0 +1,50 @@ +// Copyright 2016 Citra Emulator Project +// Licensed under GPLv2 or any later version +// Refer to the license.txt file included. + +#pragma once + +#include +#include + +#include "common/common_types.h" + +namespace Codec { + +/// A variable length buffer of signed PCM16 stereo samples. +using StereoBuffer16 = std::vector>; + +/// See: Codec::DecodeADPCM +struct ADPCMState { + // Two historical samples from previous processed buffer, + // required for ADPCM decoding + s16 yn1; ///< y[n-1] + s16 yn2; ///< y[n-2] +}; + +/** + * @param data Pointer to buffer that contains ADPCM data to decode + * @param sample_count Length of buffer in terms of number of samples + * @param adpcm_coeff ADPCM coefficients + * @param state ADPCM state, this is updated with new state + * @return Decoded stereo signed PCM16 data, sample_count in length + */ +StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, const std::array& adpcm_coeff, ADPCMState& state); + +/** + * @param num_channels Number of channels + * @param data Pointer to buffer that contains PCM8 data to decode + * @param sample_count Length of buffer in terms of number of samples + * @return Decoded stereo signed PCM16 data, sample_count in length + */ +StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count); + +/** + * @param num_channels Number of channels + * @param data Pointer to buffer that contains PCM16 data to decode + * @param sample_count Length of buffer in terms of number of samples + * @return Decoded stereo signed PCM16 data, sample_count in length + */ +StereoBuffer16 DecodePCM16(const unsigned num_channels, const u8* const data, const size_t sample_count); + +};