// Copyright 2016 Citra Emulator Project // Licensed under GPLv2 or any later version // Refer to the license.txt file included. #include #include #include #include "audio_core/audio_types.h" #include "audio_core/codec.h" #include "common/assert.h" #include "common/common_types.h" namespace AudioCore { 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 = std::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] >> 4]); ret[outputi].fill(sample1); outputi++; const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]); ret[outputi].fill(sample2); outputi++; datai++; } } state.yn1 = yn1; state.yn2 = yn2; return ret; } StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count) { ASSERT(num_channels == 1 || num_channels == 2); const auto decode_sample = [](u8 sample) { return static_cast(static_cast(sample) << 8); }; StereoBuffer16 ret(sample_count); if (num_channels == 1) { for (size_t i = 0; i < sample_count; i++) { ret[i].fill(decode_sample(data[i])); } } else { for (size_t i = 0; i < sample_count; i++) { ret[i][0] = decode_sample(data[i * 2 + 0]); ret[i][1] = decode_sample(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 { for (size_t i = 0; i < sample_count; ++i) { std::memcpy(&ret[i], data + i * sizeof(s16) * 2, 2 * sizeof(s16)); } } return ret; } } // namespace Codec } // namespace AudioCore