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	audio_core: Port codec code from Citra for ADPCM decoding.
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				| @ -4,6 +4,8 @@ add_library(audio_core STATIC | ||||
|     buffer.h | ||||
|     cubeb_sink.cpp | ||||
|     cubeb_sink.h | ||||
|     codec.cpp | ||||
|     codec.h | ||||
|     null_sink.h | ||||
|     stream.cpp | ||||
|     stream.h | ||||
|  | ||||
							
								
								
									
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								src/audio_core/codec.cpp
									
									
									
									
									
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								src/audio_core/codec.cpp
									
									
									
									
									
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							| @ -0,0 +1,77 @@ | ||||
| // Copyright 2018 yuzu Emulator Project
 | ||||
| // Licensed under GPLv2 or any later version
 | ||||
| // Refer to the license.txt file included.
 | ||||
| 
 | ||||
| #include <algorithm> | ||||
| 
 | ||||
| #include "audio_core/codec.h" | ||||
| 
 | ||||
| namespace AudioCore::Codec { | ||||
| 
 | ||||
| std::vector<s16> DecodeADPCM(const u8* const data, 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 size_t FRAME_LEN = 8; | ||||
|     constexpr size_t SAMPLES_PER_FRAME = 14; | ||||
|     constexpr std::array<int, 16> SIGNED_NIBBLES = { | ||||
|         {0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1}}; | ||||
| 
 | ||||
|     const size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME; | ||||
|     const 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 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 = 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); | ||||
|         }; | ||||
| 
 | ||||
|         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] = sample1; | ||||
|             outputi++; | ||||
| 
 | ||||
|             const s16 sample2 = decode_sample(SIGNED_NIBBLES[data[datai] & 0xF]); | ||||
|             ret[outputi] = sample2; | ||||
|             outputi++; | ||||
| 
 | ||||
|             datai++; | ||||
|         } | ||||
|     } | ||||
| 
 | ||||
|     state.yn1 = yn1; | ||||
|     state.yn2 = yn2; | ||||
| 
 | ||||
|     return ret; | ||||
| } | ||||
| 
 | ||||
| } // namespace AudioCore::Codec
 | ||||
							
								
								
									
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								src/audio_core/codec.h
									
									
									
									
									
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								src/audio_core/codec.h
									
									
									
									
									
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							| @ -0,0 +1,44 @@ | ||||
| // Copyright 2018 yuzu Emulator Project
 | ||||
| // Licensed under GPLv2 or any later version
 | ||||
| // Refer to the license.txt file included.
 | ||||
| 
 | ||||
| #pragma once | ||||
| 
 | ||||
| #include <array> | ||||
| #include <vector> | ||||
| 
 | ||||
| #include "common/common_types.h" | ||||
| 
 | ||||
| namespace AudioCore::Codec { | ||||
| 
 | ||||
| enum class PcmFormat : u32 { | ||||
|     Invalid = 0, | ||||
|     Int8 = 1, | ||||
|     Int16 = 2, | ||||
|     Int24 = 3, | ||||
|     Int32 = 4, | ||||
|     PcmFloat = 5, | ||||
|     Adpcm = 6, | ||||
| }; | ||||
| 
 | ||||
| /// 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]
 | ||||
| }; | ||||
| 
 | ||||
| using ADPCM_Coeff = std::array<s16, 16>; | ||||
| 
 | ||||
| /**
 | ||||
|  * @param data Pointer to buffer that contains ADPCM data to decode | ||||
|  * @param size Size of buffer in bytes | ||||
|  * @param coeff ADPCM coefficients | ||||
|  * @param state ADPCM state, this is updated with new state | ||||
|  * @return Decoded stereo signed PCM16 data, sample_count in length | ||||
|  */ | ||||
| std::vector<s16> DecodeADPCM(const u8* const data, size_t size, const ADPCM_Coeff& coeff, | ||||
|                              ADPCMState& state); | ||||
| 
 | ||||
| }; // namespace AudioCore::Codec
 | ||||
| @ -4,6 +4,8 @@ | ||||
| 
 | ||||
| #include <array> | ||||
| #include <vector> | ||||
| 
 | ||||
| #include "audio_core/codec.h" | ||||
| #include "common/logging/log.h" | ||||
| #include "core/core.h" | ||||
| #include "core/hle/ipc_helpers.h" | ||||
| @ -200,7 +202,7 @@ void AudOutU::OpenAudioOutImpl(Kernel::HLERequestContext& ctx) { | ||||
|     rb.Push(RESULT_SUCCESS); | ||||
|     rb.Push<u32>(DefaultSampleRate); | ||||
|     rb.Push<u32>(params.channel_count); | ||||
|     rb.Push<u32>(static_cast<u32>(PcmFormat::Int16)); | ||||
|     rb.Push<u32>(static_cast<u32>(AudioCore::Codec::PcmFormat::Int16)); | ||||
|     rb.Push<u32>(static_cast<u32>(AudioState::Stopped)); | ||||
|     rb.PushIpcInterface<Audio::IAudioOut>(audio_out_interface); | ||||
| } | ||||
|  | ||||
| @ -38,16 +38,6 @@ private: | ||||
| 
 | ||||
|     void ListAudioOutsImpl(Kernel::HLERequestContext& ctx); | ||||
|     void OpenAudioOutImpl(Kernel::HLERequestContext& ctx); | ||||
| 
 | ||||
|     enum class PcmFormat : u32 { | ||||
|         Invalid = 0, | ||||
|         Int8 = 1, | ||||
|         Int16 = 2, | ||||
|         Int24 = 3, | ||||
|         Int32 = 4, | ||||
|         PcmFloat = 5, | ||||
|         Adpcm = 6, | ||||
|     }; | ||||
| }; | ||||
| 
 | ||||
| } // namespace Service::Audio
 | ||||
|  | ||||
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