1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
|
// 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, std::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 std::size_t FRAME_LEN = 8;
constexpr std::size_t SAMPLES_PER_FRAME = 14;
static constexpr std::array<int, 16> SIGNED_NIBBLES{
0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1,
};
const std::size_t sample_count = (size / FRAME_LEN) * SAMPLES_PER_FRAME;
const std::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 std::size_t NUM_FRAMES =
(sample_count + (SAMPLES_PER_FRAME - 1)) / SAMPLES_PER_FRAME; // Round up.
for (std::size_t framei = 0; framei < NUM_FRAMES; framei++) {
const int frame_header = data[framei * FRAME_LEN];
const int scale = 1 << (frame_header & 0xF);
const auto idx = static_cast<size_t>((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);
};
std::size_t outputi = framei * SAMPLES_PER_FRAME;
std::size_t datai = framei * FRAME_LEN + 1;
for (std::size_t i = 0; i < SAMPLES_PER_FRAME && outputi < sample_count; i += 2) {
const s16 sample1 = decode_sample(SIGNED_NIBBLES[static_cast<u32>(data[datai] >> 4)]);
ret[outputi] = sample1;
outputi++;
const s16 sample2 = decode_sample(SIGNED_NIBBLES[static_cast<u32>(data[datai] & 0xF)]);
ret[outputi] = sample2;
outputi++;
datai++;
}
}
state.yn1 = static_cast<s16>(yn1);
state.yn2 = static_cast<s16>(yn2);
return ret;
}
} // namespace AudioCore::Codec
|
