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
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
|
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include "audio_core/codec.h"
#include <array>
#include <cstddef>
#include <cstring>
#include <vector>
#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<s16, 16>& 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<int, 16> 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] >> 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;
}
static s16 SignExtendS8(u8 x) {
// The data is actually signed PCM8.
// We sign extend this to signed PCM16.
return static_cast<s16>(static_cast<s8>(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;
}
};
|