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
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
|
/**
* @file windgen.h
* @brief Templated wind noise generation
*
* $LicenseInfo:firstyear=2002&license=viewerlgpl$
* Second Life Viewer Source Code
* Copyright (C) 2010, Linden Research, Inc.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation;
* version 2.1 of the License only.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
* Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
* $/LicenseInfo$
*/
#ifndef WINDGEN_H
#define WINDGEN_H
#include "llcommon.h"
template <class MIXBUFFERFORMAT_T>
class LLWindGen
{
public:
LLWindGen(const U32 sample_rate = 44100) :
mTargetGain(0.f),
mTargetFreq(100.f),
mTargetPanGainR(0.5f),
mInputSamplingRate(sample_rate),
mSubSamples(2),
mFilterBandWidth(50.f),
mBuf0(0.0f),
mBuf1(0.0f),
mBuf2(0.0f),
mY0(0.0f),
mY1(0.0f),
mCurrentGain(0.f),
mCurrentFreq(100.f),
mCurrentPanGainR(0.5f),
mLastSample(0.f)
{
mSamplePeriod = (F32)mSubSamples / (F32)mInputSamplingRate;
mB2 = expf(-F_TWO_PI * mFilterBandWidth * mSamplePeriod);
}
const U32 getInputSamplingRate() { return mInputSamplingRate; }
// newbuffer = the buffer passed from the previous DSP unit.
// numsamples = length in samples-per-channel at this mix time.
// NOTE: generates L/R interleaved stereo
MIXBUFFERFORMAT_T* windGenerate(MIXBUFFERFORMAT_T *newbuffer, int numsamples)
{
MIXBUFFERFORMAT_T *cursamplep = newbuffer;
// Filter coefficients
F32 a0 = 0.0f, b1 = 0.0f;
// No need to clip at normal volumes
bool clip = mCurrentGain > 2.0f;
bool interp_freq = false;
//if the frequency isn't changing much, we don't need to interpolate in the inner loop
if (llabs(mTargetFreq - mCurrentFreq) < (mCurrentFreq * 0.112))
{
// calculate resonant filter coefficients
mCurrentFreq = mTargetFreq;
b1 = (-4.0f * mB2) / (1.0f + mB2) * cosf(F_TWO_PI * (mCurrentFreq * mSamplePeriod));
a0 = (1.0f - mB2) * sqrtf(1.0f - (b1 * b1) / (4.0f * mB2));
}
else
{
interp_freq = true;
}
while (numsamples)
{
F32 next_sample;
// Start with white noise
// This expression is fragile, rearrange it and it will break!
next_sample = (F32)rand() * (1.0f / (F32)(RAND_MAX / (U16_MAX / 8))) + (F32)(S16_MIN / 8);
// Apply a pinking filter
// Magic numbers taken from PKE method at http://www.firstpr.com.au/dsp/pink-noise/
mBuf0 = mBuf0 * 0.99765f + next_sample * 0.0990460f;
mBuf1 = mBuf1 * 0.96300f + next_sample * 0.2965164f;
mBuf2 = mBuf2 * 0.57000f + next_sample * 1.0526913f;
next_sample = mBuf0 + mBuf1 + mBuf2 + next_sample * 0.1848f;
if (interp_freq)
{
// calculate and interpolate resonant filter coefficients
mCurrentFreq = (0.999f * mCurrentFreq) + (0.001f * mTargetFreq);
b1 = (-4.0f * mB2) / (1.0f + mB2) * cosf(F_TWO_PI * (mCurrentFreq * mSamplePeriod));
a0 = (1.0f - mB2) * sqrtf(1.0f - (b1 * b1) / (4.0f * mB2));
}
// Apply a resonant low-pass filter on the pink noise
next_sample = a0 * next_sample - b1 * mY0 - mB2 * mY1;
mY1 = mY0;
mY0 = next_sample;
mCurrentGain = (0.999f * mCurrentGain) + (0.001f * mTargetGain);
mCurrentPanGainR = (0.999f * mCurrentPanGainR) + (0.001f * mTargetPanGainR);
// For a 3dB pan law use:
// next_sample *= mCurrentGain * ((mCurrentPanGainR*(mCurrentPanGainR-1)*1.652+1.413);
next_sample *= mCurrentGain;
// delta is used to interpolate between synthesized samples
F32 delta = (next_sample - mLastSample) / (F32)mSubSamples;
// Fill the audio buffer, clipping if necessary
for (U8 i=mSubSamples; i && numsamples; --i, --numsamples)
{
mLastSample = mLastSample + delta;
S32 sample_right = (S32)(mLastSample * mCurrentPanGainR);
S32 sample_left = (S32)mLastSample - sample_right;
if (!clip)
{
*cursamplep = (MIXBUFFERFORMAT_T)sample_left;
++cursamplep;
*cursamplep = (MIXBUFFERFORMAT_T)sample_right;
++cursamplep;
}
else
{
*cursamplep = (MIXBUFFERFORMAT_T)llclamp(sample_left, (S32)S16_MIN, (S32)S16_MAX);
++cursamplep;
*cursamplep = (MIXBUFFERFORMAT_T)llclamp(sample_right, (S32)S16_MIN, (S32)S16_MAX);
++cursamplep;
}
}
}
return newbuffer;
}
public:
F32 mTargetGain;
F32 mTargetFreq;
F32 mTargetPanGainR;
private:
U32 mInputSamplingRate;
U8 mSubSamples;
F32 mSamplePeriod;
F32 mFilterBandWidth;
F32 mB2;
F32 mBuf0;
F32 mBuf1;
F32 mBuf2;
F32 mY0;
F32 mY1;
F32 mCurrentGain;
F32 mCurrentFreq;
F32 mCurrentPanGainR;
F32 mLastSample;
};
#endif
|
