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/**
* @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"
#include "llrand.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; }
const F32 getNextSample();
const F32 getClampedSample(bool clamp, F32 sample);
// 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 = getNextSample();
// 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;
MIXBUFFERFORMAT_T sample_right = (MIXBUFFERFORMAT_T)getClampedSample(clip, mLastSample * mCurrentPanGainR);
MIXBUFFERFORMAT_T sample_left = (MIXBUFFERFORMAT_T)getClampedSample(clip, mLastSample - (F32)sample_right);
*cursamplep = sample_left;
++cursamplep;
*cursamplep = sample_right;
++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;
};
template<class T> inline const F32 LLWindGen<T>::getNextSample() { return (F32)rand() * (1.0f / (F32)(RAND_MAX / (U16_MAX / 8))) + (F32)(S16_MIN / 8); }
template<> inline const F32 LLWindGen<F32>::getNextSample() { return ll_frand()-.5f; }
template<class T> inline const F32 LLWindGen<T>::getClampedSample(bool clamp, F32 sample) { return clamp ? (F32)llclamp((S32)sample,(S32)S16_MIN,(S32)S16_MAX) : sample; }
template<> inline const F32 LLWindGen<F32>::getClampedSample(bool clamp, F32 sample) { return sample; }
#endif
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