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/**
* @file windgen.h
* @brief Templated wind noise generation
*
* $LicenseInfo:firstyear=2002&license=viewergpl$
*
* Copyright (c) 2002-2009, Linden Research, Inc.
*
* Second Life Viewer Source Code
* The source code in this file ("Source Code") is provided by Linden Lab
* to you under the terms of the GNU General Public License, version 2.0
* ("GPL"), unless you have obtained a separate licensing agreement
* ("Other License"), formally executed by you and Linden Lab. Terms of
* the GPL can be found in doc/GPL-license.txt in this distribution, or
* online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
*
* There are special exceptions to the terms and conditions of the GPL as
* it is applied to this Source Code. View the full text of the exception
* in the file doc/FLOSS-exception.txt in this software distribution, or
* online at
* http://secondlifegrid.net/programs/open_source/licensing/flossexception
*
* By copying, modifying or distributing this software, you acknowledge
* that you have read and understood your obligations described above,
* and agree to abide by those obligations.
*
* ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
* WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
* COMPLETENESS OR PERFORMANCE.
* $/LicenseInfo$
*/
#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)
{
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
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