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-rw-r--r--indra/llaudio/llwindgen.h202
1 files changed, 81 insertions, 121 deletions
diff --git a/indra/llaudio/llwindgen.h b/indra/llaudio/llwindgen.h
index b9cecb60a1..847bfa6e9d 100644
--- a/indra/llaudio/llwindgen.h
+++ b/indra/llaudio/llwindgen.h
@@ -2,175 +2,135 @@
* @file windgen.h
* @brief Templated wind noise generation
*
- * $LicenseInfo:firstyear=2002&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
+ * $LicenseInfo:firstyear=2002&license=viewergpl$
+ *
+ * Copyright (c) 2002-2009, 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.
+ * 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
*
- * 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.
+ * 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
*
- * 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
+ * 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.
*
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * 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"
+#include "llrand.h"
template <class MIXBUFFERFORMAT_T>
class LLWindGen
{
public:
- LLWindGen(const U32 sample_rate = 44100) :
+ LLWindGen() :
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),
+ mbuf0(0.0),
+ mbuf1(0.0),
+ mbuf2(0.0),
+ mbuf3(0.0),
+ mbuf4(0.0),
+ mbuf5(0.0),
+ mY0(0.0),
+ mY1(0.0),
mCurrentGain(0.f),
mCurrentFreq(100.f),
- mCurrentPanGainR(0.5f),
- mLastSample(0.f)
- {
- mSamplePeriod = (F32)mSubSamples / (F32)mInputSamplingRate;
- mB2 = expf(-F_TWO_PI * mFilterBandWidth * mSamplePeriod);
- }
+ mCurrentPanGainR(0.5f) {};
+
+ static const U32 getInputSamplingRate() {return mInputSamplingRate;}
- const U32 getInputSamplingRate() { return mInputSamplingRate; }
-
// newbuffer = the buffer passed from the previous DSP unit.
// numsamples = length in samples-per-channel at this mix time.
+ // stride = number of bytes between start of each sample.
// NOTE: generates L/R interleaved stereo
- MIXBUFFERFORMAT_T* windGenerate(MIXBUFFERFORMAT_T *newbuffer, int numsamples)
+ MIXBUFFERFORMAT_T* windGenerate(MIXBUFFERFORMAT_T *newbuffer, int numsamples, int stride)
{
- MIXBUFFERFORMAT_T *cursamplep = newbuffer;
-
- // Filter coefficients
- F32 a0 = 0.0f, b1 = 0.0f;
+ U8 *cursamplep = (U8*)newbuffer;
- // No need to clip at normal volumes
- bool clip = mCurrentGain > 2.0f;
+ double bandwidth = 50.0F;
+ double a0,b1,b2;
- 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;
- }
+ // calculate resonant filter coeffs
+ b2 = exp(-(F_TWO_PI) * (bandwidth / mInputSamplingRate));
- while (numsamples)
+ while (numsamples--)
{
- F32 next_sample;
+ mCurrentFreq = (float)((0.999 * mCurrentFreq) + (0.001 * mTargetFreq));
+ mCurrentGain = (float)((0.999 * mCurrentGain) + (0.001 * mTargetGain));
+ mCurrentPanGainR = (float)((0.999 * mCurrentPanGainR) + (0.001 * mTargetPanGainR));
+ b1 = (-4.0 * b2) / (1.0 + b2) * cos(F_TWO_PI * (mCurrentFreq / mInputSamplingRate));
+ a0 = (1.0 - b2) * sqrt(1.0 - (b1 * b1) / (4.0 * b2));
+ double nextSample;
- // 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);
+ // start with white noise
+ nextSample = ll_frand(2.0f) - 1.0f;
- // 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;
+ // apply pinking filter
+ mbuf0 = 0.997f * mbuf0 + 0.0126502f * nextSample;
+ mbuf1 = 0.985f * mbuf1 + 0.0139083f * nextSample;
+ mbuf2 = 0.950f * mbuf2 + 0.0205439f * nextSample;
+ mbuf3 = 0.850f * mbuf3 + 0.0387225f * nextSample;
+ mbuf4 = 0.620f * mbuf4 + 0.0465932f * nextSample;
+ mbuf5 = 0.250f * mbuf5 + 0.1093477f * nextSample;
- next_sample = mBuf0 + mBuf1 + mBuf2 + next_sample * 0.1848f;
+ nextSample = mbuf0 + mbuf1 + mbuf2 + mbuf3 + mbuf4 + mbuf5;
- 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;
+ // do a resonant filter on the noise
+ nextSample = (double)( a0 * nextSample - b1 * mY0 - b2 * mY1 );
mY1 = mY0;
- mY0 = next_sample;
-
- mCurrentGain = (0.999f * mCurrentGain) + (0.001f * mTargetGain);
- mCurrentPanGainR = (0.999f * mCurrentPanGainR) + (0.001f * mTargetPanGainR);
+ mY0 = nextSample;
- // For a 3dB pan law use:
- // next_sample *= mCurrentGain * ((mCurrentPanGainR*(mCurrentPanGainR-1)*1.652+1.413);
- next_sample *= mCurrentGain;
+ nextSample *= mCurrentGain;
- // delta is used to interpolate between synthesized samples
- F32 delta = (next_sample - mLastSample) / (F32)mSubSamples;
+ MIXBUFFERFORMAT_T sample;
- // 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;
- }
- }
+ sample = llfloor(((F32)nextSample*32768.f*(1.0f - mCurrentPanGainR))+0.5f);
+ *(MIXBUFFERFORMAT_T*)cursamplep = llclamp(sample, (MIXBUFFERFORMAT_T)-32768, (MIXBUFFERFORMAT_T)32767);
+ cursamplep += stride;
+
+ sample = llfloor(((F32)nextSample*32768.f*mCurrentPanGainR)+0.5f);
+ *(MIXBUFFERFORMAT_T*)cursamplep = llclamp(sample, (MIXBUFFERFORMAT_T)-32768, (MIXBUFFERFORMAT_T)32767);
+ cursamplep += stride;
}
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;
-
+ static const U32 mInputSamplingRate = 44100;
+ F64 mbuf0;
+ F64 mbuf1;
+ F64 mbuf2;
+ F64 mbuf3;
+ F64 mbuf4;
+ F64 mbuf5;
+ F64 mY0;
+ F64 mY1;
F32 mCurrentGain;
F32 mCurrentFreq;
F32 mCurrentPanGainR;
- F32 mLastSample;
};
#endif
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