#824 Process source files in bulk: replace tabs with spaces, convert CRLF to LF, and trim trailing whitespaces as needed

1 files changed, 134 insertions, 134 deletions

diff --git a/indra/llaudio/llwindgen.h b/indra/llaudio/llwindgen.h
index ec58f76f5f..57a14d902d 100644
--- a/indra/llaudio/llwindgen.h
+++ b/indra/llaudio/llwindgen.h
@@ -1,25 +1,25 @@
-/** 
+/**
  * @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
  * $/LicenseInfo$
  */
@@ -33,137 +33,137 @@ 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;
-	}
-	
+    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;
-	
+    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;
+    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); }