Clean up whitespace in qa/ in preparation for .editorconfig. (#418)

* Clean up whitespace in qa/ in preparation for .editorconfig. Convert tabs to 4 spaces. Indent by 4 spaces. Strip trailing whitespace. Ensure EOL at EOF.

qa/loopback/src/audio_analyzer.h

@@ -26,13 +26,13 @@
  */
 
 /*
- * The text above constitutes the entire PortAudio license; however, 
+ * The text above constitutes the entire PortAudio license; however,
  * the PortAudio community also makes the following non-binding requests:
  *
  * Any person wishing to distribute modifications to the Software is
  * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also 
- * requested that these non-binding requests be included along with the 
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
  * license above.
  */
 
@@ -46,44 +46,44 @@
 
 typedef struct PaQaSineGenerator_s
 {
-	double    phase;
-	double    phaseIncrement;
-	double    frequency;
-	double    amplitude;
+    double    phase;
+    double    phaseIncrement;
+    double    frequency;
+    double    amplitude;
 } PaQaSineGenerator;
 
-/** Container for a monophonic audio sample in memory. */ 
+/** Container for a monophonic audio sample in memory. */
 typedef struct PaQaRecording_s
 {
-	/** Maximum number of frames that can fit in the allocated buffer. */
-	int       maxFrames;
-	float    *buffer;
-	/** Actual number of valid frames in the buffer. */
-	int       numFrames;
-	int       sampleRate;
+    /** Maximum number of frames that can fit in the allocated buffer. */
+    int       maxFrames;
+    float    *buffer;
+    /** Actual number of valid frames in the buffer. */
+    int       numFrames;
+    int       sampleRate;
 } PaQaRecording;
 
 typedef struct PaQaTestTone_s
 {
-	int       samplesPerFrame;
-	int       startDelay;
-	double    sampleRate;
-	double    frequency;
-	double    amplitude;
+    int       samplesPerFrame;
+    int       startDelay;
+    double    sampleRate;
+    double    frequency;
+    double    amplitude;
 } PaQaTestTone;
 
 typedef struct PaQaAnalysisResult_s
 {
-	int       valid;
-	/** Latency in samples from output to input. */
-	double    latency;
-	double    amplitudeRatio;
-	double    popAmplitude;
-	double    popPosition;
-	double    numDroppedFrames;
-	double    droppedFramesPosition;
-	double    numAddedFrames;
-	double    addedFramesPosition;
+    int       valid;
+    /** Latency in samples from output to input. */
+    double    latency;
+    double    amplitudeRatio;
+    double    popAmplitude;
+    double    popPosition;
+    double    numDroppedFrames;
+    double    droppedFramesPosition;
+    double    numAddedFrames;
+    double    addedFramesPosition;
 } PaQaAnalysisResult;
 
 
@@ -101,7 +101,7 @@ void PaQa_EraseBuffer( float *buffer, int numFrames, int samplesPerFrame );
 void PaQa_MixSine( PaQaSineGenerator *generator, float *buffer, int numSamples, int stride );
 
 void PaQa_WriteSine( float *buffer, int numSamples, int stride,
-					double frequency, double amplitude );
+                     double frequency, double amplitude );
 
 /**
  * Generate a signal with a sharp edge in the middle that can be recognized despite some phase shift.
@@ -133,12 +133,12 @@ void PaQa_SetupSineGenerator( PaQaSineGenerator *generator, double frequency, do
  * Allocate memory for containing a mono audio signal. Set up recording for writing.
  */
  int PaQa_InitializeRecording( PaQaRecording *recording, int maxSamples, int sampleRate );
- 
+
 /**
 * Free memory allocated by PaQa_InitializeRecording.
  */
  void PaQa_TerminateRecording( PaQaRecording *recording );
- 
+
 /**
  * Apply a biquad filter to the audio from the input recording and write it to the output recording.
  */
@@ -146,29 +146,29 @@ void PaQa_FilterRecording( PaQaRecording *input, PaQaRecording *output, BiquadFi
 
 
 int PaQa_SaveRecordingToWaveFile( PaQaRecording *recording, const char *filename );
- 
+
 /**
  * @param stride is the spacing of samples to skip in the input buffer. To use every samples pass 1. To use every other sample pass 2.
  */
 int PaQa_WriteRecording( PaQaRecording *recording, float *buffer, int numSamples, int stride );
- 
+
 /** Write zeros into a recording. */
 int PaQa_WriteSilence( PaQaRecording *recording, int numSamples );
- 
+
 int PaQa_RecordFreeze( PaQaRecording *recording, int numSamples );
 
 double PaQa_CorrelateSine( PaQaRecording *recording, double frequency, double frameRate,
-						  int startFrame, int numSamples, double *phasePtr );
+                           int startFrame, int numSamples, double *phasePtr );
 
 double PaQa_FindFirstMatch( PaQaRecording *recording, float *buffer, int numSamples, double tolerance  );
 
-/** 
+/**
  * Estimate the original amplitude of a clipped sine wave by measuring
  * its average slope at the zero crossings.
  */
 double PaQa_MeasureSineAmplitudeBySlope( PaQaRecording *recording,
-										double frequency, double frameRate,
-										int startFrame, int numFrames );
+                                         double frequency, double frameRate,
+                                         int startFrame, int numFrames );
 
 double PaQa_MeasureRootMeanSquare( float *buffer, int numFrames );
 

qa/loopback/src/biquad_filter.c

@@ -4,9 +4,10 @@
 #include "biquad_filter.h"
 
 /**
- *	Unit_BiquadFilter implements a second order IIR filter. 
-	Here is the equation that we use for this filter:
-     y(n) = a0*x(n) + a1*x(n-1)  + a2*x(n-2) - b1*y(n-1)  - b2*y(n-2)
+ *  Unit_BiquadFilter implements a second order IIR filter.
+ *
+ *  Here is the equation that we use for this filter:
+ *      y(n) = a0*x(n) + a1*x(n-1)  + a2*x(n-2) - b1*y(n-1)  - b2*y(n-2)
  *
  * @author (C) 2002 Phil Burk, SoftSynth.com, All Rights Reserved
  */
@@ -17,17 +18,17 @@
 */
 static void BiquadFilter_CalculateCommon( BiquadFilter *filter, double ratio, double Q )
 {
-	double omega;
-	
-	memset( filter, 0, sizeof(BiquadFilter) );
+    double omega;
+
+    memset( filter, 0, sizeof(BiquadFilter) );
 
 /* Don't let frequency get too close to Nyquist or filter will blow up. */
-	if( ratio >= 0.499 ) ratio = 0.499; 
-	omega = 2.0 * (double)FILTER_PI * ratio;
+    if( ratio >= 0.499 ) ratio = 0.499;
+    omega = 2.0 * (double)FILTER_PI * ratio;
 
-	filter->cos_omega = (double) cos( omega );
-	filter->sin_omega = (double) sin( omega );
-	filter->alpha = filter->sin_omega / (2.0 * Q);
+    filter->cos_omega = (double) cos( omega );
+    filter->sin_omega = (double) sin( omega );
+    filter->alpha = filter->sin_omega / (2.0 * Q);
 }
 
 /*********************************************************************************
@@ -35,21 +36,21 @@ static void BiquadFilter_CalculateCommon( BiquadFilter *filter, double ratio, do
  */
 void BiquadFilter_SetupHighPass( BiquadFilter *filter, double ratio, double Q )
 {
-	double    scalar, opc;
-	
-	if( ratio  < BIQUAD_MIN_RATIO )  ratio  = BIQUAD_MIN_RATIO;
-	if( Q < BIQUAD_MIN_Q ) Q = BIQUAD_MIN_Q;
-	
-	BiquadFilter_CalculateCommon( filter, ratio, Q );
-	
-	scalar = 1.0 / (1.0 + filter->alpha);
-	opc = (1.0 + filter->cos_omega);
-	
-	filter->a0 = opc * 0.5 * scalar;
-	filter->a1 =  - opc * scalar;
+    double    scalar, opc;
+
+    if( ratio  < BIQUAD_MIN_RATIO )  ratio  = BIQUAD_MIN_RATIO;
+    if( Q < BIQUAD_MIN_Q ) Q = BIQUAD_MIN_Q;
+
+    BiquadFilter_CalculateCommon( filter, ratio, Q );
+
+    scalar = 1.0 / (1.0 + filter->alpha);
+    opc = (1.0 + filter->cos_omega);
+
+    filter->a0 = opc * 0.5 * scalar;
+    filter->a1 =  - opc * scalar;
     filter->a2 = filter->a0;
-	filter->b1 = -2.0 * filter->cos_omega * scalar;
-	filter->b2 = (1.0 - filter->alpha) * scalar;
+    filter->b1 = -2.0 * filter->cos_omega * scalar;
+    filter->b2 = (1.0 - filter->alpha) * scalar;
 }
 
 
@@ -58,21 +59,21 @@ void BiquadFilter_SetupHighPass( BiquadFilter *filter, double ratio, double Q )
  */
 void BiquadFilter_SetupNotch( BiquadFilter *filter, double ratio, double Q )
 {
-	double    scalar, opc;
-	
-	if( ratio  < BIQUAD_MIN_RATIO )  ratio  = BIQUAD_MIN_RATIO;
-	if( Q < BIQUAD_MIN_Q ) Q = BIQUAD_MIN_Q;
-	
-	BiquadFilter_CalculateCommon( filter, ratio, Q );
-	
-	scalar = 1.0 / (1.0 + filter->alpha);
-	opc = (1.0 + filter->cos_omega);
-	
-	filter->a0 = scalar;
-	filter->a1 =  -2.0 * filter->cos_omega * scalar;
+    double    scalar, opc;
+
+    if( ratio  < BIQUAD_MIN_RATIO )  ratio  = BIQUAD_MIN_RATIO;
+    if( Q < BIQUAD_MIN_Q ) Q = BIQUAD_MIN_Q;
+
+    BiquadFilter_CalculateCommon( filter, ratio, Q );
+
+    scalar = 1.0 / (1.0 + filter->alpha);
+    opc = (1.0 + filter->cos_omega);
+
+    filter->a0 = scalar;
+    filter->a1 =  -2.0 * filter->cos_omega * scalar;
     filter->a2 = filter->a0;
-	filter->b1 = filter->a1;
-	filter->b2 = (1.0 - filter->alpha) * scalar;
+    filter->b1 = filter->a1;
+    filter->b2 = (1.0 - filter->alpha) * scalar;
 }
 
 /*****************************************************************
@@ -80,43 +81,43 @@ void BiquadFilter_SetupNotch( BiquadFilter *filter, double ratio, double Q )
 */
 void BiquadFilter_Filter( BiquadFilter *filter, float *inputs, float *outputs, int numSamples )
 {
-	int i;
+    int i;
     double xn, yn;
-	// Pull values from structure to speed up the calculation.
-	double a0 = filter->a0;
-	double a1 = filter->a1;
-	double a2 = filter->a2;
-	double b1 = filter->b1;
-	double b2 = filter->b2;
-	double xn1 = filter->xn1;
-	double xn2 = filter->xn2;
-	double yn1 = filter->yn1;
-	double yn2 = filter->yn2;
+    // Pull values from structure to speed up the calculation.
+    double a0 = filter->a0;
+    double a1 = filter->a1;
+    double a2 = filter->a2;
+    double b1 = filter->b1;
+    double b2 = filter->b2;
+    double xn1 = filter->xn1;
+    double xn2 = filter->xn2;
+    double yn1 = filter->yn1;
+    double yn2 = filter->yn2;
 
-	for( i=0; i<numSamples; i++)
-	{
-		// Generate outputs by filtering inputs.
-		xn = inputs[i];
-		yn = (a0 * xn) + (a1 * xn1) + (a2 * xn2) - (b1 * yn1) - (b2 * yn2);
-		outputs[i] = yn;
+    for( i=0; i<numSamples; i++)
+    {
+        // Generate outputs by filtering inputs.
+        xn = inputs[i];
+        yn = (a0 * xn) + (a1 * xn1) + (a2 * xn2) - (b1 * yn1) - (b2 * yn2);
+        outputs[i] = yn;
 
-		// Delay input and output values.
+        // Delay input and output values.
         xn2 = xn1;
         xn1 = xn;
         yn2 = yn1;
         yn1 = yn;
-		
-		if( (i & 7) == 0 )
-		{
-			// Apply a small bipolar impulse to filter to prevent arithmetic underflow.
-			// Underflows can cause the FPU to interrupt the CPU.
-			yn1 += (double) 1.0E-26;
-			yn2 -= (double) 1.0E-26;
-		}
-	}
-	
-	filter->xn1 = xn1;
-	filter->xn2 = xn2;
-	filter->yn1 = yn1;
-	filter->yn2 = yn2;		
+
+        if( (i & 7) == 0 )
+        {
+            // Apply a small bipolar impulse to filter to prevent arithmetic underflow.
+            // Underflows can cause the FPU to interrupt the CPU.
+            yn1 += (double) 1.0E-26;
+            yn2 -= (double) 1.0E-26;
+        }
+    }
+
+    filter->xn1 = xn1;
+    filter->xn2 = xn2;
+    filter->yn1 = yn1;
+    filter->yn2 = yn2;
 }

qa/loopback/src/biquad_filter.h

@@ -3,7 +3,7 @@
 
 
 /**
- *	Unit_BiquadFilter implements a second order IIR filter. 
+ * Unit_BiquadFilter implements a second order IIR filter.
  *
  * @author (C) 2002 Phil Burk, SoftSynth.com, All Rights Reserved
  */
@@ -14,20 +14,20 @@
 typedef struct BiquadFilter_s
 {
     double      xn1;    // storage for delayed signals
-	double      xn2;
-	double      yn1;
-	double      yn2;
+    double      xn2;
+    double      yn1;
+    double      yn2;
 
-	double      a0;    // coefficients
-	double      a1;
-	double      a2;
+    double      a0;     // coefficients
+    double      a1;
+    double      a2;
 
-	double      b1;
-	double      b2;
+    double      b1;
+    double      b2;
 
-	double      cos_omega;
-	double      sin_omega;
-	double      alpha;
+    double      cos_omega;
+    double      sin_omega;
+    double      alpha;
 } BiquadFilter;
 
 void BiquadFilter_SetupHighPass( BiquadFilter *filter, double ratio, double Q );

qa/loopback/src/paqa_tools.c

@@ -26,13 +26,13 @@
  */
 
 /*
- * The text above constitutes the entire PortAudio license; however, 
+ * The text above constitutes the entire PortAudio license; however,
  * the PortAudio community also makes the following non-binding requests:
  *
  * Any person wishing to distribute modifications to the Software is
  * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also 
- * requested that these non-binding requests be included along with the 
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
  * license above.
  */
 
@@ -43,129 +43,129 @@
 void PaQa_ListAudioDevices(void)
 {
     int     i, numDevices;
-    const   PaDeviceInfo *deviceInfo;		
-    numDevices = Pa_GetDeviceCount();    
+    const   PaDeviceInfo *deviceInfo;
+    numDevices = Pa_GetDeviceCount();
     for( i=0; i<numDevices; i++ )
     {
         deviceInfo = Pa_GetDeviceInfo( i );
         printf( "#%d: ", i );
         printf( "%2d in", deviceInfo->maxInputChannels  );
         printf( ", %2d out", deviceInfo->maxOutputChannels  );
-		printf( ",  %s", deviceInfo->name );
+        printf( ",  %s", deviceInfo->name );
         printf( ", on %s\n",  Pa_GetHostApiInfo( deviceInfo->hostApi )->name );
-	}
+    }
 }
 
 /*******************************************************************/
 void PaQa_ConvertToFloat( const void *input, int numSamples, PaSampleFormat inFormat, float *output )
 {
-	int i;
-	switch( inFormat )
-	{
-		case paUInt8:
-		{
-			unsigned char *data = (unsigned char *)input;
-			for( i=0; i<numSamples; i++ )
-			{
-				int value = *data++;
-				value -= 128;
-				*output++ = value / 128.0f;
-			}
-		}
-			break;
-			
-		case paInt8:
-		{
-			char *data = (char *)input;
-			for( i=0; i<numSamples; i++ )
-			{
-				int value = *data++;
-				*output++ = value / 128.0f;
-			}
-		}
-			break;
-			
-		case paInt16:
-		{
-			short *data = (short *)input;
-			for( i=0; i<numSamples; i++ )
-			{
-				*output++ = *data++ / 32768.0f;
-			}
-		}
-			break;
-			
-		case paInt32:
-		{
-			int *data = (int *)input;
-			for( i=0; i<numSamples; i++ )
-			{
-				int value = (*data++) >> 8;
-				float fval = (float) (value / ((double) 0x00800000));
-				*output++ = fval;
-			}
-		}
-			break;
-	}
-	
+    int i;
+    switch( inFormat )
+    {
+        case paUInt8:
+        {
+            unsigned char *data = (unsigned char *)input;
+            for( i=0; i<numSamples; i++ )
+            {
+                int value = *data++;
+                value -= 128;
+                *output++ = value / 128.0f;
+            }
+        }
+            break;
+
+        case paInt8:
+        {
+            char *data = (char *)input;
+            for( i=0; i<numSamples; i++ )
+            {
+                int value = *data++;
+                *output++ = value / 128.0f;
+            }
+        }
+            break;
+
+        case paInt16:
+        {
+            short *data = (short *)input;
+            for( i=0; i<numSamples; i++ )
+            {
+                *output++ = *data++ / 32768.0f;
+            }
+        }
+            break;
+
+        case paInt32:
+        {
+            int *data = (int *)input;
+            for( i=0; i<numSamples; i++ )
+            {
+                int value = (*data++) >> 8;
+                float fval = (float) (value / ((double) 0x00800000));
+                *output++ = fval;
+            }
+        }
+            break;
+    }
+
 }
 
 /*******************************************************************/
 void PaQa_ConvertFromFloat( const float *input, int numSamples, PaSampleFormat outFormat, void *output )
 {
-	int i;
-	switch( outFormat )
-	{
-		case paUInt8:
-		{
-			unsigned char *data = (unsigned char *)output;
-			for( i=0; i<numSamples; i++ )
-			{
-				float value = *input++;
-				int byte = ((int) (value * 127)) + 128;
-				*data++ = (unsigned char) byte;
-			}
-		}
-			break;
-			
-		case paInt8:
-		{
-			char *data = (char *)output;
-			for( i=0; i<numSamples; i++ )
-			{
-				float value = *input++;
-				int byte = (int) (value * 127);
-				*data++ = (char) byte;
-			}
-		}
-			break;
-			
-		case paInt16:
-		{
-			short *data = (short *)output;
-			for( i=0; i<numSamples; i++ )
-			{
-				float value = *input++;
-				// Use asymmetric conversion to avoid clipping.
-				short sval = value * 32767.0;
-				*data++ = sval;
-			}
-		}
-			break;
-			
-		case paInt32:
-		{
-			int *data = (int *)output;
-			for( i=0; i<numSamples; i++ )
-			{
-				float value = *input++;
-				// Use asymmetric conversion to avoid clipping.
-				int ival = value * ((double) 0x007FFFF0);
-				ival = ival << 8;
-				*data++ = ival;
-			}
-		}
-			break;
-	}
-	
+    int i;
+    switch( outFormat )
+    {
+        case paUInt8:
+        {
+            unsigned char *data = (unsigned char *)output;
+            for( i=0; i<numSamples; i++ )
+            {
+                float value = *input++;
+                int byte = ((int) (value * 127)) + 128;
+                *data++ = (unsigned char) byte;
+            }
+        }
+            break;
+
+        case paInt8:
+        {
+            char *data = (char *)output;
+            for( i=0; i<numSamples; i++ )
+            {
+                float value = *input++;
+                int byte = (int) (value * 127);
+                *data++ = (char) byte;
+            }
+        }
+            break;
+
+        case paInt16:
+        {
+            short *data = (short *)output;
+            for( i=0; i<numSamples; i++ )
+            {
+                float value = *input++;
+                // Use asymmetric conversion to avoid clipping.
+                short sval = value * 32767.0;
+                *data++ = sval;
+            }
+        }
+            break;
+
+        case paInt32:
+        {
+            int *data = (int *)output;
+            for( i=0; i<numSamples; i++ )
+            {
+                float value = *input++;
+                // Use asymmetric conversion to avoid clipping.
+                int ival = value * ((double) 0x007FFFF0);
+                ival = ival << 8;
+                *data++ = ival;
+            }
+        }
+            break;
+    }
+
 }

qa/loopback/src/paqa_tools.h

@@ -26,13 +26,13 @@
  */
 
 /*
- * The text above constitutes the entire PortAudio license; however, 
+ * The text above constitutes the entire PortAudio license; however,
  * the PortAudio community also makes the following non-binding requests:
  *
  * Any person wishing to distribute modifications to the Software is
  * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also 
- * requested that these non-binding requests be included along with the 
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
  * license above.
  */
 

qa/loopback/src/qa_tools.h

@@ -26,13 +26,13 @@
  */
 
 /*
- * The text above constitutes the entire PortAudio license; however, 
+ * The text above constitutes the entire PortAudio license; however,
  * the PortAudio community also makes the following non-binding requests:
  *
  * Any person wishing to distribute modifications to the Software is
  * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also 
- * requested that these non-binding requests be included along with the 
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
  * license above.
  */
 
@@ -43,32 +43,32 @@ extern int g_testsPassed;
 extern int g_testsFailed;
 
 #define QA_ASSERT_TRUE( message, flag ) \
-	if( !(flag) ) \
-	{ \
-		printf( "%s:%d - ERROR - %s\n", __FILE__, __LINE__, message ); \
-		g_testsFailed++; \
-		goto error; \
-	} \
-	else g_testsPassed++;
+    if( !(flag) ) \
+    { \
+        printf( "%s:%d - ERROR - %s\n", __FILE__, __LINE__, message ); \
+        g_testsFailed++; \
+        goto error; \
+    } \
+    else g_testsPassed++;
 
 
 #define QA_ASSERT_EQUALS( message, expected, actual ) \
-	if( ((expected) != (actual)) ) \
-	{ \
-		printf( "%s:%d - ERROR - %s, expected %d, got %d\n", __FILE__, __LINE__, message, expected, actual ); \
-		g_testsFailed++; \
-		goto error; \
-	} \
-	else g_testsPassed++;
+    if( ((expected) != (actual)) ) \
+    { \
+        printf( "%s:%d - ERROR - %s, expected %d, got %d\n", __FILE__, __LINE__, message, expected, actual ); \
+        g_testsFailed++; \
+        goto error; \
+    } \
+    else g_testsPassed++;
 
 #define QA_ASSERT_CLOSE( message, expected, actual, tolerance ) \
-	if (fabs((expected)-(actual))>(tolerance)) \
-	{ \
-		printf( "%s:%d - ERROR - %s, expected %f, got %f, tol=%f\n", __FILE__, __LINE__, message, ((double)(expected)), ((double)(actual)), ((double)(tolerance)) ); \
-		g_testsFailed++; \
-		goto error; \
-	} \
-	else g_testsPassed++;
+    if (fabs((expected)-(actual))>(tolerance)) \
+    { \
+        printf( "%s:%d - ERROR - %s, expected %f, got %f, tol=%f\n", __FILE__, __LINE__, message, ((double)(expected)), ((double)(actual)), ((double)(tolerance)) ); \
+        g_testsFailed++; \
+        goto error; \
+    } \
+    else g_testsPassed++;
 
 #define QA_ASSERT_CLOSE_INT( message, expected, actual, tolerance ) \
     if (abs((expected)-(actual))>(tolerance)) \

qa/loopback/src/test_audio_analyzer.h

@@ -26,13 +26,13 @@
  */
 
 /*
- * The text above constitutes the entire PortAudio license; however, 
+ * The text above constitutes the entire PortAudio license; however,
  * the PortAudio community also makes the following non-binding requests:
  *
  * Any person wishing to distribute modifications to the Software is
  * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also 
- * requested that these non-binding requests be included along with the 
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
  * license above.
  */
 

qa/loopback/src/write_wav.c

@@ -25,13 +25,13 @@
  */
 
 /*
- * The text above constitutes the entire PortAudio license; however, 
+ * The text above constitutes the entire PortAudio license; however,
  * the PortAudio community also makes the following non-binding requests:
  *
  * Any person wishing to distribute modifications to the Software is
  * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also 
- * requested that these non-binding requests be included along with the 
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
  * license above.
  */
 
@@ -47,32 +47,32 @@
 /* Write long word data to a little endian format byte array. */
 static void WriteLongLE( unsigned char **addrPtr, unsigned long data )
 {
-	unsigned char *addr = *addrPtr;
-	*addr++ =  (unsigned char) data;
-	*addr++ =  (unsigned char) (data>>8);
-	*addr++ =  (unsigned char) (data>>16);
-	*addr++ =  (unsigned char) (data>>24);
-	*addrPtr = addr;
+    unsigned char *addr = *addrPtr;
+    *addr++ =  (unsigned char) data;
+    *addr++ =  (unsigned char) (data>>8);
+    *addr++ =  (unsigned char) (data>>16);
+    *addr++ =  (unsigned char) (data>>24);
+    *addrPtr = addr;
 }
 
 /* Write short word data to a little endian format byte array. */
 static void WriteShortLE( unsigned char **addrPtr,  unsigned short data )
 {
-	unsigned char *addr = *addrPtr;
-	*addr++ =  (unsigned char) data;
-	*addr++ =  (unsigned char) (data>>8);
-	*addrPtr = addr;
+    unsigned char *addr = *addrPtr;
+    *addr++ =  (unsigned char) data;
+    *addr++ =  (unsigned char) (data>>8);
+    *addrPtr = addr;
 }
 
 /* Write IFF ChunkType data to a byte array. */
 static void WriteChunkType( unsigned char **addrPtr, unsigned long cktyp )
 {
-	unsigned char *addr = *addrPtr;
-	*addr++ =  (unsigned char) (cktyp>>24);
-	*addr++ =  (unsigned char) (cktyp>>16);
-	*addr++ =  (unsigned char) (cktyp>>8);
-	*addr++ =  (unsigned char) cktyp;
-	*addrPtr = addr;
+    unsigned char *addr = *addrPtr;
+    *addr++ =  (unsigned char) (cktyp>>24);
+    *addr++ =  (unsigned char) (cktyp>>16);
+    *addr++ =  (unsigned char) (cktyp>>8);
+    *addr++ =  (unsigned char) cktyp;
+    *addrPtr = addr;
 }
 
 #define WAV_HEADER_SIZE (4 + 4 + 4 + /* RIFF+size+WAVE */ \
@@ -87,14 +87,14 @@ static void WriteChunkType( unsigned char **addrPtr, unsigned long cktyp )
  */
 long Audio_WAV_OpenWriter( WAV_Writer *writer, const char *fileName, int frameRate, int samplesPerFrame )
 {
-	unsigned int  bytesPerSecond;
+    unsigned int  bytesPerSecond;
     unsigned char header[ WAV_HEADER_SIZE ];
-	unsigned char *addr = header;
+    unsigned char *addr = header;
     int numWritten;
-	
+
     writer->dataSize = 0;
     writer->dataSizeOffset = 0;
-	
+
     writer->fid = fopen( fileName, "wb" );
     if( writer->fid == NULL )
     {
@@ -102,35 +102,35 @@ long Audio_WAV_OpenWriter( WAV_Writer *writer, const char *fileName, int frameRa
     }
 
 /* Write RIFF header. */
-	WriteChunkType( &addr, RIFF_ID );
+    WriteChunkType( &addr, RIFF_ID );
 
 /* Write RIFF size as zero for now. Will patch later. */
-	WriteLongLE( &addr, 0 );
+    WriteLongLE( &addr, 0 );
 
 /* Write WAVE form ID. */
-	WriteChunkType( &addr, WAVE_ID );
+    WriteChunkType( &addr, WAVE_ID );
 
 /* Write format chunk based on AudioSample structure. */
-	WriteChunkType( &addr, FMT_ID );
+    WriteChunkType( &addr, FMT_ID );
     WriteLongLE( &addr, 16 );
     WriteShortLE( &addr, WAVE_FORMAT_PCM );
-		bytesPerSecond = frameRate * samplesPerFrame * sizeof( short);
-	WriteShortLE( &addr, (short) samplesPerFrame );
-	WriteLongLE( &addr, frameRate );
-	WriteLongLE( &addr,  bytesPerSecond );
-	WriteShortLE( &addr, (short) (samplesPerFrame * sizeof( short)) ); /* bytesPerBlock */
-	WriteShortLE( &addr, (short) 16 ); /* bits per sample */
+        bytesPerSecond = frameRate * samplesPerFrame * sizeof( short);
+    WriteShortLE( &addr, (short) samplesPerFrame );
+    WriteLongLE( &addr, frameRate );
+    WriteLongLE( &addr,  bytesPerSecond );
+    WriteShortLE( &addr, (short) (samplesPerFrame * sizeof( short)) ); /* bytesPerBlock */
+    WriteShortLE( &addr, (short) 16 ); /* bits per sample */
 
 /* Write ID and size for 'data' chunk. */
-	WriteChunkType( &addr, DATA_ID );
+    WriteChunkType( &addr, DATA_ID );
 /* Save offset so we can patch it later. */
     writer->dataSizeOffset = (int) (addr - header);
-	WriteLongLE( &addr, 0 );
+    WriteLongLE( &addr, 0 );
 
     numWritten = fwrite( header, 1, sizeof(header), writer->fid );
     if( numWritten != sizeof(header) ) return -1;
 
-	return (int) numWritten;
+    return (int) numWritten;
 }
 
 /*********************************************************************************
@@ -138,31 +138,31 @@ long Audio_WAV_OpenWriter( WAV_Writer *writer, const char *fileName, int frameRa
  * Returns bytes written or negative error code.
  */
 long Audio_WAV_WriteShorts( WAV_Writer *writer,
-		short *samples,
-		int numSamples
-		)
+        short *samples,
+        int numSamples
+        )
 {
-	unsigned char buffer[2];
+    unsigned char buffer[2];
     unsigned char *bufferPtr;
-	int i;
-	short *p = samples;
+    int i;
+    short *p = samples;
     int numWritten;
     int bytesWritten;
-	if( numSamples <= 0 )
-	{
-		return -1;
-	}
+    if( numSamples <= 0 )
+    {
+        return -1;
+    }
 
     for( i=0; i<numSamples; i++ )
-	{
+    {
         bufferPtr = buffer;
-		WriteShortLE( &bufferPtr, *p++ );
+        WriteShortLE( &bufferPtr, *p++ );
         numWritten = fwrite( buffer, 1, sizeof( buffer), writer->fid );
         if( numWritten != sizeof(buffer) ) return -1;
-	}
+    }
     bytesWritten = numSamples * sizeof(short);
     writer->dataSize += bytesWritten;
-	return (int) bytesWritten;
+    return (int) bytesWritten;
 }
 
 /*********************************************************************************
@@ -171,7 +171,7 @@ long Audio_WAV_WriteShorts( WAV_Writer *writer,
  */
 long Audio_WAV_CloseWriter( WAV_Writer *writer )
 {
-	unsigned char buffer[4];
+    unsigned char buffer[4];
     unsigned char *bufferPtr;
     int numWritten;
     int riffSize;
@@ -212,7 +212,7 @@ int main( void )
 #define NUM_SAMPLES  (200)
     short data[NUM_SAMPLES];
     short saw = 0;
-    
+
     for( i=0; i<NUM_SAMPLES; i++ )
     {
         data[i] = saw;

qa/loopback/src/write_wav.h

@@ -25,13 +25,13 @@
  */
 
 /*
- * The text above constitutes the entire PortAudio license; however, 
+ * The text above constitutes the entire PortAudio license; however,
  * the PortAudio community also makes the following non-binding requests:
  *
  * Any person wishing to distribute modifications to the Software is
  * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also 
- * requested that these non-binding requests be included along with the 
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
  * license above.
  */
 #ifndef _WAV_WRITER_H
@@ -65,7 +65,7 @@ extern "C" {
 #define WAVE_FORMAT_PCM        (1)
 #define WAVE_FORMAT_IMA_ADPCM  (0x0011)
 
-	
+
 typedef struct WAV_Writer_s
 {
     FILE *fid;
@@ -86,9 +86,9 @@ long Audio_WAV_OpenWriter( WAV_Writer *writer, const char *fileName, int frameRa
  * Returns bytes written or negative error code.
  */
 long Audio_WAV_WriteShorts( WAV_Writer *writer,
-		short *samples,
-		int numSamples
-		);
+        short *samples,
+        int numSamples
+        );
 
 /*********************************************************************************
  * Close WAV file.