add plibsys

13 files changed, 0 insertions, 4162 deletions

diff --git a/portaudio/qa/loopback/README.txt b/portaudio/qa/loopback/README.txt
deleted file mode 100644
index 5ad0280..0000000
--- a/portaudio/qa/loopback/README.txt
+++ /dev/null
@@ -1,92 +0,0 @@
-README for PortAudio Loopback Test
-
-Copyright (c) 1999-2010 Phil Burk and Ross Bencina
-See complete license at end of file.
-
-This folder contains code for a single executable that does a standalone test of PortAudio.
-It does not require a human to listen to the result. Instead it listens to itself using
-a loopback cable connected between the audio output and the audio input. Special pop detectors
-and phase analysers can detect errors in the audio stream.
-
-This test can be run from a script as part of a nightly build and test.
-
---- How to Build the Loopback Test ---
-
-The loopback test is not normally built by the makefile.
-To build the loopback test, enter:
-
-  ./configure && make loopback
-  
-This will build the "bin/paloopback" executable.
-  
---- How To Run Test ---
-
-Connect stereo cables from one or more output audio devices to audio input devices. 
-The test will scan all the ports and find the cables.
-
-Adjust the volume levels of the hardware so you get a decent signal that will not clip.
-
-Run the test from the command line with the following options:
-
-  -i# Input device ID. Will scan for loopback if not specified.
-  -o# Output device ID. Will scan for loopback if not specified.
-  -r# Sample Rate in Hz. Will use multiple common rates if not specified.
-  -s# Size of callback buffer in frames, framesPerBuffer.
-  -w  Save bad recordings in a WAV file.
-  -dDir  Path for Directory for WAV files. Default is current directory.
-  -m  Just test the DSP Math code and not the audio devices.
-
-If the -w option is set then any tests that fail will save the recording of the broken
-channel in a WAV file. The files will be numbered and shown in the report.
-
---- ToDo ---
-
-* Add check for harmonic and enharmonic distortion.
-* Measure min/max peak values.
-* Detect DC bias.
-* Test against matrix of devices/APIs and settings.
-* Detect mono vs stereo loopback.
-* More command line options
-   --quick
-   --latency
-   --duration
-* Automated build and test script with cron job.
-* Test on Windows.
-
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2008 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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 
- * license above.
- */
diff --git a/portaudio/qa/loopback/src/audio_analyzer.c b/portaudio/qa/loopback/src/audio_analyzer.c
deleted file mode 100644
index fbdd631..0000000
--- a/portaudio/qa/loopback/src/audio_analyzer.c
+++ /dev/null
@@ -1,706 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <assert.h>
-#include <math.h>
-#include "qa_tools.h"
-#include "audio_analyzer.h"
-#include "write_wav.h"
-
-#define PAQA_POP_THRESHOLD  (0.04)
-
-/*==========================================================================================*/
-double PaQa_GetNthFrequency( double baseFrequency, int index )
-{
-    // Use 13 tone equal tempered scale because it does not generate harmonic ratios.
-    return baseFrequency * pow( 2.0, index / 13.0 );
-}
-
-/*==========================================================================================*/
-void PaQa_EraseBuffer( float *buffer, int numFrames, int samplesPerFrame )
-{
-    int i;
-    int numSamples = numFrames * samplesPerFrame;
-    for( i=0; i<numSamples; i++ )
-    {
-        *buffer++ = 0.0;
-    }
-}
-
-/*==========================================================================================*/
-void PaQa_SetupSineGenerator( PaQaSineGenerator *generator, double frequency, double amplitude, double frameRate )
-{
-    generator->phase = 0.0;
-    generator->amplitude = amplitude;
-    generator->frequency = frequency;
-    generator->phaseIncrement = 2.0 * frequency * MATH_PI / frameRate;
-}
-
-/*==========================================================================================*/
-void PaQa_MixSine( PaQaSineGenerator *generator, float *buffer, int numSamples, int stride )
-{
-    int i;
-    for( i=0; i<numSamples; i++ )
-    {
-        float value = sinf( (float) generator->phase ) * generator->amplitude;
-        *buffer += value; // Mix with existing value.
-        buffer += stride;
-        // Advance phase and wrap around.
-        generator->phase += generator->phaseIncrement;
-        if (generator->phase > MATH_TWO_PI)
-        {
-            generator->phase -= MATH_TWO_PI;
-        }
-    }
-}
-
-/*==========================================================================================*/
-void PaQa_GenerateCrackDISABLED( float *buffer, int numSamples, int stride )
-{
-    int i;
-    int offset = numSamples/2;
-    for( i=0; i<numSamples; i++ )
-    {
-        float phase = (MATH_TWO_PI * 0.5 * (i - offset)) / numSamples;
-        float cosp = cosf( phase );
-        float cos2 = cosp * cosp;
-        // invert second half of signal
-        float value = (i < offset) ? cos2 : (0-cos2);
-        *buffer = value;
-        buffer += stride;
-    }
-}
-
-
-/*==========================================================================================*/
-int PaQa_InitializeRecording( PaQaRecording *recording, int maxFrames, int frameRate )
-{
-    int numBytes = maxFrames * sizeof(float);
-    recording->buffer = (float*)malloc(numBytes);
-    QA_ASSERT_TRUE( "Allocate recording buffer.", (recording->buffer != NULL) );
-    recording->maxFrames = maxFrames;    recording->sampleRate = frameRate;
-    recording->numFrames = 0;
-    return 0;
-error:
-    return 1;
-}
-
-/*==========================================================================================*/
-void PaQa_TerminateRecording( PaQaRecording *recording )
-{
-    if (recording->buffer != NULL)
-    {
-        free( recording->buffer );
-        recording->buffer = NULL;
-    }
-    recording->maxFrames = 0;
-}
-
-/*==========================================================================================*/
-int PaQa_WriteRecording( PaQaRecording *recording, float *buffer, int numFrames, int stride )
-{
-    int i;
-    int framesToWrite;
-    float *data = &recording->buffer[recording->numFrames];
-
-    framesToWrite = numFrames;
-    if ((framesToWrite + recording->numFrames) > recording->maxFrames)
-    {
-        framesToWrite = recording->maxFrames - recording->numFrames;
-    }
-
-    for( i=0; i<framesToWrite; i++ )
-    {
-        *data++ = *buffer;
-        buffer += stride;
-    }
-    recording->numFrames += framesToWrite;
-    return (recording->numFrames >= recording->maxFrames);
-}
-
-/*==========================================================================================*/
-int PaQa_WriteSilence( PaQaRecording *recording, int numFrames )
-{
-    int i;
-    int framesToRecord;
-    float *data = &recording->buffer[recording->numFrames];
-
-    framesToRecord = numFrames;
-    if ((framesToRecord + recording->numFrames) > recording->maxFrames)
-    {
-        framesToRecord = recording->maxFrames - recording->numFrames;
-    }
-
-    for( i=0; i<framesToRecord; i++ )
-    {
-        *data++ = 0.0f;
-    }
-    recording->numFrames += framesToRecord;
-    return (recording->numFrames >= recording->maxFrames);
-}
-
-/*==========================================================================================*/
-int PaQa_RecordFreeze( PaQaRecording *recording, int numFrames )
-{
-    int i;
-    int framesToRecord;
-    float *data = &recording->buffer[recording->numFrames];
-
-    framesToRecord = numFrames;
-    if ((framesToRecord + recording->numFrames) > recording->maxFrames)
-    {
-        framesToRecord = recording->maxFrames - recording->numFrames;
-    }
-
-    for( i=0; i<framesToRecord; i++ )
-    {
-        // Copy old value forward as if the signal had frozen.
-        data[i] = data[i-1];
-    }
-    recording->numFrames += framesToRecord;
-    return (recording->numFrames >= recording->maxFrames);
-}
-
-/*==========================================================================================*/
-/**
- * Write recording to WAV file.
- */
-int PaQa_SaveRecordingToWaveFile( PaQaRecording *recording, const char *filename )
-{
-    WAV_Writer writer;
-    int result = 0;
-#define NUM_SAMPLES  (200)
-    short data[NUM_SAMPLES];
-    const int samplesPerFrame = 1;
-    int numLeft = recording->numFrames;
-    float *buffer = &recording->buffer[0];
-
-    result =  Audio_WAV_OpenWriter( &writer, filename, recording->sampleRate, samplesPerFrame );
-    if( result < 0 ) goto error;
-
-    while( numLeft > 0 )
-    {
-        int i;
-        int numToSave = (numLeft > NUM_SAMPLES) ? NUM_SAMPLES : numLeft;
-        // Convert double samples to shorts.
-        for( i=0; i<numToSave; i++ )
-        {
-            double fval = *buffer++;
-            // Convert float to int and clip to short range.
-            int ival = fval * 32768.0;
-            if( ival > 32767 ) ival = 32767;
-            else if( ival < -32768 ) ival = -32768;
-            data[i] = ival;
-        }
-        result =  Audio_WAV_WriteShorts( &writer, data, numToSave );
-        if( result < 0 ) goto error;
-        numLeft -= numToSave;
-    }
-
-    result =  Audio_WAV_CloseWriter( &writer );
-    if( result < 0 ) goto error;
-
-    return 0;
-
-error:
-    printf("ERROR: result = %d\n", result );
-    return result;
-#undef NUM_SAMPLES
-}
-
-/*==========================================================================================*/
-
-double PaQa_MeasureCrossingSlope( float *buffer, int numFrames )
-{
-    int i;
-    double slopeTotal = 0.0;
-    int slopeCount = 0;
-    float previous;
-    double averageSlope = 0.0;
-
-    previous = buffer[0];
-    for( i=1; i<numFrames; i++ )
-    {
-        float current = buffer[i];
-        if( (current > 0.0) && (previous < 0.0) )
-        {
-            double delta = current - previous;
-            slopeTotal += delta;
-            slopeCount += 1;
-        }
-        previous = current;
-    }
-    if( slopeCount > 0 )
-    {
-        averageSlope = slopeTotal / slopeCount;
-    }
-    return averageSlope;
-}
-
-/*==========================================================================================*/
-/*
- * We can't just measure the peaks cuz they may be clipped.
- * But the zero crossing should be intact.
- * The measured slope of a sine wave at zero should be:
- *
- *   slope = sin( 2PI * frequency / sampleRate )
- *
- */
-double PaQa_MeasureSineAmplitudeBySlope( PaQaRecording *recording,
-                                         double frequency, double frameRate,
-                                         int startFrame, int numFrames )
-{
-    float *buffer = &recording->buffer[startFrame];
-    double measuredSlope = PaQa_MeasureCrossingSlope( buffer, numFrames );
-    double unitySlope = sin( MATH_TWO_PI * frequency / frameRate );
-    double estimatedAmplitude = measuredSlope / unitySlope;
-    return estimatedAmplitude;
-}
-
-/*==========================================================================================*/
-double PaQa_CorrelateSine( PaQaRecording *recording, double frequency, double frameRate,
-                           int startFrame, int numFrames, double *phasePtr )
-{
-    double magnitude = 0.0;
-    int numLeft = numFrames;
-    double phase = 0.0;
-    double phaseIncrement = 2.0 * MATH_PI * frequency / frameRate;
-    double sinAccumulator = 0.0;
-    double cosAccumulator = 0.0;
-    float *data = &recording->buffer[startFrame];
-
-    QA_ASSERT_TRUE( "startFrame out of bounds", (startFrame < recording->numFrames) );
-    QA_ASSERT_TRUE( "numFrames out of bounds", ((startFrame+numFrames) <= recording->numFrames) );
-
-    while( numLeft > 0 )
-    {
-        double sample = (double) *data++;
-        sinAccumulator += sample * sin( phase );
-        cosAccumulator += sample * cos( phase );
-        phase += phaseIncrement;
-        if (phase > MATH_TWO_PI)
-        {
-            phase -= MATH_TWO_PI;
-        }
-        numLeft -= 1;
-    }
-    sinAccumulator = sinAccumulator / numFrames;
-    cosAccumulator = cosAccumulator / numFrames;
-    // TODO Why do I have to multiply by 2.0? Need it to make result come out right.
-    magnitude = 2.0 * sqrt( (sinAccumulator * sinAccumulator) + (cosAccumulator * cosAccumulator ));
-    if( phasePtr != NULL )
-    {
-        double phase = atan2( cosAccumulator, sinAccumulator );
-        *phasePtr = phase;
-    }
-    return magnitude;
-error:
-    return -1.0;
-}
-
-/*==========================================================================================*/
-void PaQa_FilterRecording( PaQaRecording *input, PaQaRecording *output, BiquadFilter *filter )
-{
-    int numToFilter = (input->numFrames > output->maxFrames) ? output->maxFrames : input->numFrames;
-    BiquadFilter_Filter( filter, &input->buffer[0], &output->buffer[0], numToFilter );
-    output->numFrames = numToFilter;
-}
-
-/*==========================================================================================*/
-/** Scan until we get a correlation of a single that goes over the tolerance level,
- * peaks then drops to half the peak.
- * Look for inverse correlation as well.
- */
-double PaQa_FindFirstMatch( PaQaRecording *recording, float *buffer, int numFrames, double threshold  )
-{
-    int ic,is;
-    // How many buffers will fit in the recording?
-    int maxCorrelations = recording->numFrames - numFrames;
-    double maxSum = 0.0;
-    int peakIndex = -1;
-    double inverseMaxSum = 0.0;
-    int inversePeakIndex = -1;
-    double location = -1.0;
-
-    QA_ASSERT_TRUE( "numFrames out of bounds", (numFrames < recording->numFrames) );
-
-    for( ic=0; ic<maxCorrelations; ic++ )
-    {
-        int pastPeak;
-        int inversePastPeak;
-
-        double sum = 0.0;
-        // Correlate buffer against the recording.
-        float *recorded = &recording->buffer[ ic ];
-        for( is=0; is<numFrames; is++ )
-        {
-            float s1 = buffer[is];
-            float s2 = *recorded++;
-            sum += s1 * s2;
-        }
-        if( (sum > maxSum) )
-        {
-            maxSum = sum;
-            peakIndex = ic;
-        }
-        if( ((-sum) > inverseMaxSum) )
-        {
-            inverseMaxSum = -sum;
-            inversePeakIndex = ic;
-        }
-        pastPeak = (maxSum > threshold) && (sum < 0.5*maxSum);
-        inversePastPeak = (inverseMaxSum > threshold) && ((-sum) < 0.5*inverseMaxSum);
-        //printf("PaQa_FindFirstMatch: ic = %4d, sum = %8f, maxSum = %8f, inverseMaxSum = %8f\n", ic, sum, maxSum, inverseMaxSum );
-        if( pastPeak && inversePastPeak )
-        {
-            if( maxSum > inverseMaxSum )
-            {
-                location = peakIndex;
-            }
-            else
-            {
-                location = inversePeakIndex;
-            }
-            break;
-        }
-
-    }
-    //printf("PaQa_FindFirstMatch: location = %4d\n", (int)location );
-    return location;
-error:
-    return -1.0;
-}
-
-/*==========================================================================================*/
-// Measure the area under the curve by summing absolute value of each value.
-double PaQa_MeasureArea( float *buffer, int numFrames, int stride  )
-{
-    int is;
-    double area = 0.0;
-    for( is=0; is<numFrames; is++ )
-    {
-        area += fabs( *buffer );
-        buffer += stride;
-    }
-    return area;
-}
-
-/*==========================================================================================*/
-// Measure the area under the curve by summing absolute value of each value.
-double PaQa_MeasureRootMeanSquare( float *buffer, int numFrames )
-{
-    int is;
-    double area = 0.0;
-    double root;
-    for( is=0; is<numFrames; is++ )
-    {
-        float value = *buffer++;
-        area += value * value;
-    }
-    root = sqrt( area );
-    return root / numFrames;
-}
-
-
-/*==========================================================================================*/
-// Compare the amplitudes of these two signals.
-// Return ratio of recorded signal over buffer signal.
-
-double PaQa_CompareAmplitudes( PaQaRecording *recording, int startAt, float *buffer, int numFrames )
-{
-    QA_ASSERT_TRUE( "startAt+numFrames out of bounds", ((startAt+numFrames) < recording->numFrames) );
-
-    {
-        double recordedArea = PaQa_MeasureArea( &recording->buffer[startAt], numFrames, 1 );
-        double bufferArea = PaQa_MeasureArea( buffer, numFrames, 1 );
-        if( bufferArea == 0.0 ) return 100000000.0;
-        return recordedArea / bufferArea;
-    }
-error:
-    return -1.0;
-}
-
-
-/*==========================================================================================*/
-double PaQa_ComputePhaseDifference( double phase1, double phase2 )
-{
-    double delta = phase1 - phase2;
-    while( delta > MATH_PI )
-    {
-        delta -= MATH_TWO_PI;
-    }
-    while( delta < -MATH_PI )
-    {
-        delta += MATH_TWO_PI;
-    }
-    return delta;
-}
-
-/*==========================================================================================*/
-int PaQa_MeasureLatency( PaQaRecording *recording, PaQaTestTone *testTone, PaQaAnalysisResult *analysisResult )
-{
-    double threshold;
-    PaQaSineGenerator generator;
-#define MAX_BUFFER_SIZE 2048
-    float buffer[MAX_BUFFER_SIZE];
-    double period = testTone->sampleRate / testTone->frequency;
-    int cycleSize = (int) (period + 0.5);
-    //printf("PaQa_AnalyseRecording: frequency = %8f, frameRate = %8f, period = %8f, cycleSize = %8d\n",
-    //       testTone->frequency, testTone->sampleRate, period, cycleSize );
-    analysisResult->latency = -1;
-    analysisResult->valid = (0);
-
-    // Set up generator to find matching first cycle.
-    QA_ASSERT_TRUE( "cycleSize out of bounds", (cycleSize < MAX_BUFFER_SIZE) );
-    PaQa_SetupSineGenerator( &generator, testTone->frequency, testTone->amplitude, testTone->sampleRate );
-    PaQa_EraseBuffer( buffer, cycleSize, testTone->samplesPerFrame );
-    PaQa_MixSine( &generator, buffer, cycleSize, testTone->samplesPerFrame );
-
-    threshold = cycleSize * 0.02;
-    analysisResult->latency = PaQa_FindFirstMatch( recording, buffer, cycleSize, threshold );
-    QA_ASSERT_TRUE( "Could not find the start of the signal.", (analysisResult->latency >= 0) );
-    analysisResult->amplitudeRatio = PaQa_CompareAmplitudes( recording, analysisResult->latency, buffer, cycleSize );
-    return 0;
-error:
-    return -1;
-}
-
-/*==========================================================================================*/
-// Apply cosine squared window.
-void PaQa_FadeInRecording( PaQaRecording *recording, int startFrame, int count )
-{
-    int is;
-    double phase = 0.5 * MATH_PI;
-    // Advance a quarter wave
-    double phaseIncrement = 0.25 * 2.0 * MATH_PI / count;
-
-    assert( startFrame >= 0 );
-    assert( count > 0 );
-
-    /* Zero out initial part of the recording. */
-    for( is=0; is<startFrame; is++ )
-    {
-        recording->buffer[ is ] = 0.0f;
-    }
-    /* Fade in where signal begins. */
-    for( is=0; is<count; is++ )
-    {
-        double c = cos( phase );
-        double w = c * c;
-        float x = recording->buffer[ is + startFrame ];
-        float y = x * w;
-        //printf("FADE %d : w=%f, x=%f, y=%f\n", is, w, x, y );
-        recording->buffer[ is + startFrame ] = y;
-
-        phase += phaseIncrement;
-    }
-}
-
-
-/*==========================================================================================*/
-/** Apply notch filter and high pass filter then detect remaining energy.
- */
-int PaQa_DetectPop( PaQaRecording *recording, PaQaTestTone *testTone, PaQaAnalysisResult *analysisResult )
-{
-    int result = 0;
-    int i;
-    double maxAmplitude;
-    int maxPosition;
-
-    PaQaRecording     notchOutput = { 0 };
-    BiquadFilter      notchFilter;
-
-    PaQaRecording     hipassOutput = { 0 };
-    BiquadFilter      hipassFilter;
-
-    int frameRate = (int) recording->sampleRate;
-
-    analysisResult->popPosition = -1;
-    analysisResult->popAmplitude = 0.0;
-
-    result = PaQa_InitializeRecording( &notchOutput, recording->numFrames, frameRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    result = PaQa_InitializeRecording( &hipassOutput, recording->numFrames, frameRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    // Use notch filter to remove test tone.
-    BiquadFilter_SetupNotch( &notchFilter, testTone->frequency / frameRate, 0.5 );
-    PaQa_FilterRecording( recording, &notchOutput, &notchFilter );
-    //result = PaQa_SaveRecordingToWaveFile( &notchOutput, "notch_output.wav" );
-    //QA_ASSERT_EQUALS( "PaQa_SaveRecordingToWaveFile failed", 0, result );
-
-    // Apply fade-in window.
-    PaQa_FadeInRecording( &notchOutput, (int) analysisResult->latency, 500 );
-
-    // Use high pass to accentuate the edges of a pop. At higher frequency!
-    BiquadFilter_SetupHighPass( &hipassFilter, 2.0 * testTone->frequency / frameRate, 0.5 );
-    PaQa_FilterRecording( &notchOutput, &hipassOutput, &hipassFilter );
-    //result = PaQa_SaveRecordingToWaveFile( &hipassOutput, "hipass_output.wav" );
-    //QA_ASSERT_EQUALS( "PaQa_SaveRecordingToWaveFile failed", 0, result );
-
-    // Scan remaining signal looking for peak.
-    maxAmplitude = 0.0;
-    maxPosition = -1;
-    for( i=(int) analysisResult->latency; i<hipassOutput.numFrames; i++ )
-    {
-        float x = hipassOutput.buffer[i];
-        float mag = fabs( x );
-        if( mag > maxAmplitude )
-        {
-            maxAmplitude = mag;
-            maxPosition = i;
-        }
-    }
-
-    if( maxAmplitude > PAQA_POP_THRESHOLD )
-    {
-        analysisResult->popPosition = maxPosition;
-        analysisResult->popAmplitude = maxAmplitude;
-    }
-
-    PaQa_TerminateRecording( &notchOutput );
-    PaQa_TerminateRecording( &hipassOutput );
-    return 0;
-
-error:
-    PaQa_TerminateRecording( &notchOutput );
-    PaQa_TerminateRecording( &hipassOutput );
-    return -1;
-}
-
-/*==========================================================================================*/
-int PaQa_DetectPhaseError( PaQaRecording *recording, PaQaTestTone *testTone, PaQaAnalysisResult *analysisResult )
-{
-    int i;
-    double period = testTone->sampleRate / testTone->frequency;
-    int cycleSize = (int) (period + 0.5);
-
-    double maxAddedFrames = 0.0;
-    double maxDroppedFrames = 0.0;
-
-    double previousPhase = 0.0;
-    double previousFrameError = 0;
-    int loopCount = 0;
-    int skip = cycleSize;
-    int windowSize = cycleSize;
-
-    // Scan recording starting with first cycle, looking for phase errors.
-    analysisResult->numDroppedFrames = 0.0;
-    analysisResult->numAddedFrames = 0.0;
-    analysisResult->droppedFramesPosition = -1.0;
-    analysisResult->addedFramesPosition = -1.0;
-
-    for( i=analysisResult->latency; i<(recording->numFrames - windowSize); i += skip )
-    {
-        double expectedPhase = previousPhase + (skip * MATH_TWO_PI / period);
-        double expectedPhaseIncrement = PaQa_ComputePhaseDifference( expectedPhase, previousPhase );
-
-        double phase = 666.0;
-        double mag = PaQa_CorrelateSine( recording, testTone->frequency, testTone->sampleRate, i, windowSize, &phase );
-        if( (loopCount > 1) && (mag > 0.0) )
-        {
-            double phaseDelta = PaQa_ComputePhaseDifference( phase, previousPhase );
-            double phaseError = PaQa_ComputePhaseDifference( phaseDelta, expectedPhaseIncrement );
-            // Convert phaseError to equivalent number of frames.
-            double frameError = period * phaseError / MATH_TWO_PI;
-            double consecutiveFrameError = frameError + previousFrameError;
-//            if( fabs(frameError) > 0.01 )
-//            {
-//                printf("FFFFFFFFFFFFF frameError = %f, at %d\n", frameError, i );
-//            }
-            if( consecutiveFrameError > 0.8 )
-            {
-                double droppedFrames = consecutiveFrameError;
-                if (droppedFrames > (maxDroppedFrames * 1.001))
-                {
-                    analysisResult->numDroppedFrames = droppedFrames;
-                    analysisResult->droppedFramesPosition = i + (windowSize/2);
-                    maxDroppedFrames = droppedFrames;
-                }
-            }
-            else if( consecutiveFrameError < -0.8 )
-            {
-                double addedFrames = 0 - consecutiveFrameError;
-                if (addedFrames > (maxAddedFrames * 1.001))
-                {
-                    analysisResult->numAddedFrames = addedFrames;
-                    analysisResult->addedFramesPosition = i + (windowSize/2);
-                    maxAddedFrames = addedFrames;
-                }
-            }
-            previousFrameError = frameError;
-
-
-            //if( i<8000 )
-            //{
-            //    printf("%d: phase = %8f, expected = %8f, delta = %8f, frameError = %8f\n", i, phase, expectedPhaseIncrement, phaseDelta, frameError );
-            //}
-        }
-        previousPhase = phase;
-        loopCount += 1;
-    }
-    return 0;
-}
-
-/*==========================================================================================*/
-int PaQa_AnalyseRecording( PaQaRecording *recording, PaQaTestTone *testTone, PaQaAnalysisResult *analysisResult )
-{
-    int result = 0;
-
-    memset( analysisResult, 0, sizeof(PaQaAnalysisResult) );
-    result = PaQa_MeasureLatency( recording, testTone, analysisResult );
-    QA_ASSERT_EQUALS( "latency measurement", 0, result );
-
-    if( (analysisResult->latency >= 0) && (analysisResult->amplitudeRatio > 0.1) )
-    {
-        analysisResult->valid = (1);
-
-        result = PaQa_DetectPop( recording, testTone, analysisResult );
-        QA_ASSERT_EQUALS( "detect pop", 0, result );
-
-        result = PaQa_DetectPhaseError( recording, testTone, analysisResult );
-        QA_ASSERT_EQUALS( "detect phase error", 0, result );
-    }
-    return 0;
-error:
-    return -1;
-}
diff --git a/portaudio/qa/loopback/src/audio_analyzer.h b/portaudio/qa/loopback/src/audio_analyzer.h
deleted file mode 100644
index 8d9f1ee..0000000
--- a/portaudio/qa/loopback/src/audio_analyzer.h
+++ /dev/null
@@ -1,187 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#ifndef _AUDIO_ANALYZER_H
-#define _AUDIO_ANALYZER_H
-
-#include "biquad_filter.h"
-
-#define MATH_PI  (3.141592653589793238462643)
-#define MATH_TWO_PI  (2.0 * MATH_PI)
-
-typedef struct PaQaSineGenerator_s
-{
-    double    phase;
-    double    phaseIncrement;
-    double    frequency;
-    double    amplitude;
-} PaQaSineGenerator;
-
-/** 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;
-} PaQaRecording;
-
-typedef struct PaQaTestTone_s
-{
-    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;
-} PaQaAnalysisResult;
-
-
-/*================================================================*/
-/*================= General DSP Tools ============================*/
-/*================================================================*/
-/**
- * Calculate Nth frequency of a series for use in testing multiple channels.
- * Series should avoid harmonic overlap between channels.
- */
-double PaQa_GetNthFrequency( double baseFrequency, int index );
-
-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 );
-
-/**
- * Generate a signal with a sharp edge in the middle that can be recognized despite some phase shift.
- */
-void PaQa_GenerateCrack( float *buffer, int numSamples, int stride );
-
-double PaQa_ComputePhaseDifference( double phase1, double phase2 );
-
-/**
- * Measure the area under the curve by summing absolute value of each value.
- */
-double PaQa_MeasureArea( float *buffer, int numFrames, int stride  );
-
-/**
- * Measure slope of the positive zero crossings.
- */
-double PaQa_MeasureCrossingSlope( float *buffer, int numFrames );
-
-
-/**
- * Prepare an oscillator that can generate a sine tone for testing.
- */
-void PaQa_SetupSineGenerator( PaQaSineGenerator *generator, double frequency, double amplitude, double frameRate );
-
-/*================================================================*/
-/*================= Recordings ===================================*/
-/*================================================================*/
-/**
- * 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.
- */
-void PaQa_FilterRecording( PaQaRecording *input, PaQaRecording *output, BiquadFilter *filter );
-
-
-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 );
-
-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 PaQa_MeasureRootMeanSquare( float *buffer, int numFrames );
-
-/**
- * Compare the amplitudes of these two signals.
- * Return ratio of recorded signal over buffer signal.
- */
-double PaQa_CompareAmplitudes( PaQaRecording *recording, int startAt, float *buffer, int numSamples );
-
-/**
- * Analyse a recording of a sine wave.
- * Measure latency and look for dropped frames, etc.
- */
-int PaQa_AnalyseRecording( PaQaRecording *recording, PaQaTestTone *testTone, PaQaAnalysisResult *analysisResult );
-
-#endif /* _AUDIO_ANALYZER_H */
diff --git a/portaudio/qa/loopback/src/biquad_filter.c b/portaudio/qa/loopback/src/biquad_filter.c
deleted file mode 100755
index 1715fa3..0000000
--- a/portaudio/qa/loopback/src/biquad_filter.c
+++ /dev/null
@@ -1,123 +0,0 @@
-#include <math.h>
-#include <string.h>
-
-#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)
- *
- * @author (C) 2002 Phil Burk, SoftSynth.com, All Rights Reserved
- */
-
-#define FILTER_PI  (3.141592653589793238462643)
-/***********************************************************
-** Calculate coefficients common to many parametric biquad filters.
-*/
-static void BiquadFilter_CalculateCommon( BiquadFilter *filter, double ratio, double Q )
-{
-    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;
-
-    filter->cos_omega = (double) cos( omega );
-    filter->sin_omega = (double) sin( omega );
-    filter->alpha = filter->sin_omega / (2.0 * Q);
-}
-
-/*********************************************************************************
- ** Calculate coefficients for Highpass filter.
- */
-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;
-    filter->a2 = filter->a0;
-    filter->b1 = -2.0 * filter->cos_omega * scalar;
-    filter->b2 = (1.0 - filter->alpha) * scalar;
-}
-
-
-/*********************************************************************************
- ** Calculate coefficients for Notch filter.
- */
-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;
-    filter->a2 = filter->a0;
-    filter->b1 = filter->a1;
-    filter->b2 = (1.0 - filter->alpha) * scalar;
-}
-
-/*****************************************************************
-** Perform core IIR filter calculation without permutation.
-*/
-void BiquadFilter_Filter( BiquadFilter *filter, float *inputs, float *outputs, int numSamples )
-{
-    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;
-
-    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.
-        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;
-}
diff --git a/portaudio/qa/loopback/src/biquad_filter.h b/portaudio/qa/loopback/src/biquad_filter.h
deleted file mode 100755
index 0895aba..0000000
--- a/portaudio/qa/loopback/src/biquad_filter.h
+++ /dev/null
@@ -1,38 +0,0 @@
-#ifndef _BIQUADFILTER_H
-#define _BIQUADFILTER_H
-
-
-/**
- * Unit_BiquadFilter implements a second order IIR filter.
- *
- * @author (C) 2002 Phil Burk, SoftSynth.com, All Rights Reserved
- */
-
-#define BIQUAD_MIN_RATIO     (0.000001)
-#define BIQUAD_MIN_Q         (0.00001)
-
-typedef struct BiquadFilter_s
-{
-    double      xn1;    // storage for delayed signals
-    double      xn2;
-    double      yn1;
-    double      yn2;
-
-    double      a0;     // coefficients
-    double      a1;
-    double      a2;
-
-    double      b1;
-    double      b2;
-
-    double      cos_omega;
-    double      sin_omega;
-    double      alpha;
-} BiquadFilter;
-
-void BiquadFilter_SetupHighPass( BiquadFilter *filter, double ratio, double Q );
-void BiquadFilter_SetupNotch( BiquadFilter *filter, double ratio, double Q );
-
-void BiquadFilter_Filter( BiquadFilter *filter, float *inputs, float *outputs, int numSamples );
-
-#endif
diff --git a/portaudio/qa/loopback/src/paqa.c b/portaudio/qa/loopback/src/paqa.c
deleted file mode 100644
index 5eb6283..0000000
--- a/portaudio/qa/loopback/src/paqa.c
+++ /dev/null
@@ -1,1601 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <memory.h>
-#include <math.h>
-#include <string.h>
-
-#include "portaudio.h"
-
-#include "qa_tools.h"
-
-#include "paqa_tools.h"
-#include "audio_analyzer.h"
-#include "test_audio_analyzer.h"
-
-/** Accumulate counts for how many tests pass or fail. */
-int g_testsPassed = 0;
-int g_testsFailed = 0;
-
-#define MAX_NUM_GENERATORS                   (8)
-#define MAX_NUM_RECORDINGS                   (8)
-#define MAX_BACKGROUND_NOISE_RMS             (0.0004)
-#define LOOPBACK_DETECTION_DURATION_SECONDS  (0.8)
-#define DEFAULT_FRAMES_PER_BUFFER            (0)
-#define PAQA_WAIT_STREAM_MSEC                (100)
-#define PAQA_TEST_DURATION                   (1.2)
-
-// Use two separate streams instead of one full duplex stream.
-#define PAQA_FLAG_TWO_STREAMS       (1<<0)
-// Use bloching read/write for loopback.
-#define PAQA_FLAG_USE_BLOCKING_IO   (1<<1)
-
-const char * s_FlagOnNames[] =
-{
-    "Two Streams (Half Duplex)",
-    "Blocking Read/Write"
-};
-
-const char * s_FlagOffNames[] =
-{
-    "One Stream (Full Duplex)",
-    "Callback"
-};
-
-
-/** Parameters that describe a single test run. */
-typedef struct TestParameters_s
-{
-    PaStreamParameters inputParameters;
-    PaStreamParameters outputParameters;
-    double             sampleRate;
-    int                samplesPerFrame;
-    int                framesPerBuffer;
-    int                maxFrames;
-    double             baseFrequency;
-    double             amplitude;
-    PaStreamFlags      streamFlags;  // paClipOff, etc
-    int                flags;        // PAQA_FLAG_TWO_STREAMS, PAQA_FLAG_USE_BLOCKING_IO
-} TestParameters;
-
-typedef struct LoopbackContext_s
-{
-    // Generate a unique signal on each channel.
-    PaQaSineGenerator  generators[MAX_NUM_GENERATORS];
-    // Record each channel individually.
-    PaQaRecording      recordings[MAX_NUM_RECORDINGS];
-
-    // Reported by the stream after it's opened
-    PaTime             streamInfoInputLatency;
-    PaTime             streamInfoOutputLatency;
-
-    // Measured at runtime.
-    volatile int       callbackCount; // incremented for each callback
-    volatile int       inputBufferCount; // incremented if input buffer not NULL
-    int                inputUnderflowCount;
-    int                inputOverflowCount;
-
-    volatile int       outputBufferCount; // incremented if output buffer not NULL
-    int                outputOverflowCount;
-    int                outputUnderflowCount;
-
-    // Measure whether input or output is lagging behind.
-    volatile int       minInputOutputDelta;
-    volatile int       maxInputOutputDelta;
-
-    int                minFramesPerBuffer;
-    int                maxFramesPerBuffer;
-    int                primingCount;
-    TestParameters    *test;
-    volatile int       done;
-} LoopbackContext;
-
-typedef struct UserOptions_s
-{
-    int           sampleRate;
-    int           framesPerBuffer;
-    int           inputLatency;
-    int           outputLatency;
-    int           saveBadWaves;
-    int           verbose;
-    int           waveFileCount;
-    const char   *waveFilePath;
-    PaDeviceIndex inputDevice;
-    PaDeviceIndex outputDevice;
-} UserOptions;
-
-#define BIG_BUFFER_SIZE  (sizeof(float) * 2 * 2 * 1024)
-static unsigned char g_ReadWriteBuffer[BIG_BUFFER_SIZE];
-
-#define MAX_CONVERSION_SAMPLES   (2 * 32 * 1024)
-#define CONVERSION_BUFFER_SIZE  (sizeof(float) * 2 * MAX_CONVERSION_SAMPLES)
-static unsigned char g_ConversionBuffer[CONVERSION_BUFFER_SIZE];
-
-/*******************************************************************/
-static int RecordAndPlaySinesCallback( const void *inputBuffer, void *outputBuffer,
-                        unsigned long framesPerBuffer,
-                        const PaStreamCallbackTimeInfo* timeInfo,
-                        PaStreamCallbackFlags statusFlags,
-                        void *userData )
-{
-    int i;
-    LoopbackContext *loopbackContext = (LoopbackContext *) userData;
-
-
-    loopbackContext->callbackCount += 1;
-    if( statusFlags & paInputUnderflow ) loopbackContext->inputUnderflowCount += 1;
-    if( statusFlags & paInputOverflow ) loopbackContext->inputOverflowCount += 1;
-    if( statusFlags & paOutputUnderflow ) loopbackContext->outputUnderflowCount += 1;
-    if( statusFlags & paOutputOverflow ) loopbackContext->outputOverflowCount += 1;
-    if( statusFlags & paPrimingOutput ) loopbackContext->primingCount += 1;
-    if( framesPerBuffer > loopbackContext->maxFramesPerBuffer )
-    {
-        loopbackContext->maxFramesPerBuffer = framesPerBuffer;
-    }
-    if( framesPerBuffer < loopbackContext->minFramesPerBuffer )
-    {
-        loopbackContext->minFramesPerBuffer = framesPerBuffer;
-    }
-
-    /* This may get called with NULL inputBuffer during initial setup.
-     * We may also use the same callback with output only streams.
-     */
-    if( inputBuffer != NULL)
-    {
-        int channelsPerFrame = loopbackContext->test->inputParameters.channelCount;
-        float *in = (float *)inputBuffer;
-        PaSampleFormat inFormat = loopbackContext->test->inputParameters.sampleFormat;
-
-        loopbackContext->inputBufferCount += 1;
-
-        if( inFormat != paFloat32 )
-        {
-            int samplesToConvert = framesPerBuffer * channelsPerFrame;
-            in = (float *) g_ConversionBuffer;
-            if( samplesToConvert > MAX_CONVERSION_SAMPLES )
-            {
-                // Hack to prevent buffer overflow.
-                // @todo Loop with small buffer instead of failing.
-                printf("Format conversion buffer too small!\n");
-                return paComplete;
-            }
-            PaQa_ConvertToFloat( inputBuffer, samplesToConvert, inFormat, (float *) g_ConversionBuffer );
-        }
-
-        // Read each channel from the buffer.
-        for( i=0; i<channelsPerFrame; i++ )
-        {
-            loopbackContext->done |= PaQa_WriteRecording( &loopbackContext->recordings[i],
-                                        in + i,
-                                        framesPerBuffer,
-                                        channelsPerFrame );
-        }
-    }
-
-    if( outputBuffer != NULL )
-    {
-        int channelsPerFrame = loopbackContext->test->outputParameters.channelCount;
-        float *out = (float *)outputBuffer;
-        PaSampleFormat outFormat = loopbackContext->test->outputParameters.sampleFormat;
-
-        loopbackContext->outputBufferCount += 1;
-
-        if( outFormat != paFloat32 )
-        {
-            // If we need to convert then mix to the g_ConversionBuffer and then convert into the PA outputBuffer.
-            out = (float *) g_ConversionBuffer;
-        }
-
-        PaQa_EraseBuffer( out, framesPerBuffer, channelsPerFrame );
-        for( i=0; i<channelsPerFrame; i++ )
-        {
-            PaQa_MixSine( &loopbackContext->generators[i],
-                         out + i,
-                         framesPerBuffer,
-                         channelsPerFrame );
-        }
-
-        if( outFormat != paFloat32 )
-        {
-            int samplesToConvert = framesPerBuffer * channelsPerFrame;
-            if( samplesToConvert > MAX_CONVERSION_SAMPLES )
-            {
-                printf("Format conversion buffer too small!\n");
-                return paComplete;
-            }
-            PaQa_ConvertFromFloat( out, framesPerBuffer * channelsPerFrame, outFormat, outputBuffer );
-        }
-
-    }
-
-    // Measure whether the input or output are lagging behind.
-    // Don't measure lag at end.
-    if( !loopbackContext->done )
-    {
-        int inputOutputDelta = loopbackContext->inputBufferCount - loopbackContext->outputBufferCount;
-        if( loopbackContext->maxInputOutputDelta < inputOutputDelta )
-        {
-            loopbackContext->maxInputOutputDelta = inputOutputDelta;
-        }
-        if( loopbackContext->minInputOutputDelta > inputOutputDelta )
-        {
-            loopbackContext->minInputOutputDelta = inputOutputDelta;
-        }
-    }
-
-    return loopbackContext->done ? paComplete : paContinue;
-}
-
-static void CopyStreamInfoToLoopbackContext( LoopbackContext *loopbackContext, PaStream *inputStream, PaStream *outputStream )
-{
-    const PaStreamInfo *inputStreamInfo = Pa_GetStreamInfo( inputStream );
-    const PaStreamInfo *outputStreamInfo = Pa_GetStreamInfo( outputStream );
-
-    loopbackContext->streamInfoInputLatency = inputStreamInfo ? inputStreamInfo->inputLatency : -1;
-    loopbackContext->streamInfoOutputLatency = outputStreamInfo ? outputStreamInfo->outputLatency : -1;
-}
-
-/*******************************************************************/
-/**
- * Open a full duplex audio stream.
- * Generate sine waves on the output channels and record the input channels.
- * Then close the stream.
- * @return 0 if OK or negative error.
- */
-int PaQa_RunLoopbackFullDuplex( LoopbackContext *loopbackContext )
-{
-    PaStream *stream = NULL;
-    PaError err = 0;
-    TestParameters *test = loopbackContext->test;
-    loopbackContext->done = 0;
-    // Use one full duplex stream.
-    err = Pa_OpenStream(
-                    &stream,
-                    &test->inputParameters,
-                    &test->outputParameters,
-                    test->sampleRate,
-                    test->framesPerBuffer,
-                    paClipOff, /* we won't output out of range samples so don't bother clipping them */
-                    RecordAndPlaySinesCallback,
-                    loopbackContext );
-    if( err != paNoError ) goto error;
-
-    CopyStreamInfoToLoopbackContext( loopbackContext, stream, stream );
-
-    err = Pa_StartStream( stream );
-    if( err != paNoError ) goto error;
-
-    // Wait for stream to finish.
-    while( loopbackContext->done == 0 )
-    {
-        Pa_Sleep(PAQA_WAIT_STREAM_MSEC);
-    }
-
-    err = Pa_StopStream( stream );
-    if( err != paNoError ) goto error;
-
-    err = Pa_CloseStream( stream );
-    if( err != paNoError ) goto error;
-
-    return 0;
-
-error:
-    return err;
-}
-
-/*******************************************************************/
-/**
- * Open two audio streams, one for input and one for output.
- * Generate sine waves on the output channels and record the input channels.
- * Then close the stream.
- * @return 0 if OK or paTimedOut.
- */
-
-int PaQa_WaitForStream( LoopbackContext *loopbackContext )
-{
-    int timeoutMSec = 1000 * PAQA_TEST_DURATION * 2;
-
-    // Wait for stream to finish or timeout.
-    while( (loopbackContext->done == 0) && (timeoutMSec > 0) )
-    {
-        Pa_Sleep(PAQA_WAIT_STREAM_MSEC);
-        timeoutMSec -= PAQA_WAIT_STREAM_MSEC;
-    }
-
-    if( loopbackContext->done == 0 )
-    {
-        printf("ERROR - stream completion timed out!");
-        return paTimedOut;
-    }
-    return 0;
-}
-
-/*******************************************************************/
-/**
- * Open two audio streams, one for input and one for output.
- * Generate sine waves on the output channels and record the input channels.
- * Then close the stream.
- * @return 0 if OK or negative error.
- */
-int PaQa_RunLoopbackHalfDuplex( LoopbackContext *loopbackContext )
-{
-    PaStream *inStream = NULL;
-    PaStream *outStream = NULL;
-    PaError err = 0;
-    int timedOut = 0;
-    TestParameters *test = loopbackContext->test;
-    loopbackContext->done = 0;
-
-    // Use two half duplex streams.
-    err = Pa_OpenStream(
-                        &inStream,
-                        &test->inputParameters,
-                        NULL,
-                        test->sampleRate,
-                        test->framesPerBuffer,
-                        test->streamFlags,
-                        RecordAndPlaySinesCallback,
-                        loopbackContext );
-    if( err != paNoError ) goto error;
-    err = Pa_OpenStream(
-                        &outStream,
-                        NULL,
-                        &test->outputParameters,
-                        test->sampleRate,
-                        test->framesPerBuffer,
-                        test->streamFlags,
-                        RecordAndPlaySinesCallback,
-                        loopbackContext );
-    if( err != paNoError ) goto error;
-
-    CopyStreamInfoToLoopbackContext( loopbackContext, inStream, outStream );
-
-    err = Pa_StartStream( inStream );
-    if( err != paNoError ) goto error;
-
-    // Start output later so we catch the beginning of the waveform.
-    err = Pa_StartStream( outStream );
-    if( err != paNoError ) goto error;
-
-    timedOut = PaQa_WaitForStream( loopbackContext );
-
-    err = Pa_StopStream( inStream );
-    if( err != paNoError ) goto error;
-
-    err = Pa_StopStream( outStream );
-    if( err != paNoError ) goto error;
-
-    err = Pa_CloseStream( inStream );
-    if( err != paNoError ) goto error;
-
-    err = Pa_CloseStream( outStream );
-    if( err != paNoError ) goto error;
-
-    return timedOut;
-
-error:
-    return err;
-}
-
-
-/*******************************************************************/
-/**
- * Open one audio streams, just for input.
- * Record background level.
- * Then close the stream.
- * @return 0 if OK or negative error.
- */
-int PaQa_RunInputOnly( LoopbackContext *loopbackContext )
-{
-    PaStream *inStream = NULL;
-    PaError err = 0;
-    int timedOut = 0;
-    TestParameters *test = loopbackContext->test;
-    loopbackContext->done = 0;
-
-    // Just open an input stream.
-    err = Pa_OpenStream(
-                        &inStream,
-                        &test->inputParameters,
-                        NULL,
-                        test->sampleRate,
-                        test->framesPerBuffer,
-                        paClipOff, /* We won't output out of range samples so don't bother clipping them. */
-                        RecordAndPlaySinesCallback,
-                        loopbackContext );
-    if( err != paNoError ) goto error;
-
-    err = Pa_StartStream( inStream );
-    if( err != paNoError ) goto error;
-
-    timedOut = PaQa_WaitForStream( loopbackContext );
-
-    err = Pa_StopStream( inStream );
-    if( err != paNoError ) goto error;
-
-    err = Pa_CloseStream( inStream );
-    if( err != paNoError ) goto error;
-
-    return timedOut;
-
-error:
-    return err;
-}
-
-/*******************************************************************/
-static int RecordAndPlayBlockingIO( PaStream *inStream,
-                                      PaStream *outStream,
-                                      LoopbackContext *loopbackContext
-                                      )
-{
-    int i;
-    float *in = (float *)g_ReadWriteBuffer;
-    float *out = (float *)g_ReadWriteBuffer;
-    PaError err;
-    int done = 0;
-    long available;
-    const long maxPerBuffer = 64;
-    TestParameters *test = loopbackContext->test;
-    long framesPerBuffer = test->framesPerBuffer;
-    if( framesPerBuffer <= 0 )
-    {
-        framesPerBuffer = maxPerBuffer; // bigger values might run past end of recording
-    }
-
-    // Read in audio.
-    err = Pa_ReadStream( inStream, in, framesPerBuffer );
-    // Ignore an overflow on the first read.
-    //if( !((loopbackContext->callbackCount == 0) && (err == paInputOverflowed)) )
-    if( err != paInputOverflowed )
-    {
-        QA_ASSERT_EQUALS( "Pa_ReadStream failed", paNoError, err );
-    }
-    else
-    {
-        loopbackContext->inputOverflowCount += 1;
-    }
-
-
-    // Save in a recording.
-    for( i=0; i<loopbackContext->test->inputParameters.channelCount; i++ )
-    {
-        done |= PaQa_WriteRecording( &loopbackContext->recordings[i],
-                 in + i,
-                 framesPerBuffer,
-                 loopbackContext->test->inputParameters.channelCount );
-    }
-
-    // Synthesize audio.
-    available = Pa_GetStreamWriteAvailable( outStream );
-    if( available > (2*framesPerBuffer) ) available = (2*framesPerBuffer);
-    PaQa_EraseBuffer( out, available, loopbackContext->test->outputParameters.channelCount );
-    for( i=0; i<loopbackContext->test->outputParameters.channelCount; i++ )
-    {
-        PaQa_MixSine( &loopbackContext->generators[i],
-                  out + i,
-                  available,
-                  loopbackContext->test->outputParameters.channelCount );
-    }
-
-    // Write out audio.
-    err = Pa_WriteStream( outStream, out, available );
-    // Ignore an underflow on the first write.
-    //if( !((loopbackContext->callbackCount == 0) && (err == paOutputUnderflowed)) )
-    if( err != paOutputUnderflowed )
-    {
-        QA_ASSERT_EQUALS( "Pa_WriteStream failed", paNoError, err );
-    }
-    else
-    {
-        loopbackContext->outputUnderflowCount += 1;
-    }
-
-
-    loopbackContext->callbackCount += 1;
-
-    return done;
-error:
-    return err;
-}
-
-
-/*******************************************************************/
-/**
- * Open two audio streams with non-blocking IO.
- * Generate sine waves on the output channels and record the input channels.
- * Then close the stream.
- * @return 0 if OK or negative error.
- */
-int PaQa_RunLoopbackHalfDuplexBlockingIO( LoopbackContext *loopbackContext )
-{
-    PaStream *inStream = NULL;
-    PaStream *outStream = NULL;
-    PaError err = 0;
-    TestParameters *test = loopbackContext->test;
-
-    // Use two half duplex streams.
-    err = Pa_OpenStream(
-                        &inStream,
-                        &test->inputParameters,
-                        NULL,
-                        test->sampleRate,
-                        test->framesPerBuffer,
-                        paClipOff, /* we won't output out of range samples so don't bother clipping them */
-                        NULL, // causes non-blocking IO
-                        NULL );
-    if( err != paNoError ) goto error1;
-    err = Pa_OpenStream(
-                        &outStream,
-                        NULL,
-                        &test->outputParameters,
-                        test->sampleRate,
-                        test->framesPerBuffer,
-                        paClipOff, /* we won't output out of range samples so don't bother clipping them */
-                        NULL, // causes non-blocking IO
-                        NULL );
-    if( err != paNoError ) goto error2;
-
-    CopyStreamInfoToLoopbackContext( loopbackContext, inStream, outStream );
-
-    err = Pa_StartStream( outStream );
-    if( err != paNoError ) goto error3;
-
-    err = Pa_StartStream( inStream );
-    if( err != paNoError ) goto error3;
-
-    while( err == 0 )
-    {
-        err = RecordAndPlayBlockingIO( inStream, outStream, loopbackContext );
-        if( err < 0 ) goto error3;
-    }
-
-    err = Pa_StopStream( inStream );
-    if( err != paNoError ) goto error3;
-
-    err = Pa_StopStream( outStream );
-    if( err != paNoError ) goto error3;
-
-    err = Pa_CloseStream( outStream );
-    if( err != paNoError ) goto error2;
-
-    err = Pa_CloseStream( inStream );
-    if( err != paNoError ) goto error1;
-
-
-    return 0;
-
-error3:
-    Pa_CloseStream( outStream );
-error2:
-    Pa_CloseStream( inStream );
-error1:
-    return err;
-}
-
-
-/*******************************************************************/
-/**
- * Open one audio stream with non-blocking IO.
- * Generate sine waves on the output channels and record the input channels.
- * Then close the stream.
- * @return 0 if OK or negative error.
- */
-int PaQa_RunLoopbackFullDuplexBlockingIO( LoopbackContext *loopbackContext )
-{
-    PaStream *stream = NULL;
-    PaError err = 0;
-    TestParameters *test = loopbackContext->test;
-
-    // Use one full duplex stream.
-    err = Pa_OpenStream(
-                        &stream,
-                        &test->inputParameters,
-                        &test->outputParameters,
-                        test->sampleRate,
-                        test->framesPerBuffer,
-                        paClipOff, /* we won't output out of range samples so don't bother clipping them */
-                        NULL, // causes non-blocking IO
-                        NULL );
-    if( err != paNoError ) goto error1;
-
-    CopyStreamInfoToLoopbackContext( loopbackContext, stream, stream );
-
-    err = Pa_StartStream( stream );
-    if( err != paNoError ) goto error2;
-
-    while( err == 0 )
-    {
-        err = RecordAndPlayBlockingIO( stream, stream, loopbackContext );
-        if( err < 0 ) goto error2;
-    }
-
-    err = Pa_StopStream( stream );
-    if( err != paNoError ) goto error2;
-
-
-    err = Pa_CloseStream( stream );
-    if( err != paNoError ) goto error1;
-
-
-    return 0;
-
-error2:
-    Pa_CloseStream( stream );
-error1:
-    return err;
-}
-
-
-/*******************************************************************/
-/**
- * Run some kind of loopback test.
- * @return 0 if OK or negative error.
- */
-int PaQa_RunLoopback( LoopbackContext *loopbackContext )
-{
-    PaError err = 0;
-    TestParameters *test = loopbackContext->test;
-
-
-    if( test->flags & PAQA_FLAG_TWO_STREAMS )
-    {
-        if( test->flags & PAQA_FLAG_USE_BLOCKING_IO )
-        {
-            err = PaQa_RunLoopbackHalfDuplexBlockingIO( loopbackContext );
-        }
-        else
-        {
-            err = PaQa_RunLoopbackHalfDuplex( loopbackContext );
-        }
-    }
-    else
-    {
-        if( test->flags & PAQA_FLAG_USE_BLOCKING_IO )
-        {
-            err = PaQa_RunLoopbackFullDuplexBlockingIO( loopbackContext );
-        }
-        else
-        {
-            err = PaQa_RunLoopbackFullDuplex( loopbackContext );
-        }
-    }
-
-    if( err != paNoError )
-    {
-        printf("PortAudio error = %s\n", Pa_GetErrorText( err ) );
-    }
-    return err;
-}
-
-/*******************************************************************/
-static int PaQa_SaveTestResultToWaveFile( UserOptions *userOptions, PaQaRecording *recording )
-{
-    if( userOptions->saveBadWaves )
-    {
-        char filename[256];
-#ifdef WIN32
-        _snprintf( filename, sizeof(filename), "%s\\paloopback_%d.wav", userOptions->waveFilePath, userOptions->waveFileCount++ );
-#else
-        snprintf( filename, sizeof(filename), "%s/paloopback_%d.wav", userOptions->waveFilePath, userOptions->waveFileCount++ );
-#endif
-        printf( "\"%s\", ", filename );
-        return PaQa_SaveRecordingToWaveFile( recording, filename );
-    }
-    return 0;
-}
-
-/*******************************************************************/
-static int PaQa_SetupLoopbackContext( LoopbackContext *loopbackContextPtr, TestParameters *testParams )
-{
-    int i;
-    // Setup loopback context.
-    memset( loopbackContextPtr, 0, sizeof(LoopbackContext) );
-    loopbackContextPtr->test = testParams;
-    for( i=0; i<testParams->samplesPerFrame; i++ )
-    {
-        int err = PaQa_InitializeRecording( &loopbackContextPtr->recordings[i], testParams->maxFrames, testParams->sampleRate );
-        QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", paNoError, err );
-    }
-    for( i=0; i<testParams->samplesPerFrame; i++ )
-    {
-        PaQa_SetupSineGenerator( &loopbackContextPtr->generators[i], PaQa_GetNthFrequency( testParams->baseFrequency, i ),
-                                testParams->amplitude, testParams->sampleRate );
-    }
-    loopbackContextPtr->minFramesPerBuffer = 0x0FFFFFFF;
-    return 0;
-error:
-    return -1;
-}
-
-/*******************************************************************/
-static void PaQa_TeardownLoopbackContext( LoopbackContext *loopbackContextPtr )
-{
-    int i;
-    if( loopbackContextPtr->test != NULL )
-    {
-        for( i=0; i<loopbackContextPtr->test->samplesPerFrame; i++ )
-        {
-            PaQa_TerminateRecording( &loopbackContextPtr->recordings[i] );
-        }
-    }
-}
-
-/*******************************************************************/
-static void PaQa_PrintShortErrorReport( PaQaAnalysisResult *analysisResultPtr, int channel )
-{
-    printf("channel %d ", channel);
-    if( analysisResultPtr->popPosition > 0 )
-    {
-        printf("POP %0.3f at %d, ", (double)analysisResultPtr->popAmplitude, (int)analysisResultPtr->popPosition );
-    }
-    else
-    {
-        if( analysisResultPtr->addedFramesPosition > 0 )
-        {
-            printf("ADD %d at %d ", (int)analysisResultPtr->numAddedFrames, (int)analysisResultPtr->addedFramesPosition );
-        }
-
-        if( analysisResultPtr->droppedFramesPosition > 0 )
-        {
-            printf("DROP %d at %d ", (int)analysisResultPtr->numDroppedFrames, (int)analysisResultPtr->droppedFramesPosition );
-        }
-    }
-}
-
-/*******************************************************************/
-static void PaQa_PrintFullErrorReport( PaQaAnalysisResult *analysisResultPtr, int channel )
-{
-    printf("\n=== Loopback Analysis ===================\n");
-    printf("             channel: %d\n", channel );
-    printf("             latency: %10.3f\n", analysisResultPtr->latency );
-    printf("      amplitudeRatio: %10.3f\n", (double)analysisResultPtr->amplitudeRatio );
-    printf("         popPosition: %10.3f\n", (double)analysisResultPtr->popPosition );
-    printf("        popAmplitude: %10.3f\n", (double)analysisResultPtr->popAmplitude );
-    printf("    num added frames: %10.3f\n", analysisResultPtr->numAddedFrames );
-    printf("     added frames at: %10.3f\n", analysisResultPtr->addedFramesPosition );
-    printf("  num dropped frames: %10.3f\n", analysisResultPtr->numDroppedFrames );
-    printf("   dropped frames at: %10.3f\n", analysisResultPtr->droppedFramesPosition );
-}
-
-/*******************************************************************/
-/**
- * Test loopback connection using the given parameters.
- * @return number of channels with glitches, or negative error.
- */
-static int PaQa_SingleLoopBackTest( UserOptions *userOptions, TestParameters *testParams )
-{
-    int i;
-    LoopbackContext loopbackContext;
-    PaError err = paNoError;
-    PaQaTestTone testTone;
-    PaQaAnalysisResult analysisResult;
-    int numBadChannels = 0;
-
-    printf("| %5d | %6d | ", ((int)(testParams->sampleRate+0.5)), testParams->framesPerBuffer );
-    fflush(stdout);
-
-    testTone.samplesPerFrame = testParams->samplesPerFrame;
-    testTone.sampleRate = testParams->sampleRate;
-    testTone.amplitude = testParams->amplitude;
-    testTone.startDelay = 0;
-
-    err = PaQa_SetupLoopbackContext( &loopbackContext, testParams );
-    if( err ) return err;
-
-    err = PaQa_RunLoopback( &loopbackContext );
-    QA_ASSERT_TRUE("loopback did not run", (loopbackContext.callbackCount > 1) );
-
-    printf( "%7.2f %7.2f %7.2f | ",
-           loopbackContext.streamInfoInputLatency * 1000.0,
-           loopbackContext.streamInfoOutputLatency * 1000.0,
-           (loopbackContext.streamInfoInputLatency + loopbackContext.streamInfoOutputLatency) * 1000.0
-           );
-
-    printf( "%4d/%4d/%4d, %4d/%4d/%4d | ",
-           loopbackContext.inputOverflowCount,
-           loopbackContext.inputUnderflowCount,
-           loopbackContext.inputBufferCount,
-           loopbackContext.outputOverflowCount,
-           loopbackContext.outputUnderflowCount,
-           loopbackContext.outputBufferCount
-           );
-
-    // Analyse recording to detect glitches.
-    for( i=0; i<testParams->samplesPerFrame; i++ )
-    {
-        double freq = PaQa_GetNthFrequency( testParams->baseFrequency, i );
-        testTone.frequency = freq;
-
-        PaQa_AnalyseRecording(  &loopbackContext.recordings[i], &testTone, &analysisResult );
-
-        if( i==0 )
-        {
-            double latencyMSec;
-
-            printf( "%4d-%4d | ",
-                   loopbackContext.minFramesPerBuffer,
-                   loopbackContext.maxFramesPerBuffer
-                   );
-
-            latencyMSec = 1000.0 * analysisResult.latency / testParams->sampleRate;
-            printf("%7.2f | ", latencyMSec );
-
-        }
-
-        if( analysisResult.valid )
-        {
-            int badChannel = ( (analysisResult.popPosition > 0)
-                       || (analysisResult.addedFramesPosition > 0)
-                       || (analysisResult.droppedFramesPosition > 0) );
-
-            if( badChannel )
-            {
-                if( userOptions->verbose )
-                {
-                    PaQa_PrintFullErrorReport( &analysisResult, i );
-                }
-                else
-                {
-                    PaQa_PrintShortErrorReport( &analysisResult, i );
-                }
-                PaQa_SaveTestResultToWaveFile( userOptions, &loopbackContext.recordings[i] );
-            }
-            numBadChannels += badChannel;
-        }
-        else
-        {
-            printf( "[%d] No or low signal, ampRatio = %f", i, analysisResult.amplitudeRatio );
-            numBadChannels += 1;
-        }
-
-    }
-    if( numBadChannels == 0 )
-    {
-        printf( "OK" );
-    }
-
-    // Print the # errors so far to make it easier to see where the error occurred.
-    printf( " - #errs = %d\n", g_testsFailed );
-
-    PaQa_TeardownLoopbackContext( &loopbackContext );
-    if( numBadChannels > 0 )
-    {
-        g_testsFailed += 1;
-    }
-    return numBadChannels;
-
-error:
-    PaQa_TeardownLoopbackContext( &loopbackContext );
-    printf( "\n" );
-    g_testsFailed += 1;
-    return err;
-}
-
-/*******************************************************************/
-static void PaQa_SetDefaultTestParameters( TestParameters *testParamsPtr, PaDeviceIndex inputDevice, PaDeviceIndex outputDevice )
-{
-    memset( testParamsPtr, 0, sizeof(TestParameters) );
-
-    testParamsPtr->samplesPerFrame = 2;
-    testParamsPtr->amplitude = 0.5;
-    testParamsPtr->sampleRate = 44100;
-    testParamsPtr->maxFrames = (int) (PAQA_TEST_DURATION * testParamsPtr->sampleRate);
-    testParamsPtr->framesPerBuffer = DEFAULT_FRAMES_PER_BUFFER;
-    testParamsPtr->baseFrequency = 200.0;
-    testParamsPtr->flags = PAQA_FLAG_TWO_STREAMS;
-    testParamsPtr->streamFlags = paClipOff; /* we won't output out of range samples so don't bother clipping them */
-
-    testParamsPtr->inputParameters.device = inputDevice;
-    testParamsPtr->inputParameters.sampleFormat = paFloat32;
-    testParamsPtr->inputParameters.channelCount = testParamsPtr->samplesPerFrame;
-    testParamsPtr->inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputDevice )->defaultLowInputLatency;
-    //testParamsPtr->inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputDevice )->defaultHighInputLatency;
-
-    testParamsPtr->outputParameters.device = outputDevice;
-    testParamsPtr->outputParameters.sampleFormat = paFloat32;
-    testParamsPtr->outputParameters.channelCount = testParamsPtr->samplesPerFrame;
-    testParamsPtr->outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputDevice )->defaultLowOutputLatency;
-    //testParamsPtr->outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputDevice )->defaultHighOutputLatency;
-}
-
-/*******************************************************************/
-static void PaQa_OverrideTestParameters( TestParameters *testParamsPtr,  UserOptions *userOptions )
-{
-    // Check to see if a specific value was requested.
-    if( userOptions->sampleRate >= 0 )
-    {
-        testParamsPtr->sampleRate = userOptions->sampleRate;
-        testParamsPtr->maxFrames = (int) (PAQA_TEST_DURATION * testParamsPtr->sampleRate);
-    }
-    if( userOptions->framesPerBuffer >= 0 )
-    {
-        testParamsPtr->framesPerBuffer = userOptions->framesPerBuffer;
-    }
-    if( userOptions->inputLatency >= 0 )
-    {
-        testParamsPtr->inputParameters.suggestedLatency = userOptions->inputLatency * 0.001;
-    }
-    if( userOptions->outputLatency >= 0 )
-    {
-        testParamsPtr->outputParameters.suggestedLatency = userOptions->outputLatency * 0.001;
-    }
-    printf( "   Running with suggested latency (msec): input = %5.2f, out = %5.2f\n",
-        (testParamsPtr->inputParameters.suggestedLatency * 1000.0),
-        (testParamsPtr->outputParameters.suggestedLatency * 1000.0) );
-}
-
-/*******************************************************************/
-/**
- * Run a series of tests on this loopback connection.
- * @return number of bad channel results
- */
-static int PaQa_AnalyzeLoopbackConnection( UserOptions *userOptions, PaDeviceIndex inputDevice, PaDeviceIndex outputDevice )
-{
-    int iFlags;
-    int iRate;
-    int iSize;
-    int iFormat;
-    int savedValue;
-    TestParameters testParams;
-    const PaDeviceInfo *inputDeviceInfo = Pa_GetDeviceInfo( inputDevice );
-    const PaDeviceInfo *outputDeviceInfo = Pa_GetDeviceInfo( outputDevice );
-    int totalBadChannels = 0;
-
-    // test half duplex first because it is more likely to work.
-    int flagSettings[] = { PAQA_FLAG_TWO_STREAMS, 0 };
-    int numFlagSettings = (sizeof(flagSettings)/sizeof(int));
-
-    double sampleRates[] = { 8000.0, 11025.0, 16000.0, 22050.0, 32000.0, 44100.0, 48000.0, 96000.0 };
-    int numRates = (sizeof(sampleRates)/sizeof(double));
-
-    // framesPerBuffer==0 means PA decides on the buffer size.
-    int framesPerBuffers[] = { 0, 16, 32, 40, 64, 100, 128, 256, 512, 1024 };
-    int numBufferSizes = (sizeof(framesPerBuffers)/sizeof(int));
-
-    PaSampleFormat sampleFormats[] = { paFloat32, paUInt8, paInt8, paInt16, paInt32 };
-    const char *sampleFormatNames[] = { "paFloat32", "paUInt8", "paInt8", "paInt16", "paInt32" };
-    int numSampleFormats = (sizeof(sampleFormats)/sizeof(PaSampleFormat));
-
-    printf( "=============== Analysing Loopback %d to %d =====================\n", outputDevice, inputDevice  );
-    printf( "   Devices: %s => %s\n", outputDeviceInfo->name, inputDeviceInfo->name);
-
-    PaQa_SetDefaultTestParameters( &testParams, inputDevice, outputDevice );
-
-    PaQa_OverrideTestParameters( &testParams, userOptions );
-
-    // Loop though combinations of audio parameters.
-    for( iFlags=0; iFlags<numFlagSettings; iFlags++ )
-    {
-        int numRuns = 0;
-
-        testParams.flags = flagSettings[iFlags];
-        printf( "\n************ Mode = %s ************\n",
-               (( testParams.flags & 1 ) ? s_FlagOnNames[0] : s_FlagOffNames[0]) );
-
-        printf("|-   requested  -|-  stream info latency  -|- measured ------------------------------\n");
-        printf("|-sRate-|-fr/buf-|- in    - out   - total -|- over/under/calls for in, out -|- frm/buf -|-latency-|- channel results -\n");
-
-        // Loop though various sample rates.
-        if( userOptions->sampleRate < 0 )
-        {
-            savedValue = testParams.sampleRate;
-            for( iRate=0; iRate<numRates; iRate++ )
-            {
-                int numBadChannels;
-
-                // SAMPLE RATE
-                testParams.sampleRate = sampleRates[iRate];
-                testParams.maxFrames = (int) (PAQA_TEST_DURATION * testParams.sampleRate);
-
-                numBadChannels = PaQa_SingleLoopBackTest( userOptions, &testParams );
-                totalBadChannels += numBadChannels;
-            }
-            testParams.sampleRate = savedValue;
-            testParams.maxFrames = (int) (PAQA_TEST_DURATION * testParams.sampleRate);
-            printf( "\n" );
-            numRuns += 1;
-        }
-
-        // Loop through various buffer sizes.
-        if( userOptions->framesPerBuffer < 0 )
-        {
-            savedValue = testParams.framesPerBuffer;
-            for( iSize=0; iSize<numBufferSizes; iSize++ )
-            {
-                int numBadChannels;
-
-                // BUFFER SIZE
-                testParams.framesPerBuffer = framesPerBuffers[iSize];
-
-                numBadChannels = PaQa_SingleLoopBackTest( userOptions, &testParams );
-                totalBadChannels += numBadChannels;
-            }
-            testParams.framesPerBuffer = savedValue;
-            printf( "\n" );
-            numRuns += 1;
-        }
-        // Run one with single parameters in case we did not do a series.
-        if( numRuns == 0 )
-        {
-            int numBadChannels = PaQa_SingleLoopBackTest( userOptions, &testParams );
-            totalBadChannels += numBadChannels;
-        }
-    }
-
-    printf("\nTest Sample Formats using Half Duplex IO -----\n" );
-
-    PaQa_SetDefaultTestParameters( &testParams, inputDevice, outputDevice );
-    testParams.flags = PAQA_FLAG_TWO_STREAMS;
-    for( iFlags= 0; iFlags<4; iFlags++ )
-    {
-        // Cycle through combinations of flags.
-        testParams.streamFlags = 0;
-        if( iFlags & 1 ) testParams.streamFlags |= paClipOff;
-        if( iFlags & 2 ) testParams.streamFlags |= paDitherOff;
-
-        for( iFormat=0; iFormat<numSampleFormats; iFormat++ )
-        {
-            int numBadChannels;
-            PaSampleFormat format = sampleFormats[ iFormat ];
-            testParams.inputParameters.sampleFormat = format;
-            testParams.outputParameters.sampleFormat = format;
-            printf("Sample format = %d = %s, PaStreamFlags = 0x%02X\n", (int) format, sampleFormatNames[iFormat], (unsigned int) testParams.streamFlags );
-            numBadChannels = PaQa_SingleLoopBackTest( userOptions, &testParams );
-            totalBadChannels += numBadChannels;
-        }
-    }
-    printf( "\n" );
-    printf( "****************************************\n");
-
-    return totalBadChannels;
-}
-
-/*******************************************************************/
-int PaQa_CheckForClippedLoopback( LoopbackContext *loopbackContextPtr )
-{
-    int clipped = 0;
-    TestParameters *testParamsPtr = loopbackContextPtr->test;
-
-    // Start in the middle assuming past latency.
-    int startFrame = testParamsPtr->maxFrames/2;
-    int numFrames = testParamsPtr->maxFrames/2;
-
-    // Check to see if the signal is clipped.
-    double amplitudeLeft = PaQa_MeasureSineAmplitudeBySlope( &loopbackContextPtr->recordings[0],
-                                                            testParamsPtr->baseFrequency, testParamsPtr->sampleRate,
-                                                            startFrame, numFrames );
-    double gainLeft = amplitudeLeft / testParamsPtr->amplitude;
-    double amplitudeRight = PaQa_MeasureSineAmplitudeBySlope( &loopbackContextPtr->recordings[1],
-                                                             testParamsPtr->baseFrequency, testParamsPtr->sampleRate,
-                                                             startFrame, numFrames );
-    double gainRight = amplitudeLeft / testParamsPtr->amplitude;
-    printf("   Loop gain: left = %f, right = %f\n", gainLeft, gainRight );
-
-    if( (amplitudeLeft > 1.0 ) || (amplitudeRight > 1.0) )
-    {
-        printf("ERROR - loop gain is too high. Should be around than 1.0. Please lower output level and/or input gain.\n" );
-        clipped = 1;
-    }
-    return clipped;
-}
-
-/*******************************************************************/
-int PaQa_MeasureBackgroundNoise( LoopbackContext *loopbackContextPtr, double *rmsPtr )
-{
-    int result = 0;
-    *rmsPtr = 0.0;
-    // Rewind so we can record some input.
-    loopbackContextPtr->recordings[0].numFrames = 0;
-    loopbackContextPtr->recordings[1].numFrames = 0;
-    result = PaQa_RunInputOnly( loopbackContextPtr );
-    if( result == 0 )
-    {
-        double leftRMS = PaQa_MeasureRootMeanSquare( loopbackContextPtr->recordings[0].buffer,
-                                                    loopbackContextPtr->recordings[0].numFrames );
-        double rightRMS = PaQa_MeasureRootMeanSquare( loopbackContextPtr->recordings[1].buffer,
-                                                     loopbackContextPtr->recordings[1].numFrames );
-        *rmsPtr = (leftRMS + rightRMS) / 2.0;
-    }
-    return result;
-}
-
-/*******************************************************************/
-/**
- * Output a sine wave then try to detect it on input.
- *
- * @return 1 if loopback connected, 0 if not, or negative error.
- */
-int PaQa_CheckForLoopBack( UserOptions *userOptions, PaDeviceIndex inputDevice, PaDeviceIndex outputDevice )
-{
-    TestParameters testParams;
-    LoopbackContext loopbackContext;
-    const PaDeviceInfo *inputDeviceInfo;
-    const PaDeviceInfo *outputDeviceInfo;
-    PaError err = paNoError;
-    double minAmplitude;
-    int loopbackIsConnected;
-    int startFrame, numFrames;
-    double magLeft, magRight;
-
-    inputDeviceInfo = Pa_GetDeviceInfo( inputDevice );
-    if( inputDeviceInfo == NULL )
-    {
-        printf("ERROR - Pa_GetDeviceInfo for input returned NULL.\n");
-        return paInvalidDevice;
-    }
-    if( inputDeviceInfo->maxInputChannels < 2 )
-    {
-        return 0;
-    }
-
-    outputDeviceInfo = Pa_GetDeviceInfo( outputDevice );
-    if( outputDeviceInfo == NULL )
-    {
-        printf("ERROR - Pa_GetDeviceInfo for output returned NULL.\n");
-        return paInvalidDevice;
-    }
-    if( outputDeviceInfo->maxOutputChannels < 2 )
-    {
-        return 0;
-    }
-
-    printf( "Look for loopback cable between \"%s\" => \"%s\"\n", outputDeviceInfo->name, inputDeviceInfo->name);
-
-    printf( "   Default suggested input latency (msec): low = %5.2f, high = %5.2f\n",
-        (inputDeviceInfo->defaultLowInputLatency * 1000.0),
-        (inputDeviceInfo->defaultHighInputLatency * 1000.0) );
-    printf( "   Default suggested output latency (msec): low = %5.2f, high = %5.2f\n",
-        (outputDeviceInfo->defaultLowOutputLatency * 1000.0),
-        (outputDeviceInfo->defaultHighOutputLatency * 1000.0) );
-
-    PaQa_SetDefaultTestParameters( &testParams, inputDevice, outputDevice );
-
-    PaQa_OverrideTestParameters( &testParams, userOptions );
-
-    testParams.maxFrames = (int) (LOOPBACK_DETECTION_DURATION_SECONDS * testParams.sampleRate);
-    minAmplitude = testParams.amplitude / 4.0;
-
-    // Check to see if the selected formats are supported.
-    if( Pa_IsFormatSupported( &testParams.inputParameters, NULL, testParams.sampleRate ) != paFormatIsSupported )
-    {
-        printf( "Input not supported for this format!\n" );
-        return 0;
-    }
-    if( Pa_IsFormatSupported( NULL, &testParams.outputParameters, testParams.sampleRate ) != paFormatIsSupported )
-    {
-        printf( "Output not supported for this format!\n" );
-        return 0;
-    }
-
-    PaQa_SetupLoopbackContext( &loopbackContext, &testParams );
-
-    if( inputDevice == outputDevice )
-    {
-        // Use full duplex if checking for loopback on one device.
-        testParams.flags &= ~PAQA_FLAG_TWO_STREAMS;
-    }
-    else
-    {
-        // Use half duplex if checking for loopback on two different device.
-        testParams.flags = PAQA_FLAG_TWO_STREAMS;
-    }
-    err = PaQa_RunLoopback( &loopbackContext );
-    QA_ASSERT_TRUE("loopback detection callback did not run", (loopbackContext.callbackCount > 1) );
-
-    // Analyse recording to see if we captured the output.
-    // Start in the middle assuming past latency.
-    startFrame = testParams.maxFrames/2;
-    numFrames = testParams.maxFrames/2;
-    magLeft = PaQa_CorrelateSine( &loopbackContext.recordings[0],
-                                    loopbackContext.generators[0].frequency,
-                                    testParams.sampleRate,
-                                    startFrame, numFrames, NULL );
-    magRight = PaQa_CorrelateSine( &loopbackContext.recordings[1],
-                                    loopbackContext.generators[1].frequency,
-                                    testParams.sampleRate,
-                                    startFrame, numFrames, NULL );
-    printf("   Amplitudes: left = %f, right = %f\n", magLeft, magRight );
-
-    // Check for backwards cable.
-    loopbackIsConnected = ((magLeft > minAmplitude) && (magRight > minAmplitude));
-
-    if( !loopbackIsConnected )
-    {
-        double magLeftReverse = PaQa_CorrelateSine( &loopbackContext.recordings[0],
-                                                   loopbackContext.generators[1].frequency,
-                                                   testParams.sampleRate,
-                                                   startFrame, numFrames, NULL );
-
-        double magRightReverse = PaQa_CorrelateSine( &loopbackContext.recordings[1],
-                                                    loopbackContext.generators[0].frequency,
-                                                    testParams.sampleRate,
-                                                    startFrame, numFrames, NULL );
-
-        if ((magLeftReverse > minAmplitude) && (magRightReverse>minAmplitude))
-        {
-            printf("ERROR - You seem to have the left and right channels swapped on the loopback cable!\n");
-        }
-    }
-    else
-    {
-        double rms = 0.0;
-        if( PaQa_CheckForClippedLoopback( &loopbackContext ) )
-        {
-            // Clipped so don't use this loopback.
-            loopbackIsConnected = 0;
-        }
-
-        err = PaQa_MeasureBackgroundNoise( &loopbackContext, &rms );
-        printf("   Background noise = %f\n", rms );
-        if( err )
-        {
-            printf("ERROR - Could not measure background noise on this input!\n");
-            loopbackIsConnected = 0;
-        }
-        else if( rms > MAX_BACKGROUND_NOISE_RMS )
-        {
-            printf("ERROR - There is too much background noise on this input!\n");
-            loopbackIsConnected = 0;
-        }
-    }
-
-    PaQa_TeardownLoopbackContext( &loopbackContext );
-    return loopbackIsConnected;
-
-error:
-    PaQa_TeardownLoopbackContext( &loopbackContext );
-    return err;
-}
-
-/*******************************************************************/
-/**
- * If there is a loopback connection then run the analysis.
- */
-static int CheckLoopbackAndScan( UserOptions *userOptions,
-                                PaDeviceIndex iIn, PaDeviceIndex iOut )
-{
-    int loopbackConnected = PaQa_CheckForLoopBack( userOptions, iIn, iOut );
-    if( loopbackConnected > 0 )
-    {
-        PaQa_AnalyzeLoopbackConnection( userOptions, iIn, iOut );
-        return 1;
-    }
-    return 0;
-}
-
-/*******************************************************************/
-/**
- * Scan every combination of output to input device.
- * If a loopback is found the analyse the combination.
- * The scan can be overridden using the -i and -o command line options.
- */
-static int ScanForLoopback(UserOptions *userOptions)
-{
-    PaDeviceIndex iIn,iOut;
-    int  numLoopbacks = 0;
-    int  numDevices;
-    numDevices = Pa_GetDeviceCount();
-
-    // If both devices are specified then just use that combination.
-    if ((userOptions->inputDevice >= 0) && (userOptions->outputDevice >= 0))
-    {
-        numLoopbacks += CheckLoopbackAndScan( userOptions, userOptions->inputDevice, userOptions->outputDevice );
-    }
-    else if (userOptions->inputDevice >= 0)
-    {
-        // Just scan for output.
-        for( iOut=0; iOut<numDevices; iOut++ )
-        {
-            numLoopbacks += CheckLoopbackAndScan( userOptions, userOptions->inputDevice, iOut );
-        }
-    }
-    else if (userOptions->outputDevice >= 0)
-    {
-        // Just scan for input.
-        for( iIn=0; iIn<numDevices; iIn++ )
-        {
-            numLoopbacks += CheckLoopbackAndScan( userOptions, iIn, userOptions->outputDevice );
-        }
-    }
-    else
-    {
-        // Scan both.
-        for( iOut=0; iOut<numDevices; iOut++ )
-        {
-            for( iIn=0; iIn<numDevices; iIn++ )
-            {
-                numLoopbacks += CheckLoopbackAndScan( userOptions, iIn, iOut );
-            }
-        }
-    }
-    QA_ASSERT_TRUE( "No good loopback cable found.", (numLoopbacks > 0) );
-    return numLoopbacks;
-
-error:
-    return -1;
-}
-
-/*==========================================================================================*/
-int TestSampleFormatConversion( void )
-{
-    int i;
-    const float floatInput[] = { 1.0, 0.5, -0.5, -1.0 };
-
-    const char charInput[] = { 127, 64, -64, -128 };
-    const unsigned char ucharInput[] = { 255, 128+64, 64, 0 };
-    const short shortInput[] = { 32767, 32768/2, -32768/2, -32768 };
-    const int intInput[] = { 2147483647, 2147483647/2, -1073741824 /*-2147483648/2 doesn't work in msvc*/, -2147483648 };
-
-    float floatOutput[4];
-    short shortOutput[4];
-    int intOutput[4];
-    unsigned char ucharOutput[4];
-    char charOutput[4];
-
-    QA_ASSERT_EQUALS("int must be 32-bit", 4, (int) sizeof(int) );
-    QA_ASSERT_EQUALS("short must be 16-bit", 2, (int) sizeof(short) );
-
-    // from Float ======
-    PaQa_ConvertFromFloat( floatInput, 4, paUInt8, ucharOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE_INT( "paFloat32 -> paUInt8 -> error", ucharInput[i], ucharOutput[i], 1 );
-    }
-
-    PaQa_ConvertFromFloat( floatInput, 4, paInt8, charOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE_INT( "paFloat32 -> paInt8 -> error", charInput[i], charOutput[i], 1 );
-    }
-
-    PaQa_ConvertFromFloat( floatInput, 4, paInt16, shortOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE_INT( "paFloat32 -> paInt16 error", shortInput[i], shortOutput[i], 1 );
-    }
-
-    PaQa_ConvertFromFloat( floatInput, 4, paInt32, intOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE_INT( "paFloat32 -> paInt32 error", intInput[i], intOutput[i], 0x00010000 );
-    }
-
-
-    // to Float ======
-    memset( floatOutput, 0, sizeof(floatOutput) );
-    PaQa_ConvertToFloat( ucharInput, 4, paUInt8, floatOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE( "paUInt8 -> paFloat32 error", floatInput[i], floatOutput[i], 0.01 );
-    }
-
-    memset( floatOutput, 0, sizeof(floatOutput) );
-    PaQa_ConvertToFloat( charInput, 4, paInt8, floatOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE( "paInt8 -> paFloat32 error", floatInput[i], floatOutput[i], 0.01 );
-    }
-
-    memset( floatOutput, 0, sizeof(floatOutput) );
-    PaQa_ConvertToFloat( shortInput, 4, paInt16, floatOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE( "paInt16 -> paFloat32 error", floatInput[i], floatOutput[i], 0.001 );
-    }
-
-    memset( floatOutput, 0, sizeof(floatOutput) );
-    PaQa_ConvertToFloat( intInput, 4, paInt32, floatOutput );
-    for( i=0; i<4; i++ )
-    {
-        QA_ASSERT_CLOSE( "paInt32 -> paFloat32 error", floatInput[i], floatOutput[i], 0.00001 );
-    }
-
-    return 0;
-
-error:
-    return -1;
-}
-
-
-/*******************************************************************/
-void usage( const char *name )
-{
-    printf("%s [-i# -o# -l# -r# -s# -m -w -dDir]\n", name);
-    printf("  -i# - Input device ID. Will scan for loopback cable if not specified.\n");
-    printf("  -o# - Output device ID. Will scan for loopback if not specified.\n");
-    printf("  -l# - Latency for both input and output in milliseconds.\n");
-    printf("  --inputLatency # Input latency in milliseconds.\n");
-    printf("  --outputLatency # Output latency in milliseconds.\n");
-    printf("  -r# - Sample Rate in Hz.  Will use multiple common rates if not specified.\n");
-    printf("  -s# - Size of callback buffer in frames, framesPerBuffer. Will use common values if not specified.\n");
-    printf("  -w  - Save bad recordings in a WAV file.\n");
-    printf("  -dDir - Path for Directory for WAV files. Default is current directory.\n");
-    printf("  -m  - Just test the DSP Math code and not the audio devices.\n");
-    printf("  -v  - Verbose reports.\n");
-}
-
-/*******************************************************************/
-int main( int argc, char **argv )
-{
-    int i;
-    UserOptions userOptions;
-    int result = 0;
-    int justMath = 0;
-    char *executableName = argv[0];
-
-    printf("PortAudio LoopBack Test built " __DATE__ " at " __TIME__ "\n");
-
-    if( argc > 1 ){
-        printf("running with arguments:");
-        for(i=1; i < argc; ++i )
-            printf(" %s", argv[i] );
-        printf("\n");
-    }else{
-        printf("running with no arguments\n");
-    }
-
-    memset(&userOptions, 0, sizeof(userOptions));
-    userOptions.inputDevice = paNoDevice;
-    userOptions.outputDevice = paNoDevice;
-    userOptions.sampleRate = -1;
-    userOptions.framesPerBuffer = -1;
-    userOptions.inputLatency = -1;
-    userOptions.outputLatency = -1;
-    userOptions.waveFilePath = ".";
-
-    // Process arguments. Skip name of executable.
-    i = 1;
-    while( i<argc )
-    {
-        char *arg = argv[i];
-        if( arg[0] == '-' )
-        {
-            switch(arg[1])
-            {
-                case 'i':
-                    userOptions.inputDevice = atoi(&arg[2]);
-                    break;
-                case 'o':
-                    userOptions.outputDevice = atoi(&arg[2]);
-                    break;
-                case 'l':
-                    userOptions.inputLatency = userOptions.outputLatency = atoi(&arg[2]);
-                    break;
-                case 'r':
-                    userOptions.sampleRate = atoi(&arg[2]);
-                    break;
-                case 's':
-                    userOptions.framesPerBuffer = atoi(&arg[2]);
-                    break;
-
-                case 'm':
-                    printf("Option -m set so just testing math and not the audio devices.\n");
-                    justMath = 1;
-                    break;
-
-                case 'w':
-                    userOptions.saveBadWaves = 1;
-                    break;
-                case 'd':
-                    userOptions.waveFilePath = &arg[2];
-                    break;
-
-                case 'v':
-                    userOptions.verbose = 1;
-                    break;
-
-                case 'h':
-                    usage( executableName );
-                    exit(0);
-                    break;
-
-                case '-':
-                {
-                    if( strcmp( &arg[2], "inputLatency" ) == 0 )
-                    {
-                        i += 1;
-                        userOptions.inputLatency = atoi(argv[i]);
-                    }
-                    else if( strcmp( &arg[2], "outputLatency" ) == 0 )
-                    {
-                        i += 1;
-                        userOptions.outputLatency = atoi(argv[i]);
-                    }
-                    else
-                    {
-                        printf("Illegal option: %s\n", arg);
-                        usage( executableName );
-                        exit(1);
-                    }
-
-                }
-                    break;
-
-
-                default:
-                    printf("Illegal option: %s\n", arg);
-                    usage( executableName );
-                    exit(1);
-                    break;
-            }
-        }
-        else
-        {
-            printf("Illegal argument: %s\n", arg);
-            usage( executableName );
-            exit(1);
-
-        }
-        i += 1;
-    }
-
-    result = PaQa_TestAnalyzer();
-
-    // Test sample format conversion tool.
-    result = TestSampleFormatConversion();
-
-    if( (result == 0) && (justMath == 0) )
-    {
-        Pa_Initialize();
-        printf( "PortAudio version number = %d\nPortAudio version text = '%s'\n",
-               Pa_GetVersion(), Pa_GetVersionText() );
-        printf( "=============== PortAudio Devices ========================\n" );
-        PaQa_ListAudioDevices();
-        if( Pa_GetDeviceCount() == 0 )
-            printf( "no devices found.\n" );
-
-        printf( "=============== Detect Loopback ==========================\n" );
-        ScanForLoopback(&userOptions);
-
-        Pa_Terminate();
-    }
-
-    if (g_testsFailed == 0)
-    {
-        printf("PortAudio QA SUCCEEDED! %d tests passed, %d tests failed\n", g_testsPassed, g_testsFailed );
-        return 0;
-
-    }
-    else
-    {
-        printf("PortAudio QA FAILED! %d tests passed, %d tests failed\n", g_testsPassed, g_testsFailed );
-        return 1;
-    }
-}
diff --git a/portaudio/qa/loopback/src/paqa_tools.c b/portaudio/qa/loopback/src/paqa_tools.c
deleted file mode 100644
index 2e44c63..0000000
--- a/portaudio/qa/loopback/src/paqa_tools.c
+++ /dev/null
@@ -1,171 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#include "paqa_tools.h"
-
-
-/*******************************************************************/
-void PaQa_ListAudioDevices(void)
-{
-    int     i, numDevices;
-    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( ", 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;
-    }
-
-}
-
-/*******************************************************************/
-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;
-    }
-
-}
diff --git a/portaudio/qa/loopback/src/paqa_tools.h b/portaudio/qa/loopback/src/paqa_tools.h
deleted file mode 100644
index 77f6a25..0000000
--- a/portaudio/qa/loopback/src/paqa_tools.h
+++ /dev/null
@@ -1,52 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#ifndef _PAQA_TOOLS_H
-#define _PAQA_TOOLS_H
-
-
-#include <stdio.h>
-#include "portaudio.h"
-
-void PaQa_ListAudioDevices(void);
-
-void PaQa_ConvertToFloat( const void *input, int numSamples, PaSampleFormat inFormat, float *output );
-
-void PaQa_ConvertFromFloat( const float *input, int numSamples, PaSampleFormat outFormat, void *output );
-
-#endif /* _PAQA_TOOLS_H */
diff --git a/portaudio/qa/loopback/src/qa_tools.h b/portaudio/qa/loopback/src/qa_tools.h
deleted file mode 100755
index 9b2debd..0000000
--- a/portaudio/qa/loopback/src/qa_tools.h
+++ /dev/null
@@ -1,83 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#ifndef _QA_TOOLS_H
-#define _QA_TOOLS_H
-
-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++;
-
-
-#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++;
-
-#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++;
-
-#define QA_ASSERT_CLOSE_INT( message, expected, actual, tolerance ) \
-    if (abs((expected)-(actual))>(tolerance)) \
-    { \
-        printf( "%s:%d - ERROR - %s, expected %d, got %d, tol=%d\n", __FILE__, __LINE__, message, ((int)(expected)), ((int)(actual)), ((int)(tolerance)) ); \
-        g_testsFailed++; \
-        goto error; \
-    } \
-    else g_testsPassed++;
-
-
-#endif
diff --git a/portaudio/qa/loopback/src/test_audio_analyzer.c b/portaudio/qa/loopback/src/test_audio_analyzer.c
deleted file mode 100644
index 82fa859..0000000
--- a/portaudio/qa/loopback/src/test_audio_analyzer.c
+++ /dev/null
@@ -1,718 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include "qa_tools.h"
-#include "audio_analyzer.h"
-#include "test_audio_analyzer.h"
-#include "write_wav.h"
-#include "biquad_filter.h"
-
-#define FRAMES_PER_BLOCK  (64)
-#define PRINT_REPORTS  0
-
-#define TEST_SAVED_WAVE  (0)
-
-/*==========================================================================================*/
-/**
- * Detect a single tone.
- */
-static int TestSingleMonoTone( void )
-{
-    int result = 0;
-    PaQaSineGenerator generator;
-    PaQaRecording     recording;
-    float buffer[FRAMES_PER_BLOCK];
-    double sampleRate = 44100.0;
-    int maxFrames = ((int)sampleRate) * 1;
-    int samplesPerFrame = 1;
-    int stride = 1;
-    int done = 0;
-
-    double freq = 234.5;
-    double amp = 0.5;
-
-    double mag1, mag2;
-
-    // Setup a sine oscillator.
-    PaQa_SetupSineGenerator( &generator, freq, amp, sampleRate );
-
-    result = PaQa_InitializeRecording( &recording, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    done = 0;
-    while (!done)
-    {
-        PaQa_EraseBuffer( buffer, FRAMES_PER_BLOCK, samplesPerFrame );
-        PaQa_MixSine( &generator, buffer, FRAMES_PER_BLOCK, stride );
-        done = PaQa_WriteRecording( &recording, buffer, FRAMES_PER_BLOCK, samplesPerFrame );
-    }
-
-    mag1 = PaQa_CorrelateSine( &recording, freq, sampleRate, 0, recording.numFrames, NULL );
-    QA_ASSERT_CLOSE( "exact frequency match", amp, mag1, 0.01 );
-
-    mag2 = PaQa_CorrelateSine( &recording, freq * 1.23, sampleRate, 0, recording.numFrames, NULL );
-    QA_ASSERT_CLOSE( "wrong frequency", 0.0, mag2, 0.01 );
-
-    PaQa_TerminateRecording( &recording );
-    return 0;
-
-error:
-    PaQa_TerminateRecording( &recording);
-    return 1;
-
-}
-
-/*==========================================================================================*/
-/**
- * Mix multiple tones and then detect them.
- */
-
-static int TestMixedMonoTones( void )
-{
-    int i;
-    int result = 0;
-#define NUM_TONES (5)
-    PaQaSineGenerator generators[NUM_TONES];
-    PaQaRecording     recording;
-    float buffer[FRAMES_PER_BLOCK];
-    double sampleRate = 44100.0;
-    int maxFrames = ((int)sampleRate) * 1;
-    int samplesPerFrame = 1;
-
-    double baseFreq = 234.5;
-    double amp = 0.1;
-
-    double mag2;
-
-    int stride = samplesPerFrame;
-    int done = 0;
-
-    // Setup a sine oscillator.
-    for( i=0; i<NUM_TONES; i++ )
-    {
-        PaQa_SetupSineGenerator( &generators[i], PaQa_GetNthFrequency( baseFreq, i ), amp, sampleRate );
-    }
-
-    result = PaQa_InitializeRecording( &recording, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    done = 0;
-    while (!done)
-    {
-        PaQa_EraseBuffer( buffer, FRAMES_PER_BLOCK, samplesPerFrame );
-        for( i=0; i<NUM_TONES; i++ )
-        {
-            PaQa_MixSine( &generators[i], buffer, FRAMES_PER_BLOCK, stride );
-        }
-        done = PaQa_WriteRecording( &recording, buffer, FRAMES_PER_BLOCK, samplesPerFrame );
-    }
-
-    for( i=0; i<NUM_TONES; i++ )
-    {
-        double mag = PaQa_CorrelateSine( &recording, PaQa_GetNthFrequency( baseFreq, i), sampleRate, 0, recording.numFrames, NULL );
-        QA_ASSERT_CLOSE( "exact frequency match", amp, mag, 0.01 );
-    }
-
-    mag2 = PaQa_CorrelateSine( &recording, baseFreq * 0.87, sampleRate, 0, recording.numFrames, NULL );
-    QA_ASSERT_CLOSE( "wrong frequency", 0.0, mag2, 0.01 );
-
-    PaQa_TerminateRecording( &recording );
-    return 0;
-
-error:
-    PaQa_TerminateRecording( &recording);
-    return 1;
-
-}
-
-
-/*==========================================================================================*/
-/**
- * Generate a recording with added or dropped frames.
- */
-
-static void MakeRecordingWithAddedFrames( PaQaRecording *recording, PaQaTestTone *testTone, int glitchPosition, int framesToAdd )
-{
-    PaQaSineGenerator generator;
-#define BUFFER_SIZE 512
-    float buffer[BUFFER_SIZE];
-
-    int frameCounter = testTone->startDelay;
-
-    int stride = 1;
-    // Record some initial silence.
-    int done = PaQa_WriteSilence( recording, testTone->startDelay );
-
-    // Setup a sine oscillator.
-    PaQa_SetupSineGenerator( &generator, testTone->frequency, testTone->amplitude, testTone->sampleRate );
-
-    while (!done)
-    {
-        int framesThisLoop = BUFFER_SIZE;
-
-        if( frameCounter == glitchPosition )
-        {
-            if( framesToAdd > 0 )
-            {
-                // Record some frozen data without advancing the sine generator.
-                done = PaQa_RecordFreeze( recording, framesToAdd );
-                frameCounter += framesToAdd;
-            }
-            else if( framesToAdd < 0 )
-            {
-                // Advance sine generator a few frames.
-                PaQa_MixSine( &generator, buffer, 0 - framesToAdd, stride );
-            }
-
-        }
-        else if( (frameCounter < glitchPosition) && ((frameCounter + framesThisLoop) > glitchPosition) )
-        {
-            // Go right up to the glitchPosition.
-            framesThisLoop = glitchPosition - frameCounter;
-        }
-
-        if( framesThisLoop > 0 )
-        {
-            PaQa_EraseBuffer( buffer, framesThisLoop, testTone->samplesPerFrame );
-            PaQa_MixSine( &generator, buffer, framesThisLoop, stride );
-            done = PaQa_WriteRecording( recording, buffer, framesThisLoop, testTone->samplesPerFrame );
-        }
-        frameCounter += framesThisLoop;
-    }
-}
-
-
-/*==========================================================================================*/
-/**
- * Generate a clean recording.
- */
-
-static void MakeCleanRecording( PaQaRecording *recording, PaQaTestTone *testTone )
-{
-    PaQaSineGenerator generator;
-#define BUFFER_SIZE 512
-    float buffer[BUFFER_SIZE];
-
-    int stride = 1;
-    // Record some initial silence.
-    int done = PaQa_WriteSilence( recording, testTone->startDelay );
-
-    // Setup a sine oscillator.
-    PaQa_SetupSineGenerator( &generator, testTone->frequency, testTone->amplitude, testTone->sampleRate );
-
-    // Generate recording with good phase.
-    while (!done)
-    {
-        PaQa_EraseBuffer( buffer, BUFFER_SIZE, testTone->samplesPerFrame );
-        PaQa_MixSine( &generator, buffer, BUFFER_SIZE, stride );
-        done = PaQa_WriteRecording( recording, buffer, BUFFER_SIZE, testTone->samplesPerFrame );
-    }
-}
-
-/*==========================================================================================*/
-/**
- * Generate a recording with pop.
- */
-
-static void MakeRecordingWithPop( PaQaRecording *recording, PaQaTestTone *testTone, int popPosition, int popWidth, double popAmplitude )
-{
-    int i;
-
-    MakeCleanRecording( recording, testTone );
-
-    // Apply glitch to good recording.
-    if( (popPosition + popWidth) >= recording->numFrames )
-    {
-        popWidth = (recording->numFrames - popPosition) - 1;
-    }
-
-    for( i=0; i<popWidth; i++ )
-    {
-        float good = recording->buffer[i+popPosition];
-        float bad = (good > 0.0) ? (good - popAmplitude) : (good + popAmplitude);
-        recording->buffer[i+popPosition] = bad;
-    }
-}
-
-/*==========================================================================================*/
-/**
- * Detect one phase error in a recording.
- */
-static int TestDetectSinglePhaseError( double sampleRate, int cycleSize, int latencyFrames, int glitchPosition, int framesAdded )
-{
-    int result = 0;
-    PaQaRecording     recording;
-    PaQaTestTone testTone;
-    PaQaAnalysisResult analysisResult = { 0.0 };
-    int framesDropped = 0;
-    int maxFrames = ((int)sampleRate) * 2;
-
-    testTone.samplesPerFrame = 1;
-    testTone.sampleRate = sampleRate;
-    testTone.frequency = sampleRate / cycleSize;
-    testTone.amplitude = 0.5;
-    testTone.startDelay = latencyFrames;
-
-    result = PaQa_InitializeRecording( &recording, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    MakeRecordingWithAddedFrames( &recording, &testTone, glitchPosition, framesAdded );
-
-    PaQa_AnalyseRecording( &recording, &testTone, &analysisResult );
-
-    if( framesAdded < 0 )
-    {
-        framesDropped = -framesAdded;
-        framesAdded = 0;
-    }
-
-#if PRINT_REPORTS
-    printf("\n=== Dropped Frame Analysis ===================\n");
-    printf("                        expected      actual\n");
-    printf("             latency: %10.3f  %10.3f\n", (double)latencyFrames, analysisResult.latency );
-    printf("    num added frames: %10.3f  %10.3f\n", (double)framesAdded, analysisResult.numAddedFrames );
-    printf("     added frames at: %10.3f  %10.3f\n", (double)glitchPosition, analysisResult.addedFramesPosition );
-    printf("  num dropped frames: %10.3f  %10.3f\n", (double)framesDropped, analysisResult.numDroppedFrames );
-    printf("   dropped frames at: %10.3f  %10.3f\n", (double)glitchPosition, analysisResult.droppedFramesPosition );
-#endif
-
-    QA_ASSERT_CLOSE( "PaQa_AnalyseRecording latency", latencyFrames, analysisResult.latency, 0.5 );
-    QA_ASSERT_CLOSE( "PaQa_AnalyseRecording framesAdded", framesAdded, analysisResult.numAddedFrames, 1.0 );
-    QA_ASSERT_CLOSE( "PaQa_AnalyseRecording framesDropped", framesDropped, analysisResult.numDroppedFrames, 1.0 );
-//    QA_ASSERT_CLOSE( "PaQa_AnalyseRecording glitchPosition", glitchPosition, analysisResult.glitchPosition, cycleSize );
-
-    PaQa_TerminateRecording( &recording );
-    return 0;
-
-error:
-    PaQa_TerminateRecording( &recording);
-    return 1;
-}
-
-/*==========================================================================================*/
-/**
- * Test various dropped sample scenarios.
- */
-static int TestDetectPhaseErrors( void )
-{
-    int result;
-
-    result = TestDetectSinglePhaseError( 44100, 200, 477, -1, 0 );
-    if( result < 0 ) return result;
-/*
-    result = TestDetectSinglePhaseError( 44100, 200, 77, -1, 0 );
-    if( result < 0 ) return result;
-
-    result = TestDetectSinglePhaseError( 44100, 200, 83, 3712, 9 );
-    if( result < 0 ) return result;
-
-    result = TestDetectSinglePhaseError( 44100, 280, 83, 3712, 27 );
-    if( result < 0 ) return result;
-
-    result = TestDetectSinglePhaseError( 44100, 200, 234, 3712, -9 );
-    if( result < 0 ) return result;
-
-    result = TestDetectSinglePhaseError( 44100, 200, 2091, 8923, -2 );
-    if( result < 0 ) return result;
-
-    result = TestDetectSinglePhaseError( 44100, 120, 1782, 5772, -18 );
-    if( result < 0 ) return result;
-
-    // Note that if the frequency is too high then it is hard to detect single dropped frames.
-    result = TestDetectSinglePhaseError( 44100, 200, 500, 4251, -1 );
-    if( result < 0 ) return result;
-*/
-    return 0;
-}
-
-/*==========================================================================================*/
-/**
- * Detect one pop in a recording.
- */
-static int TestDetectSinglePop( double sampleRate, int cycleSize, int latencyFrames, int popPosition, int popWidth, double popAmplitude )
-{
-    int result = 0;
-    PaQaRecording     recording;
-    PaQaTestTone testTone;
-    PaQaAnalysisResult analysisResult = { 0.0 };
-    int maxFrames = ((int)sampleRate) * 2;
-
-    testTone.samplesPerFrame = 1;
-    testTone.sampleRate = sampleRate;
-    testTone.frequency = sampleRate / cycleSize;
-    testTone.amplitude = 0.5;
-    testTone.startDelay = latencyFrames;
-
-    result = PaQa_InitializeRecording( &recording, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    MakeRecordingWithPop( &recording, &testTone, popPosition, popWidth, popAmplitude );
-
-    PaQa_AnalyseRecording( &recording, &testTone, &analysisResult );
-
-#if PRINT_REPORTS
-    printf("\n=== Pop Analysis ===================\n");
-    printf("                        expected      actual\n");
-    printf("             latency: %10.3f  %10.3f\n", (double)latencyFrames, analysisResult.latency );
-    printf("         popPosition: %10.3f  %10.3f\n", (double)popPosition, analysisResult.popPosition );
-    printf("        popAmplitude: %10.3f  %10.3f\n", popAmplitude, analysisResult.popAmplitude );
-    printf("           cycleSize: %6d\n", cycleSize );
-    printf("    num added frames: %10.3f\n", analysisResult.numAddedFrames );
-    printf("     added frames at: %10.3f\n", analysisResult.addedFramesPosition );
-    printf("  num dropped frames: %10.3f\n", analysisResult.numDroppedFrames );
-    printf("   dropped frames at: %10.3f\n", analysisResult.droppedFramesPosition );
-#endif
-
-    QA_ASSERT_CLOSE( "PaQa_AnalyseRecording latency", latencyFrames, analysisResult.latency, 0.5 );
-    QA_ASSERT_CLOSE( "PaQa_AnalyseRecording popPosition", popPosition, analysisResult.popPosition, 10 );
-    if( popWidth > 0 )
-    {
-        QA_ASSERT_CLOSE( "PaQa_AnalyseRecording popAmplitude", popAmplitude, analysisResult.popAmplitude, 0.1 * popAmplitude  );
-    }
-
-    PaQa_TerminateRecording( &recording );
-    return 0;
-
-error:
-    PaQa_SaveRecordingToWaveFile( &recording, "bad_recording.wav" );
-    PaQa_TerminateRecording( &recording);
-    return 1;
-}
-
-/*==========================================================================================*/
-/**
- * Analyse recording with a DC offset.
- */
-static int TestSingleInitialSpike( double sampleRate, int stepPosition, int cycleSize, int latencyFrames, double stepAmplitude )
-{
-    int i;
-    int result = 0;
-    // Account for highpass filter offset.
-    int expectedLatency = latencyFrames + 1;
-    PaQaRecording     recording;
-
-    PaQaRecording     hipassOutput = { 0 };
-    BiquadFilter      hipassFilter;
-
-    PaQaTestTone testTone;
-    PaQaAnalysisResult analysisResult = { 0.0 };
-    int maxFrames = ((int)sampleRate) * 2;
-
-    testTone.samplesPerFrame = 1;
-    testTone.sampleRate = sampleRate;
-    testTone.frequency = sampleRate / cycleSize;
-    testTone.amplitude = -0.5;
-    testTone.startDelay = latencyFrames;
-
-    result = PaQa_InitializeRecording( &recording, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    result = PaQa_InitializeRecording( &hipassOutput, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    MakeCleanRecording( &recording, &testTone );
-
-    // Apply DC step.
-    for( i=stepPosition; i<recording.numFrames; i++ )
-    {
-        recording.buffer[i] += stepAmplitude;
-    }
-
-    // Use high pass as a DC blocker!
-    BiquadFilter_SetupHighPass( &hipassFilter, 10.0 / sampleRate, 0.5 );
-    PaQa_FilterRecording( &recording, &hipassOutput, &hipassFilter );
-
-    testTone.amplitude = 0.5;
-    PaQa_AnalyseRecording( &hipassOutput, &testTone, &analysisResult );
-
-#if PRINT_REPORTS
-    printf("\n=== InitialSpike Analysis ===================\n");
-    printf("                        expected      actual\n");
-    printf("             latency: %10.3f  %10.3f\n", (double)expectedLatency, analysisResult.latency );
-    printf("         popPosition: %10.3f\n", analysisResult.popPosition );
-    printf("        popAmplitude: %10.3f\n", analysisResult.popAmplitude );
-    printf("      amplitudeRatio: %10.3f\n", analysisResult.amplitudeRatio );
-    printf("           cycleSize: %6d\n", cycleSize );
-    printf("    num added frames: %10.3f\n", analysisResult.numAddedFrames );
-    printf("     added frames at: %10.3f\n", analysisResult.addedFramesPosition );
-    printf("  num dropped frames: %10.3f\n", analysisResult.numDroppedFrames );
-    printf("   dropped frames at: %10.3f\n", analysisResult.droppedFramesPosition );
-#endif
-
-    QA_ASSERT_CLOSE( "PaQa_AnalyseRecording latency", expectedLatency, analysisResult.latency, 4.0 );
-    QA_ASSERT_EQUALS( "PaQa_AnalyseRecording no pop from step", -1, (int) analysisResult.popPosition );
-    PaQa_TerminateRecording( &recording );
-    PaQa_TerminateRecording( &hipassOutput );
-    return 0;
-
-error:
-    PaQa_SaveRecordingToWaveFile( &recording, "bad_step_original.wav" );
-    PaQa_SaveRecordingToWaveFile( &hipassOutput, "bad_step_hipass.wav" );
-    PaQa_TerminateRecording( &recording);
-    PaQa_TerminateRecording( &hipassOutput );
-    return 1;
-}
-
-/*==========================================================================================*/
-/**
- * Test various dropped sample scenarios.
- */
-static int TestDetectPops( void )
-{
-    int result;
-
-    // No pop.
-    result = TestDetectSinglePop( 44100, 200, 477, -1, 0, 0.0 );
-    if( result < 0 ) return result;
-
-    // short pop
-    result = TestDetectSinglePop( 44100, 300, 810, 3987, 1, 0.5 );
-    if( result < 0 ) return result;
-
-    // medium long pop
-    result = TestDetectSinglePop( 44100, 300, 810, 9876, 5, 0.5 );
-    if( result < 0 ) return result;
-
-    // short tiny pop
-    result = TestDetectSinglePop( 44100, 250, 810, 5672, 1, 0.05 );
-    if( result < 0 ) return result;
-
-
-    return 0;
-}
-
-/*==========================================================================================*/
-/**
- * Test analysis when there is a DC offset step before the sine signal.
- */
-static int TestInitialSpike( void )
-{
-    int result;
-
-//( double sampleRate, int stepPosition, int cycleSize, int latencyFrames, double stepAmplitude )
-    // No spike.
-    result = TestSingleInitialSpike( 44100, 32, 100, 537, 0.0 );
-    if( result < 0 ) return result;
-
-    // Small spike.
-    result = TestSingleInitialSpike( 44100, 32, 100, 537, 0.1 );
-    if( result < 0 ) return result;
-
-    // short pop like Ross's error.
-    result = TestSingleInitialSpike( 8000, 32, 42, 2000, 0.1 );
-    if( result < 0 ) return result;
-
-    // Medium spike.
-    result = TestSingleInitialSpike( 44100, 40, 190, 3000, 0.5 );
-    if( result < 0 ) return result;
-
-    // Spike near sine.
-    //result = TestSingleInitialSpike( 44100, 2900, 140, 3000, 0.1 );
-    if( result < 0 ) return result;
-
-
-    return 0;
-}
-
-
-#if TEST_SAVED_WAVE
-/*==========================================================================================*/
-/**
- * Simple test that writes a sawtooth waveform to a file.
- */
-static int TestSavedWave()
-{
-    int i,j;
-    WAV_Writer writer;
-    int result = 0;
-#define NUM_SAMPLES  (200)
-    short data[NUM_SAMPLES];
-    short saw = 0;
-
-
-    result =  Audio_WAV_OpenWriter( &writer, "test_sawtooth.wav", 44100, 1 );
-    if( result < 0 ) goto error;
-
-    for( i=0; i<15; i++ )
-    {
-        for( j=0; j<NUM_SAMPLES; j++ )
-        {
-            data[j] = saw;
-            saw += 293;
-        }
-        result =  Audio_WAV_WriteShorts( &writer, data, NUM_SAMPLES );
-        if( result < 0 ) goto error;
-    }
-
-    result =  Audio_WAV_CloseWriter( &writer );
-    if( result < 0 ) goto error;
-
-
-    return 0;
-
-error:
-    printf("ERROR: result = %d\n", result );
-    return result;
-}
-#endif /* TEST_SAVED_WAVE */
-
-/*==========================================================================================*/
-/**
- * Easy way to generate a sine tone recording.
- */
-void PaQa_FillWithSine( PaQaRecording *recording, double sampleRate, double freq, double amp )
-{
-    PaQaSineGenerator generator;
-    float buffer[FRAMES_PER_BLOCK];
-    int samplesPerFrame = 1;
-    int stride = 1;
-    int done = 0;
-
-    // Setup a sine oscillator.
-    PaQa_SetupSineGenerator( &generator, freq, amp, sampleRate );
-
-    done = 0;
-    while (!done)
-    {
-        PaQa_EraseBuffer( buffer, FRAMES_PER_BLOCK, samplesPerFrame );
-        PaQa_MixSine( &generator, buffer, FRAMES_PER_BLOCK, stride );
-        done = PaQa_WriteRecording( recording, buffer, FRAMES_PER_BLOCK, samplesPerFrame );
-    }
-
-}
-
-/*==========================================================================================*/
-/**
- * Generate a tone then knock it out using a filter.
- * Also check using filter slightly off tune to see if some energy gets through.
- */
-static int TestNotchFilter( void )
-{
-    int result = 0;
-    PaQaRecording     original = { 0 };
-    PaQaRecording     filtered = { 0 };
-    BiquadFilter      notchFilter;
-    double sampleRate = 44100.0;
-    int maxFrames = ((int)sampleRate) * 1;
-
-    double freq = 234.5;
-    double amp = 0.5;
-
-    double mag1, mag2, mag3;
-
-    result = PaQa_InitializeRecording( &original, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    PaQa_FillWithSine( &original, sampleRate, freq, amp );
-
-    //result = PaQa_SaveRecordingToWaveFile( &original, "original.wav" );
-    //QA_ASSERT_EQUALS( "PaQa_SaveRecordingToWaveFile failed", 0, result );
-
-    mag1 = PaQa_CorrelateSine( &original, freq, sampleRate, 0, original.numFrames, NULL );
-    QA_ASSERT_CLOSE( "exact frequency match", amp, mag1, 0.01 );
-
-    // Filter with exact frequency.
-    result = PaQa_InitializeRecording( &filtered, maxFrames, (int) sampleRate );
-    QA_ASSERT_EQUALS( "PaQa_InitializeRecording failed", 0, result );
-
-    BiquadFilter_SetupNotch( &notchFilter, freq / sampleRate, 0.5 );
-    PaQa_FilterRecording( &original, &filtered, &notchFilter );
-    result = PaQa_SaveRecordingToWaveFile( &filtered, "filtered1.wav" );
-    QA_ASSERT_EQUALS( "PaQa_SaveRecordingToWaveFile failed", 0, result );
-
-    mag2 = PaQa_CorrelateSine( &filtered, freq, sampleRate, 0, filtered.numFrames, NULL );
-    QA_ASSERT_CLOSE( "should eliminate tone", 0.0, mag2, 0.01 );
-
-    // Filter with mismatched frequency.
-    BiquadFilter_SetupNotch( &notchFilter, 1.07 * freq / sampleRate, 2.0 );
-    PaQa_FilterRecording( &original, &filtered, &notchFilter );
-
-    //result = PaQa_SaveRecordingToWaveFile( &filtered, "badfiltered.wav" );
-    //QA_ASSERT_EQUALS( "PaQa_SaveRecordingToWaveFile failed", 0, result );
-
-    mag3 = PaQa_CorrelateSine( &filtered, freq, sampleRate, 0, filtered.numFrames, NULL );
-    QA_ASSERT_CLOSE( "should eliminate tone", amp*0.26, mag3, 0.01 );
-
-
-    PaQa_TerminateRecording( &original );
-    PaQa_TerminateRecording( &filtered );
-    return 0;
-
-error:
-    PaQa_TerminateRecording( &original);
-    PaQa_TerminateRecording( &filtered );
-    return 1;
-
-}
-
-/*==========================================================================================*/
-/**
- */
-int PaQa_TestAnalyzer( void )
-{
-    int result;
-
-#if TEST_SAVED_WAVE
-    // Write a simple wave file.
-    if ((result = TestSavedWave()) != 0) return result;
-#endif /* TEST_SAVED_WAVE */
-
-    // Generate single tone and verify presence.
-    if ((result = TestSingleMonoTone()) != 0) return result;
-
-    // Generate prime series of tones and verify presence.
-    if ((result = TestMixedMonoTones()) != 0) return result;
-
-    // Detect dropped or added samples in a sine wave recording.
-    if ((result = TestDetectPhaseErrors()) != 0) return result;
-
-    // Test to see if notch filter can knock out the test tone.
-    if ((result = TestNotchFilter()) != 0) return result;
-
-    // Detect pops that get back in phase.
-    if ((result = TestDetectPops()) != 0) return result;
-
-    // Test to see if the latency detector can be tricked like it was on Ross' Windows machine.
-    if ((result = TestInitialSpike()) != 0) return result;
-
-
-    return 0;
-}
diff --git a/portaudio/qa/loopback/src/test_audio_analyzer.h b/portaudio/qa/loopback/src/test_audio_analyzer.h
deleted file mode 100644
index bfe073f..0000000
--- a/portaudio/qa/loopback/src/test_audio_analyzer.h
+++ /dev/null
@@ -1,46 +0,0 @@
-
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-#ifndef _TEST_AUDIO_ANALYZER_H
-#define _TEST_AUDIO_ANALYZER_H
-
-/** Test the audio analyzer by itself without any PortAudio calls. */
-int PaQa_TestAnalyzer( void );
-
-
-#endif /* _TEST_AUDIO_ANALYZER_H */
diff --git a/portaudio/qa/loopback/src/write_wav.c b/portaudio/qa/loopback/src/write_wav.c
deleted file mode 100755
index aa5ee21..0000000
--- a/portaudio/qa/loopback/src/write_wav.c
+++ /dev/null
@@ -1,242 +0,0 @@
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-
-/**
-  * Very simple WAV file writer for saving captured audio.
-  */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include "write_wav.h"
-
-
-/* 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;
-}
-
-/* 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;
-}
-
-/* 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;
-}
-
-#define WAV_HEADER_SIZE (4 + 4 + 4 + /* RIFF+size+WAVE */ \
-        4 + 4 + 16 + /* fmt chunk */ \
-        4 + 4 ) /* data chunk */
-
-
-/*********************************************************************************
- * Open named file and write WAV header to the file.
- * The header includes the DATA chunk type and size.
- * Returns number of bytes written to file or negative error code.
- */
-long Audio_WAV_OpenWriter( WAV_Writer *writer, const char *fileName, int frameRate, int samplesPerFrame )
-{
-    unsigned int  bytesPerSecond;
-    unsigned char header[ WAV_HEADER_SIZE ];
-    unsigned char *addr = header;
-    int numWritten;
-
-    writer->dataSize = 0;
-    writer->dataSizeOffset = 0;
-
-    writer->fid = fopen( fileName, "wb" );
-    if( writer->fid == NULL )
-    {
-        return -1;
-    }
-
-/* Write RIFF header. */
-    WriteChunkType( &addr, RIFF_ID );
-
-/* Write RIFF size as zero for now. Will patch later. */
-    WriteLongLE( &addr, 0 );
-
-/* Write WAVE form ID. */
-    WriteChunkType( &addr, WAVE_ID );
-
-/* Write format chunk based on AudioSample structure. */
-    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 */
-
-/* Write ID and size for 'data' chunk. */
-    WriteChunkType( &addr, DATA_ID );
-/* Save offset so we can patch it later. */
-    writer->dataSizeOffset = (int) (addr - header);
-    WriteLongLE( &addr, 0 );
-
-    numWritten = fwrite( header, 1, sizeof(header), writer->fid );
-    if( numWritten != sizeof(header) ) return -1;
-
-    return (int) numWritten;
-}
-
-/*********************************************************************************
- * Write to the data chunk portion of a WAV file.
- * Returns bytes written or negative error code.
- */
-long Audio_WAV_WriteShorts( WAV_Writer *writer,
-        short *samples,
-        int numSamples
-        )
-{
-    unsigned char buffer[2];
-    unsigned char *bufferPtr;
-    int i;
-    short *p = samples;
-    int numWritten;
-    int bytesWritten;
-    if( numSamples <= 0 )
-    {
-        return -1;
-    }
-
-    for( i=0; i<numSamples; i++ )
-    {
-        bufferPtr = buffer;
-        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;
-}
-
-/*********************************************************************************
- * Close WAV file.
- * Update chunk sizes so it can be read by audio applications.
- */
-long Audio_WAV_CloseWriter( WAV_Writer *writer )
-{
-    unsigned char buffer[4];
-    unsigned char *bufferPtr;
-    int numWritten;
-    int riffSize;
-
-    /* Go back to beginning of file and update DATA size */
-    int result = fseek( writer->fid, writer->dataSizeOffset, SEEK_SET );
-    if( result < 0 ) return result;
-
-    bufferPtr = buffer;
-    WriteLongLE( &bufferPtr, writer->dataSize );
-    numWritten = fwrite( buffer, 1, sizeof( buffer), writer->fid );
-    if( numWritten != sizeof(buffer) ) return -1;
-
-    /* Update RIFF size */
-    result = fseek( writer->fid, 4, SEEK_SET );
-    if( result < 0 ) return result;
-
-    riffSize = writer->dataSize + (WAV_HEADER_SIZE - 8);
-    bufferPtr = buffer;
-    WriteLongLE( &bufferPtr, riffSize );
-    numWritten = fwrite( buffer, 1, sizeof( buffer), writer->fid );
-    if( numWritten != sizeof(buffer) ) return -1;
-
-    fclose( writer->fid );
-    writer->fid = NULL;
-    return writer->dataSize;
-}
-
-/*********************************************************************************
- * Simple test that write a sawtooth waveform to a file.
- */
-#if 0
-int main( void )
-{
-    int i;
-    WAV_Writer writer;
-    int result;
-#define NUM_SAMPLES  (200)
-    short data[NUM_SAMPLES];
-    short saw = 0;
-
-    for( i=0; i<NUM_SAMPLES; i++ )
-    {
-        data[i] = saw;
-        saw += 293;
-    }
-
-
-    result =  Audio_WAV_OpenWriter( &writer, "rendered_midi.wav", 44100, 1 );
-    if( result < 0 ) goto error;
-
-    for( i=0; i<15; i++ )
-    {
-        result =  Audio_WAV_WriteShorts( &writer, data, NUM_SAMPLES );
-        if( result < 0 ) goto error;
-    }
-
-    result =  Audio_WAV_CloseWriter( &writer );
-    if( result < 0 ) goto error;
-
-
-    return 0;
-
-error:
-    printf("ERROR: result = %d\n", result );
-    return result;
-}
-#endif
diff --git a/portaudio/qa/loopback/src/write_wav.h b/portaudio/qa/loopback/src/write_wav.h
deleted file mode 100755
index 3560055..0000000
--- a/portaudio/qa/loopback/src/write_wav.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * PortAudio Portable Real-Time Audio Library
- * Latest Version at: http://www.portaudio.com
- *
- * Copyright (c) 1999-2010 Phil Burk and Ross Bencina
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * 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
- * license above.
- */
-#ifndef _WAV_WRITER_H
-#define _WAV_WRITER_H
-
-/*
- * WAV file writer.
- *
- * Author: Phil Burk
- */
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/* Define WAV Chunk and FORM types as 4 byte integers. */
-#define RIFF_ID   (('R'<<24) | ('I'<<16) | ('F'<<8) | 'F')
-#define WAVE_ID   (('W'<<24) | ('A'<<16) | ('V'<<8) | 'E')
-#define FMT_ID    (('f'<<24) | ('m'<<16) | ('t'<<8) | ' ')
-#define DATA_ID   (('d'<<24) | ('a'<<16) | ('t'<<8) | 'a')
-#define FACT_ID   (('f'<<24) | ('a'<<16) | ('c'<<8) | 't')
-
-/* Errors returned by Audio_ParseSampleImage_WAV */
-#define WAV_ERR_CHUNK_SIZE     (-1)   /* Chunk size is illegal or past file size. */
-#define WAV_ERR_FILE_TYPE      (-2)   /* Not a WAV file. */
-#define WAV_ERR_ILLEGAL_VALUE  (-3)   /* Illegal or unsupported value. Eg. 927 bits/sample */
-#define WAV_ERR_FORMAT_TYPE    (-4)   /* Unsupported format, eg. compressed. */
-#define WAV_ERR_TRUNCATED      (-5)   /* End of file missing. */
-
-/* WAV PCM data format ID */
-#define WAVE_FORMAT_PCM        (1)
-#define WAVE_FORMAT_IMA_ADPCM  (0x0011)
-
-
-typedef struct WAV_Writer_s
-{
-    FILE *fid;
-    /* Offset in file for data size. */
-    int   dataSizeOffset;
-    int   dataSize;
-} WAV_Writer;
-
-/*********************************************************************************
- * Open named file and write WAV header to the file.
- * The header includes the DATA chunk type and size.
- * Returns number of bytes written to file or negative error code.
- */
-long Audio_WAV_OpenWriter( WAV_Writer *writer, const char *fileName, int frameRate, int samplesPerFrame );
-
-/*********************************************************************************
- * Write to the data chunk portion of a WAV file.
- * Returns bytes written or negative error code.
- */
-long Audio_WAV_WriteShorts( WAV_Writer *writer,
-        short *samples,
-        int numSamples
-        );
-
-/*********************************************************************************
- * Close WAV file.
- * Update chunk sizes so it can be read by audio applications.
- */
-long Audio_WAV_CloseWriter( WAV_Writer *writer );
-
-#ifdef __cplusplus
-};
-#endif
-
-#endif /* _WAV_WRITER_H */