add plibsys

12 files changed, 4070 insertions, 0 deletions

diff --git a/3rdparty/portaudio/qa/loopback/src/audio_analyzer.c b/3rdparty/portaudio/qa/loopback/src/audio_analyzer.c
new file mode 100644
index 0000000..fbdd631
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/audio_analyzer.c
@@ -0,0 +1,706 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/audio_analyzer.h b/3rdparty/portaudio/qa/loopback/src/audio_analyzer.h
new file mode 100644
index 0000000..8d9f1ee
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/audio_analyzer.h
@@ -0,0 +1,187 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/biquad_filter.c b/3rdparty/portaudio/qa/loopback/src/biquad_filter.c
new file mode 100755
index 0000000..1715fa3
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/biquad_filter.c
@@ -0,0 +1,123 @@
+#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/3rdparty/portaudio/qa/loopback/src/biquad_filter.h b/3rdparty/portaudio/qa/loopback/src/biquad_filter.h
new file mode 100755
index 0000000..0895aba
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/biquad_filter.h
@@ -0,0 +1,38 @@
+#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/3rdparty/portaudio/qa/loopback/src/paqa.c b/3rdparty/portaudio/qa/loopback/src/paqa.c
new file mode 100644
index 0000000..5eb6283
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/paqa.c
@@ -0,0 +1,1601 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/paqa_tools.c b/3rdparty/portaudio/qa/loopback/src/paqa_tools.c
new file mode 100644
index 0000000..2e44c63
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/paqa_tools.c
@@ -0,0 +1,171 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/paqa_tools.h b/3rdparty/portaudio/qa/loopback/src/paqa_tools.h
new file mode 100644
index 0000000..77f6a25
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/paqa_tools.h
@@ -0,0 +1,52 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/qa_tools.h b/3rdparty/portaudio/qa/loopback/src/qa_tools.h
new file mode 100755
index 0000000..9b2debd
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/qa_tools.h
@@ -0,0 +1,83 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.c b/3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.c
new file mode 100644
index 0000000..82fa859
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.c
@@ -0,0 +1,718 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.h b/3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.h
new file mode 100644
index 0000000..bfe073f
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.h
@@ -0,0 +1,46 @@
+
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/write_wav.c b/3rdparty/portaudio/qa/loopback/src/write_wav.c
new file mode 100755
index 0000000..aa5ee21
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/write_wav.c
@@ -0,0 +1,242 @@
+/*
+ * 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/3rdparty/portaudio/qa/loopback/src/write_wav.h b/3rdparty/portaudio/qa/loopback/src/write_wav.h
new file mode 100755
index 0000000..3560055
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/src/write_wav.h
@@ -0,0 +1,103 @@
+/*
+ * 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 */