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-rw-r--r--3rdparty/portaudio/qa/loopback/README.txt92
-rw-r--r--3rdparty/portaudio/qa/loopback/src/audio_analyzer.c706
-rw-r--r--3rdparty/portaudio/qa/loopback/src/audio_analyzer.h187
-rwxr-xr-x3rdparty/portaudio/qa/loopback/src/biquad_filter.c123
-rwxr-xr-x3rdparty/portaudio/qa/loopback/src/biquad_filter.h38
-rw-r--r--3rdparty/portaudio/qa/loopback/src/paqa.c1601
-rw-r--r--3rdparty/portaudio/qa/loopback/src/paqa_tools.c171
-rw-r--r--3rdparty/portaudio/qa/loopback/src/paqa_tools.h52
-rwxr-xr-x3rdparty/portaudio/qa/loopback/src/qa_tools.h83
-rw-r--r--3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.c718
-rw-r--r--3rdparty/portaudio/qa/loopback/src/test_audio_analyzer.h46
-rwxr-xr-x3rdparty/portaudio/qa/loopback/src/write_wav.c242
-rwxr-xr-x3rdparty/portaudio/qa/loopback/src/write_wav.h103
13 files changed, 4162 insertions, 0 deletions
diff --git a/3rdparty/portaudio/qa/loopback/README.txt b/3rdparty/portaudio/qa/loopback/README.txt
new file mode 100644
index 0000000..5ad0280
--- /dev/null
+++ b/3rdparty/portaudio/qa/loopback/README.txt
@@ -0,0 +1,92 @@
+README for PortAudio Loopback Test
+
+Copyright (c) 1999-2010 Phil Burk and Ross Bencina
+See complete license at end of file.
+
+This folder contains code for a single executable that does a standalone test of PortAudio.
+It does not require a human to listen to the result. Instead it listens to itself using
+a loopback cable connected between the audio output and the audio input. Special pop detectors
+and phase analysers can detect errors in the audio stream.
+
+This test can be run from a script as part of a nightly build and test.
+
+--- How to Build the Loopback Test ---
+
+The loopback test is not normally built by the makefile.
+To build the loopback test, enter:
+
+ ./configure && make loopback
+
+This will build the "bin/paloopback" executable.
+
+--- How To Run Test ---
+
+Connect stereo cables from one or more output audio devices to audio input devices.
+The test will scan all the ports and find the cables.
+
+Adjust the volume levels of the hardware so you get a decent signal that will not clip.
+
+Run the test from the command line with the following options:
+
+ -i# Input device ID. Will scan for loopback if not specified.
+ -o# Output device ID. Will scan for loopback if not specified.
+ -r# Sample Rate in Hz. Will use multiple common rates if not specified.
+ -s# Size of callback buffer in frames, framesPerBuffer.
+ -w Save bad recordings in a WAV file.
+ -dDir Path for Directory for WAV files. Default is current directory.
+ -m Just test the DSP Math code and not the audio devices.
+
+If the -w option is set then any tests that fail will save the recording of the broken
+channel in a WAV file. The files will be numbered and shown in the report.
+
+--- ToDo ---
+
+* Add check for harmonic and enharmonic distortion.
+* Measure min/max peak values.
+* Detect DC bias.
+* Test against matrix of devices/APIs and settings.
+* Detect mono vs stereo loopback.
+* More command line options
+ --quick
+ --latency
+ --duration
+* Automated build and test script with cron job.
+* Test on Windows.
+
+
+/*
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Copyright (c) 1999-2008 Phil Burk and Ross Bencina
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
diff --git a/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 */
generated by cgit v1.2.1 (git 2.18.0) at 2024-06-30 23:02:28 +0000