Merge "aaudio_loopback: rectify and low pass filter" (eacf0137) · Commits · e / os / android_frameworks_av

media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h

+309 −174

Original line number	Diff line number	Diff line
		@@ -34,12 +34,123 @@

		// Tag for machine readable results as property = value pairs
		#define LOOPBACK_RESULT_TAG "RESULT: "
		#define LOOPBACK_SAMPLE_RATE 48000

		#define MILLIS_PER_SECOND 1000
		constexpr int32_t kDefaultSampleRate = 48000;
		constexpr int32_t kMillisPerSecond = 1000;
		constexpr int32_t kMinLatencyMillis = 4; // arbitrary and very low
		constexpr int32_t kMaxLatencyMillis = 400; // arbitrary and generous
		constexpr double kMaxEchoGain = 10.0; // based on experiments, otherwise too noisy
		constexpr double kMinimumConfidence = 0.5;

		#define MAX_ZEROTH_PARTIAL_BINS 40
		constexpr double MAX_ECHO_GAIN = 10.0; // based on experiments, otherwise autocorrelation too noisy
		static void printAudioScope(float sample) {
		const int maxStars = 80; // arbitrary, fits on one line
		char c = '*';
		if (sample < -1.0) {
		sample = -1.0;
		c = '$';
		} else if (sample > 1.0) {
		sample = 1.0;
		c = '$';
		}
		int numSpaces = (int) (((sample + 1.0) * 0.5) * maxStars);
		for (int i = 0; i < numSpaces; i++) {
		putchar(' ');
		}
		printf("%c\n", c);
		}

		/*

		FIR filter designed with
		http://t-filter.appspot.com

		sampling frequency: 48000 Hz

		* 0 Hz - 8000 Hz
		gain = 1.2
		desired ripple = 5 dB
		actual ripple = 5.595266169703693 dB

		* 12000 Hz - 20000 Hz
		gain = 0
		desired attenuation = -40 dB
		actual attenuation = -37.58691566571914 dB

		*/

		#define FILTER_TAP_NUM 11

		static const float sFilterTaps8000[FILTER_TAP_NUM] = {
		-0.05944219353343189f,
		-0.07303434839503208f,
		-0.037690487672689066f,
		0.1870480506596512f,
		0.3910337357836833f,
		0.5333672385425637f,
		0.3910337357836833f,
		0.1870480506596512f,
		-0.037690487672689066f,
		-0.07303434839503208f,
		-0.05944219353343189f
		};

		class LowPassFilter {
		public:

		/*
		* Filter one input sample.
		* @return filtered output
		*/
		float filter(float input) {
		float output = 0.0f;
		mX[mCursor] = input;
		// Index backwards over x.
		int xIndex = mCursor + FILTER_TAP_NUM;
		// Write twice so we avoid having to wrap in the middle of the convolution.
		mX[xIndex] = input;
		for (int i = 0; i < FILTER_TAP_NUM; i++) {
		output += sFilterTaps8000[i] * mX[xIndex--];
		}
		if (++mCursor >= FILTER_TAP_NUM) {
		mCursor = 0;
		}
		return output;
		}

		/**
		* @return true if PASSED
		*/
		bool test() {
		// Measure the impulse of the filter at different phases so we exercise
		// all the wraparound cases in the FIR.
		for (int offset = 0; offset < (FILTER_TAP_NUM * 2); offset++ ) {
		// printf("LowPassFilter: cursor = %d\n", mCursor);
		// Offset by one each time.
		if (filter(0.0f) != 0.0f) {
		printf("ERROR: filter should return 0.0 before impulse response\n");
		return false;
		}
		for (int i = 0; i < FILTER_TAP_NUM; i++) {
		float output = filter((i == 0) ? 1.0f : 0.0f); // impulse
		if (output != sFilterTaps8000[i]) {
		printf("ERROR: filter should return impulse response\n");
		return false;
		}
		}
		for (int i = 0; i < FILTER_TAP_NUM; i++) {
		if (filter(0.0f) != 0.0f) {
		printf("ERROR: filter should return 0.0 after impulse response\n");
		return false;
		}
		}
		}
		return true;
		}

		private:
		float mX[FILTER_TAP_NUM * 2]{}; // twice as big as needed to avoid wrapping
		int32_t mCursor = 0;
		};

		// A narrow impulse seems to have better immunity against over estimating the
		// latency due to detecting subharmonics by the auto-correlator.
		@@ -78,6 +189,12 @@ private:
		int64_t mSeed = 99887766;
		};


		typedef struct LatencyReport_s {
		double latencyInFrames;
		double confidence;
		} LatencyReport;

		static double calculateCorrelation(const float *a,
		const float *b,
		int windowSize)
		@@ -101,127 +218,75 @@ static double calculateCorrelation(const float *a,
		return correlation;
		}

		static int calculateCorrelations(const float *haystack, int haystackSize,
		const float *needle, int needleSize,
		float *results, int resultSize)
		{
		int maxCorrelations = haystackSize - needleSize;
		int numCorrelations = std::min(maxCorrelations, resultSize);

		for (int ic = 0; ic < numCorrelations; ic++) {
		double correlation = calculateCorrelation(&haystack[ic], needle, needleSize);
		results[ic] = correlation;
		}

		return numCorrelations;
		}

		/==========================================================================================/
		/**
		* Scan until we get a correlation of a single scan that goes over the tolerance level,
		* peaks then drops back down.
		*/
		static double findFirstMatch(const float *haystack, int haystackSize,
		const float *needle, int needleSize, double threshold )
		{
		int ic;
		// How many correlations can we calculate?
		int numCorrelations = haystackSize - needleSize;
		double maxCorrelation = 0.0;
		int peakIndex = -1;
		double location = -1.0;
		const double backThresholdScaler = 0.5;

		for (ic = 0; ic < numCorrelations; ic++) {
		double correlation = calculateCorrelation(&haystack[ic], needle, needleSize);

		if( (correlation > maxCorrelation) ) {
		maxCorrelation = correlation;
		peakIndex = ic;
		}

		//printf("PaQa_FindFirstMatch: ic = %4d, correlation = %8f, maxSum = %8f\n",
		// ic, correlation, maxSum );
		// Are we past what we were looking for?
		if((maxCorrelation > threshold) && (correlation < backThresholdScaler * maxCorrelation)) {
		location = peakIndex;
		break;
		}
		}

		return location;
		static int measureLatencyFromEchos(const float *data,
		int32_t numFloats,
		int32_t sampleRate,
		LatencyReport *report) {
		// Allocate results array
		const int minReasonableLatencyFrames = sampleRate * kMinLatencyMillis / kMillisPerSecond;
		const int maxReasonableLatencyFrames = sampleRate * kMaxLatencyMillis / kMillisPerSecond;
		int32_t maxCorrelationSize = maxReasonableLatencyFrames * 3;
		int numCorrelations = std::min(numFloats, maxCorrelationSize);
		float *correlations = new float[numCorrelations]{};
		float *harmonicSums = new float[numCorrelations]{};

		// Perform sliding auto-correlation.
		// Skip first frames to avoid huge peak at zero offset.
		for (int i = minReasonableLatencyFrames; i < numCorrelations; i++) {
		int32_t remaining = numFloats - i;
		float correlation = (float) calculateCorrelation(&data[i], data, remaining);
		correlations[i] = correlation;
		// printf("correlation[%d] = %f\n", ic, correlation);
		}

		typedef struct LatencyReport_s {
		double latencyInFrames;
		double confidence;
		} LatencyReport;

		// Apply a technique similar to Harmonic Product Spectrum Analysis to find echo fundamental.
		// Using first echo instead of the original impulse for a better match.
		static int measureLatencyFromEchos(const float *haystack, int haystackSize,
		const float *needle, int needleSize,
		LatencyReport *report) {
		const double threshold = 0.1;
		// printf("%s: haystackSize = %d, needleSize = %d\n", __func__, haystackSize, needleSize);

		// Find first peak
		int first = (int) (findFirstMatch(haystack,
		haystackSize,
		needle,
		needleSize,
		threshold) + 0.5);

		// Use first echo as the needle for the other echos because
		// it will be more similar.
		needle = &haystack[first];
		int again = (int) (findFirstMatch(haystack,
		haystackSize,
		needle,
		needleSize,
		threshold) + 0.5);
		first = again;

		// Allocate results array
		int remaining = haystackSize - first;
		const int maxReasonableLatencyFrames = 48000 * 2; // arbitrary but generous value
		int numCorrelations = std::min(remaining, maxReasonableLatencyFrames);
		float *correlations = new float[numCorrelations];
		float *harmonicSums = new float[numCorrelations](); // set to zero

		// Generate correlation for every position.
		numCorrelations = calculateCorrelations(&haystack[first], remaining,
		needle, needleSize,
		correlations, numCorrelations);

		// Add higher harmonics mapped onto lower harmonics.
		// This reinforces the "fundamental" echo.
		// Add higher harmonics mapped onto lower harmonics. This reinforces the "fundamental" echo.
		const int numEchoes = 8;
		for (int partial = 1; partial < numEchoes; partial++) {
		for (int i = 0; i < numCorrelations; i++) {
		for (int i = minReasonableLatencyFrames; i < numCorrelations; i++) {
		harmonicSums[i / partial] += correlations[i] / partial;
		}
		}

		// Find highest peak in correlation array.
		float maxCorrelation = 0.0;
		float sumOfPeaks = 0.0;
		int peakIndex = 0;
		const int skip = MAX_ZEROTH_PARTIAL_BINS; // skip low bins
		for (int i = skip; i < numCorrelations; i++) {
		for (int i = 0; i < numCorrelations; i++) {
		if (harmonicSums[i] > maxCorrelation) {
		maxCorrelation = harmonicSums[i];
		sumOfPeaks += maxCorrelation;
		peakIndex = i;
		// printf("maxCorrelation = %f at %d\n", maxCorrelation, peakIndex);
		}
		}

		report->latencyInFrames = peakIndex;
		if (sumOfPeaks < 0.0001) {
		/*
		{
		int32_t topPeak = peakIndex * 7 / 2;
		for (int i = 0; i < topPeak; i++) {
		float sample = harmonicSums[i];
		printf("%4d: %7.5f ", i, sample);
		printAudioScope(sample);
		}
		}
		*/

		// Calculate confidence.
		if (maxCorrelation < 0.001) {
		report->confidence = 0.0;
		} else {
		report->confidence = maxCorrelation / sumOfPeaks;
		// Compare peak to average value around peak.
		int32_t numSamples = std::min(numCorrelations, peakIndex * 2);
		if (numSamples <= 0) {
		report->confidence = 0.0;
		} else {
		double sum = 0.0;
		for (int i = 0; i < numSamples; i++) {
		sum += harmonicSums[i];
		}
		const double average = sum / numSamples;
		const double ratio = average / maxCorrelation; // will be < 1.0
		report->confidence = 1.0 - sqrt(ratio);
		}
		}

		delete[] correlations;
		@@ -317,7 +382,9 @@ public:
		}

		assert(info.channels == 1);
		assert(info.format == SF_FORMAT_FLOAT);

		setSampleRate(info.samplerate);
		allocate(info.frames);
		mFrameCounter = sf_readf_float(sndFile, mData, info.frames);

		@@ -325,11 +392,49 @@ public:
		return mFrameCounter;
		}

		/**
		* Square the samples so they are all positive and so the peaks are emphasized.
		*/
		void square() {
		for (int i = 0; i < mFrameCounter; i++) {
		const float sample = mData[i];
		mData[i] = sample * sample;
		}
		}

		/**
		* Low pass filter the recording using a simple FIR filter.
		* Note that the lowpass filter cutoff tracks the sample rate.
		* That is OK because the impulse width is a fixed number of samples.
		*/
		void lowPassFilter() {
		for (int i = 0; i < mFrameCounter; i++) {
		mData[i] = mLowPassFilter.filter(mData[i]);
		}
		}

		/**
		* Remove DC offset using a one-pole one-zero IIR filter.
		*/
		void dcBlocker() {
		const float R = 0.996; // narrow notch at zero Hz
		float x1 = 0.0;
		float y1 = 0.0;
		for (int i = 0; i < mFrameCounter; i++) {
		const float x = mData[i];
		const float y = x - x1 + (R * y1);
		mData[i] = y;
		y1 = y;
		x1 = x;
		}
		}

		private:
		float *mData = nullptr;
		int32_t mFrameCounter = 0;
		int32_t mMaxFrames = 0;
		int32_t mSampleRate = 48000; // common default
		int32_t mSampleRate = kDefaultSampleRate; // common default
		LowPassFilter mLowPassFilter;
		};

		// ====================================================================================
		@@ -349,8 +454,12 @@ public:

		virtual void printStatus() {};

		virtual int getResult() {
		return -1;
		int32_t getResult() {
		return mResult;
		}

		void setResult(int32_t result) {
		mResult = result;
		}

		virtual bool isDone() {
		@@ -379,7 +488,7 @@ public:
		static float measurePeakAmplitude(float *inputData, int inputChannelCount, int numFrames) {
		float peak = 0.0f;
		for (int i = 0; i < numFrames; i++) {
		float pos = fabs(*inputData);
		const float pos = fabs(*inputData);
		if (pos > peak) {
		peak = pos;
		}
		@@ -390,7 +499,8 @@ public:


		private:
		int32_t mSampleRate = LOOPBACK_SAMPLE_RATE;
		int32_t mSampleRate = kDefaultSampleRate;
		int32_t mResult = 0;
		};

		class PeakDetector {
		@@ -409,24 +519,6 @@ private:
		float mPrevious = 0.0f;
		};


		static void printAudioScope(float sample) {
		const int maxStars = 80; // arbitrary, fits on one line
		char c = '*';
		if (sample < -1.0) {
		sample = -1.0;
		c = '$';
		} else if (sample > 1.0) {
		sample = 1.0;
		c = '$';
		}
		int numSpaces = (int) (((sample + 1.0) * 0.5) * maxStars);
		for (int i = 0; i < numSpaces; i++) {
		putchar(' ');
		}
		printf("%c\n", c);
		}

		// ====================================================================================
		/**
		* Measure latency given a loopback stream data.
		@@ -447,17 +539,13 @@ public:
		}

		void reset() override {
		mDownCounter = 200;
		mDownCounter = getSampleRate() / 2;
		mLoopCounter = 0;
		mMeasuredLoopGain = 0.0f;
		mEchoGain = 1.0f;
		mState = STATE_INITIAL_SILENCE;
		}

		virtual int getResult() {
		return mState == STATE_DONE ? 0 : -1;
		}

		virtual bool isDone() {
		return mState == STATE_DONE \|\| mState == STATE_FAILED;
		}
		@@ -470,34 +558,57 @@ public:
		return mEchoGain;
		}

		void report() override {
		bool testLowPassFilter() {
		LowPassFilter filter;
		return filter.test();
		}

		void report() override {
		printf("EchoAnalyzer ---------------\n");
		if (getResult() != 0) {
		printf(LOOPBACK_RESULT_TAG "result = %d\n", getResult());
		return;
		}

		// printf("LowPassFilter test %s\n", testLowPassFilter() ? "PASSED" : "FAILED");

		printf(LOOPBACK_RESULT_TAG "measured.gain = %8f\n", mMeasuredLoopGain);
		printf(LOOPBACK_RESULT_TAG "echo.gain = %8f\n", mEchoGain);
		printf(LOOPBACK_RESULT_TAG "test.state = %8d\n", mState);
		printf(LOOPBACK_RESULT_TAG "test.state.name = %8s\n", convertStateToText(mState));

		if (mState == STATE_WAITING_FOR_SILENCE) {
		printf("WARNING - Stuck waiting for silence. Input may be too noisy!\n");
		}
		if (mMeasuredLoopGain >= 0.9999) {
		setResult(ERROR_NOISY);
		} else if (mMeasuredLoopGain >= 0.9999) {
		printf(" ERROR - clipping, turn down volume slightly\n");
		setResult(ERROR_CLIPPING);
		} else if (mState != STATE_DONE && mState != STATE_GATHERING_ECHOS) {
		printf("WARNING - Bad state. Check volume on device.\n");
		setResult(ERROR_INVALID_STATE);
		} else {
		const float *needle = s_Impulse;
		int needleSize = (int) (sizeof(s_Impulse) / sizeof(float));
		float *haystack = mAudioRecording.getData();
		int haystackSize = mAudioRecording.size();
		measureLatencyFromEchos(haystack, haystackSize, needle, needleSize, &mLatencyReport);
		if (mLatencyReport.confidence < 0.01) {
		printf(" ERROR - confidence too low = %f\n", mLatencyReport.confidence);
		} else {
		double latencyMillis = 1000.0 * mLatencyReport.latencyInFrames / getSampleRate();
		// Cleanup the signal to improve the auto-correlation.
		mAudioRecording.dcBlocker();
		mAudioRecording.square();
		mAudioRecording.lowPassFilter();

		printf("Please wait several seconds for auto-correlation to complete.\n");
		measureLatencyFromEchos(mAudioRecording.getData(),
		mAudioRecording.size(),
		getSampleRate(),
		&mLatencyReport);

		double latencyMillis = kMillisPerSecond * (double) mLatencyReport.latencyInFrames
		/ getSampleRate();
		printf(LOOPBACK_RESULT_TAG "latency.frames = %8.2f\n",
		mLatencyReport.latencyInFrames);
		printf(LOOPBACK_RESULT_TAG "latency.msec = %8.2f\n",
		latencyMillis);
		printf(LOOPBACK_RESULT_TAG "latency.confidence = %8.6f\n",
		mLatencyReport.confidence);
		if (mLatencyReport.confidence < kMinimumConfidence) {
		printf(" ERROR - confidence too low!\n");
		setResult(ERROR_CONFIDENCE);
		}
		}
		}
		@@ -523,6 +634,11 @@ public:
		sendImpulses(outputData, outputChannelCount, kImpulseSizeInFrames);
		}

		// @return number of frames for a typical block of processing
		int32_t getBlockFrames() {
		return getSampleRate() / 8;
		}

		void process(float *inputData, int inputChannelCount,
		float *outputData, int outputChannelCount,
		int numFrames) override {
		@@ -540,10 +656,11 @@ public:
		for (int i = 0; i < numSamples; i++) {
		outputData[i] = 0;
		}
		if (mDownCounter-- <= 0) {
		mDownCounter -= numFrames;
		if (mDownCounter <= 0) {
		nextState = STATE_MEASURING_GAIN;
		//printf("%5d: switch to STATE_MEASURING_GAIN\n", mLoopCounter);
		mDownCounter = 8;
		mDownCounter = getBlockFrames() * 2;
		}
		break;

		@@ -552,15 +669,16 @@ public:
		peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
		// If we get several in a row then go to next state.
		if (peak > mPulseThreshold) {
		if (mDownCounter-- <= 0) {
		mDownCounter -= numFrames;
		if (mDownCounter <= 0) {
		//printf("%5d: switch to STATE_WAITING_FOR_SILENCE, measured peak = %f\n",
		// mLoopCounter, peak);
		mDownCounter = 8;
		mDownCounter = getBlockFrames();
		mMeasuredLoopGain = peak; // assumes original pulse amplitude is one
		mSilenceThreshold = peak * 0.1; // scale silence to measured pulse
		// Calculate gain that will give us a nice decaying echo.
		mEchoGain = mDesiredEchoGain / mMeasuredLoopGain;
		if (mEchoGain > MAX_ECHO_GAIN) {
		if (mEchoGain > kMaxEchoGain) {
		printf("ERROR - loop gain too low. Increase the volume.\n");
		nextState = STATE_FAILED;
		} else {
		@@ -568,7 +686,7 @@ public:
		}
		}
		} else if (numFrames > kImpulseSizeInFrames){ // ignore short callbacks
		mDownCounter = 8;
		mDownCounter = getBlockFrames();
		}
		break;

		@@ -581,13 +699,14 @@ public:
		peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
		// If we get several in a row then go to next state.
		if (peak < mSilenceThreshold) {
		if (mDownCounter-- <= 0) {
		mDownCounter -= numFrames;
		if (mDownCounter <= 0) {
		nextState = STATE_SENDING_PULSE;
		//printf("%5d: switch to STATE_SENDING_PULSE\n", mLoopCounter);
		mDownCounter = 8;
		mDownCounter = getBlockFrames();
		}
		} else {
		mDownCounter = 8;
		mDownCounter = getBlockFrames();
		}
		break;

		@@ -620,11 +739,11 @@ public:
		}
		if (numWritten < numFrames) {
		nextState = STATE_DONE;
		//printf("%5d: switch to STATE_DONE\n", mLoopCounter);
		}
		break;

		case STATE_DONE:
		case STATE_FAILED:
		default:
		break;
		}
		@@ -638,11 +757,22 @@ public:
		}

		int load(const char *fileName) override {
		return mAudioRecording.load(fileName);
		int result = mAudioRecording.load(fileName);
		setSampleRate(mAudioRecording.getSampleRate());
		mState = STATE_DONE;
		return result;
		}

		private:

		enum error_code {
		ERROR_OK = 0,
		ERROR_NOISY = -99,
		ERROR_CLIPPING,
		ERROR_CONFIDENCE,
		ERROR_INVALID_STATE
		};

		enum echo_state {
		STATE_INITIAL_SILENCE,
		STATE_MEASURING_GAIN,
		@@ -708,10 +838,6 @@ private:
		class SineAnalyzer : public LoopbackProcessor {
		public:

		virtual int getResult() {
		return mState == STATE_LOCKED ? 0 : -1;
		}

		void report() override {
		printf("SineAnalyzer ------------------\n");
		printf(LOOPBACK_RESULT_TAG "peak.amplitude = %8f\n", mPeakAmplitude);
		@@ -724,10 +850,9 @@ public:
		float signalToNoiseDB = 10.0 * log(signalToNoise);
		printf(LOOPBACK_RESULT_TAG "signal.to.noise.db = %8.2f\n", signalToNoiseDB);
		if (signalToNoiseDB < MIN_SNRATIO_DB) {
		printf("WARNING - signal to noise ratio is too low! < %d dB\n", MIN_SNRATIO_DB);
		printf("ERROR - signal to noise ratio is too low! < %d dB. Adjust volume.\n", MIN_SNRATIO_DB);
		setResult(ERROR_NOISY);
		}
		printf(LOOPBACK_RESULT_TAG "phase.offset = %8.5f\n", mPhaseOffset);
		printf(LOOPBACK_RESULT_TAG "ref.phase = %8.5f\n", mPhase);
		printf(LOOPBACK_RESULT_TAG "frames.accumulated = %8d\n", mFramesAccumulated);
		printf(LOOPBACK_RESULT_TAG "sine.period = %8d\n", mSinePeriod);
		printf(LOOPBACK_RESULT_TAG "test.state = %8d\n", mState);
		@@ -736,10 +861,15 @@ public:
		bool gotLock = (mState == STATE_LOCKED) \|\| (mGlitchCount > 0);
		if (!gotLock) {
		printf("ERROR - failed to lock on reference sine tone\n");
		setResult(ERROR_NO_LOCK);
		} else {
		// Only print if meaningful.
		printf(LOOPBACK_RESULT_TAG "glitch.count = %8d\n", mGlitchCount);
		printf(LOOPBACK_RESULT_TAG "max.glitch = %8f\n", mMaxGlitchDelta);
		if (mGlitchCount > 0) {
		printf("ERROR - number of glitches > 0\n");
		setResult(ERROR_GLITCHES);
		}
		}
		}

		@@ -913,6 +1043,13 @@ public:

		private:

		enum error_code {
		OK,
		ERROR_NO_LOCK = -80,
		ERROR_GLITCHES,
		ERROR_NOISY
		};

		enum sine_state_t {
		STATE_IDLE,
		STATE_MEASURE_NOISE,
		@@ -963,8 +1100,6 @@ private:
		sine_state_t mState = STATE_IDLE;
		};


		#undef LOOPBACK_SAMPLE_RATE
		#undef LOOPBACK_RESULT_TAG

		#endif /* AAUDIO_EXAMPLES_LOOPBACK_ANALYSER_H */

media/libaaudio/examples/loopback/src/loopback.cpp

+46 −29

File changed.

Preview size limit exceeded, changes collapsed.

media/libaaudio/examples/utils/AAudioArgsParser.h

+4 −2

Original line number	Diff line number	Diff line
		@@ -130,10 +130,12 @@ public:
		}

		int32_t getBufferCapacity() const {
		printf("%s() returns %d\n", __func__, mBufferCapacity);
		return mBufferCapacity;
		}

		void setBufferCapacity(int32_t frames) {
		printf("%s(%d)\n", __func__, frames);
		mBufferCapacity = frames;
		}

		@@ -197,7 +199,7 @@ public:
		* @param builder
		*/
		void applyParameters(AAudioStreamBuilder *builder) const {
		AAudioStreamBuilder_setBufferCapacityInFrames(builder, mBufferCapacity);
		AAudioStreamBuilder_setBufferCapacityInFrames(builder, getBufferCapacity());
		AAudioStreamBuilder_setChannelCount(builder, mChannelCount);
		AAudioStreamBuilder_setDeviceId(builder, mDeviceId);
		AAudioStreamBuilder_setFormat(builder, mFormat);
		@@ -281,7 +283,7 @@ public:
		if (strlen(arg) > 2) {
		policy = atoi(&arg[2]);
		}
		if (!AAudio_setMMapPolicy(policy)) {
		if (AAudio_setMMapPolicy(policy) != AAUDIO_OK) {
		printf("ERROR: invalid MMAP policy mode %i\n", policy);
		}
		} break;