aaudio_loopback: rectify and low pass filter
Help the autocorrelator recognize the peaks easier.
Reduce false detection of suboctaves.
Better timing for devices with high latency.
Improve error reporting.
Set input capacity based on output burst.
Bug: 119032743
Test: see repro case in bug
Test: aaudio_loopback -te -pl
Change-Id: I921c81db882f46ede8988cbdbebda0765e775086
diff --git a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
index 64deb60..9711b86 100644
--- a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
+++ b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
@@ -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 @@
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 @@
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;
-}
-
-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;
-
+static int measureLatencyFromEchos(const float *data,
+ int32_t numFloats,
+ int32_t sampleRate,
+ LatencyReport *report) {
// 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
+ 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]{};
- // Generate correlation for every position.
- numCorrelations = calculateCorrelations(&haystack[first], remaining,
- needle, needleSize,
- correlations, 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);
+ }
- // Add higher harmonics mapped onto lower harmonics.
- // This reinforces the "fundamental" echo.
+ // Apply a technique similar to Harmonic Product Spectrum Analysis to find echo fundamental.
+ // 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 @@
}
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 @@
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
+ float *mData = nullptr;
+ int32_t mFrameCounter = 0;
+ int32_t mMaxFrames = 0;
+ int32_t mSampleRate = kDefaultSampleRate; // common default
+ LowPassFilter mLowPassFilter;
};
// ====================================================================================
@@ -349,8 +454,12 @@
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 @@
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 @@
private:
- int32_t mSampleRate = LOOPBACK_SAMPLE_RATE;
+ int32_t mSampleRate = kDefaultSampleRate;
+ int32_t mResult = 0;
};
class PeakDetector {
@@ -409,24 +519,6 @@
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 @@
}
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 @@
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();
- 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);
+ // 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 @@
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 @@
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 @@
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 @@
}
}
} else if (numFrames > kImpulseSizeInFrames){ // ignore short callbacks
- mDownCounter = 8;
+ mDownCounter = getBlockFrames();
}
break;
@@ -581,13 +699,14 @@
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 @@
}
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 @@
}
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 @@
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 @@
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 @@
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 @@
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 @@
sine_state_t mState = STATE_IDLE;
};
-
-#undef LOOPBACK_SAMPLE_RATE
#undef LOOPBACK_RESULT_TAG
#endif /* AAUDIO_EXAMPLES_LOOPBACK_ANALYSER_H */
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 124efd8..2a02b20 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -35,11 +35,14 @@
#include "AAudioExampleUtils.h"
#include "LoopbackAnalyzer.h"
+// V0.4.00 = rectify and low-pass filter the echos, use auto-correlation on entire echo
+#define APP_VERSION "0.4.00"
+
// Tag for machine readable results as property = value pairs
#define RESULT_TAG "RESULT: "
#define FILENAME_ALL "/data/loopback_all.wav"
#define FILENAME_ECHOS "/data/loopback_echos.wav"
-#define APP_VERSION "0.3.00"
+#define FILENAME_PROCESSED "/data/loopback_processed.wav"
constexpr int kLogPeriodMillis = 1000;
constexpr int kNumInputChannels = 1;
@@ -173,7 +176,8 @@
int64_t inputFramesWritten = AAudioStream_getFramesWritten(myData->inputStream);
int64_t inputFramesRead = AAudioStream_getFramesRead(myData->inputStream);
int64_t framesAvailable = inputFramesWritten - inputFramesRead;
- actualFramesRead = readFormattedData(myData, numFrames);
+
+ actualFramesRead = readFormattedData(myData, numFrames); // READ
if (actualFramesRead < 0) {
result = AAUDIO_CALLBACK_RESULT_STOP;
} else {
@@ -193,6 +197,7 @@
}
myData->insufficientReadCount++;
myData->insufficientReadFrames += numFrames - actualFramesRead; // deficit
+ // printf("Error insufficientReadCount = %d\n",(int)myData->insufficientReadCount);
}
int32_t numSamples = actualFramesRead * myData->actualInputChannelCount;
@@ -355,6 +360,9 @@
printf("%s - Audio loopback using AAudio V" APP_VERSION "\n", argv[0]);
+ // Use LOW_LATENCY as the default to match input default.
+ argParser.setPerformanceMode(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+
for (int i = 1; i < argc; i++) {
const char *arg = argv[i];
if (argParser.parseArg(arg)) {
@@ -403,7 +411,7 @@
}
int32_t requestedDuration = argParser.getDurationSeconds();
- int32_t requestedDurationMillis = requestedDuration * MILLIS_PER_SECOND;
+ int32_t requestedDurationMillis = requestedDuration * kMillisPerSecond;
int32_t timeMillis = 0;
int32_t recordingDuration = std::min(60 * 5, requestedDuration);
@@ -420,9 +428,11 @@
loopbackData.loopbackProcessor = &loopbackData.echoAnalyzer;
int read = loopbackData.loopbackProcessor->load(FILENAME_ECHOS);
- printf("main() read %d mono samples from %s on Android device\n", read, FILENAME_ECHOS);
+ printf("main() read %d mono samples from %s on Android device, rate = %d\n",
+ read, FILENAME_ECHOS,
+ loopbackData.loopbackProcessor->getSampleRate());
loopbackData.loopbackProcessor->report();
- return 0;
+ goto report_result;
}
break;
default:
@@ -580,45 +590,6 @@
printf("input error = %d = %s\n",
loopbackData.inputError, AAudio_convertResultToText(loopbackData.inputError));
- if (loopbackData.inputError == AAUDIO_OK) {
- if (testMode == TEST_SINE_MAGNITUDE) {
- printAudioGraph(loopbackData.audioRecording, 200);
- // Print again so we don't have to scroll past waveform.
- printf("OUTPUT Stream ----------------------------------------\n");
- argParser.compareWithStream(outputStream);
- printf("INPUT Stream ----------------------------------------\n");
- argParser.compareWithStream(inputStream);
- }
-
- loopbackData.loopbackProcessor->report();
- }
-
- {
- int32_t framesRead = AAudioStream_getFramesRead(inputStream);
- int32_t framesWritten = AAudioStream_getFramesWritten(inputStream);
- printf("Callback Results ---------------------------------------- INPUT\n");
- printf(" input overruns = %8d\n", AAudioStream_getXRunCount(inputStream));
- printf(" framesWritten = %8d\n", framesWritten);
- printf(" framesRead = %8d\n", framesRead);
- printf(" myFramesRead = %8d\n", (int) loopbackData.framesReadTotal);
- printf(" written - read = %8d\n", (int) (framesWritten - framesRead));
- printf(" insufficient # = %8d\n", (int) loopbackData.insufficientReadCount);
- if (loopbackData.insufficientReadCount > 0) {
- printf(" insuffic. frames = %8d\n", (int) loopbackData.insufficientReadFrames);
- }
- }
- {
- int32_t framesRead = AAudioStream_getFramesRead(outputStream);
- int32_t framesWritten = AAudioStream_getFramesWritten(outputStream);
- printf("Callback Results ---------------------------------------- OUTPUT\n");
- printf(" output underruns = %8d\n", AAudioStream_getXRunCount(outputStream));
- printf(" myFramesWritten = %8d\n", (int) loopbackData.framesWrittenTotal);
- printf(" framesWritten = %8d\n", framesWritten);
- printf(" framesRead = %8d\n", framesRead);
- printf(" min numFrames = %8d\n", (int) loopbackData.minNumFrames);
- printf(" max numFrames = %8d\n", (int) loopbackData.maxNumFrames);
- }
-
written = loopbackData.loopbackProcessor->save(FILENAME_ECHOS);
if (written > 0) {
printf("main() wrote %8d mono samples to \"%s\" on Android device\n",
@@ -631,10 +602,46 @@
written, FILENAME_ALL);
}
- if (loopbackData.loopbackProcessor->getResult() < 0) {
- printf("ERROR: LOOPBACK PROCESSING FAILED. Maybe because the volume was too low.\n");
- result = loopbackData.loopbackProcessor->getResult();
+ if (loopbackData.inputError == AAUDIO_OK) {
+ if (testMode == TEST_SINE_MAGNITUDE) {
+ printAudioGraph(loopbackData.audioRecording, 200);
+ }
+
+ loopbackData.loopbackProcessor->report();
}
+
+ {
+ int32_t framesRead = AAudioStream_getFramesRead(inputStream);
+ int32_t framesWritten = AAudioStream_getFramesWritten(inputStream);
+ const int64_t framesAvailable = framesWritten - framesRead;
+ printf("Callback Results ---------------------------------------- INPUT\n");
+ printf(" input overruns = %8d\n", AAudioStream_getXRunCount(inputStream));
+ printf(" framesWritten = %8d\n", framesWritten);
+ printf(" framesRead = %8d\n", framesRead);
+ printf(" myFramesRead = %8d\n", (int) loopbackData.framesReadTotal);
+ printf(" written - read = %8d\n", (int) framesAvailable);
+ printf(" insufficient # = %8d\n", (int) loopbackData.insufficientReadCount);
+ if (loopbackData.insufficientReadCount > 0) {
+ printf(" insuffic. frames = %8d\n", (int) loopbackData.insufficientReadFrames);
+ }
+ int32_t actualInputCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
+ if (framesAvailable > 2 * actualInputCapacity) {
+ printf(" WARNING: written - read > 2*capacity !\n");
+ }
+ }
+
+ {
+ int32_t framesRead = AAudioStream_getFramesRead(outputStream);
+ int32_t framesWritten = AAudioStream_getFramesWritten(outputStream);
+ printf("Callback Results ---------------------------------------- OUTPUT\n");
+ printf(" output underruns = %8d\n", AAudioStream_getXRunCount(outputStream));
+ printf(" myFramesWritten = %8d\n", (int) loopbackData.framesWrittenTotal);
+ printf(" framesWritten = %8d\n", framesWritten);
+ printf(" framesRead = %8d\n", framesRead);
+ printf(" min numFrames = %8d\n", (int) loopbackData.minNumFrames);
+ printf(" max numFrames = %8d\n", (int) loopbackData.maxNumFrames);
+ }
+
if (loopbackData.insufficientReadCount > 3) {
printf("ERROR: LOOPBACK PROCESSING FAILED. insufficientReadCount too high\n");
result = AAUDIO_ERROR_UNAVAILABLE;
@@ -646,13 +653,23 @@
delete[] loopbackData.inputFloatData;
delete[] loopbackData.inputShortData;
+report_result:
+ written = loopbackData.loopbackProcessor->save(FILENAME_PROCESSED);
+ if (written > 0) {
+ printf("main() wrote %8d processed samples to \"%s\" on Android device\n",
+ written, FILENAME_PROCESSED);
+ }
+
+ if (loopbackData.loopbackProcessor->getResult() < 0) {
+ result = loopbackData.loopbackProcessor->getResult();
+ }
printf(RESULT_TAG "result = %d \n", result); // machine readable
printf("result is %s\n", AAudio_convertResultToText(result)); // human readable
if (result != AAUDIO_OK) {
- printf("FAILURE\n");
+ printf("TEST FAILED\n");
return EXIT_FAILURE;
} else {
- printf("SUCCESS\n");
+ printf("TEST PASSED\n");
return EXIT_SUCCESS;
}
}
diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index ece9e6a..a5dc55f 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h
@@ -130,10 +130,12 @@
}
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 @@
* @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 @@
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;