Merge "aaudio_loopback: glitch locations, hang callback"
diff --git a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
index 9711b86..8eb70b1 100644
--- a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
+++ b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
@@ -310,7 +310,7 @@
}
// Write SHORT data from the first channel.
- int write(int16_t *inputData, int inputChannelCount, int numFrames) {
+ int32_t write(int16_t *inputData, int32_t inputChannelCount, int32_t numFrames) {
// stop at end of buffer
if ((mFrameCounter + numFrames) > mMaxFrames) {
numFrames = mMaxFrames - mFrameCounter;
@@ -322,7 +322,7 @@
}
// Write FLOAT data from the first channel.
- int write(float *inputData, int inputChannelCount, int numFrames) {
+ int32_t write(float *inputData, int32_t inputChannelCount, int32_t numFrames) {
// stop at end of buffer
if ((mFrameCounter + numFrames) > mMaxFrames) {
numFrames = mMaxFrames - mFrameCounter;
@@ -333,7 +333,7 @@
return numFrames;
}
- int size() {
+ int32_t size() {
return mFrameCounter;
}
@@ -443,9 +443,14 @@
virtual ~LoopbackProcessor() = default;
+ enum process_result {
+ PROCESS_RESULT_OK,
+ PROCESS_RESULT_GLITCH
+ };
+
virtual void reset() {}
- virtual void process(float *inputData, int inputChannelCount,
+ virtual process_result process(float *inputData, int inputChannelCount,
float *outputData, int outputChannelCount,
int numFrames) = 0;
@@ -639,7 +644,7 @@
return getSampleRate() / 8;
}
- void process(float *inputData, int inputChannelCount,
+ process_result process(float *inputData, int inputChannelCount,
float *outputData, int outputChannelCount,
int numFrames) override {
int channelsValid = std::min(inputChannelCount, outputChannelCount);
@@ -750,6 +755,7 @@
mState = nextState;
mLoopCounter++;
+ return PROCESS_RESULT_OK;
}
int save(const char *fileName) override {
@@ -896,9 +902,10 @@
* @param inputData contains microphone data with sine signal feedback
* @param outputData contains the reference sine wave
*/
- void process(float *inputData, int inputChannelCount,
+ process_result process(float *inputData, int inputChannelCount,
float *outputData, int outputChannelCount,
int numFrames) override {
+ process_result result = PROCESS_RESULT_OK;
mProcessCount++;
float peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
@@ -978,6 +985,7 @@
mMaxGlitchDelta = std::max(mMaxGlitchDelta, absDiff);
if (absDiff > mTolerance) {
mGlitchCount++;
+ result = PROCESS_RESULT_GLITCH;
//printf("%5d: Got a glitch # %d, predicted = %f, actual = %f\n",
// mFrameCounter, mGlitchCount, predicted, sample);
mState = STATE_IMMUNE;
@@ -1018,6 +1026,7 @@
mFrameCounter++;
}
+ return result;
}
void resetAccumulator() {
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 3de1514..75d425f 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -34,6 +34,7 @@
#include "AAudioSimpleRecorder.h"
#include "AAudioExampleUtils.h"
#include "LoopbackAnalyzer.h"
+#include "../../utils/AAudioExampleUtils.h"
// V0.4.00 = rectify and low-pass filter the echos, use auto-correlation on entire echo
#define APP_VERSION "0.4.00"
@@ -47,10 +48,14 @@
constexpr int kLogPeriodMillis = 1000;
constexpr int kNumInputChannels = 1;
constexpr int kNumCallbacksToDrain = 20;
+constexpr int kNumCallbacksToNotRead = 0; // let input fill back up
constexpr int kNumCallbacksToDiscard = 20;
+constexpr int kDefaultHangTimeMillis = 50;
+constexpr int kMaxGlitchEventsToSave = 32;
struct LoopbackData {
AAudioStream *inputStream = nullptr;
+ AAudioStream *outputStream = nullptr;
int32_t inputFramesMaximum = 0;
int16_t *inputShortData = nullptr;
float *inputFloatData = nullptr;
@@ -58,6 +63,7 @@
int32_t actualInputChannelCount = 0;
int32_t actualOutputChannelCount = 0;
int32_t numCallbacksToDrain = kNumCallbacksToDrain;
+ int32_t numCallbacksToNotRead = kNumCallbacksToNotRead;
int32_t numCallbacksToDiscard = kNumCallbacksToDiscard;
int32_t minNumFrames = INT32_MAX;
int32_t maxNumFrames = 0;
@@ -65,6 +71,9 @@
int32_t insufficientReadFrames = 0;
int32_t framesReadTotal = 0;
int32_t framesWrittenTotal = 0;
+ int32_t hangPeriodMillis = 5 * 1000; // time between hangs
+ int32_t hangCountdownFrames = 5 * 48000; // frames til next hang
+ int32_t hangTimeMillis = 0; // 0 for no hang
bool isDone = false;
aaudio_result_t inputError = AAUDIO_OK;
@@ -74,6 +83,29 @@
EchoAnalyzer echoAnalyzer;
AudioRecording audioRecording;
LoopbackProcessor *loopbackProcessor;
+
+ int32_t glitchFrames[kMaxGlitchEventsToSave];
+ int32_t numGlitchEvents = 0;
+
+ void hangIfRequested(int32_t numFrames) {
+ if (hangTimeMillis > 0) {
+ hangCountdownFrames -= numFrames;
+ if (hangCountdownFrames <= 0) {
+ const int64_t startNanos = getNanoseconds();
+ usleep(hangTimeMillis * 1000);
+ const int64_t endNanos = getNanoseconds();
+ const int32_t elapsedMicros = (int32_t)
+ ((endNanos - startNanos) / 1000);
+ printf("callback hanging for %d millis, actual = %d micros\n",
+ hangTimeMillis, elapsedMicros);
+ hangCountdownFrames = (int64_t) hangPeriodMillis
+ * AAudioStream_getSampleRate(outputStream)
+ / 1000;
+ }
+ }
+
+
+ }
};
static void convertPcm16ToFloat(const int16_t *source,
@@ -166,6 +198,9 @@
myData->numCallbacksToDrain--;
}
+ } else if (myData->numCallbacksToNotRead > 0) {
+ // Let the input fill up a bit so we are not so close to the write pointer.
+ myData->numCallbacksToNotRead--;
} else if (myData->numCallbacksToDiscard > 0) {
// Ignore. Allow the input to fill back up to equilibrium with the output.
actualFramesRead = readFormattedData(myData, numFrames);
@@ -175,6 +210,7 @@
myData->numCallbacksToDiscard--;
} else {
+ myData->hangIfRequested(numFrames);
int32_t numInputBytes = numFrames * myData->actualInputChannelCount * sizeof(float);
memset(myData->inputFloatData, 0 /* value */, numInputBytes);
@@ -191,7 +227,7 @@
if (actualFramesRead < numFrames) {
if(actualFramesRead < (int32_t) framesAvailable) {
- printf("insufficient but numFrames = %d"
+ printf("insufficient for no reason, numFrames = %d"
", actualFramesRead = %d"
", inputFramesWritten = %d"
", inputFramesRead = %d"
@@ -212,16 +248,25 @@
if (myData->actualInputFormat == AAUDIO_FORMAT_PCM_I16) {
convertPcm16ToFloat(myData->inputShortData, myData->inputFloatData, numSamples);
}
- // Save for later.
- myData->audioRecording.write(myData->inputFloatData,
- myData->actualInputChannelCount,
- numFrames);
+
// Analyze the data.
- myData->loopbackProcessor->process(myData->inputFloatData,
+ LoopbackProcessor::process_result procResult = myData->loopbackProcessor->process(myData->inputFloatData,
myData->actualInputChannelCount,
outputData,
myData->actualOutputChannelCount,
numFrames);
+
+ if (procResult == LoopbackProcessor::PROCESS_RESULT_GLITCH) {
+ if (myData->numGlitchEvents < kMaxGlitchEventsToSave) {
+ myData->glitchFrames[myData->numGlitchEvents++] = myData->audioRecording.size();
+ }
+ }
+
+ // Save for later.
+ myData->audioRecording.write(myData->inputFloatData,
+ myData->actualInputChannelCount,
+ actualFramesRead);
+
myData->isDone = myData->loopbackProcessor->isDone();
if (myData->isDone) {
result = AAUDIO_CALLBACK_RESULT_STOP;
@@ -249,6 +294,7 @@
printf(" -C{channels} number of input channels\n");
printf(" -F{0,1,2} input format, 1=I16, 2=FLOAT\n");
printf(" -g{gain} recirculating loopback gain\n");
+ printf(" -h{hangMillis} occasionally hang in the callback\n");
printf(" -P{inPerf} set input AAUDIO_PERFORMANCE_MODE*\n");
printf(" n for _NONE\n");
printf(" l for _LATENCY\n");
@@ -307,9 +353,7 @@
return testMode;
}
-void printAudioGraph(AudioRecording &recording, int numSamples) {
- int32_t start = recording.size() / 2;
- int32_t end = start + numSamples;
+void printAudioGraphRegion(AudioRecording &recording, int32_t start, int32_t end) {
if (end >= recording.size()) {
end = recording.size() - 1;
}
@@ -360,6 +404,7 @@
int testMode = TEST_ECHO_LATENCY;
double gain = 1.0;
+ int hangTimeMillis = 0;
// Make printf print immediately so that debug info is not stuck
// in a buffer if we hang or crash.
@@ -389,6 +434,15 @@
case 'g':
gain = atof(&arg[2]);
break;
+ case 'h':
+ // Was there a number after the "-h"?
+ if (arg[2]) {
+ hangTimeMillis = atoi(&arg[2]);
+ } else {
+ // If no number then use the default.
+ hangTimeMillis = kDefaultHangTimeMillis;
+ }
+ break;
case 'P':
inputPerformanceLevel = parsePerformanceMode(arg[2]);
break;
@@ -453,7 +507,7 @@
fprintf(stderr, "ERROR - player.open() returned %d\n", result);
exit(1);
}
- outputStream = player.getStream();
+ outputStream = loopbackData.outputStream = player.getStream();
actualOutputFormat = AAudioStream_getFormat(outputStream);
if (actualOutputFormat != AAUDIO_FORMAT_PCM_FLOAT) {
@@ -487,20 +541,24 @@
}
inputStream = loopbackData.inputStream = recorder.getStream();
- {
- int32_t actualCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
- result = AAudioStream_setBufferSizeInFrames(inputStream, actualCapacity);
- if (result < 0) {
- fprintf(stderr, "ERROR - AAudioStream_setBufferSizeInFrames() returned %d\n", result);
- goto finish;
- } else {}
- }
-
argParser.compareWithStream(inputStream);
- // If the input stream is too small then we cannot satisfy the output callback.
{
int32_t actualCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
+ (void) AAudioStream_setBufferSizeInFrames(inputStream, actualCapacity);
+
+ if (testMode == TEST_SINE_MAGNITUDE) {
+ result = AAudioStream_setBufferSizeInFrames(outputStream, actualCapacity);
+ if (result < 0) {
+ fprintf(stderr, "ERROR - AAudioStream_setBufferSizeInFrames(output) returned %d\n",
+ result);
+ goto finish;
+ } else {
+ printf("Output buffer size set to match input capacity = %d frames.\n", result);
+ }
+ }
+
+ // If the input stream is too small then we cannot satisfy the output callback.
if (actualCapacity < 2 * outputFramesPerBurst) {
fprintf(stderr, "ERROR - input capacity < 2 * outputFramesPerBurst\n");
goto finish;
@@ -525,6 +583,8 @@
loopbackData.loopbackProcessor->reset();
+ loopbackData.hangTimeMillis = hangTimeMillis;
+
// Start OUTPUT first so INPUT does not overflow.
result = player.start();
if (result != AAUDIO_OK) {
@@ -611,7 +671,17 @@
if (loopbackData.inputError == AAUDIO_OK) {
if (testMode == TEST_SINE_MAGNITUDE) {
- printAudioGraph(loopbackData.audioRecording, 200);
+ if (loopbackData.numGlitchEvents > 0) {
+ // Graph around the first glitch if there is one.
+ const int32_t start = loopbackData.glitchFrames[0] - 8;
+ const int32_t end = start + outputFramesPerBurst + 8 + 8;
+ printAudioGraphRegion(loopbackData.audioRecording, start, end);
+ } else {
+ // Or graph the middle of the signal.
+ const int32_t start = loopbackData.audioRecording.size() / 2;
+ const int32_t end = start + 200;
+ printAudioGraphRegion(loopbackData.audioRecording, start, end);
+ }
}
loopbackData.loopbackProcessor->report();
@@ -661,6 +731,11 @@
delete[] loopbackData.inputShortData;
report_result:
+
+ for (int i = 0; i < loopbackData.numGlitchEvents; i++) {
+ printf(" glitch at frame %d\n", loopbackData.glitchFrames[i]);
+ }
+
written = loopbackData.loopbackProcessor->save(FILENAME_PROCESSED);
if (written > 0) {
printf("main() wrote %8d processed samples to \"%s\" on Android device\n",
diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index a5dc55f..f5ed7aa 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h
@@ -130,12 +130,10 @@
}
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;
}