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Commit 90ece66a authored by TreeHugger Robot's avatar TreeHugger Robot Committed by Android (Google) Code Review
Browse files

Merge "aaudio_loopback: glitch locations, hang callback"

parents 3f58b119 a98c1aa6
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+15 −6
Original line number Diff line number Diff line
@@ -310,7 +310,7 @@ public:
    }

    // 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 @@ public:
    }

    // 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 @@ public:
        return numFrames;
    }

    int size() {
    int32_t size() {
        return mFrameCounter;
    }

@@ -443,9 +443,14 @@ public:
    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 @@ public:
        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 @@ public:

        mState = nextState;
        mLoopCounter++;
        return PROCESS_RESULT_OK;
    }

    int save(const char *fileName) override {
@@ -896,9 +902,10 @@ public:
     * @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 @@ public:
                    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 @@ public:

            mFrameCounter++;
        }
        return result;
    }

    void resetAccumulator() {
+96 −21
Original line number Diff line number Diff line
@@ -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 @@ struct LoopbackData {
    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 @@ struct LoopbackData {
    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 @@ struct LoopbackData {
    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 @@ static aaudio_data_callback_result_t MyDataCallbackProc(
            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 @@ static aaudio_data_callback_result_t MyDataCallbackProc(
        myData->numCallbacksToDiscard--;

    } else {
        myData->hangIfRequested(numFrames);

        int32_t numInputBytes = numFrames * myData->actualInputChannelCount * sizeof(float);
        memset(myData->inputFloatData, 0 /* value */, numInputBytes);
@@ -191,7 +227,7 @@ static aaudio_data_callback_result_t MyDataCallbackProc(

            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 @@ static aaudio_data_callback_result_t MyDataCallbackProc(
            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 @@ static void usage() {
    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 @@ static int parseTestMode(char c) {
    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 main(int argc, const char **argv)

    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 @@ int main(int argc, const char **argv)
                    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 @@ int main(int argc, const char **argv)
        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 @@ int main(int argc, const char **argv)
    }
    inputStream = loopbackData.inputStream = recorder.getStream();

    argParser.compareWithStream(inputStream);

    {
        int32_t actualCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
        result = AAudioStream_setBufferSizeInFrames(inputStream, actualCapacity);
        (void) AAudioStream_setBufferSizeInFrames(inputStream, actualCapacity);

        if (testMode == TEST_SINE_MAGNITUDE) {
            result = AAudioStream_setBufferSizeInFrames(outputStream, actualCapacity);
            if (result < 0) {
            fprintf(stderr, "ERROR -  AAudioStream_setBufferSizeInFrames() returned %d\n", result);
                fprintf(stderr, "ERROR -  AAudioStream_setBufferSizeInFrames(output) returned %d\n",
                        result);
                goto finish;
        } else {}
            } else {
                printf("Output buffer size set to match input capacity = %d frames.\n", result);
            }
        }

    argParser.compareWithStream(inputStream);

        // If the input stream is too small then we cannot satisfy the output callback.
    {
        int32_t actualCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
        if (actualCapacity < 2 * outputFramesPerBurst) {
            fprintf(stderr, "ERROR - input capacity < 2 * outputFramesPerBurst\n");
            goto finish;
@@ -525,6 +583,8 @@ int main(int argc, const char **argv)

    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 @@ int main(int argc, const char **argv)

    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 @@ finish:
    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",
+0 −2
Original line number Diff line number Diff line
@@ -130,12 +130,10 @@ 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;
    }