diff --git a/media/libaaudio/examples/loopback/Android.bp b/media/libaaudio/examples/loopback/Android.bp
index fa8fdc9..5b7d956 100644
--- a/media/libaaudio/examples/loopback/Android.bp
+++ b/media/libaaudio/examples/loopback/Android.bp
@@ -3,6 +3,10 @@
     gtest: false,
     srcs: ["src/loopback.cpp"],
     cflags: ["-Wall", "-Werror"],
-    shared_libs: ["libaaudio"],
+    static_libs: ["libsndfile"],
+    shared_libs: [
+        "libaaudio",
+        "libaudioutils",
+        ],
     header_libs: ["libaaudio_example_utils"],
 }
diff --git a/media/libaaudio/examples/loopback/jni/Android.mk b/media/libaaudio/examples/loopback/jni/Android.mk
index 1fe3def..aebe877 100644
--- a/media/libaaudio/examples/loopback/jni/Android.mk
+++ b/media/libaaudio/examples/loopback/jni/Android.mk
@@ -10,6 +10,7 @@
 # NDK recommends using this kind of relative path instead of an absolute path.
 LOCAL_SRC_FILES:= ../src/loopback.cpp
 LOCAL_CFLAGS := -Wall -Werror
-LOCAL_SHARED_LIBRARIES := libaaudio
+LOCAL_STATIC_LIBRARIES := libsndfile
+LOCAL_SHARED_LIBRARIES := libaaudio libaudioutils
 LOCAL_MODULE := aaudio_loopback
 include $(BUILD_EXECUTABLE)
diff --git a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
index 276b45f..b83851a 100644
--- a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
+++ b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
@@ -30,6 +30,8 @@
 #include <stdlib.h>
 #include <unistd.h>
 
+#include <audio_utils/sndfile.h>
+
 // Tag for machine readable results as property = value pairs
 #define LOOPBACK_RESULT_TAG      "RESULT: "
 #define LOOPBACK_SAMPLE_RATE     48000
@@ -37,6 +39,7 @@
 #define MILLIS_PER_SECOND        1000
 
 #define MAX_ZEROTH_PARTIAL_BINS  40
+constexpr double MAX_ECHO_GAIN = 10.0; // based on experiments, otherwise autocorrelation too noisy
 
 static const float s_Impulse[] = {
         0.0f, 0.0f, 0.0f, 0.0f, 0.2f, // silence on each side of the impulse
@@ -156,6 +159,8 @@
                             const float *needle, int needleSize,
                             LatencyReport *report) {
     const double threshold = 0.1;
+    printf("measureLatencyFromEchos: haystackSize = %d, needleSize = %d\n",
+           haystackSize, needleSize);
 
     // Find first peak
     int first = (int) (findFirstMatch(haystack,
@@ -173,7 +178,7 @@
                                       needleSize,
                                       threshold) + 0.5);
 
-    printf("first = %d, again at %d\n", first, again);
+    printf("measureLatencyFromEchos: first = %d, again at %d\n", first, again);
     first = again;
 
     // Allocate results array
@@ -270,37 +275,60 @@
         return mData;
     }
 
+    void setSampleRate(int32_t sampleRate) {
+        mSampleRate = sampleRate;
+    }
+
+    int32_t getSampleRate() {
+        return mSampleRate;
+    }
+
     int save(const char *fileName, bool writeShorts = true) {
+        SNDFILE *sndFile = nullptr;
         int written = 0;
-        const int chunkSize = 64;
-        FILE *fid = fopen(fileName, "wb");
-        if (fid == NULL) {
+        SF_INFO info = {
+                .frames = mFrameCounter,
+                .samplerate = mSampleRate,
+                .channels = 1,
+                .format = SF_FORMAT_WAV | (writeShorts ? SF_FORMAT_PCM_16 : SF_FORMAT_FLOAT)
+        };
+
+        sndFile = sf_open(fileName, SFM_WRITE, &info);
+        if (sndFile == nullptr) {
+            printf("AudioRecording::save(%s) failed to open file\n", fileName);
             return -errno;
         }
 
-        if (writeShorts) {
-            int16_t buffer[chunkSize];
-            int32_t framesLeft = mFrameCounter;
-            int32_t cursor = 0;
-            while (framesLeft) {
-                int32_t framesToWrite = framesLeft < chunkSize ? framesLeft : chunkSize;
-                for (int i = 0; i < framesToWrite; i++) {
-                    buffer[i] = (int16_t) (mData[cursor++] * 32767);
-                }
-                written += fwrite(buffer, sizeof(int16_t), framesToWrite, fid);
-                framesLeft -= framesToWrite;
-            }
-        } else {
-            written = (int) fwrite(mData, sizeof(float), mFrameCounter, fid);
-        }
-        fclose(fid);
+        written = sf_writef_float(sndFile, mData, mFrameCounter);
+
+        sf_close(sndFile);
         return written;
     }
 
+    int load(const char *fileName) {
+        SNDFILE *sndFile = nullptr;
+        SF_INFO info;
+
+        sndFile = sf_open(fileName, SFM_READ, &info);
+        if (sndFile == nullptr) {
+            printf("AudioRecording::load(%s) failed to open file\n", fileName);
+            return -errno;
+        }
+
+        assert(info.channels == 1);
+
+        allocate(info.frames);
+        mFrameCounter = sf_readf_float(sndFile, mData, info.frames);
+
+        sf_close(sndFile);
+        return mFrameCounter;
+    }
+
 private:
     float  *mData = nullptr;
     int32_t mFrameCounter = 0;
     int32_t mMaxFrames = 0;
+    int32_t mSampleRate = 48000; // common default
 };
 
 // ====================================================================================
@@ -320,11 +348,25 @@
 
     virtual void printStatus() {};
 
+    virtual int getResult() {
+        return -1;
+    }
+
     virtual bool isDone() {
         return false;
     }
 
-    void setSampleRate(int32_t sampleRate) {
+    virtual int save(const char *fileName) {
+        (void) fileName;
+        return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+
+    virtual int load(const char *fileName) {
+        (void) fileName;
+        return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+
+    virtual void setSampleRate(int32_t sampleRate) {
         mSampleRate = sampleRate;
     }
 
@@ -395,7 +437,13 @@
 public:
 
     EchoAnalyzer() : LoopbackProcessor() {
-        audioRecorder.allocate(2 * LOOPBACK_SAMPLE_RATE);
+        mAudioRecording.allocate(2 * getSampleRate());
+        mAudioRecording.setSampleRate(getSampleRate());
+    }
+
+    void setSampleRate(int32_t sampleRate) override {
+        LoopbackProcessor::setSampleRate(sampleRate);
+        mAudioRecording.setSampleRate(sampleRate);
     }
 
     void reset() override {
@@ -406,8 +454,12 @@
         mState = STATE_INITIAL_SILENCE;
     }
 
+    virtual int getResult() {
+        return mState == STATE_DONE ? 0 : -1;
+    }
+
     virtual bool isDone() {
-        return mState == STATE_DONE;
+        return mState == STATE_DONE || mState == STATE_FAILED;
     }
 
     void setGain(float gain) {
@@ -423,31 +475,24 @@
         printf("EchoAnalyzer ---------------\n");
         printf(LOOPBACK_RESULT_TAG "measured.gain          = %f\n", mMeasuredLoopGain);
         printf(LOOPBACK_RESULT_TAG "echo.gain              = %f\n", mEchoGain);
-        printf(LOOPBACK_RESULT_TAG "frame.count            = %d\n", mFrameCounter);
         printf(LOOPBACK_RESULT_TAG "test.state             = %d\n", mState);
         if (mMeasuredLoopGain >= 0.9999) {
             printf("   ERROR - clipping, turn down volume slightly\n");
         } else {
             const float *needle = s_Impulse;
             int needleSize = (int) (sizeof(s_Impulse) / sizeof(float));
-            float *haystack = audioRecorder.getData();
-            int haystackSize = audioRecorder.size();
-            measureLatencyFromEchos(haystack, haystackSize, needle, needleSize, &latencyReport);
-            if (latencyReport.confidence < 0.01) {
-                printf("   ERROR - confidence too low = %f\n", latencyReport.confidence);
+            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 * latencyReport.latencyInFrames / getSampleRate();
-                printf(LOOPBACK_RESULT_TAG "latency.frames        = %8.2f\n", latencyReport.latencyInFrames);
+                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", latencyReport.confidence);
+                printf(LOOPBACK_RESULT_TAG "latency.confidence    = %8.6f\n", mLatencyReport.confidence);
             }
         }
-
-        {
-#define ECHO_FILENAME "/data/oboe_echo.raw"
-            int written = audioRecorder.save(ECHO_FILENAME);
-            printf("Echo wrote %d mono samples to %s on Android device\n", written, ECHO_FILENAME);
-        }
     }
 
     void printStatus() override {
@@ -491,13 +536,18 @@
                 // If we get several in a row then go to next state.
                 if (peak > mPulseThreshold) {
                     if (mDownCounter-- <= 0) {
-                        nextState = STATE_WAITING_FOR_SILENCE;
                         //printf("%5d: switch to STATE_WAITING_FOR_SILENCE, measured peak = %f\n",
                         //       mLoopCounter, peak);
                         mDownCounter = 8;
                         mMeasuredLoopGain = peak;  // assumes original pulse amplitude is one
                         // Calculate gain that will give us a nice decaying echo.
                         mEchoGain = mDesiredEchoGain / mMeasuredLoopGain;
+                        if (mEchoGain > MAX_ECHO_GAIN) {
+                            printf("ERROR - loop gain too low. Increase the volume.\n");
+                            nextState = STATE_FAILED;
+                        } else {
+                            nextState = STATE_WAITING_FOR_SILENCE;
+                        }
                     }
                 } else {
                     mDownCounter = 8;
@@ -524,14 +574,14 @@
                 break;
 
             case STATE_SENDING_PULSE:
-                audioRecorder.write(inputData, inputChannelCount, numFrames);
+                mAudioRecording.write(inputData, inputChannelCount, numFrames);
                 sendImpulse(outputData, outputChannelCount);
                 nextState = STATE_GATHERING_ECHOS;
                 //printf("%5d: switch to STATE_GATHERING_ECHOS\n", mLoopCounter);
                 break;
 
             case STATE_GATHERING_ECHOS:
-                numWritten = audioRecorder.write(inputData, inputChannelCount, numFrames);
+                numWritten = mAudioRecording.write(inputData, inputChannelCount, numFrames);
                 peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
                 if (peak > mMeasuredLoopGain) {
                     mMeasuredLoopGain = peak;  // AGC might be raising gain so adjust it on the fly.
@@ -565,6 +615,14 @@
         mLoopCounter++;
     }
 
+    int save(const char *fileName) override {
+        return mAudioRecording.save(fileName);
+    }
+
+    int load(const char *fileName) override {
+        return mAudioRecording.load(fileName);
+    }
+
 private:
 
     enum echo_state_t {
@@ -573,22 +631,22 @@
         STATE_WAITING_FOR_SILENCE,
         STATE_SENDING_PULSE,
         STATE_GATHERING_ECHOS,
-        STATE_DONE
+        STATE_DONE,
+        STATE_FAILED
     };
 
-    int           mDownCounter = 500;
-    int           mLoopCounter = 0;
-    float         mPulseThreshold = 0.02f;
-    float         mSilenceThreshold = 0.002f;
-    float         mMeasuredLoopGain = 0.0f;
-    float         mDesiredEchoGain = 0.95f;
-    float         mEchoGain = 1.0f;
-    echo_state_t  mState = STATE_INITIAL_SILENCE;
-    int32_t       mFrameCounter = 0;
+    int             mDownCounter = 500;
+    int             mLoopCounter = 0;
+    float           mPulseThreshold = 0.02f;
+    float           mSilenceThreshold = 0.002f;
+    float           mMeasuredLoopGain = 0.0f;
+    float           mDesiredEchoGain = 0.95f;
+    float           mEchoGain = 1.0f;
+    echo_state_t    mState = STATE_INITIAL_SILENCE;
 
-    AudioRecording     audioRecorder;
-    LatencyReport      latencyReport;
-    PeakDetector       mPeakDetector;
+    AudioRecording  mAudioRecording; // contains only the input after the gain detection burst
+    LatencyReport   mLatencyReport;
+    // PeakDetector    mPeakDetector;
 };
 
 
@@ -602,6 +660,10 @@
 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     = %7.5f\n", mPeakAmplitude);
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index ac6024e..d23d907 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -37,10 +37,10 @@
 
 // Tag for machine readable results as property = value pairs
 #define RESULT_TAG              "RESULT: "
-#define SAMPLE_RATE             48000
 #define NUM_SECONDS             5
 #define NUM_INPUT_CHANNELS      1
-#define FILENAME                "/data/oboe_input.raw"
+#define FILENAME_ALL            "/data/loopback_all.wav"
+#define FILENAME_ECHOS          "/data/loopback_echos.wav"
 #define APP_VERSION             "0.1.22"
 
 struct LoopbackData {
@@ -61,7 +61,7 @@
 
     SineAnalyzer       sineAnalyzer;
     EchoAnalyzer       echoAnalyzer;
-    AudioRecording     audioRecorder;
+    AudioRecording     audioRecording;
     LoopbackProcessor *loopbackProcessor;
 };
 
@@ -126,7 +126,7 @@
             result = AAUDIO_CALLBACK_RESULT_STOP;
         } else if (framesRead > 0) {
 
-            myData->audioRecorder.write(myData->inputData,
+            myData->audioRecording.write(myData->inputData,
                                         myData->actualInputChannelCount,
                                         numFrames);
 
@@ -176,7 +176,8 @@
     printf("              p for _POWER_SAVING\n");
     printf("          -t{test}          select test mode\n");
     printf("              m for sine magnitude\n");
-    printf("              e for echo latency (default)\n\n");
+    printf("              e for echo latency (default)\n");
+    printf("              f for file latency, analyzes %s\n\n", FILENAME_ECHOS);
     printf("          -x                use EXCLUSIVE mode for output\n");
     printf("          -X                use EXCLUSIVE mode for input\n");
     printf("Example:  aaudio_loopback -n2 -pl -Pl -x\n");
@@ -205,6 +206,7 @@
 enum {
     TEST_SINE_MAGNITUDE = 0,
     TEST_ECHO_LATENCY,
+    TEST_FILE_LATENCY,
 };
 
 static int parseTestMode(char c) {
@@ -217,6 +219,9 @@
         case 'e':
             testMode = TEST_ECHO_LATENCY;
             break;
+        case 'f':
+            testMode = TEST_FILE_LATENCY;
+            break;
         default:
             printf("ERROR in value test mode %c\n", c);
             break;
@@ -254,13 +259,13 @@
 int main(int argc, const char **argv)
 {
 
-    AAudioArgsParser     argParser;
-    AAudioSimplePlayer   player;
-    AAudioSimpleRecorder recorder;
-    LoopbackData         loopbackData;
-    AAudioStream        *outputStream = nullptr;
+    AAudioArgsParser      argParser;
+    AAudioSimplePlayer    player;
+    AAudioSimpleRecorder  recorder;
+    LoopbackData          loopbackData;
+    AAudioStream         *outputStream = nullptr;
 
-    aaudio_result_t      result = AAUDIO_OK;
+    aaudio_result_t       result = AAUDIO_OK;
     aaudio_sharing_mode_t requestedInputSharingMode     = AAUDIO_SHARING_MODE_SHARED;
     int                   requestedInputChannelCount = NUM_INPUT_CHANNELS;
     const aaudio_format_t requestedInputFormat = AAUDIO_FORMAT_PCM_I16;
@@ -268,6 +273,7 @@
     aaudio_format_t       actualInputFormat;
     aaudio_format_t       actualOutputFormat;
     aaudio_performance_mode_t inputPerformanceLevel = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
+    int32_t               actualSampleRate = 0;
 
     int testMode = TEST_ECHO_LATENCY;
     double gain = 1.0;
@@ -324,7 +330,6 @@
 
     int32_t requestedDuration = argParser.getDurationSeconds();
     int32_t recordingDuration = std::min(60, requestedDuration);
-    loopbackData.audioRecorder.allocate(recordingDuration * SAMPLE_RATE);
 
     switch(testMode) {
         case TEST_SINE_MAGNITUDE:
@@ -334,6 +339,16 @@
             loopbackData.echoAnalyzer.setGain(gain);
             loopbackData.loopbackProcessor = &loopbackData.echoAnalyzer;
             break;
+        case TEST_FILE_LATENCY: {
+            loopbackData.echoAnalyzer.setGain(gain);
+
+            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);
+            loopbackData.loopbackProcessor->report();
+            return 0;
+        }
+            break;
         default:
             exit(1);
             break;
@@ -344,7 +359,7 @@
     result = player.open(argParser, MyDataCallbackProc, MyErrorCallbackProc, &loopbackData);
     if (result != AAUDIO_OK) {
         fprintf(stderr, "ERROR -  player.open() returned %d\n", result);
-        goto finish;
+        exit(1);
     }
     outputStream = player.getStream();
     argParser.compareWithStream(outputStream);
@@ -352,6 +367,10 @@
     actualOutputFormat = AAudioStream_getFormat(outputStream);
     assert(actualOutputFormat == AAUDIO_FORMAT_PCM_FLOAT);
 
+    actualSampleRate = AAudioStream_getSampleRate(outputStream);
+    loopbackData.audioRecording.allocate(recordingDuration * actualSampleRate);
+    loopbackData.audioRecording.setSampleRate(actualSampleRate);
+
     printf("INPUT stream ----------------------------------------\n");
     // Use different parameters for the input.
     argParser.setNumberOfBursts(AAUDIO_UNSPECIFIED);
@@ -380,7 +399,7 @@
 
     // Allocate a buffer for the audio data.
     loopbackData.inputFramesMaximum = 32 * framesPerBurst;
-    loopbackData.inputBuffersToDiscard = 100;
+    loopbackData.inputBuffersToDiscard = 200;
 
     loopbackData.inputData = new int16_t[loopbackData.inputFramesMaximum
                                          * loopbackData.actualInputChannelCount];
@@ -436,25 +455,31 @@
         }
     }
 
-    printf("input error = %d = %s\n",
-                loopbackData.inputError, AAudio_convertResultToText(loopbackData.inputError));
+    if (loopbackData.loopbackProcessor->getResult() < 0) {
+        printf("Test failed!\n");
+    } else {
+        printf("input error = %d = %s\n",
+               loopbackData.inputError, AAudio_convertResultToText(loopbackData.inputError));
 
-    printf("AAudioStream_getXRunCount %d\n", AAudioStream_getXRunCount(outputStream));
-    printf("framesRead    = %8d\n", (int) AAudioStream_getFramesRead(outputStream));
-    printf("framesWritten = %8d\n", (int) AAudioStream_getFramesWritten(outputStream));
-    printf("min numFrames = %8d\n", (int) loopbackData.minNumFrames);
-    printf("max numFrames = %8d\n", (int) loopbackData.maxNumFrames);
+        printf("AAudioStream_getXRunCount %d\n", AAudioStream_getXRunCount(outputStream));
+        printf("framesRead    = %8d\n", (int) AAudioStream_getFramesRead(outputStream));
+        printf("framesWritten = %8d\n", (int) AAudioStream_getFramesWritten(outputStream));
+        printf("min numFrames = %8d\n", (int) loopbackData.minNumFrames);
+        printf("max numFrames = %8d\n", (int) loopbackData.maxNumFrames);
 
-    if (loopbackData.inputError == AAUDIO_OK) {
-        if (testMode == TEST_SINE_MAGNITUDE) {
-            printAudioGraph(loopbackData.audioRecorder, 200);
+        if (loopbackData.inputError == AAUDIO_OK) {
+            if (testMode == TEST_SINE_MAGNITUDE) {
+                printAudioGraph(loopbackData.audioRecording, 200);
+            }
+            loopbackData.loopbackProcessor->report();
         }
-        loopbackData.loopbackProcessor->report();
-    }
 
-    {
-        int written = loopbackData.audioRecorder.save(FILENAME);
-        printf("main() wrote %d mono samples to %s on Android device\n", written, FILENAME);
+        int written = loopbackData.loopbackProcessor->save(FILENAME_ECHOS);
+        printf("main() wrote %d mono samples to %s on Android device\n", written,
+               FILENAME_ECHOS);
+        printf("main() loopbackData.audioRecording.getSampleRate() = %d\n", loopbackData.audioRecording.getSampleRate());
+        written = loopbackData.audioRecording.save(FILENAME_ALL);
+        printf("main() wrote %d mono samples to %s on Android device\n", written, FILENAME_ALL);
     }
 
 finish:
diff --git a/media/libaaudio/examples/utils/AAudioSimplePlayer.h b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
index 3fafecf..54b77ba 100644
--- a/media/libaaudio/examples/utils/AAudioSimplePlayer.h
+++ b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
@@ -170,7 +170,6 @@
 
     aaudio_result_t close() {
         if (mStream != nullptr) {
-            printf("call AAudioStream_close(%p)\n", mStream);  fflush(stdout);
             AAudioStream_close(mStream);
             mStream = nullptr;
         }
diff --git a/media/libaaudio/examples/utils/AAudioSimpleRecorder.h b/media/libaaudio/examples/utils/AAudioSimpleRecorder.h
index 1344273..869fad0 100644
--- a/media/libaaudio/examples/utils/AAudioSimpleRecorder.h
+++ b/media/libaaudio/examples/utils/AAudioSimpleRecorder.h
@@ -178,7 +178,6 @@
 
     aaudio_result_t close() {
         if (mStream != nullptr) {
-            printf("call AAudioStream_close(%p)\n", mStream);  fflush(stdout);
             AAudioStream_close(mStream);
             mStream = nullptr;
         }