diff --git a/media/libaaudio/src/binding/IAAudioService.cpp b/media/libaaudio/src/binding/IAAudioService.cpp
index b8ef611..ef4a51f 100644
--- a/media/libaaudio/src/binding/IAAudioService.cpp
+++ b/media/libaaudio/src/binding/IAAudioService.cpp
@@ -45,11 +45,9 @@
         Parcel data, reply;
         // send command
         data.writeInterfaceToken(IAAudioService::getInterfaceDescriptor());
-        ALOGV("BpAAudioService::client openStream --------------------");
         // request.dump();
         request.writeToParcel(&data);
         status_t err = remote()->transact(OPEN_STREAM, data, &reply);
-        ALOGV("BpAAudioService::client openStream returned %d", err);
         if (err != NO_ERROR) {
             ALOGE("BpAAudioService::client openStream transact failed %d", err);
             return AAudioConvert_androidToAAudioResult(err);
@@ -57,6 +55,7 @@
         // parse reply
         aaudio_handle_t stream;
         err = reply.readInt32(&stream);
+        ALOGD("BpAAudioService::client openStream returned stream = 0x%08x", stream);
         if (err != NO_ERROR) {
             ALOGE("BpAAudioService::client transact(OPEN_STREAM) readInt %d", err);
             return AAudioConvert_androidToAAudioResult(err);
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 5cb642b..0684ed6 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -246,3 +246,7 @@
     return (int32_t)mDataQueue->getBufferCapacityInFrames();
 }
 
+void AudioEndpoint::dump() const {
+    ALOGD("AudioEndpoint: data readCounter  = %lld", (long long) mDataQueue->getReadCounter());
+    ALOGD("AudioEndpoint: data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());
+}
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index 53ba033..e7c6916 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -91,6 +91,8 @@
 
     int32_t getBufferCapacityInFrames() const;
 
+    void dump() const;
+
 private:
     android::FifoBuffer    *mUpCommandQueue;
     android::FifoBuffer    *mDataQueue;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 768d9db..b59c445 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -63,8 +63,8 @@
         , mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
         , mFramesPerBurst(16)
         , mStreamVolume(1.0f)
-        , mServiceInterface(serviceInterface)
-        , mInService(inService) {
+        , mInService(inService)
+        , mServiceInterface(serviceInterface) {
 }
 
 AudioStreamInternal::~AudioStreamInternal() {
@@ -154,13 +154,13 @@
         if (getDataCallbackProc()) {
             mCallbackFrames = builder.getFramesPerDataCallback();
             if (mCallbackFrames > getBufferCapacity() / 2) {
-                ALOGE("AudioStreamInternal.open(): framesPerCallback too large = %d, capacity = %d",
+                ALOGE("AudioStreamInternal::open(): framesPerCallback too big = %d, capacity = %d",
                       mCallbackFrames, getBufferCapacity());
                 mServiceInterface.closeStream(mServiceStreamHandle);
                 return AAUDIO_ERROR_OUT_OF_RANGE;
 
             } else if (mCallbackFrames < 0) {
-                ALOGE("AudioStreamInternal.open(): framesPerCallback negative");
+                ALOGE("AudioStreamInternal::open(): framesPerCallback negative");
                 mServiceInterface.closeStream(mServiceStreamHandle);
                 return AAUDIO_ERROR_OUT_OF_RANGE;
 
@@ -185,7 +185,7 @@
 }
 
 aaudio_result_t AudioStreamInternal::close() {
-    ALOGD("AudioStreamInternal.close(): mServiceStreamHandle = 0x%08X",
+    ALOGD("AudioStreamInternal::close(): mServiceStreamHandle = 0x%08X",
              mServiceStreamHandle);
     if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
         // Don't close a stream while it is running.
@@ -201,12 +201,14 @@
                 result, AAudio_convertResultToText(result));
             }
         }
+        setState(AAUDIO_STREAM_STATE_CLOSING);
         aaudio_handle_t serviceStreamHandle = mServiceStreamHandle;
         mServiceStreamHandle = AAUDIO_HANDLE_INVALID;
 
         mServiceInterface.closeStream(serviceStreamHandle);
         delete[] mCallbackBuffer;
         mCallbackBuffer = nullptr;
+        setState(AAUDIO_STREAM_STATE_CLOSED);
         return mEndPointParcelable.close();
     } else {
         return AAUDIO_ERROR_INVALID_HANDLE;
@@ -228,15 +230,20 @@
 aaudio_result_t AudioStreamInternal::requestStart()
 {
     int64_t startTime;
-    ALOGD("AudioStreamInternal(): start()");
+    ALOGD("AudioStreamInternal()::requestStart()");
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
+    if (isActive()) {
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    aaudio_stream_state_t originalState = getState();
+
+    setState(AAUDIO_STREAM_STATE_STARTING);
+    aaudio_result_t result = AAudioConvert_androidToAAudioResult(startWithStatus());
 
     startTime = AudioClock::getNanoseconds();
     mClockModel.start(startTime);
-    setState(AAUDIO_STREAM_STATE_STARTING);
-    aaudio_result_t result = AAudioConvert_androidToAAudioResult(startWithStatus());
 
     if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) {
         // Launch the callback loop thread.
@@ -246,6 +253,9 @@
         mCallbackEnabled.store(true);
         result = createThread(periodNanos, aaudio_callback_thread_proc, this);
     }
+    if (result != AAUDIO_OK) {
+        setState(originalState);
+    }
     return result;
 }
 
@@ -279,7 +289,7 @@
 aaudio_result_t AudioStreamInternal::requestPauseInternal()
 {
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
-        ALOGE("AudioStreamInternal(): requestPauseInternal() mServiceStreamHandle invalid = 0x%08X",
+        ALOGE("AudioStreamInternal::requestPauseInternal() mServiceStreamHandle invalid = 0x%08X",
               mServiceStreamHandle);
         return AAUDIO_ERROR_INVALID_STATE;
     }
@@ -301,7 +311,7 @@
 
 aaudio_result_t AudioStreamInternal::requestFlush() {
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
-        ALOGE("AudioStreamInternal(): requestFlush() mServiceStreamHandle invalid = 0x%08X",
+        ALOGE("AudioStreamInternal::requestFlush() mServiceStreamHandle invalid = 0x%08X",
               mServiceStreamHandle);
         return AAUDIO_ERROR_INVALID_STATE;
     }
@@ -312,13 +322,14 @@
 
 // TODO for Play only
 void AudioStreamInternal::onFlushFromServer() {
-    ALOGD("AudioStreamInternal(): onFlushFromServer()");
     int64_t readCounter = mAudioEndpoint.getDataReadCounter();
     int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();
 
     // Bump offset so caller does not see the retrograde motion in getFramesRead().
     int64_t framesFlushed = writeCounter - readCounter;
     mFramesOffsetFromService += framesFlushed;
+    ALOGD("AudioStreamInternal::onFlushFromServer() readN = %lld, writeN = %lld, offset = %lld",
+          (long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);
 
     // Flush written frames by forcing writeCounter to readCounter.
     // This is because we cannot move the read counter in the hardware.
@@ -328,7 +339,7 @@
 aaudio_result_t AudioStreamInternal::requestStopInternal()
 {
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
-        ALOGE("AudioStreamInternal(): requestStopInternal() mServiceStreamHandle invalid = 0x%08X",
+        ALOGE("AudioStreamInternal::requestStopInternal() mServiceStreamHandle invalid = 0x%08X",
               mServiceStreamHandle);
         return AAUDIO_ERROR_INVALID_STATE;
     }
@@ -370,7 +381,7 @@
                            int64_t *timeNanoseconds) {
     // TODO Generate in server and pass to client. Return latest.
     int64_t time = AudioClock::getNanoseconds();
-    *framePosition = mClockModel.convertTimeToPosition(time);
+    *framePosition = mClockModel.convertTimeToPosition(time) + mFramesOffsetFromService;
     // TODO Get a more accurate timestamp from the service. This code just adds a fudge factor.
     *timeNanoseconds = time + (6 * AAUDIO_NANOS_PER_MILLISECOND);
     return AAUDIO_OK;
@@ -383,31 +394,28 @@
     return processCommands();
 }
 
-#if LOG_TIMESTAMPS
-static void AudioStreamInternal_logTimestamp(AAudioServiceMessage &command) {
+void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
     static int64_t oldPosition = 0;
     static int64_t oldTime = 0;
     int64_t framePosition = command.timestamp.position;
     int64_t nanoTime = command.timestamp.timestamp;
-    ALOGD("AudioStreamInternal() timestamp says framePosition = %08lld at nanoTime %lld",
+    ALOGD("AudioStreamInternal: timestamp says framePosition = %08lld at nanoTime %lld",
          (long long) framePosition,
          (long long) nanoTime);
     int64_t nanosDelta = nanoTime - oldTime;
     if (nanosDelta > 0 && oldTime > 0) {
         int64_t framesDelta = framePosition - oldPosition;
         int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
-        ALOGD("AudioStreamInternal() - framesDelta = %08lld", (long long) framesDelta);
-        ALOGD("AudioStreamInternal() - nanosDelta = %08lld", (long long) nanosDelta);
-        ALOGD("AudioStreamInternal() - measured rate = %lld", (long long) rate);
+        ALOGD("AudioStreamInternal: framesDelta = %08lld, nanosDelta = %08lld, rate = %lld",
+              (long long) framesDelta, (long long) nanosDelta, (long long) rate);
     }
     oldPosition = framePosition;
     oldTime = nanoTime;
 }
-#endif
 
 aaudio_result_t AudioStreamInternal::onTimestampFromServer(AAudioServiceMessage *message) {
 #if LOG_TIMESTAMPS
-    AudioStreamInternal_logTimestamp(*message);
+    logTimestamp(*message);
 #endif
     processTimestamp(message->timestamp.position, message->timestamp.timestamp);
     return AAUDIO_OK;
@@ -417,47 +425,48 @@
     aaudio_result_t result = AAUDIO_OK;
     switch (message->event.event) {
         case AAUDIO_SERVICE_EVENT_STARTED:
-            ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STARTED");
+            ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_STARTED");
             if (getState() == AAUDIO_STREAM_STATE_STARTING) {
                 setState(AAUDIO_STREAM_STATE_STARTED);
             }
             break;
         case AAUDIO_SERVICE_EVENT_PAUSED:
-            ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_PAUSED");
+            ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_PAUSED");
             if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
                 setState(AAUDIO_STREAM_STATE_PAUSED);
             }
             break;
         case AAUDIO_SERVICE_EVENT_STOPPED:
-            ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_STOPPED");
+            ALOGD("AudioStreamInternal::onEventFromServergot() AAUDIO_SERVICE_EVENT_STOPPED");
             if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
                 setState(AAUDIO_STREAM_STATE_STOPPED);
             }
             break;
         case AAUDIO_SERVICE_EVENT_FLUSHED:
-            ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_FLUSHED");
+            ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_FLUSHED");
             if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
                 setState(AAUDIO_STREAM_STATE_FLUSHED);
                 onFlushFromServer();
             }
             break;
         case AAUDIO_SERVICE_EVENT_CLOSED:
-            ALOGD("processCommands() got AAUDIO_SERVICE_EVENT_CLOSED");
+            ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_CLOSED");
             setState(AAUDIO_STREAM_STATE_CLOSED);
             break;
         case AAUDIO_SERVICE_EVENT_DISCONNECTED:
             result = AAUDIO_ERROR_DISCONNECTED;
             setState(AAUDIO_STREAM_STATE_DISCONNECTED);
-            ALOGW("WARNING - processCommands() AAUDIO_SERVICE_EVENT_DISCONNECTED");
+            ALOGW("WARNING - AudioStreamInternal::onEventFromServer()"
+                          " AAUDIO_SERVICE_EVENT_DISCONNECTED");
             break;
         case AAUDIO_SERVICE_EVENT_VOLUME:
             mStreamVolume = (float)message->event.dataDouble;
             doSetVolume();
-            ALOGD("processCommands() AAUDIO_SERVICE_EVENT_VOLUME %lf",
+            ALOGD("AudioStreamInternal::onEventFromServer() AAUDIO_SERVICE_EVENT_VOLUME %lf",
                      message->event.dataDouble);
             break;
         default:
-            ALOGW("WARNING - processCommands() Unrecognized event = %d",
+            ALOGW("WARNING - AudioStreamInternal::onEventFromServer() Unrecognized event = %d",
                  (int) message->event.event);
             break;
     }
@@ -499,27 +508,27 @@
 {
     const char * traceName = (mInService) ? "aaWrtS" : "aaWrtC";
     ATRACE_BEGIN(traceName);
-    aaudio_result_t result = AAUDIO_OK;
-    int32_t loopCount = 0;
-    uint8_t* audioData = (uint8_t*)buffer;
-    int64_t currentTimeNanos = AudioClock::getNanoseconds();
-    int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
-    int32_t framesLeft = numFrames;
-
     int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
     if (ATRACE_ENABLED()) {
         const char * traceName = (mInService) ? "aaFullS" : "aaFullC";
         ATRACE_INT(traceName, fullFrames);
     }
 
+    aaudio_result_t result = AAUDIO_OK;
+    int32_t loopCount = 0;
+    uint8_t* audioData = (uint8_t*)buffer;
+    int64_t currentTimeNanos = AudioClock::getNanoseconds();
+    const int64_t entryTimeNanos = currentTimeNanos;
+    const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
+    int32_t framesLeft = numFrames;
+
     // Loop until all the data has been processed or until a timeout occurs.
     while (framesLeft > 0) {
-        // The call to processDataNow() will not block. It will just read as much as it can.
+        // The call to processDataNow() will not block. It will just process as much as it can.
         int64_t wakeTimeNanos = 0;
         aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft,
                                                   currentTimeNanos, &wakeTimeNanos);
         if (framesProcessed < 0) {
-            ALOGE("AudioStreamInternal::processData() loop: framesProcessed = %d", framesProcessed);
             result = framesProcessed;
             break;
         }
@@ -537,12 +546,16 @@
             if (wakeTimeNanos > deadlineNanos) {
                 // If we time out, just return the framesWritten so far.
                 // TODO remove after we fix the deadline bug
-                ALOGE("AudioStreamInternal::processData(): timed out after %lld nanos",
+                ALOGW("AudioStreamInternal::processData(): entered at %lld nanos, currently %lld",
+                      (long long) entryTimeNanos, (long long) currentTimeNanos);
+                ALOGW("AudioStreamInternal::processData(): timed out after %lld nanos",
                       (long long) timeoutNanoseconds);
-                ALOGE("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos",
+                ALOGW("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos",
                       (long long) wakeTimeNanos, (long long) deadlineNanos);
-                ALOGE("AudioStreamInternal::processData(): past deadline by %d micros",
+                ALOGW("AudioStreamInternal::processData(): past deadline by %d micros",
                       (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
+                mClockModel.dump();
+                mAudioEndpoint.dump();
                 break;
             }
 
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 47aedad..c2c6419 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -125,6 +125,8 @@
 
     aaudio_result_t onTimestampFromServer(AAudioServiceMessage *message);
 
+    void logTimestamp(AAudioServiceMessage &message);
+
     // Calculate timeout for an operation involving framesPerOperation.
     int64_t calculateReasonableTimeout(int32_t framesPerOperation);
 
@@ -155,6 +157,9 @@
     uint8_t                 *mCallbackBuffer = nullptr;
     int32_t                  mCallbackFrames = 0;
 
+    // The service uses this for SHARED mode.
+    bool                     mInService = false;  // Is this running in the client or the service?
+
 private:
     /*
      * Asynchronous write with data conversion.
@@ -172,8 +177,6 @@
     EndpointDescriptor       mEndpointDescriptor; // buffer description with resolved addresses
     AAudioServiceInterface  &mServiceInterface;   // abstract interface to the service
 
-    // The service uses this for SHARED mode.
-    bool                     mInService = false;  // Is this running in the client or the service?
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 93693bd..22f8bd1 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -14,10 +14,11 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG (mInService ? "AAudioService" : "AAudio")
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
+#include <algorithm>
 #include <aaudio/AAudio.h>
 
 #include "client/AudioStreamInternalCapture.h"
@@ -155,29 +156,27 @@
 
     int32_t framesProcessed = numFrames - framesLeft;
     mAudioEndpoint.advanceReadIndex(framesProcessed);
-    incrementFramesRead(framesProcessed);
 
     //ALOGD("AudioStreamInternalCapture::readNowWithConversion() returns %d", framesProcessed);
     return framesProcessed;
 }
 
-int64_t AudioStreamInternalCapture::getFramesWritten()
-{
-    int64_t frames =
-            mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
-            + mFramesOffsetFromService;
-    // Prevent retrograde motion.
-    if (frames < mLastFramesWritten) {
-        frames = mLastFramesWritten;
+int64_t AudioStreamInternalCapture::getFramesWritten() {
+    int64_t framesWrittenHardware;
+    if (isActive()) {
+        framesWrittenHardware = mClockModel.convertTimeToPosition(AudioClock::getNanoseconds());
     } else {
-        mLastFramesWritten = frames;
+        framesWrittenHardware = mAudioEndpoint.getDataWriteCounter();
     }
-    //ALOGD("AudioStreamInternalCapture::getFramesWritten() returns %lld", (long long)frames);
-    return frames;
+    // Prevent retrograde motion.
+    mLastFramesWritten = std::max(mLastFramesWritten,
+                                  framesWrittenHardware + mFramesOffsetFromService);
+    //ALOGD("AudioStreamInternalCapture::getFramesWritten() returns %lld",
+    //      (long long)mLastFramesWritten);
+    return mLastFramesWritten;
 }
 
-int64_t AudioStreamInternalCapture::getFramesRead()
-{
+int64_t AudioStreamInternalCapture::getFramesRead() {
     int64_t frames = mAudioEndpoint.getDataWriteCounter()
                                + mFramesOffsetFromService;
     //ALOGD("AudioStreamInternalCapture::getFramesRead() returns %lld", (long long)frames);
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index fc9766f..76ecbf9 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG (mInService ? "AAudioService" : "AAudio")
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -51,15 +51,17 @@
     }
 
     if (mAudioEndpoint.isFreeRunning()) {
-        //ALOGD("AudioStreamInternal::processDataNow() - update read counter");
         // Update data queue based on the timing model.
         int64_t estimatedReadCounter = mClockModel.convertTimeToPosition(currentNanoTime);
+        // ALOGD("AudioStreamInternal::processDataNow() - estimatedReadCounter = %d", (int)estimatedReadCounter);
         mAudioEndpoint.setDataReadCounter(estimatedReadCounter);
     }
-    // TODO else query from endpoint cuz set by actual reader, maybe
 
     // If the read index passed the write index then consider it an underrun.
     if (mAudioEndpoint.getFullFramesAvailable() < 0) {
+        ALOGV("AudioStreamInternal::processDataNow() - XRun! write = %d, read = %d",
+              (int)mAudioEndpoint.getDataWriteCounter(),
+              (int)mAudioEndpoint.getDataReadCounter());
         mXRunCount++;
     }
 
@@ -201,9 +203,6 @@
     int32_t framesWritten = numFrames - framesLeft;
     mAudioEndpoint.advanceWriteIndex(framesWritten);
 
-    if (framesWritten > 0) {
-        incrementFramesWritten(framesWritten);
-    }
     // ALOGD("AudioStreamInternal::writeNowWithConversion() returns %d", framesWritten);
     return framesWritten;
 }
@@ -211,25 +210,29 @@
 
 int64_t AudioStreamInternalPlay::getFramesRead()
 {
-    int64_t framesRead =
-            mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
-            + mFramesOffsetFromService;
+    int64_t framesReadHardware;
+    if (isActive()) {
+        framesReadHardware = mClockModel.convertTimeToPosition(AudioClock::getNanoseconds());
+    } else {
+        framesReadHardware = mAudioEndpoint.getDataReadCounter();
+    }
+    int64_t framesRead = framesReadHardware + mFramesOffsetFromService;
     // Prevent retrograde motion.
     if (framesRead < mLastFramesRead) {
         framesRead = mLastFramesRead;
     } else {
         mLastFramesRead = framesRead;
     }
-    ALOGD("AudioStreamInternal::getFramesRead() returns %lld", (long long)framesRead);
+    //ALOGD("AudioStreamInternalPlay::getFramesRead() returns %lld", (long long)framesRead);
     return framesRead;
 }
 
 int64_t AudioStreamInternalPlay::getFramesWritten()
 {
-    int64_t getFramesWritten = mAudioEndpoint.getDataWriteCounter()
+    int64_t framesWritten = mAudioEndpoint.getDataWriteCounter()
                                + mFramesOffsetFromService;
-    ALOGD("AudioStreamInternal::getFramesWritten() returns %lld", (long long)getFramesWritten);
-    return getFramesWritten;
+    //ALOGD("AudioStreamInternalPlay::getFramesWritten() returns %lld", (long long)framesWritten);
+    return framesWritten;
 }
 
 
diff --git a/media/libaaudio/src/client/IsochronousClockModel.cpp b/media/libaaudio/src/client/IsochronousClockModel.cpp
index 1de33bb..73f4c1d 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.cpp
+++ b/media/libaaudio/src/client/IsochronousClockModel.cpp
@@ -41,6 +41,13 @@
 IsochronousClockModel::~IsochronousClockModel() {
 }
 
+void IsochronousClockModel::setPositionAndTime(int64_t framePosition, int64_t nanoTime) {
+    ALOGV("IsochronousClockModel::setPositionAndTime(%lld, %lld)",
+          (long long) framePosition, (long long) nanoTime);
+    mMarkerFramePosition = framePosition;
+    mMarkerNanoTime = nanoTime;
+}
+
 void IsochronousClockModel::start(int64_t nanoTime) {
     ALOGD("IsochronousClockModel::start(nanos = %lld)\n", (long long) nanoTime);
     mMarkerNanoTime = nanoTime;
@@ -49,8 +56,8 @@
 
 void IsochronousClockModel::stop(int64_t nanoTime) {
     ALOGD("IsochronousClockModel::stop(nanos = %lld)\n", (long long) nanoTime);
-    mMarkerNanoTime = nanoTime;
-    mMarkerFramePosition = convertTimeToPosition(nanoTime); // TODO should we do this?
+    setPositionAndTime(convertTimeToPosition(nanoTime), nanoTime);
+    // TODO should we set position?
     mState = STATE_STOPPED;
 }
 
@@ -79,15 +86,13 @@
     case STATE_STOPPED:
         break;
     case STATE_STARTING:
-        mMarkerFramePosition = framePosition;
-        mMarkerNanoTime = nanoTime;
+        setPositionAndTime(framePosition, nanoTime);
         mState = STATE_SYNCING;
         break;
     case STATE_SYNCING:
         // This will handle a burst of rapid transfer at the beginning.
         if (nanosDelta < expectedNanosDelta) {
-            mMarkerFramePosition = framePosition;
-            mMarkerNanoTime = nanoTime;
+            setPositionAndTime(framePosition, nanoTime);
         } else {
 //            ALOGD("processTimestamp() - advance to STATE_RUNNING");
             mState = STATE_RUNNING;
@@ -98,17 +103,15 @@
             // Earlier than expected timestamp.
             // This data is probably more accurate so use it.
             // or we may be drifting due to a slow HW clock.
-            mMarkerFramePosition = framePosition;
-            mMarkerNanoTime = nanoTime;
 //            ALOGD("processTimestamp() - STATE_RUNNING - %d < %d micros - EARLY",
 //                 (int) (nanosDelta / 1000), (int)(expectedNanosDelta / 1000));
+            setPositionAndTime(framePosition, nanoTime);
         } else if (nanosDelta > (expectedNanosDelta + mMaxLatenessInNanos)) {
             // Later than expected timestamp.
-            mMarkerFramePosition = framePosition;
-            mMarkerNanoTime = nanoTime - mMaxLatenessInNanos;
 //            ALOGD("processTimestamp() - STATE_RUNNING - %d > %d + %d micros - LATE",
 //                 (int) (nanosDelta / 1000), (int)(expectedNanosDelta / 1000),
 //                 (int) (mMaxLatenessInNanos / 1000));
+            setPositionAndTime(framePosition - mFramesPerBurst,  nanoTime - mMaxLatenessInNanos);
         }
         break;
     default:
@@ -131,8 +134,7 @@
     mMaxLatenessInNanos = (nanosLate > MIN_LATENESS_NANOS) ? nanosLate : MIN_LATENESS_NANOS;
 }
 
-int64_t IsochronousClockModel::convertDeltaPositionToTime(
-        int64_t framesDelta) const {
+int64_t IsochronousClockModel::convertDeltaPositionToTime(int64_t framesDelta) const {
     return (AAUDIO_NANOS_PER_SECOND * framesDelta) / mSampleRate;
 }
 
@@ -171,3 +173,12 @@
 //         (long long) framesDelta, mFramesPerBurst);
     return position;
 }
+
+void IsochronousClockModel::dump() const {
+    ALOGD("IsochronousClockModel::mMarkerFramePosition = %lld", (long long) mMarkerFramePosition);
+    ALOGD("IsochronousClockModel::mMarkerNanoTime      = %lld", (long long) mMarkerNanoTime);
+    ALOGD("IsochronousClockModel::mSampleRate          = %6d", mSampleRate);
+    ALOGD("IsochronousClockModel::mFramesPerBurst      = %6d", mFramesPerBurst);
+    ALOGD("IsochronousClockModel::mMaxLatenessInNanos  = %6d", mMaxLatenessInNanos);
+    ALOGD("IsochronousClockModel::mState               = %6d", mState);
+}
diff --git a/media/libaaudio/src/client/IsochronousClockModel.h b/media/libaaudio/src/client/IsochronousClockModel.h
index 0314f55..585f53a 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.h
+++ b/media/libaaudio/src/client/IsochronousClockModel.h
@@ -43,6 +43,8 @@
      */
     void setSampleRate(int32_t sampleRate);
 
+    void setPositionAndTime(int64_t framePosition, int64_t nanoTime);
+
     int32_t getSampleRate() const {
         return mSampleRate;
     }
@@ -86,6 +88,8 @@
      */
     int64_t convertDeltaTimeToPosition(int64_t nanosDelta) const;
 
+    void dump() const;
+
 private:
     enum clock_model_state_t {
         STATE_STOPPED,
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 76f98fa..3f5de77 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -253,8 +253,10 @@
 AAUDIO_API aaudio_result_t  AAudioStream_requestStart(AAudioStream* stream)
 {
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
-    ALOGD("AAudioStream_requestStart(%p)", stream);
-    return audioStream->requestStart();
+    ALOGD("AAudioStream_requestStart(%p) called --------------", stream);
+    aaudio_result_t result = audioStream->requestStart();
+    ALOGD("AAudioStream_requestStart(%p) returned ------------", stream);
+    return result;
 }
 
 AAUDIO_API aaudio_result_t  AAudioStream_requestPause(AAudioStream* stream)
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 5a05c0e..19b08c4 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -36,6 +36,22 @@
     setPeriodNanoseconds(0);
 }
 
+static const char *AudioStream_convertSharingModeToShortText(aaudio_sharing_mode_t sharingMode) {
+    const char *result;
+    switch (sharingMode) {
+        case AAUDIO_SHARING_MODE_EXCLUSIVE:
+            result = "EX";
+            break;
+        case AAUDIO_SHARING_MODE_SHARED:
+            result = "SH";
+            break;
+        default:
+            result = "?!";
+            break;
+    }
+    return result;
+}
+
 aaudio_result_t AudioStream::open(const AudioStreamBuilder& builder)
 {
     // Call here as well because the AAudioService will call this without calling build().
@@ -62,8 +78,9 @@
     mErrorCallbackUserData = builder.getErrorCallbackUserData();
 
     // This is very helpful for debugging in the future. Please leave it in.
-    ALOGI("AudioStream::open() rate = %d, channels = %d, format = %d, sharing = %d, dir = %s",
-          mSampleRate, mSamplesPerFrame, mFormat, mSharingMode,
+    ALOGI("AudioStream::open() rate = %d, channels = %d, format = %d, sharing = %s, dir = %s",
+          mSampleRate, mSamplesPerFrame, mFormat,
+          AudioStream_convertSharingModeToShortText(mSharingMode),
           (getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "OUTPUT" : "INPUT");
     ALOGI("AudioStream::open() device = %d, perfMode = %d, callbackFrames = %d",
           mDeviceId, mPerformanceMode, mFramesPerDataCallback);
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 39c9f9c..7c7fcc5 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -186,13 +186,9 @@
         return AAudioConvert_formatToSizeInBytes(mFormat);
     }
 
-    virtual int64_t getFramesWritten() {
-        return mFramesWritten.get();
-    }
+    virtual int64_t getFramesWritten() = 0;
 
-    virtual int64_t getFramesRead() {
-        return mFramesRead.get();
-    }
+    virtual int64_t getFramesRead() = 0;
 
     AAudioStream_dataCallback getDataCallbackProc() const {
         return mDataCallbackProc;
@@ -232,13 +228,6 @@
 
 protected:
 
-    virtual int64_t incrementFramesWritten(int32_t frames) {
-        return mFramesWritten.increment(frames);
-    }
-
-    virtual int64_t incrementFramesRead(int32_t frames) {
-        return mFramesRead.increment(frames);
-    }
 
     /**
      * This should not be called after the open() call.
@@ -281,8 +270,6 @@
     std::atomic<bool>    mCallbackEnabled;
 
 protected:
-    MonotonicCounter     mFramesWritten;
-    MonotonicCounter     mFramesRead;
 
     void setPeriodNanoseconds(int64_t periodNanoseconds) {
         mPeriodNanoseconds.store(periodNanoseconds, std::memory_order_release);
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index 0ded8e1..d2ef3c7 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -74,6 +74,15 @@
 
     virtual int64_t incrementClientFrameCounter(int32_t frames)  = 0;
 
+
+    virtual int64_t getFramesWritten() override {
+        return mFramesWritten.get();
+    }
+
+    virtual int64_t getFramesRead() override {
+        return mFramesRead.get();
+    }
+
 protected:
 
     class StreamDeviceCallback : public android::AudioSystem::AudioDeviceCallback
@@ -103,6 +112,17 @@
     void onStart() { mCallbackEnabled.store(true); }
     void onStop() { mCallbackEnabled.store(false); }
 
+    int64_t incrementFramesWritten(int32_t frames) {
+        return mFramesWritten.increment(frames);
+    }
+
+    int64_t incrementFramesRead(int32_t frames) {
+        return mFramesRead.increment(frames);
+    }
+
+    MonotonicCounter           mFramesWritten;
+    MonotonicCounter           mFramesRead;
+
     FixedBlockAdapter         *mBlockAdapter = nullptr;
     aaudio_wrapping_frames_t   mPositionWhenStarting = 0;
     int32_t                    mCallbackBufferSize = 0;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 7e39908..77f31e2 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -423,7 +423,7 @@
     default:
         break;
     }
-    return AudioStream::getFramesRead();
+    return AudioStreamLegacy::getFramesRead();
 }
 
 aaudio_result_t AudioStreamTrack::getTimestamp(clockid_t clockId,
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
index 3dc1feb..5c6825d 100644
--- a/services/oboeservice/AAudioEndpointManager.cpp
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -52,15 +52,17 @@
             assert(false); // There are only two possible directions.
             break;
     }
-    ALOGD("AAudioEndpointManager::openEndpoint(), found %p for device = %d, dir = %d",
-          endpoint, deviceId, (int)direction);
 
     // If we can't find an existing one then open a new one.
-    if (endpoint == nullptr) {
+    if (endpoint != nullptr) {
+        ALOGD("AAudioEndpointManager::openEndpoint(), found %p for device = %d, dir = %d",
+              endpoint, deviceId, (int)direction);
+
+    } else {
         if (direction == AAUDIO_DIRECTION_INPUT) {
             AAudioServiceEndpointCapture *capture = new AAudioServiceEndpointCapture(audioService);
             if (capture->open(deviceId) != AAUDIO_OK) {
-                ALOGE("AAudioEndpointManager::openEndpoint(), open failed");
+                ALOGE("AAudioEndpointManager::openEndpoint(), open input failed");
                 delete capture;
             } else {
                 mInputs[deviceId] = capture;
@@ -69,17 +71,20 @@
         } else if (direction == AAUDIO_DIRECTION_OUTPUT) {
             AAudioServiceEndpointPlay *player = new AAudioServiceEndpointPlay(audioService);
             if (player->open(deviceId) != AAUDIO_OK) {
-                ALOGE("AAudioEndpointManager::openEndpoint(), open failed");
+                ALOGE("AAudioEndpointManager::openEndpoint(), open output failed");
                 delete player;
             } else {
                 mOutputs[deviceId] = player;
                 endpoint = player;
             }
         }
-
+        ALOGD("AAudioEndpointManager::openEndpoint(), created %p for device = %d, dir = %d",
+              endpoint, deviceId, (int)direction);
     }
 
     if (endpoint != nullptr) {
+        ALOGD("AAudioEndpointManager::openEndpoint(), sampleRate = %d, framesPerBurst = %d",
+              endpoint->getSampleRate(), endpoint->getFramesPerBurst());
         // Increment the reference count under this lock.
         endpoint->setReferenceCount(endpoint->getReferenceCount() + 1);
     }
@@ -95,9 +100,15 @@
     // Decrement the reference count under this lock.
     int32_t newRefCount = serviceEndpoint->getReferenceCount() - 1;
     serviceEndpoint->setReferenceCount(newRefCount);
+    ALOGD("AAudioEndpointManager::closeEndpoint(%p) newRefCount = %d",
+          serviceEndpoint, newRefCount);
+
+    // If no longer in use then close and delete it.
     if (newRefCount <= 0) {
         aaudio_direction_t direction = serviceEndpoint->getDirection();
-        int32_t deviceId = serviceEndpoint->getDeviceId();
+        // Track endpoints based on requested deviceId because UNSPECIFIED
+        // can change to a specific device after opening.
+        int32_t deviceId = serviceEndpoint->getRequestedDeviceId();
 
         switch (direction) {
             case AAUDIO_DIRECTION_INPUT:
@@ -109,6 +120,8 @@
         }
 
         serviceEndpoint->close();
+        ALOGD("AAudioEndpointManager::closeEndpoint() delete %p for device %d, dir = %d",
+              serviceEndpoint, deviceId, (int)direction);
         delete serviceEndpoint;
     }
 }
diff --git a/services/oboeservice/AAudioEndpointManager.h b/services/oboeservice/AAudioEndpointManager.h
index db1103d..899ea35 100644
--- a/services/oboeservice/AAudioEndpointManager.h
+++ b/services/oboeservice/AAudioEndpointManager.h
@@ -35,7 +35,7 @@
 
     /**
      * Find a service endpoint for the given deviceId and direction.
-     * If an endpoint does not already exist then it will try to create one.
+     * If an endpoint does not already exist then try to create one.
      *
      * @param deviceId
      * @param direction
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index c9b9065..b0e0a74 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -89,7 +89,7 @@
         return result;
     } else {
         aaudio_handle_t handle = mHandleTracker.put(AAUDIO_HANDLE_TYPE_STREAM, serviceStream);
-        ALOGV("AAudioService::openStream(): handle = 0x%08X", handle);
+        ALOGD("AAudioService::openStream(): handle = 0x%08X", handle);
         if (handle < 0) {
             ALOGE("AAudioService::openStream(): handle table full");
             delete serviceStream;
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index d8ae284..cc2cb44 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -46,6 +46,7 @@
 
 // Set up an EXCLUSIVE MMAP stream that will be shared.
 aaudio_result_t AAudioServiceEndpoint::open(int32_t deviceId) {
+    mRequestedDeviceId = deviceId;
     mStreamInternal = getStreamInternal();
 
     AudioStreamBuilder builder;
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index 50bf049..c271dbd 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -48,6 +48,7 @@
     aaudio_result_t stopStream(AAudioServiceStreamShared *sharedStream);
     aaudio_result_t close();
 
+    int32_t getRequestedDeviceId() const { return mRequestedDeviceId; }
     int32_t getDeviceId() const { return mStreamInternal->getDeviceId(); }
 
     aaudio_direction_t getDirection() const { return mStreamInternal->getDirection(); }
@@ -81,6 +82,7 @@
 
     AudioStreamInternal     *mStreamInternal = nullptr;
     int32_t                  mReferenceCount = 0;
+    int32_t                  mRequestedDeviceId = 0;
 };
 
 } /* namespace aaudio */
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.h b/services/oboeservice/AAudioServiceEndpointPlay.h
index b977960..89935ae 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.h
+++ b/services/oboeservice/AAudioServiceEndpointPlay.h
@@ -32,6 +32,9 @@
 
 namespace aaudio {
 
+/**
+ * Contains a mixer and a stream for writing the result of the mix.
+ */
 class AAudioServiceEndpointPlay : public AAudioServiceEndpoint {
 public:
     explicit AAudioServiceEndpointPlay(android::AAudioService &audioService);
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 8f0abc2..ee0e7ed 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -71,12 +71,11 @@
 }
 
 aaudio_result_t AAudioServiceStreamBase::pause() {
-
     sendCurrentTimestamp();
     mThreadEnabled.store(false);
     aaudio_result_t result = mAAudioThread.stop();
     if (result != AAUDIO_OK) {
-        processError();
+        processFatalError();
         return result;
     }
     sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
@@ -90,7 +89,7 @@
     mThreadEnabled.store(false);
     aaudio_result_t result = mAAudioThread.stop();
     if (result != AAUDIO_OK) {
-        processError();
+        processFatalError();
         return result;
     }
     sendServiceEvent(AAUDIO_SERVICE_EVENT_STOPPED);
@@ -126,7 +125,7 @@
     ALOGD("AAudioServiceStreamBase::run() exiting ----------------");
 }
 
-void AAudioServiceStreamBase::processError() {
+void AAudioServiceStreamBase::processFatalError() {
     sendServiceEvent(AAUDIO_SERVICE_EVENT_DISCONNECTED);
 }
 
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index ee52c39..46ceeae 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -111,7 +111,7 @@
 
     void run() override; // to implement Runnable
 
-    void processError();
+    void processFatalError();
 
 protected:
     aaudio_result_t writeUpMessageQueue(AAudioServiceMessage *command);
@@ -122,16 +122,16 @@
 
     virtual aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) = 0;
 
-    aaudio_stream_state_t               mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+    aaudio_stream_state_t   mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
 
     pid_t              mRegisteredClientThread = ILLEGAL_THREAD_ID;
 
     SharedRingBuffer*  mUpMessageQueue;
     std::mutex         mLockUpMessageQueue;
 
-    AAudioThread        mAAudioThread;
+    AAudioThread       mAAudioThread;
     // This is used by one thread to tell another thread to exit. So it must be atomic.
-    std::atomic<bool>   mThreadEnabled;
+    std::atomic<bool>  mThreadEnabled;
 
     aaudio_format_t    mAudioFormat = AAUDIO_FORMAT_UNSPECIFIED;
     int32_t            mFramesPerBurst = 0;
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index 97b9937..2f3ec27 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -138,15 +138,19 @@
         return AAUDIO_ERROR_UNAVAILABLE;
     }
 
+    if (deviceId == AAUDIO_UNSPECIFIED) {
+        ALOGW("AAudioServiceStreamMMAP::open() - openMmapStream() failed to set deviceId");
+    }
+
     // Create MMAP/NOIRQ buffer.
     int32_t minSizeFrames = configurationInput.getBufferCapacity();
-    if (minSizeFrames == 0) { // zero will get rejected
+    if (minSizeFrames <= 0) { // zero will get rejected
         minSizeFrames = AAUDIO_BUFFER_CAPACITY_MIN;
     }
     status = mMmapStream->createMmapBuffer(minSizeFrames, &mMmapBufferinfo);
     if (status != OK) {
-        ALOGE("%s: createMmapBuffer() returned status %d, return AAUDIO_ERROR_UNAVAILABLE",
-              __FILE__, status);
+        ALOGE("AAudioServiceStreamMMAP::open() - createMmapBuffer() returned status %d",
+              status);
         return AAUDIO_ERROR_UNAVAILABLE;
     } else {
         ALOGD("createMmapBuffer status %d shared_address = %p buffer_size %d burst_size %d",
@@ -181,6 +185,9 @@
     ALOGD("AAudioServiceStreamMMAP::open() original burst = %d, minMicros = %d, final burst = %d\n",
           mMmapBufferinfo.burst_size_frames, burstMinMicros, mFramesPerBurst);
 
+    ALOGD("AAudioServiceStreamMMAP::open() actual rate = %d, channels = %d, deviceId = %d\n",
+          mSampleRate, mSamplesPerFrame, deviceId);
+
     // Fill in AAudioStreamConfiguration
     configurationOutput.setSampleRate(mSampleRate);
     configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
@@ -199,7 +206,7 @@
     status_t status = mMmapStream->start(mMmapClient, &mPortHandle);
     if (status != OK) {
         ALOGE("AAudioServiceStreamMMAP::start() mMmapStream->start() returned %d", status);
-        processError();
+        processFatalError();
         result = AAudioConvert_androidToAAudioResult(status);
     } else {
         result = AAudioServiceStreamBase::start();
@@ -234,8 +241,6 @@
 aaudio_result_t AAudioServiceStreamMMAP::flush() {
     if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
     // TODO how do we flush an MMAP/NOIRQ buffer? sync pointers?
-    sendServiceEvent(AAUDIO_SERVICE_EVENT_FLUSHED);
-    mState = AAUDIO_STREAM_STATE_FLUSHED;
     return AAudioServiceStreamBase::flush();;
 }
 
@@ -244,13 +249,13 @@
                                                                 int64_t *timeNanos) {
     struct audio_mmap_position position;
     if (mMmapStream == nullptr) {
-        processError();
+        processFatalError();
         return AAUDIO_ERROR_NULL;
     }
     status_t status = mMmapStream->getMmapPosition(&position);
     if (status != OK) {
         ALOGE("sendCurrentTimestamp(): getMmapPosition() returned %d", status);
-        processError();
+        processFatalError();
         return AAudioConvert_androidToAAudioResult(status);
     } else {
         mFramesRead.update32(position.position_frames);
@@ -295,4 +300,4 @@
     parcelable.mDownDataQueueParcelable.setFramesPerBurst(mFramesPerBurst);
     parcelable.mDownDataQueueParcelable.setCapacityInFrames(mCapacityInFrames);
     return AAUDIO_OK;
-}
\ No newline at end of file
+}
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index 494b18e..f246fc02 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -34,8 +34,10 @@
 using namespace android;
 using namespace aaudio;
 
-#define MIN_BURSTS_PER_BUFFER   2
-#define MAX_BURSTS_PER_BUFFER   32
+#define MIN_BURSTS_PER_BUFFER       2
+#define DEFAULT_BURSTS_PER_BUFFER   16
+// This is an arbitrary range. TODO review.
+#define MAX_FRAMES_PER_BUFFER       (32 * 1024)
 
 AAudioServiceStreamShared::AAudioServiceStreamShared(AAudioService &audioService)
     : mAudioService(audioService)
@@ -46,12 +48,58 @@
     close();
 }
 
+int32_t AAudioServiceStreamShared::calculateBufferCapacity(int32_t requestedCapacityFrames,
+                                                           int32_t framesPerBurst) {
+
+    if (requestedCapacityFrames > MAX_FRAMES_PER_BUFFER) {
+        ALOGE("AAudioServiceStreamShared::open(), requested capacity %d > max %d",
+              requestedCapacityFrames, MAX_FRAMES_PER_BUFFER);
+        return AAUDIO_ERROR_OUT_OF_RANGE;
+    }
+
+    // Determine how many bursts will fit in the buffer.
+    int32_t numBursts;
+    if (requestedCapacityFrames == AAUDIO_UNSPECIFIED) {
+        // Use fewer bursts if default is too many.
+        if ((DEFAULT_BURSTS_PER_BUFFER * framesPerBurst) > MAX_FRAMES_PER_BUFFER) {
+            numBursts = MAX_FRAMES_PER_BUFFER / framesPerBurst;
+        } else {
+            numBursts = DEFAULT_BURSTS_PER_BUFFER;
+        }
+    } else {
+        // round up to nearest burst boundary
+        numBursts = (requestedCapacityFrames + framesPerBurst - 1) / framesPerBurst;
+    }
+
+    // Clip to bare minimum.
+    if (numBursts < MIN_BURSTS_PER_BUFFER) {
+        numBursts = MIN_BURSTS_PER_BUFFER;
+    }
+    // Check for numeric overflow.
+    if (numBursts > 0x8000 || framesPerBurst > 0x8000) {
+        ALOGE("AAudioServiceStreamShared::open(), numeric overflow, capacity = %d * %d",
+              numBursts, framesPerBurst);
+        return AAUDIO_ERROR_OUT_OF_RANGE;
+    }
+    int32_t capacityInFrames = numBursts * framesPerBurst;
+
+    // Final sanity check.
+    if (capacityInFrames > MAX_FRAMES_PER_BUFFER) {
+        ALOGE("AAudioServiceStreamShared::open(), calculated capacity %d > max %d",
+              capacityInFrames, MAX_FRAMES_PER_BUFFER);
+        return AAUDIO_ERROR_OUT_OF_RANGE;
+    }
+    ALOGD("AAudioServiceStreamShared::open(), requested capacity = %d frames, actual = %d",
+          requestedCapacityFrames, capacityInFrames);
+    return capacityInFrames;
+}
+
 aaudio_result_t AAudioServiceStreamShared::open(const aaudio::AAudioStreamRequest &request,
                      aaudio::AAudioStreamConfiguration &configurationOutput)  {
 
     aaudio_result_t result = AAudioServiceStreamBase::open(request, configurationOutput);
     if (result != AAUDIO_OK) {
-        ALOGE("AAudioServiceStreamBase open returned %d", result);
+        ALOGE("AAudioServiceStreamBase open() returned %d", result);
         return result;
     }
 
@@ -72,16 +120,18 @@
         mAudioFormat = AAUDIO_FORMAT_PCM_FLOAT;
     } else if (mAudioFormat != AAUDIO_FORMAT_PCM_FLOAT) {
         ALOGE("AAudioServiceStreamShared::open(), mAudioFormat = %d, need FLOAT", mAudioFormat);
-        return AAUDIO_ERROR_INVALID_FORMAT;
+        result = AAUDIO_ERROR_INVALID_FORMAT;
+        goto error;
     }
 
     mSampleRate = configurationInput.getSampleRate();
     if (mSampleRate == AAUDIO_UNSPECIFIED) {
         mSampleRate = mServiceEndpoint->getSampleRate();
     } else if (mSampleRate != mServiceEndpoint->getSampleRate()) {
-        ALOGE("AAudioServiceStreamShared::open(), mAudioFormat = %d, need %d",
+        ALOGE("AAudioServiceStreamShared::open(), mSampleRate = %d, need %d",
               mSampleRate, mServiceEndpoint->getSampleRate());
-        return AAUDIO_ERROR_INVALID_RATE;
+        result = AAUDIO_ERROR_INVALID_RATE;
+        goto error;
     }
 
     mSamplesPerFrame = configurationInput.getSamplesPerFrame();
@@ -90,37 +140,51 @@
     } else if (mSamplesPerFrame != mServiceEndpoint->getSamplesPerFrame()) {
         ALOGE("AAudioServiceStreamShared::open(), mSamplesPerFrame = %d, need %d",
               mSamplesPerFrame, mServiceEndpoint->getSamplesPerFrame());
-        return AAUDIO_ERROR_OUT_OF_RANGE;
+        result = AAUDIO_ERROR_OUT_OF_RANGE;
+        goto error;
     }
 
-    // Determine this stream's shared memory buffer capacity.
     mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
-    int32_t minCapacityFrames = configurationInput.getBufferCapacity();
-    int32_t numBursts = MAX_BURSTS_PER_BUFFER;
-    if (minCapacityFrames != AAUDIO_UNSPECIFIED) {
-        numBursts = (minCapacityFrames + mFramesPerBurst - 1) / mFramesPerBurst;
-        if (numBursts < MIN_BURSTS_PER_BUFFER) {
-            numBursts = MIN_BURSTS_PER_BUFFER;
-        } else if (numBursts > MAX_BURSTS_PER_BUFFER) {
-            numBursts = MAX_BURSTS_PER_BUFFER;
-        }
+    ALOGD("AAudioServiceStreamShared::open(), mSampleRate = %d, mFramesPerBurst = %d",
+          mSampleRate, mFramesPerBurst);
+
+    mCapacityInFrames = calculateBufferCapacity(configurationInput.getBufferCapacity(),
+                                     mFramesPerBurst);
+    if (mCapacityInFrames < 0) {
+        result = mCapacityInFrames; // negative error code
+        mCapacityInFrames = 0;
+        goto error;
     }
-    mCapacityInFrames = numBursts * mFramesPerBurst;
-    ALOGD("AAudioServiceStreamShared::open(), mCapacityInFrames = %d", mCapacityInFrames);
 
     // Create audio data shared memory buffer for client.
     mAudioDataQueue = new SharedRingBuffer();
-    mAudioDataQueue->allocate(calculateBytesPerFrame(), mCapacityInFrames);
+    result = mAudioDataQueue->allocate(calculateBytesPerFrame(), mCapacityInFrames);
+    if (result != AAUDIO_OK) {
+        ALOGE("AAudioServiceStreamShared::open(), could not allocate FIFO with %d frames",
+              mCapacityInFrames);
+        result = AAUDIO_ERROR_NO_MEMORY;
+        goto error;
+    }
+
+    ALOGD("AAudioServiceStreamShared::open() actual rate = %d, channels = %d, deviceId = %d",
+          mSampleRate, mSamplesPerFrame, mServiceEndpoint->getDeviceId());
 
     // Fill in configuration for client.
     configurationOutput.setSampleRate(mSampleRate);
     configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
     configurationOutput.setAudioFormat(mAudioFormat);
-    configurationOutput.setDeviceId(deviceId);
+    configurationOutput.setDeviceId(mServiceEndpoint->getDeviceId());
 
-    mServiceEndpoint->registerStream(this);
+    result = mServiceEndpoint->registerStream(this);
+    if (result != AAUDIO_OK) {
+        goto error;
+    }
 
     return AAUDIO_OK;
+
+error:
+    close();
+    return result;
 }
 
 /**
@@ -137,11 +201,11 @@
     aaudio_result_t result = endpoint->startStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::start() mServiceEndpoint returned %d", result);
-        processError();
+        processFatalError();
     } else {
         result = AAudioServiceStreamBase::start();
     }
-    return AAUDIO_OK;
+    return result;
 }
 
 /**
@@ -154,11 +218,10 @@
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    // Add this stream to the mixer.
     aaudio_result_t result = endpoint->stopStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::pause() mServiceEndpoint returned %d", result);
-        processError();
+        processFatalError();
     }
     return AAudioServiceStreamBase::pause();
 }
@@ -168,11 +231,10 @@
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    // Add this stream to the mixer.
     aaudio_result_t result = endpoint->stopStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::stop() mServiceEndpoint returned %d", result);
-        processError();
+        processFatalError();
     }
     return AAudioServiceStreamBase::stop();
 }
@@ -183,9 +245,17 @@
  * An AAUDIO_SERVICE_EVENT_FLUSHED will be sent to the client when complete.
  */
 aaudio_result_t AAudioServiceStreamShared::flush()  {
-    // TODO make sure we are paused
-    // TODO actually flush the data
-    return AAudioServiceStreamBase::flush() ;
+    AAudioServiceEndpoint *endpoint = mServiceEndpoint;
+    if (endpoint == nullptr) {
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    if (mState != AAUDIO_STREAM_STATE_PAUSED) {
+        ALOGE("AAudioServiceStreamShared::flush() stream not paused, state = %s",
+            AAudio_convertStreamStateToText(mState));
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    // Data will get flushed when the client receives the FLUSHED event.
+    return AAudioServiceStreamBase::flush();
 }
 
 aaudio_result_t AAudioServiceStreamShared::close()  {
diff --git a/services/oboeservice/AAudioServiceStreamShared.h b/services/oboeservice/AAudioServiceStreamShared.h
index dfdbbb3..35af434 100644
--- a/services/oboeservice/AAudioServiceStreamShared.h
+++ b/services/oboeservice/AAudioServiceStreamShared.h
@@ -97,6 +97,14 @@
 
     aaudio_result_t getFreeRunningPosition(int64_t *positionFrames, int64_t *timeNanos) override;
 
+    /**
+     * @param requestedCapacityFrames
+     * @param framesPerBurst
+     * @return capacity or negative error
+     */
+    static int32_t calculateBufferCapacity(int32_t requestedCapacityFrames,
+                                            int32_t framesPerBurst);
+
 private:
     android::AAudioService  &mAudioService;
     AAudioServiceEndpoint   *mServiceEndpoint = nullptr;