diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 214f888..06f66d3 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -32,19 +32,12 @@
 #define RIDICULOUSLY_LARGE_FRAME_SIZE        4096
 
 AudioEndpoint::AudioEndpoint()
-    : mUpCommandQueue(nullptr)
-    , mDataQueue(nullptr)
-    , mFreeRunning(false)
+    : mFreeRunning(false)
     , mDataReadCounter(0)
     , mDataWriteCounter(0)
 {
 }
 
-AudioEndpoint::~AudioEndpoint() {
-    delete mDataQueue;
-    delete mUpCommandQueue;
-}
-
 // TODO Consider moving to a method in RingBufferDescriptor
 static aaudio_result_t AudioEndpoint_validateQueueDescriptor(const char *type,
                                                   const RingBufferDescriptor *descriptor) {
@@ -144,7 +137,7 @@
         return AAUDIO_ERROR_INTERNAL;
     }
 
-    mUpCommandQueue = new FifoBuffer(
+    mUpCommandQueue = std::make_unique<FifoBuffer>(
             descriptor->bytesPerFrame,
             descriptor->capacityInFrames,
             descriptor->readCounterAddress,
@@ -173,7 +166,7 @@
                                   ? &mDataWriteCounter
                                   : descriptor->writeCounterAddress;
 
-    mDataQueue = new FifoBuffer(
+    mDataQueue = std::make_unique<FifoBuffer>(
             descriptor->bytesPerFrame,
             descriptor->capacityInFrames,
             readCounterAddress,
@@ -194,18 +187,15 @@
     return mDataQueue->getEmptyRoomAvailable(wrappingBuffer);
 }
 
-int32_t AudioEndpoint::getEmptyFramesAvailable()
-{
+int32_t AudioEndpoint::getEmptyFramesAvailable() {
     return mDataQueue->getEmptyFramesAvailable();
 }
 
-int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer)
-{
+int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer) {
     return mDataQueue->getFullDataAvailable(wrappingBuffer);
 }
 
-int32_t AudioEndpoint::getFullFramesAvailable()
-{
+int32_t AudioEndpoint::getFullFramesAvailable() {
     return mDataQueue->getFullFramesAvailable();
 }
 
@@ -217,29 +207,24 @@
     mDataQueue->advanceReadIndex(deltaFrames);
 }
 
-void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead)
-{
+void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead) {
     mDataQueue->setReadCounter(framesRead);
 }
 
-fifo_counter_t AudioEndpoint::getDataReadCounter()
-{
+fifo_counter_t AudioEndpoint::getDataReadCounter() const {
     return mDataQueue->getReadCounter();
 }
 
-void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead)
-{
+void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead) {
     mDataQueue->setWriteCounter(framesRead);
 }
 
-fifo_counter_t AudioEndpoint::getDataWriteCounter()
-{
+fifo_counter_t AudioEndpoint::getDataWriteCounter() const {
     return mDataQueue->getWriteCounter();
 }
 
 int32_t AudioEndpoint::setBufferSizeInFrames(int32_t requestedFrames,
-                                            int32_t *actualFrames)
-{
+                                            int32_t *actualFrames) {
     if (requestedFrames < ENDPOINT_DATA_QUEUE_SIZE_MIN) {
         requestedFrames = ENDPOINT_DATA_QUEUE_SIZE_MIN;
     }
@@ -248,19 +233,17 @@
     return AAUDIO_OK;
 }
 
-int32_t AudioEndpoint::getBufferSizeInFrames() const
-{
+int32_t AudioEndpoint::getBufferSizeInFrames() const {
     return mDataQueue->getThreshold();
 }
 
-int32_t AudioEndpoint::getBufferCapacityInFrames() const
-{
+int32_t AudioEndpoint::getBufferCapacityInFrames() const {
     return (int32_t)mDataQueue->getBufferCapacityInFrames();
 }
 
 void AudioEndpoint::dump() const {
-    ALOGD("data readCounter  = %lld", (long long) mDataQueue->getReadCounter());
-    ALOGD("data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());
+    ALOGD("data readCounter  = %lld", (long long) getDataReadCounter());
+    ALOGD("data writeCounter = %lld", (long long) getDataWriteCounter());
 }
 
 void AudioEndpoint::eraseDataMemory() {
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index f5b67e8..484d917 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -35,7 +35,6 @@
 
 public:
     AudioEndpoint();
-    virtual ~AudioEndpoint();
 
     /**
      * Configure based on the EndPointDescriptor_t.
@@ -67,11 +66,11 @@
      */
     void setDataReadCounter(android::fifo_counter_t framesRead);
 
-    android::fifo_counter_t getDataReadCounter();
+    android::fifo_counter_t getDataReadCounter() const;
 
     void setDataWriteCounter(android::fifo_counter_t framesWritten);
 
-    android::fifo_counter_t getDataWriteCounter();
+    android::fifo_counter_t getDataWriteCounter() const;
 
     /**
      * The result is not valid until after configure() is called.
@@ -94,8 +93,8 @@
     void dump() const;
 
 private:
-    android::FifoBuffer    *mUpCommandQueue;
-    android::FifoBuffer    *mDataQueue;
+    std::unique_ptr<android::FifoBuffer> mUpCommandQueue;
+    std::unique_ptr<android::FifoBuffer> mDataQueue;
     bool                    mFreeRunning;
     android::fifo_counter_t mDataReadCounter; // only used if free-running
     android::fifo_counter_t mDataWriteCounter; // only used if free-running
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 076c92d..f89cde7 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -58,7 +58,6 @@
 AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface  &serviceInterface, bool inService)
         : AudioStream()
         , mClockModel()
-        , mAudioEndpoint()
         , mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
         , mInService(inService)
         , mServiceInterface(serviceInterface)
@@ -74,7 +73,6 @@
 aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
 
     aaudio_result_t result = AAUDIO_OK;
-    int32_t capacity;
     int32_t framesPerBurst;
     int32_t framesPerHardwareBurst;
     AAudioStreamRequest request;
@@ -173,7 +171,8 @@
     }
 
     // Configure endpoint based on descriptor.
-    result = mAudioEndpoint.configure(&mEndpointDescriptor, getDirection());
+    mAudioEndpoint = std::make_unique<AudioEndpoint>();
+    result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection());
     if (result != AAUDIO_OK) {
         goto error;
     }
@@ -201,9 +200,10 @@
     }
     mFramesPerBurst = framesPerBurst; // only save good value
 
-    capacity = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
-    if (capacity < mFramesPerBurst || capacity > MAX_BUFFER_CAPACITY_IN_FRAMES) {
-        ALOGE("%s - bufferCapacity out of range = %d", __func__, capacity);
+    mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
+    if (mBufferCapacityInFrames < mFramesPerBurst
+            || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
+        ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
     }
@@ -239,7 +239,7 @@
     // You can use this offset to reduce glitching.
     // You can also use this offset to force glitching. By iterating over multiple
     // values you can reveal the distribution of the hardware timing jitter.
-    if (mAudioEndpoint.isFreeRunning()) { // MMAP?
+    if (mAudioEndpoint->isFreeRunning()) { // MMAP?
         int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT)
                 ? AAudioProperty_getOutputMMapOffsetMicros()
                 : AAudioProperty_getInputMMapOffsetMicros();
@@ -251,7 +251,7 @@
         mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;
     }
 
-    setBufferSize(capacity / 2); // Default buffer size to match Q
+    setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
 
     setState(AAUDIO_STREAM_STATE_OPEN);
 
@@ -280,6 +280,11 @@
 
         mServiceInterface.closeStream(serviceStreamHandle);
         mCallbackBuffer.reset();
+
+        // Update local frame counters so we can query them after releasing the endpoint.
+        getFramesRead();
+        getFramesWritten();
+        mAudioEndpoint.reset();
         result = mEndPointParcelable.close();
         aaudio_result_t result2 = AudioStream::release_l();
         return (result != AAUDIO_OK) ? result : result2;
@@ -538,7 +543,7 @@
         case AAUDIO_SERVICE_EVENT_DISCONNECTED:
             // Prevent hardware from looping on old data and making buzzing sounds.
             if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
-                mAudioEndpoint.eraseDataMemory();
+                mAudioEndpoint->eraseDataMemory();
             }
             result = AAUDIO_ERROR_DISCONNECTED;
             setState(AAUDIO_STREAM_STATE_DISCONNECTED);
@@ -564,7 +569,10 @@
 
     while (result == AAUDIO_OK) {
         AAudioServiceMessage message;
-        if (mAudioEndpoint.readUpCommand(&message) != 1) {
+        if (!mAudioEndpoint) {
+            break;
+        }
+        if (mAudioEndpoint->readUpCommand(&message) != 1) {
             break; // no command this time, no problem
         }
         switch (message.what) {
@@ -592,7 +600,10 @@
 
     while (result == AAUDIO_OK) {
         AAudioServiceMessage message;
-        if (mAudioEndpoint.readUpCommand(&message) != 1) {
+        if (!mAudioEndpoint) {
+            break;
+        }
+        if (mAudioEndpoint->readUpCommand(&message) != 1) {
             break; // no command this time, no problem
         }
         switch (message.what) {
@@ -625,7 +636,7 @@
     const char * fifoName = "aaRdy";
     ATRACE_BEGIN(traceName);
     if (ATRACE_ENABLED()) {
-        int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
+        int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
         ATRACE_INT(fifoName, fullFrames);
     }
 
@@ -654,7 +665,7 @@
         if (timeoutNanoseconds == 0) {
             break; // don't block
         } else if (wakeTimeNanos != 0) {
-            if (!mAudioEndpoint.isFreeRunning()) {
+            if (!mAudioEndpoint->isFreeRunning()) {
                 // If there is software on the other end of the FIFO then it may get delayed.
                 // So wake up just a little after we expect it to be ready.
                 wakeTimeNanos += mWakeupDelayNanos;
@@ -679,12 +690,12 @@
                 ALOGW("processData(): past deadline by %d micros",
                       (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
                 mClockModel.dump();
-                mAudioEndpoint.dump();
+                mAudioEndpoint->dump();
                 break;
             }
 
             if (ATRACE_ENABLED()) {
-                int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
+                int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
                 ATRACE_INT(fifoName, fullFrames);
                 int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos;
                 ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos);
@@ -696,7 +707,7 @@
     }
 
     if (ATRACE_ENABLED()) {
-        int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
+        int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
         ATRACE_INT(fifoName, fullFrames);
     }
 
@@ -730,11 +741,15 @@
         adjustedFrames = std::min(maximumSize, adjustedFrames);
     }
 
-    // Clip against the actual size from the endpoint.
-    int32_t actualFrames = 0;
-    mAudioEndpoint.setBufferSizeInFrames(maximumSize, &actualFrames);
-    // actualFrames should be <= actual maximum size of endpoint
-    adjustedFrames = std::min(actualFrames, adjustedFrames);
+    if (mAudioEndpoint) {
+        // Clip against the actual size from the endpoint.
+        int32_t actualFrames = 0;
+        // Set to maximum size so we can write extra data when ready in order to reduce glitches.
+        // The amount we keep in the buffer is controlled by mBufferSizeInFrames.
+        mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames);
+        // actualFrames should be <= actual maximum size of endpoint
+        adjustedFrames = std::min(actualFrames, adjustedFrames);
+    }
 
     mBufferSizeInFrames = adjustedFrames;
     ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
@@ -746,7 +761,7 @@
 }
 
 int32_t AudioStreamInternal::getBufferCapacity() const {
-    return mAudioEndpoint.getBufferCapacityInFrames();
+    return mBufferCapacityInFrames;
 }
 
 int32_t AudioStreamInternal::getFramesPerBurst() const {
@@ -759,5 +774,5 @@
 }
 
 bool AudioStreamInternal::isClockModelInControl() const {
-    return isActive() && mAudioEndpoint.isFreeRunning() && mClockModel.isRunning();
+    return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
 }
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 42f2889..61591b3 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -155,7 +155,8 @@
 
     IsochronousClockModel    mClockModel;      // timing model for chasing the HAL
 
-    AudioEndpoint            mAudioEndpoint;   // source for reads or sink for writes
+    std::unique_ptr<AudioEndpoint> mAudioEndpoint;   // source for reads or sink for writes
+
     aaudio_handle_t          mServiceStreamHandle; // opaque handle returned from service
 
     int32_t                  mFramesPerBurst = MIN_FRAMES_PER_BURST; // frames per HAL transfer
@@ -178,6 +179,9 @@
 
     float                    mStreamVolume = 1.0f;
 
+    int64_t                  mLastFramesWritten = 0;
+    int64_t                  mLastFramesRead = 0;
+
 private:
     /*
      * Asynchronous write with data conversion.
@@ -207,6 +211,8 @@
     int32_t                  mDeviceChannelCount = 0;
 
     int32_t                  mBufferSizeInFrames = 0; // local threshold to control latency
+    int32_t                  mBufferCapacityInFrames = 0;
+
 
 };
 
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 32cf368..9fa2e40 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -42,8 +42,8 @@
 AudioStreamInternalCapture::~AudioStreamInternalCapture() {}
 
 void AudioStreamInternalCapture::advanceClientToMatchServerPosition() {
-    int64_t readCounter = mAudioEndpoint.getDataReadCounter();
-    int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();
+    int64_t readCounter = mAudioEndpoint->getDataReadCounter();
+    int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();
 
     // Bump offset so caller does not see the retrograde motion in getFramesRead().
     int64_t offset = readCounter - writeCounter;
@@ -53,7 +53,7 @@
 
     // Force readCounter to match writeCounter.
     // This is because we cannot change the write counter in the hardware.
-    mAudioEndpoint.setDataReadCounter(writeCounter);
+    mAudioEndpoint->setDataReadCounter(writeCounter);
 }
 
 // Write the data, block if needed and timeoutMillis > 0
@@ -86,7 +86,7 @@
     }
     // If we have gotten this far then we have at least one timestamp from server.
 
-    if (mAudioEndpoint.isFreeRunning()) {
+    if (mAudioEndpoint->isFreeRunning()) {
         //ALOGD("AudioStreamInternalCapture::processDataNow() - update remote counter");
         // Update data queue based on the timing model.
         // Jitter in the DSP can cause late writes to the FIFO.
@@ -95,7 +95,7 @@
         // that the DSP could have written the data.
         int64_t estimatedRemoteCounter = mClockModel.convertLatestTimeToPosition(currentNanoTime);
         // TODO refactor, maybe use setRemoteCounter()
-        mAudioEndpoint.setDataWriteCounter(estimatedRemoteCounter);
+        mAudioEndpoint->setDataWriteCounter(estimatedRemoteCounter);
     }
 
     // This code assumes that we have already received valid timestamps.
@@ -108,8 +108,8 @@
 
     // If the capture buffer is full beyond capacity then consider it an overrun.
     // For shared streams, the xRunCount is passed up from the service.
-    if (mAudioEndpoint.isFreeRunning()
-        && mAudioEndpoint.getFullFramesAvailable() > mAudioEndpoint.getBufferCapacityInFrames()) {
+    if (mAudioEndpoint->isFreeRunning()
+        && mAudioEndpoint->getFullFramesAvailable() > mAudioEndpoint->getBufferCapacityInFrames()) {
         mXRunCount++;
         if (ATRACE_ENABLED()) {
             ATRACE_INT("aaOverRuns", mXRunCount);
@@ -143,7 +143,7 @@
                 // Calculate frame position based off of the readCounter because
                 // the writeCounter might have just advanced in the background,
                 // causing us to sleep until a later burst.
-                int64_t nextPosition = mAudioEndpoint.getDataReadCounter() + mFramesPerBurst;
+                int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + mFramesPerBurst;
                 wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
             }
                 break;
@@ -166,7 +166,7 @@
     uint8_t *destination = (uint8_t *) buffer;
     int32_t framesLeft = numFrames;
 
-    mAudioEndpoint.getFullFramesAvailable(&wrappingBuffer);
+    mAudioEndpoint->getFullFramesAvailable(&wrappingBuffer);
 
     // Read data in one or two parts.
     for (int partIndex = 0; framesLeft > 0 && partIndex < WrappingBuffer::SIZE; partIndex++) {
@@ -208,26 +208,29 @@
     }
 
     int32_t framesProcessed = numFrames - framesLeft;
-    mAudioEndpoint.advanceReadIndex(framesProcessed);
+    mAudioEndpoint->advanceReadIndex(framesProcessed);
 
     //ALOGD("readNowWithConversion() returns %d", framesProcessed);
     return framesProcessed;
 }
 
 int64_t AudioStreamInternalCapture::getFramesWritten() {
-    const int64_t framesWrittenHardware = isClockModelInControl()
-            ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
-            : mAudioEndpoint.getDataWriteCounter();
-    // Add service offset and prevent retrograde motion.
-    mLastFramesWritten = std::max(mLastFramesWritten,
-                                  framesWrittenHardware + mFramesOffsetFromService);
+    if (mAudioEndpoint) {
+        const int64_t framesWrittenHardware = isClockModelInControl()
+                ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
+                : mAudioEndpoint->getDataWriteCounter();
+        // Add service offset and prevent retrograde motion.
+        mLastFramesWritten = std::max(mLastFramesWritten,
+                                      framesWrittenHardware + mFramesOffsetFromService);
+    }
     return mLastFramesWritten;
 }
 
 int64_t AudioStreamInternalCapture::getFramesRead() {
-    int64_t frames = mAudioEndpoint.getDataReadCounter() + mFramesOffsetFromService;
-    //ALOGD("getFramesRead() returns %lld", (long long)frames);
-    return frames;
+    if (mAudioEndpoint) {
+        mLastFramesRead = mAudioEndpoint->getDataReadCounter() + mFramesOffsetFromService;
+    }
+    return mLastFramesRead;
 }
 
 // Read data from the stream and pass it to the callback for processing.
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.h b/media/libaaudio/src/client/AudioStreamInternalCapture.h
index 294dbaf..6436a53 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.h
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.h
@@ -68,8 +68,6 @@
      * @return frames written or negative error
      */
     aaudio_result_t readNowWithConversion(void *buffer, int32_t numFrames);
-
-    int64_t       mLastFramesWritten = 0; // used to prevent retrograde motion
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index b50a512..1303daf 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -87,8 +87,8 @@
 }
 
 void AudioStreamInternalPlay::advanceClientToMatchServerPosition() {
-    int64_t readCounter = mAudioEndpoint.getDataReadCounter();
-    int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();
+    int64_t readCounter = mAudioEndpoint->getDataReadCounter();
+    int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();
 
     // Bump offset so caller does not see the retrograde motion in getFramesRead().
     int64_t offset = writeCounter - readCounter;
@@ -98,7 +98,7 @@
 
     // Force writeCounter to match readCounter.
     // This is because we cannot change the read counter in the hardware.
-    mAudioEndpoint.setDataWriteCounter(readCounter);
+    mAudioEndpoint->setDataWriteCounter(readCounter);
 }
 
 void AudioStreamInternalPlay::onFlushFromServer() {
@@ -135,11 +135,11 @@
     // If we have gotten this far then we have at least one timestamp from server.
 
     // If a DMA channel or DSP is reading the other end then we have to update the readCounter.
-    if (mAudioEndpoint.isFreeRunning()) {
+    if (mAudioEndpoint->isFreeRunning()) {
         // Update data queue based on the timing model.
         int64_t estimatedReadCounter = mClockModel.convertTimeToPosition(currentNanoTime);
         // ALOGD("AudioStreamInternal::processDataNow() - estimatedReadCounter = %d", (int)estimatedReadCounter);
-        mAudioEndpoint.setDataReadCounter(estimatedReadCounter);
+        mAudioEndpoint->setDataReadCounter(estimatedReadCounter);
     }
 
     if (mNeedCatchUp.isRequested()) {
@@ -151,7 +151,7 @@
 
     // If the read index passed the write index then consider it an underrun.
     // For shared streams, the xRunCount is passed up from the service.
-    if (mAudioEndpoint.isFreeRunning() && mAudioEndpoint.getFullFramesAvailable() < 0) {
+    if (mAudioEndpoint->isFreeRunning() && mAudioEndpoint->getFullFramesAvailable() < 0) {
         mXRunCount++;
         if (ATRACE_ENABLED()) {
             ATRACE_INT("aaUnderRuns", mXRunCount);
@@ -170,7 +170,7 @@
     // Sleep if there is too much data in the buffer.
     // Calculate an ideal time to wake up.
     if (wakeTimePtr != nullptr
-            && (mAudioEndpoint.getFullFramesAvailable() >= getBufferSize())) {
+            && (mAudioEndpoint->getFullFramesAvailable() >= getBufferSize())) {
         // By default wake up a few milliseconds from now.  // TODO review
         int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
         aaudio_stream_state_t state = getState();
@@ -188,7 +188,7 @@
             {
                 // Sleep until the readCounter catches up and we only have
                 // the getBufferSize() frames of data sitting in the buffer.
-                int64_t nextReadPosition = mAudioEndpoint.getDataWriteCounter() - getBufferSize();
+                int64_t nextReadPosition = mAudioEndpoint->getDataWriteCounter() - getBufferSize();
                 wakeTime = mClockModel.convertPositionToTime(nextReadPosition);
             }
                 break;
@@ -210,7 +210,7 @@
     uint8_t *byteBuffer = (uint8_t *) buffer;
     int32_t framesLeft = numFrames;
 
-    mAudioEndpoint.getEmptyFramesAvailable(&wrappingBuffer);
+    mAudioEndpoint->getEmptyFramesAvailable(&wrappingBuffer);
 
     // Write data in one or two parts.
     int partIndex = 0;
@@ -236,24 +236,28 @@
         partIndex++;
     }
     int32_t framesWritten = numFrames - framesLeft;
-    mAudioEndpoint.advanceWriteIndex(framesWritten);
+    mAudioEndpoint->advanceWriteIndex(framesWritten);
 
     return framesWritten;
 }
 
 int64_t AudioStreamInternalPlay::getFramesRead() {
-    const int64_t framesReadHardware = isClockModelInControl()
-            ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
-            : mAudioEndpoint.getDataReadCounter();
-    // Add service offset and prevent retrograde motion.
-    mLastFramesRead = std::max(mLastFramesRead, framesReadHardware + mFramesOffsetFromService);
+    if (mAudioEndpoint) {
+        const int64_t framesReadHardware = isClockModelInControl()
+                ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
+                : mAudioEndpoint->getDataReadCounter();
+        // Add service offset and prevent retrograde motion.
+        mLastFramesRead = std::max(mLastFramesRead, framesReadHardware + mFramesOffsetFromService);
+    }
     return mLastFramesRead;
 }
 
 int64_t AudioStreamInternalPlay::getFramesWritten() {
-    const int64_t framesWritten = mAudioEndpoint.getDataWriteCounter()
-                               + mFramesOffsetFromService;
-    return framesWritten;
+    if (mAudioEndpoint) {
+        mLastFramesWritten = mAudioEndpoint->getDataWriteCounter()
+                             + mFramesOffsetFromService;
+    }
+    return mLastFramesWritten;
 }
 
 
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index cab2942..2e93157 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h
@@ -92,8 +92,6 @@
     aaudio_result_t writeNowWithConversion(const void *buffer,
                                            int32_t numFrames);
 
-    int64_t                  mLastFramesRead = 0; // used to prevent retrograde motion
-
     AAudioFlowGraph          mFlowGraph;
 
 };