diff --git a/media/libaaudio/src/binding/AAudioBinderAdapter.cpp b/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
index 6e3a1c8..42d81ca 100644
--- a/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
+++ b/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
@@ -124,4 +124,16 @@
     return result;
 }
 
+aaudio_result_t AAudioBinderAdapter::exitStandby(aaudio_handle_t streamHandle,
+                                                 AudioEndpointParcelable &endpointOut) {
+    aaudio_result_t result;
+    Endpoint endpoint;
+    Status status = mDelegate->exitStandby(streamHandle, &endpoint, &result);
+    if (!status.isOk()) {
+        result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
+    }
+    endpointOut = std::move(endpoint);
+    return result;
+}
+
 }  // namespace aaudio
diff --git a/media/libaaudio/src/binding/AAudioBinderAdapter.h b/media/libaaudio/src/binding/AAudioBinderAdapter.h
index 5e9ab57..d170783 100644
--- a/media/libaaudio/src/binding/AAudioBinderAdapter.h
+++ b/media/libaaudio/src/binding/AAudioBinderAdapter.h
@@ -57,6 +57,9 @@
     aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
                                           pid_t clientThreadId) override;
 
+    aaudio_result_t exitStandby(aaudio_handle_t streamHandle,
+                                AudioEndpointParcelable &parcelable) override;
+
 private:
     IAAudioService* const mDelegate;
 };
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.cpp b/media/libaaudio/src/binding/AAudioBinderClient.cpp
index 135bac3..8e5facc 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.cpp
+++ b/media/libaaudio/src/binding/AAudioBinderClient.cpp
@@ -201,3 +201,11 @@
 
     return service->unregisterAudioThread(streamHandle, clientThreadId);
 }
+
+aaudio_result_t AAudioBinderClient::exitStandby(aaudio_handle_t streamHandle,
+                                                AudioEndpointParcelable &endpointOut) {
+    std::shared_ptr<AAudioServiceInterface> service = getAAudioService();
+    if (service.get() == nullptr) return AAUDIO_ERROR_NO_SERVICE;
+
+    return service->exitStandby(streamHandle, endpointOut);
+}
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.h b/media/libaaudio/src/binding/AAudioBinderClient.h
index 557ced5..0968f4c 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.h
+++ b/media/libaaudio/src/binding/AAudioBinderClient.h
@@ -108,6 +108,9 @@
         return AAUDIO_ERROR_UNAVAILABLE;
     }
 
+    aaudio_result_t exitStandby(aaudio_handle_t streamHandle,
+                                AudioEndpointParcelable &endpointOut) override;
+
     void onStreamChange(aaudio_handle_t /*handle*/, int32_t /*opcode*/, int32_t /*value*/) {
         // TODO This is just a stub so we can have a client Binder to pass to the service.
         // TODO Implemented in a later CL.
diff --git a/media/libaaudio/src/binding/AAudioServiceInterface.h b/media/libaaudio/src/binding/AAudioServiceInterface.h
index bf94774..e901767 100644
--- a/media/libaaudio/src/binding/AAudioServiceInterface.h
+++ b/media/libaaudio/src/binding/AAudioServiceInterface.h
@@ -95,6 +95,16 @@
 
     virtual aaudio_result_t stopClient(aaudio_handle_t streamHandle,
                                        audio_port_handle_t clientHandle) = 0;
+
+    /**
+     * Exit the standby mode.
+     *
+     * @param streamHandle the stream handle
+     * @param parcelable contains new data queue information
+     * @return the result of the execution
+     */
+    virtual aaudio_result_t exitStandby(aaudio_handle_t streamHandle,
+                                        AudioEndpointParcelable &parcelable) = 0;
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
index dea3e4a..b1262df 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
@@ -79,6 +79,22 @@
     return index;
 }
 
+void AudioEndpointParcelable::closeDataFileDescriptor() {
+    const int32_t curDataMemoryIndex = mDownDataQueueParcelable.getSharedMemoryIndex();
+    mSharedMemories[curDataMemoryIndex].closeAndReleaseFd();
+}
+
+void AudioEndpointParcelable::updateDataFileDescriptor(
+        AudioEndpointParcelable* endpointParcelable) {
+    const int32_t curDataMemoryIndex = mDownDataQueueParcelable.getSharedMemoryIndex();
+    const int32_t newDataMemoryIndex =
+            endpointParcelable->mDownDataQueueParcelable.getSharedMemoryIndex();
+    mSharedMemories[curDataMemoryIndex].close();
+    mSharedMemories[curDataMemoryIndex].setup(
+            endpointParcelable->mSharedMemories[newDataMemoryIndex]);
+    mDownDataQueueParcelable.updateMemory(endpointParcelable->mDownDataQueueParcelable);
+}
+
 aaudio_result_t AudioEndpointParcelable::resolve(EndpointDescriptor *descriptor) {
     aaudio_result_t result = mUpMessageQueueParcelable.resolve(mSharedMemories,
                                                            &descriptor->upMessageQueueDescriptor);
@@ -92,6 +108,10 @@
     return result;
 }
 
+aaudio_result_t AudioEndpointParcelable::resolveDataQueue(RingBufferDescriptor *descriptor) {
+    return mDownDataQueueParcelable.resolve(mSharedMemories, descriptor);
+}
+
 aaudio_result_t AudioEndpointParcelable::close() {
     int err = 0;
     for (int i = 0; i < mNumSharedMemories; i++) {
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.h b/media/libaaudio/src/binding/AudioEndpointParcelable.h
index 544aa92..5d2c38f 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.h
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.h
@@ -52,7 +52,20 @@
      */
     int32_t addFileDescriptor(const android::base::unique_fd& fd, int32_t sizeInBytes);
 
+    /**
+     * Close current data file descriptor. The duplicated file descriptor will be close.
+     */
+    void closeDataFileDescriptor();
+
+    /**
+     * Update current data file descriptor with given endpoint parcelable.
+     * @param endpointParcelable an endpoint parcelable that contains new data file
+     *                           descriptor information
+     */
+    void updateDataFileDescriptor(AudioEndpointParcelable* endpointParcelable);
+
     aaudio_result_t resolve(EndpointDescriptor *descriptor);
+    aaudio_result_t resolveDataQueue(RingBufferDescriptor *descriptor);
 
     aaudio_result_t close();
 
diff --git a/media/libaaudio/src/binding/RingBufferParcelable.cpp b/media/libaaudio/src/binding/RingBufferParcelable.cpp
index fa7ca72..3bc51d0 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.cpp
+++ b/media/libaaudio/src/binding/RingBufferParcelable.cpp
@@ -33,6 +33,7 @@
         : mReadCounterParcelable(parcelable.readCounterParcelable),
           mWriteCounterParcelable(parcelable.writeCounterParcelable),
           mDataParcelable(parcelable.dataParcelable),
+          mSharedMemoryIndex(parcelable.sharedMemoryIndex),
           mBytesPerFrame(parcelable.bytesPerFrame),
           mFramesPerBurst(parcelable.framesPerBurst),
           mCapacityInFrames(parcelable.capacityInFrames),
@@ -45,6 +46,7 @@
     result.readCounterParcelable = mReadCounterParcelable.parcelable();
     result.writeCounterParcelable = mWriteCounterParcelable.parcelable();
     result.dataParcelable = mDataParcelable.parcelable();
+    result.sharedMemoryIndex = mSharedMemoryIndex;
     result.bytesPerFrame = mBytesPerFrame;
     result.framesPerBurst = mFramesPerBurst;
     result.capacityInFrames = mCapacityInFrames;
@@ -60,6 +62,7 @@
                  int32_t readCounterOffset,
                  int32_t writeCounterOffset,
                  int32_t counterSizeBytes) {
+    mSharedMemoryIndex = sharedMemoryIndex;
     mReadCounterParcelable.setup(sharedMemoryIndex, readCounterOffset, counterSizeBytes);
     mWriteCounterParcelable.setup(sharedMemoryIndex, writeCounterOffset, counterSizeBytes);
     mDataParcelable.setup(sharedMemoryIndex, dataMemoryOffset, dataSizeInBytes);
@@ -68,12 +71,13 @@
 void RingBufferParcelable::setupMemory(int32_t sharedMemoryIndex,
                  int32_t dataMemoryOffset,
                  int32_t dataSizeInBytes) {
+    mSharedMemoryIndex = sharedMemoryIndex;
     mReadCounterParcelable.setup(sharedMemoryIndex, 0, 0);
     mWriteCounterParcelable.setup(sharedMemoryIndex, 0, 0);
     mDataParcelable.setup(sharedMemoryIndex, dataMemoryOffset, dataSizeInBytes);
 }
 
-int32_t RingBufferParcelable::getBytesPerFrame() {
+int32_t RingBufferParcelable::getBytesPerFrame() const {
     return mBytesPerFrame;
 }
 
@@ -81,7 +85,7 @@
     mBytesPerFrame = bytesPerFrame;
 }
 
-int32_t RingBufferParcelable::getFramesPerBurst() {
+int32_t RingBufferParcelable::getFramesPerBurst() const {
     return mFramesPerBurst;
 }
 
@@ -89,7 +93,7 @@
     mFramesPerBurst = framesPerBurst;
 }
 
-int32_t RingBufferParcelable::getCapacityInFrames() {
+int32_t RingBufferParcelable::getCapacityInFrames() const {
     return mCapacityInFrames;
 }
 
@@ -124,6 +128,14 @@
     return AAUDIO_OK;
 }
 
+void RingBufferParcelable::updateMemory(const RingBufferParcelable& parcelable) {
+    setupMemory(mSharedMemoryIndex, 0,
+                parcelable.getCapacityInFrames() * parcelable.getBytesPerFrame());
+    setBytesPerFrame(parcelable.getBytesPerFrame());
+    setFramesPerBurst(parcelable.getFramesPerBurst());
+    setCapacityInFrames(parcelable.getCapacityInFrames());
+}
+
 aaudio_result_t RingBufferParcelable::validate() const {
     if (mCapacityInFrames < 0 || mCapacityInFrames >= 32 * 1024) {
         ALOGE("invalid mCapacityInFrames = %d", mCapacityInFrames);
diff --git a/media/libaaudio/src/binding/RingBufferParcelable.h b/media/libaaudio/src/binding/RingBufferParcelable.h
index 2508cea..29d0d86 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.h
+++ b/media/libaaudio/src/binding/RingBufferParcelable.h
@@ -46,15 +46,15 @@
                      int32_t dataMemoryOffset,
                      int32_t dataSizeInBytes);
 
-    int32_t getBytesPerFrame();
+    int32_t getBytesPerFrame() const;
 
     void setBytesPerFrame(int32_t bytesPerFrame);
 
-    int32_t getFramesPerBurst();
+    int32_t getFramesPerBurst() const;
 
     void setFramesPerBurst(int32_t framesPerBurst);
 
-    int32_t getCapacityInFrames();
+    int32_t getCapacityInFrames() const;
 
     void setCapacityInFrames(int32_t capacityInFrames);
 
@@ -62,6 +62,12 @@
 
     aaudio_result_t resolve(SharedMemoryParcelable *memoryParcels, RingBufferDescriptor *descriptor);
 
+    void updateMemory(const RingBufferParcelable& parcelable);
+
+    int32_t getSharedMemoryIndex() const {
+        return mSharedMemoryIndex;
+    }
+
     void dump();
 
     // Extract a parcelable representation of this object.
@@ -71,6 +77,7 @@
     SharedRegionParcelable  mReadCounterParcelable;
     SharedRegionParcelable  mWriteCounterParcelable;
     SharedRegionParcelable  mDataParcelable;
+    int32_t                 mSharedMemoryIndex = -1;
     int32_t                 mBytesPerFrame = 0;     // index is in frames
     int32_t                 mFramesPerBurst = 0;    // for ISOCHRONOUS queues
     int32_t                 mCapacityInFrames = 0;  // zero if unused
diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
index 3a49655..741aefc 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
@@ -64,6 +64,10 @@
     mSizeInBytes = sizeInBytes;
 }
 
+void SharedMemoryParcelable::setup(const SharedMemoryParcelable &sharedMemoryParcelable) {
+    setup(sharedMemoryParcelable.mFd, sharedMemoryParcelable.mSizeInBytes);
+}
+
 aaudio_result_t SharedMemoryParcelable::close() {
     if (mResolvedAddress != MMAP_UNRESOLVED_ADDRESS) {
         int err = munmap(mResolvedAddress, mSizeInBytes);
@@ -76,6 +80,14 @@
     return AAUDIO_OK;
 }
 
+aaudio_result_t SharedMemoryParcelable::closeAndReleaseFd() {
+    aaudio_result_t result = close();
+    if (result == AAUDIO_OK) {
+        mFd.reset();
+    }
+    return result;
+}
+
 aaudio_result_t SharedMemoryParcelable::resolveSharedMemory(const unique_fd& fd) {
     mResolvedAddress = (uint8_t *) mmap(nullptr, mSizeInBytes, PROT_READ | PROT_WRITE,
                                         MAP_SHARED, fd.get(), 0);
diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.h b/media/libaaudio/src/binding/SharedMemoryParcelable.h
index 1f2c335..7762fef 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.h
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.h
@@ -52,12 +52,16 @@
      */
     void setup(const android::base::unique_fd& fd, int32_t sizeInBytes);
 
+    void setup(const SharedMemoryParcelable& sharedMemoryParcelable);
+
     // mmap() shared memory
     aaudio_result_t resolve(int32_t offsetInBytes, int32_t sizeInBytes, void **regionAddressPtr);
 
     // munmap() any mapped memory
     aaudio_result_t close();
 
+    aaudio_result_t closeAndReleaseFd();
+
     int32_t getSizeInBytes();
 
     void dump();
diff --git a/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl b/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl
index 44d2211..485c2e2 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/IAAudioService.aidl
@@ -78,4 +78,6 @@
 
     int unregisterAudioThread(int streamHandle,
                               int clientThreadId);
+
+    int exitStandby(int streamHandle, out Endpoint endpoint);
 }
diff --git a/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl b/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl
index a58b33a..dd64493 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/RingBuffer.aidl
@@ -26,4 +26,5 @@
     int                 framesPerBurst;    // for ISOCHRONOUS queues
     int                 capacityInFrames;  // zero if unused
     int /* RingbufferFlags */ flags;  // = RingbufferFlags::NONE;
+    int                 sharedMemoryIndex;
 }
\ No newline at end of file
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 24888de..e780f4f 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -139,38 +139,49 @@
     );
 
     // ============================ data queue =============================
-    descriptor = &pEndpointDescriptor->dataQueueDescriptor;
-    ALOGV("configure() data framesPerBurst = %d", descriptor->framesPerBurst);
+    result = configureDataQueue(pEndpointDescriptor->dataQueueDescriptor, direction);
+
+    return result;
+}
+
+aaudio_result_t AudioEndpoint::configureDataQueue(const RingBufferDescriptor& descriptor,
+                                                  aaudio_direction_t direction) {
+    aaudio_result_t result = AudioEndpoint_validateQueueDescriptor("data", &descriptor);
+    if (result != AAUDIO_OK) {
+        return result;
+    }
+
+    ALOGV("configure() data framesPerBurst = %d", descriptor.framesPerBurst);
     ALOGV("configure() data readCounterAddress = %p",
-          descriptor->readCounterAddress);
+          descriptor.readCounterAddress);
 
     // An example of free running is when the other side is read or written by hardware DMA
     // or a DSP. It does not update its counter so we have to update it.
     int64_t *remoteCounter = (direction == AAUDIO_DIRECTION_OUTPUT)
-                             ? descriptor->readCounterAddress // read by other side
-                             : descriptor->writeCounterAddress; // written by other side
+                             ? descriptor.readCounterAddress // read by other side
+                             : descriptor.writeCounterAddress; // written by other side
     mFreeRunning = (remoteCounter == nullptr);
     ALOGV("configure() mFreeRunning = %d", mFreeRunning ? 1 : 0);
 
-    int64_t *readCounterAddress = (descriptor->readCounterAddress == nullptr)
+    int64_t *readCounterAddress = (descriptor.readCounterAddress == nullptr)
                                   ? &mDataReadCounter
-                                  : descriptor->readCounterAddress;
-    int64_t *writeCounterAddress = (descriptor->writeCounterAddress == nullptr)
+                                  : descriptor.readCounterAddress;
+    int64_t *writeCounterAddress = (descriptor.writeCounterAddress == nullptr)
                                   ? &mDataWriteCounter
-                                  : descriptor->writeCounterAddress;
+                                  : descriptor.writeCounterAddress;
 
     // Clear buffer to avoid an initial glitch on some devices.
-    size_t bufferSizeBytes = descriptor->capacityInFrames * descriptor->bytesPerFrame;
-    memset(descriptor->dataAddress, 0, bufferSizeBytes);
+    size_t bufferSizeBytes = descriptor.capacityInFrames * descriptor.bytesPerFrame;
+    memset(descriptor.dataAddress, 0, bufferSizeBytes);
 
     mDataQueue = std::make_unique<FifoBufferIndirect>(
-            descriptor->bytesPerFrame,
-            descriptor->capacityInFrames,
+            descriptor.bytesPerFrame,
+            descriptor.capacityInFrames,
             readCounterAddress,
             writeCounterAddress,
-            descriptor->dataAddress
+            descriptor.dataAddress
     );
-    uint32_t threshold = descriptor->capacityInFrames / 2;
+    uint32_t threshold = descriptor.capacityInFrames / 2;
     mDataQueue->setThreshold(threshold);
     return result;
 }
@@ -181,47 +192,66 @@
 }
 
 int32_t AudioEndpoint::getEmptyFramesAvailable(WrappingBuffer *wrappingBuffer) {
-    return mDataQueue->getEmptyRoomAvailable(wrappingBuffer);
+    return mDataQueue == nullptr ? 0 : mDataQueue->getEmptyRoomAvailable(wrappingBuffer);
 }
 
 int32_t AudioEndpoint::getEmptyFramesAvailable() {
-    return mDataQueue->getEmptyFramesAvailable();
+    return mDataQueue == nullptr ? 0 : mDataQueue->getEmptyFramesAvailable();
 }
 
 int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer) {
-    return mDataQueue->getFullDataAvailable(wrappingBuffer);
+    return mDataQueue == nullptr ? 0 : mDataQueue->getFullDataAvailable(wrappingBuffer);
 }
 
 int32_t AudioEndpoint::getFullFramesAvailable() {
-    return mDataQueue->getFullFramesAvailable();
+    return mDataQueue == nullptr ? 0 : mDataQueue->getFullFramesAvailable();
+}
+
+android::fifo_frames_t AudioEndpoint::read(void *buffer, android::fifo_frames_t numFrames) {
+    return mDataQueue == nullptr ? 0 : mDataQueue->read(buffer, numFrames);
+}
+
+android::fifo_frames_t AudioEndpoint::write(void *buffer, android::fifo_frames_t numFrames) {
+    return mDataQueue == nullptr ? 0 : mDataQueue->write(buffer, numFrames);
 }
 
 void AudioEndpoint::advanceWriteIndex(int32_t deltaFrames) {
-    mDataQueue->advanceWriteIndex(deltaFrames);
+    if (mDataQueue != nullptr) {
+        mDataQueue->advanceWriteIndex(deltaFrames);
+    }
 }
 
 void AudioEndpoint::advanceReadIndex(int32_t deltaFrames) {
-    mDataQueue->advanceReadIndex(deltaFrames);
+    if (mDataQueue != nullptr) {
+        mDataQueue->advanceReadIndex(deltaFrames);
+    }
 }
 
 void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead) {
-    mDataQueue->setReadCounter(framesRead);
+    if (mDataQueue != nullptr) {
+        mDataQueue->setReadCounter(framesRead);
+    }
 }
 
 fifo_counter_t AudioEndpoint::getDataReadCounter() const {
-    return mDataQueue->getReadCounter();
+    return mDataQueue == nullptr ? 0 : mDataQueue->getReadCounter();
 }
 
 void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead) {
-    mDataQueue->setWriteCounter(framesRead);
+    if (mDataQueue != nullptr) {
+        mDataQueue->setWriteCounter(framesRead);
+    }
 }
 
 fifo_counter_t AudioEndpoint::getDataWriteCounter() const {
-    return mDataQueue->getWriteCounter();
+    return mDataQueue == nullptr ? 0 : mDataQueue->getWriteCounter();
 }
 
 int32_t AudioEndpoint::setBufferSizeInFrames(int32_t requestedFrames,
                                             int32_t *actualFrames) {
+    if (mDataQueue == nullptr) {
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
     if (requestedFrames < ENDPOINT_DATA_QUEUE_SIZE_MIN) {
         requestedFrames = ENDPOINT_DATA_QUEUE_SIZE_MIN;
     }
@@ -231,11 +261,11 @@
 }
 
 int32_t AudioEndpoint::getBufferSizeInFrames() const {
-    return mDataQueue->getThreshold();
+    return mDataQueue == nullptr ? 0 : mDataQueue->getThreshold();
 }
 
 int32_t AudioEndpoint::getBufferCapacityInFrames() const {
-    return (int32_t)mDataQueue->getBufferCapacityInFrames();
+    return mDataQueue == nullptr ? 0 : (int32_t)mDataQueue->getBufferCapacityInFrames();
 }
 
 void AudioEndpoint::dump() const {
@@ -244,5 +274,7 @@
 }
 
 void AudioEndpoint::eraseDataMemory() {
-    mDataQueue->eraseMemory();
+    if (mDataQueue != nullptr) {
+        mDataQueue->eraseMemory();
+    }
 }
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index b3dbc20..01dd05a 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AAUDIO_AUDIO_ENDPOINT_H
 #define ANDROID_AAUDIO_AUDIO_ENDPOINT_H
 
+#include <mutex>
+
 #include <aaudio/AAudio.h>
 
 #include "binding/AAudioServiceMessage.h"
@@ -42,6 +44,9 @@
     aaudio_result_t configure(const EndpointDescriptor *pEndpointDescriptor,
                               aaudio_direction_t direction);
 
+    aaudio_result_t configureDataQueue(const RingBufferDescriptor &descriptor,
+                            aaudio_direction_t direction);
+
     /**
      * Read from a command passed up from the Server.
      * @return 1 if command received, 0 for no command, or negative error.
@@ -56,6 +61,10 @@
 
     int32_t getFullFramesAvailable();
 
+    android::fifo_frames_t read(void* buffer, android::fifo_frames_t numFrames);
+
+    android::fifo_frames_t write(void* buffer, android::fifo_frames_t numFrames);
+
     void advanceReadIndex(int32_t deltaFrames);
 
     void advanceWriteIndex(int32_t deltaFrames);
@@ -85,11 +94,21 @@
 
     int32_t getBufferCapacityInFrames() const;
 
+    void setThreshold(int32_t frames) {
+        mDataQueue->setThreshold(frames);
+    }
+
+    int32_t getThreshold() {
+        return mDataQueue->getThreshold();
+    }
+
     /**
      * Write zeros to the data queue memory.
      */
     void eraseDataMemory();
 
+    void freeDataQueue();
+
     void dump() const;
 
 private:
@@ -98,6 +117,8 @@
     bool                    mFreeRunning{false};
     android::fifo_counter_t mDataReadCounter{0}; // only used if free-running
     android::fifo_counter_t mDataWriteCounter{0}; // only used if free-running
+
+    std::mutex mDataQueueLock;
 };
 
 } // namespace aaudio
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index afdc2ac..9f0564f 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -81,8 +81,6 @@
 aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
 
     aaudio_result_t result = AAUDIO_OK;
-    int32_t framesPerBurst;
-    int32_t framesPerHardwareBurst;
     AAudioStreamRequest request;
     AAudioStreamConfiguration configurationOutput;
 
@@ -97,9 +95,6 @@
         return result;
     }
 
-    const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
-    int32_t burstMicros = 0;
-
     const audio_format_t requestedFormat = getFormat();
     // We have to do volume scaling. So we prefer FLOAT format.
     if (requestedFormat == AUDIO_FORMAT_DEFAULT) {
@@ -215,12 +210,28 @@
         goto error;
     }
 
-    framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
+    if ((result = configureDataInformation(builder.getFramesPerDataCallback())) != AAUDIO_OK) {
+        goto error;
+    }
+
+    setState(AAUDIO_STREAM_STATE_OPEN);
+
+    return result;
+
+error:
+    safeReleaseClose();
+    return result;
+}
+
+aaudio_result_t AudioStreamInternal::configureDataInformation(int32_t callbackFrames) {
+    int32_t framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
 
     // Scale up the burst size to meet the minimum equivalent in microseconds.
     // This is to avoid waking the CPU too often when the HW burst is very small
     // or at high sample rates.
-    framesPerBurst = framesPerHardwareBurst;
+    int32_t framesPerBurst = framesPerHardwareBurst;
+    int32_t burstMicros = 0;
+    const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
     do {
         if (burstMicros > 0) {  // skip first loop
             framesPerBurst *= 2;
@@ -233,8 +244,7 @@
     // Validate final burst size.
     if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
         ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst);
-        result = AAUDIO_ERROR_OUT_OF_RANGE;
-        goto error;
+        return AAUDIO_ERROR_OUT_OF_RANGE;
     }
     setFramesPerBurst(framesPerBurst); // only save good value
 
@@ -242,26 +252,21 @@
     if (mBufferCapacityInFrames < getFramesPerBurst()
             || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
         ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
-        result = AAUDIO_ERROR_OUT_OF_RANGE;
-        goto error;
+        return AAUDIO_ERROR_OUT_OF_RANGE;
     }
 
     mClockModel.setSampleRate(getSampleRate());
     mClockModel.setFramesPerBurst(framesPerHardwareBurst);
 
     if (isDataCallbackSet()) {
-        mCallbackFrames = builder.getFramesPerDataCallback();
+        mCallbackFrames = callbackFrames;
         if (mCallbackFrames > getBufferCapacity() / 2) {
             ALOGW("%s - framesPerCallback too big = %d, capacity = %d",
                   __func__, mCallbackFrames, getBufferCapacity());
-            result = AAUDIO_ERROR_OUT_OF_RANGE;
-            goto error;
-
+            return AAUDIO_ERROR_OUT_OF_RANGE;
         } else if (mCallbackFrames < 0) {
             ALOGW("%s - framesPerCallback negative", __func__);
-            result = AAUDIO_ERROR_OUT_OF_RANGE;
-            goto error;
-
+            return AAUDIO_ERROR_OUT_OF_RANGE;
         }
         if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
             mCallbackFrames = getFramesPerBurst();
@@ -302,14 +307,7 @@
     }
 
     setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
-
-    setState(AAUDIO_STREAM_STATE_OPEN);
-
-    return result;
-
-error:
-    safeReleaseClose();
-    return result;
+    return AAUDIO_OK;
 }
 
 // This must be called under mStreamLock.
@@ -357,6 +355,60 @@
     }
 }
 
+aaudio_result_t AudioStreamInternal::exitStandby_l() {
+    AudioEndpointParcelable endpointParcelable;
+    // The stream is in standby mode, copy all available data and then close the duplicated
+    // shared file descriptor so that it won't cause issue when the HAL try to reallocate new
+    // shared file descriptor when exiting from standby.
+    // Cache current read counter, which will be reset to new read and write counter
+    // when the new data queue and endpoint are reconfigured.
+    const android::fifo_counter_t readCounter = mAudioEndpoint->getDataReadCounter();
+    // Cache the buffer size which may be from client.
+    const int32_t previousBufferSize = mBufferSizeInFrames;
+    // Copy all available data from current data queue.
+    uint8_t buffer[getBufferCapacity() * getBytesPerFrame()];
+    android::fifo_frames_t fullFramesAvailable =
+            mAudioEndpoint->read(buffer, getBufferCapacity());
+    mEndPointParcelable.closeDataFileDescriptor();
+    aaudio_result_t result = mServiceInterface.exitStandby(
+            mServiceStreamHandle, endpointParcelable);
+    if (result != AAUDIO_OK) {
+        ALOGE("Failed to exit standby, error=%d", result);
+        goto exit;
+    }
+    // Reconstruct data queue descriptor using new shared file descriptor.
+    mEndPointParcelable.updateDataFileDescriptor(&endpointParcelable);
+    result = mEndPointParcelable.resolveDataQueue(&mEndpointDescriptor.dataQueueDescriptor);
+    if (result != AAUDIO_OK) {
+        ALOGE("Failed to resolve data queue after exiting standby, error=%d", result);
+        goto exit;
+    }
+    // Reconfigure audio endpoint with new data queue descriptor.
+    mAudioEndpoint->configureDataQueue(
+            mEndpointDescriptor.dataQueueDescriptor, getDirection());
+    // Set read and write counters with previous read counter, the later write action
+    // will make the counter at the correct place.
+    mAudioEndpoint->setDataReadCounter(readCounter);
+    mAudioEndpoint->setDataWriteCounter(readCounter);
+    result = configureDataInformation(mCallbackFrames);
+    if (result != AAUDIO_OK) {
+        ALOGE("Failed to configure data information after exiting standby, error=%d", result);
+        goto exit;
+    }
+    // Write data from previous data buffer to new endpoint.
+    if (android::fifo_frames_t framesWritten =
+                mAudioEndpoint->write(buffer, fullFramesAvailable);
+            framesWritten != fullFramesAvailable) {
+        ALOGW("Some data lost after exiting standby, frames written: %d, "
+              "frames to write: %d", framesWritten, fullFramesAvailable);
+    }
+    // Reset previous buffer size as it may be requested by the client.
+    setBufferSize(previousBufferSize);
+
+exit:
+    return result;
+}
+
 /*
  * It normally takes about 20-30 msec to start a stream on the server.
  * But the first time can take as much as 200-300 msec. The HW
@@ -393,8 +445,15 @@
     prepareBuffersForStart(); // tell subclasses to get ready
 
     aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);
-    if (result == AAUDIO_ERROR_INVALID_HANDLE) {
-        ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
+    if (result == AAUDIO_ERROR_STANDBY) {
+        // The stream is at standby mode. Need to exit standby before starting the stream.
+        result = exitStandby_l();
+        if (result == AAUDIO_OK) {
+            result = mServiceInterface.startStream(mServiceStreamHandle);
+        }
+    }
+    if (result != AAUDIO_OK) {
+        ALOGD("%s() error = %d, stream was probably stolen", __func__, result);
         // Stealing was added in R. Coerce result to improve backward compatibility.
         result = AAUDIO_ERROR_DISCONNECTED;
         setState(AAUDIO_STREAM_STATE_DISCONNECTED);
@@ -414,6 +473,7 @@
         result = createThread_l(periodNanos, aaudio_callback_thread_proc, this);
     }
     if (result != AAUDIO_OK) {
+        // TODO(b/214607638): Do we want to roll back to original state or keep as disconnected?
         setState(originalState);
     }
     return result;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index eab1382..2367572 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -184,9 +184,14 @@
     aaudio_result_t writeNowWithConversion(const void *buffer,
                                      int32_t numFrames);
 
+    // Exit the stream from standby, will reconstruct data path.
+    aaudio_result_t exitStandby_l() REQUIRES(mStreamLock);
+
     // Adjust timing model based on timestamp from service.
     void processTimestamp(uint64_t position, int64_t time);
 
+    aaudio_result_t configureDataInformation(int32_t callbackFrames);
+
     // Thread on other side of FIFO will have wakeup jitter.
     // By delaying slightly we can avoid waking up before other side is ready.
     const int32_t            mWakeupDelayNanos; // delay past typical wakeup jitter