diff --git a/drm/libmediadrm/CryptoHal.cpp b/drm/libmediadrm/CryptoHal.cpp
index 1fdc6e1..5dd2563 100644
--- a/drm/libmediadrm/CryptoHal.cpp
+++ b/drm/libmediadrm/CryptoHal.cpp
@@ -240,11 +240,12 @@
     Mutex::Autolock autoLock(mLock);
 
     int32_t seqNum = mHeapSeqNum++;
+
     int fd = heap->getHeapID();
     nativeHandle->data[0] = fd;
     auto hidlHandle = hidl_handle(nativeHandle);
     auto hidlMemory = hidl_memory("ashmem", hidlHandle, heap->getSize());
-    mHeapBases.add(seqNum, mNextBufferId);
+    mHeapBases.add(seqNum, HeapBase(mNextBufferId, heap->getSize()));
     Return<void> hResult = mPlugin->setSharedBufferBase(hidlMemory, mNextBufferId++);
     ALOGE_IF(!hResult.isOk(), "setSharedBufferBase(): remote call failed");
     return seqNum;
@@ -269,10 +270,26 @@
         return UNEXPECTED_NULL;
     }
 
-    // memory must be in the declared heap
-    CHECK(mHeapBases.indexOfKey(seqNum) >= 0);
+    // memory must be in one of the heaps that have been set
+    if (mHeapBases.indexOfKey(seqNum) < 0) {
+        return UNKNOWN_ERROR;
+    }
 
-    buffer->bufferId = mHeapBases.valueFor(seqNum);
+    // heap must be the same size as the one that was set in setHeapBase
+    if (mHeapBases.valueFor(seqNum).getSize() != heap->getSize()) {
+        android_errorWriteLog(0x534e4554, "76221123");
+        return UNKNOWN_ERROR;
+     }
+
+    // memory must be within the address space of the heap
+    if (memory->pointer() != static_cast<uint8_t *>(heap->getBase()) + memory->offset()  ||
+            heap->getSize() < memory->offset() + memory->size() ||
+            SIZE_MAX - memory->offset() < memory->size()) {
+        android_errorWriteLog(0x534e4554, "76221123");
+        return UNKNOWN_ERROR;
+    }
+
+    buffer->bufferId = mHeapBases.valueFor(seqNum).getBufferId();
     buffer->offset = offset >= 0 ? offset : 0;
     buffer->size = size;
     return OK;
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index d838975..f071a02 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -940,7 +940,7 @@
             audio_output_flags_t flags =
                     static_cast <audio_output_flags_t>(data.readInt32());
             bool hasOffloadInfo = data.readInt32() != 0;
-            audio_offload_info_t offloadInfo;
+            audio_offload_info_t offloadInfo = {};
             if (hasOffloadInfo) {
                 data.read(&offloadInfo, sizeof(audio_offload_info_t));
             }
@@ -956,7 +956,7 @@
 
         case GET_OUTPUT_FOR_ATTR: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_attributes_t attr;
+            audio_attributes_t attr = {};
             bool hasAttributes = data.readInt32() != 0;
             if (hasAttributes) {
                 data.read(&attr, sizeof(audio_attributes_t));
@@ -1024,7 +1024,7 @@
 
         case GET_INPUT_FOR_ATTR: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_attributes_t attr;
+            audio_attributes_t attr = {};
             data.read(&attr, sizeof(audio_attributes_t));
             sanetizeAudioAttributes(&attr);
             audio_io_handle_t input = (audio_io_handle_t)data.readInt32();
@@ -1125,8 +1125,11 @@
 
         case GET_OUTPUT_FOR_EFFECT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            effect_descriptor_t desc;
-            data.read(&desc, sizeof(effect_descriptor_t));
+            effect_descriptor_t desc = {};
+            if (data.read(&desc, sizeof(desc)) != NO_ERROR) {
+                android_errorWriteLog(0x534e4554, "73126106");
+            }
+            (void)sanitizeEffectDescriptor(&desc);
             audio_io_handle_t output = getOutputForEffect(&desc);
             reply->writeInt32(static_cast <int>(output));
             return NO_ERROR;
@@ -1134,8 +1137,11 @@
 
         case REGISTER_EFFECT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            effect_descriptor_t desc;
-            data.read(&desc, sizeof(effect_descriptor_t));
+            effect_descriptor_t desc = {};
+            if (data.read(&desc, sizeof(desc)) != NO_ERROR) {
+                android_errorWriteLog(0x534e4554, "73126106");
+            }
+            (void)sanitizeEffectDescriptor(&desc);
             audio_io_handle_t io = data.readInt32();
             uint32_t strategy = data.readInt32();
             audio_session_t session = (audio_session_t) data.readInt32();
@@ -1194,7 +1200,7 @@
                 count = AudioEffect::kMaxPreProcessing;
             }
             uint32_t retCount = count;
-            effect_descriptor_t *descriptors = new effect_descriptor_t[count];
+            effect_descriptor_t *descriptors = new effect_descriptor_t[count]{};
             status_t status = queryDefaultPreProcessing(audioSession, descriptors, &retCount);
             reply->writeInt32(status);
             if (status != NO_ERROR && status != NO_MEMORY) {
@@ -1213,7 +1219,7 @@
 
         case IS_OFFLOAD_SUPPORTED: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_offload_info_t info;
+            audio_offload_info_t info = {};
             data.read(&info, sizeof(audio_offload_info_t));
             bool isSupported = isOffloadSupported(info);
             reply->writeInt32(isSupported);
@@ -1268,7 +1274,7 @@
 
         case CREATE_AUDIO_PATCH: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            struct audio_patch patch;
+            struct audio_patch patch = {};
             data.read(&patch, sizeof(struct audio_patch));
             audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
             if (data.read(&handle, sizeof(audio_patch_handle_t)) != NO_ERROR) {
@@ -1284,7 +1290,7 @@
 
         case RELEASE_AUDIO_PATCH: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_patch_handle_t handle;
+            audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
             data.read(&handle, sizeof(audio_patch_handle_t));
             status_t status = releaseAudioPatch(handle);
             reply->writeInt32(status);
@@ -1323,8 +1329,9 @@
 
         case SET_AUDIO_PORT_CONFIG: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            struct audio_port_config config;
+            struct audio_port_config config = {};
             data.read(&config, sizeof(struct audio_port_config));
+            (void)sanitizeAudioPortConfig(&config);
             status_t status = setAudioPortConfig(&config);
             reply->writeInt32(status);
             return NO_ERROR;
@@ -1398,9 +1405,10 @@
 
         case START_AUDIO_SOURCE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            struct audio_port_config source;
+            struct audio_port_config source = {};
             data.read(&source, sizeof(struct audio_port_config));
-            audio_attributes_t attributes;
+            (void)sanitizeAudioPortConfig(&source);
+            audio_attributes_t attributes = {};
             data.read(&attributes, sizeof(audio_attributes_t));
             sanetizeAudioAttributes(&attributes);
             audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
@@ -1453,6 +1461,14 @@
     }
 }
 
+/** returns true if string overflow was prevented by zero termination */
+template <size_t size>
+static bool preventStringOverflow(char (&s)[size]) {
+    if (strnlen(s, size) < size) return false;
+    s[size - 1] = '\0';
+    return true;
+}
+
 void BnAudioPolicyService::sanetizeAudioAttributes(audio_attributes_t* attr)
 {
     const size_t tagsMaxSize = AUDIO_ATTRIBUTES_TAGS_MAX_SIZE;
@@ -1462,6 +1478,27 @@
     attr->tags[tagsMaxSize - 1] = '\0';
 }
 
+/** returns BAD_VALUE if sanitization was required. */
+status_t BnAudioPolicyService::sanitizeEffectDescriptor(effect_descriptor_t* desc)
+{
+    if (preventStringOverflow(desc->name)
+        | /* always */ preventStringOverflow(desc->implementor)) {
+        android_errorWriteLog(0x534e4554, "73126106"); // SafetyNet logging
+        return BAD_VALUE;
+    }
+    return NO_ERROR;
+}
+
+/** returns BAD_VALUE if sanitization was required. */
+status_t BnAudioPolicyService::sanitizeAudioPortConfig(struct audio_port_config* config)
+{
+    if (config->type == AUDIO_PORT_TYPE_DEVICE &&
+        preventStringOverflow(config->ext.device.address)) {
+        return BAD_VALUE;
+    }
+    return NO_ERROR;
+}
+
 // ----------------------------------------------------------------------------
 
 } // namespace android
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index 60ba4ba..eec3e88 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -185,6 +185,8 @@
                                     uint32_t flags = 0);
 private:
     void sanetizeAudioAttributes(audio_attributes_t* attr);
+    status_t sanitizeEffectDescriptor(effect_descriptor_t* desc);
+    status_t sanitizeAudioPortConfig(struct audio_port_config* config);
 };
 
 // ----------------------------------------------------------------------------
diff --git a/media/libmedia/include/media/CryptoHal.h b/media/libmedia/include/media/CryptoHal.h
index a5d8b43..80e181e 100644
--- a/media/libmedia/include/media/CryptoHal.h
+++ b/media/libmedia/include/media/CryptoHal.h
@@ -79,7 +79,20 @@
      */
     status_t mInitCheck;
 
-    KeyedVector<int32_t, uint32_t> mHeapBases;
+    struct HeapBase {
+        HeapBase() : mBufferId(0), mSize(0) {}
+        HeapBase(uint32_t bufferId, size_t size) :
+            mBufferId(bufferId), mSize(size) {}
+
+        uint32_t getBufferId() const {return mBufferId;}
+        size_t getSize() const {return mSize;}
+
+    private:
+        uint32_t mBufferId;
+        size_t mSize;
+    };
+
+    KeyedVector<int32_t, HeapBase> mHeapBases;
     uint32_t mNextBufferId;
     int32_t mHeapSeqNum;
 
diff --git a/media/libstagefright/ItemTable.cpp b/media/libstagefright/ItemTable.cpp
index f9ee1c4..f589f4f 100644
--- a/media/libstagefright/ItemTable.cpp
+++ b/media/libstagefright/ItemTable.cpp
@@ -1335,7 +1335,8 @@
         ALOGV("adding %s: itemId %d", image.isGrid() ? "grid" : "image", info.itemId);
 
         if (image.isGrid()) {
-            if (size > 12) {
+            // ImageGrid struct is at least 8-byte, at most 12-byte (if flags&1)
+            if (size < 8 || size > 12) {
                 return ERROR_MALFORMED;
             }
             uint8_t buf[12];
diff --git a/media/libstagefright/VideoFrameScheduler.cpp b/media/libstagefright/VideoFrameScheduler.cpp
index 03226c7..6819bba 100644
--- a/media/libstagefright/VideoFrameScheduler.cpp
+++ b/media/libstagefright/VideoFrameScheduler.cpp
@@ -129,6 +129,11 @@
         numSamplesToUse = mNumSamples;
     }
 
+    if ((period >> kPrecision) == 0 ) {
+        ALOGW("Period is 0, or after including precision is 0 - would cause div0, returning");
+        return false;
+    }
+
     int64_t sumX = 0;
     int64_t sumXX = 0;
     int64_t sumXY = 0;
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 58d8b13..e897b67 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -328,12 +328,25 @@
 }
 
 void ID3::removeUnsynchronization() {
-    for (size_t i = 0; i + 1 < mSize; ++i) {
-        if (mData[i] == 0xff && mData[i + 1] == 0x00) {
-            memmove(&mData[i + 1], &mData[i + 2], mSize - i - 2);
-            --mSize;
+
+    // This file has "unsynchronization", so we have to replace occurrences
+    // of 0xff 0x00 with just 0xff in order to get the real data.
+
+    size_t writeOffset = 1;
+    for (size_t readOffset = 1; readOffset < mSize; ++readOffset) {
+        if (mData[readOffset - 1] == 0xff && mData[readOffset] == 0x00) {
+            continue;
         }
+        // Only move data if there's actually something to move.
+        // This handles the special case of the data being only [0xff, 0x00]
+        // which should be converted to just 0xff if unsynchronization is on.
+        mData[writeOffset++] = mData[readOffset];
     }
+
+    if (writeOffset < mSize) {
+        mSize = writeOffset;
+    }
+
 }
 
 static void WriteSyncsafeInteger(uint8_t *dst, size_t x) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index fcf9070..cdaffe7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -385,6 +385,7 @@
     mSamplingRate = 0;
     mChannelMask = AUDIO_CHANNEL_NONE;
     mFormat = AUDIO_FORMAT_INVALID;
+    memset(&mGain, 0, sizeof(struct audio_gain_config));
     mGain.index = -1;
 }
 
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index 2ef6234..c19cc35 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -49,9 +49,9 @@
 
     virtual aaudio_result_t close() = 0;
 
-    virtual aaudio_result_t registerStream(android::sp<AAudioServiceStreamBase> stream);
+    aaudio_result_t registerStream(android::sp<AAudioServiceStreamBase> stream);
 
-    virtual aaudio_result_t unregisterStream(android::sp<AAudioServiceStreamBase> stream);
+    aaudio_result_t unregisterStream(android::sp<AAudioServiceStreamBase> stream);
 
     virtual aaudio_result_t startStream(android::sp<AAudioServiceStreamBase> stream,
                                         audio_port_handle_t *clientHandle) = 0;
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index ff0b037..d9b8766 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -104,6 +104,9 @@
             goto error;
         }
 
+        // This is not protected by a lock because the stream cannot be
+        // referenced until the service returns a handle to the client.
+        // So only one thread can open a stream.
         mServiceEndpoint = mEndpointManager.openEndpoint(mAudioService,
                                                          request,
                                                          sharingMode);
@@ -112,6 +115,9 @@
             result = AAUDIO_ERROR_UNAVAILABLE;
             goto error;
         }
+        // Save a weak pointer that we will use to access the endpoint.
+        mServiceEndpointWeak = mServiceEndpoint;
+
         mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
         copyFrom(*mServiceEndpoint);
     }
@@ -130,15 +136,19 @@
 
     stop();
 
-    if (mServiceEndpoint == nullptr) {
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
         result = AAUDIO_ERROR_INVALID_STATE;
     } else {
-        mServiceEndpoint->unregisterStream(this);
-        AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
-        mEndpointManager.closeEndpoint(mServiceEndpoint);
-        mServiceEndpoint.clear();
+        endpoint->unregisterStream(this);
+        AAudioEndpointManager &endpointManager = AAudioEndpointManager::getInstance();
+        endpointManager.closeEndpoint(endpoint);
+
+        // AAudioService::closeStream() prevents two threads from closing at the same time.
+        mServiceEndpoint.clear(); // endpoint will hold the pointer until this method returns.
     }
 
+
     {
         std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
         stopTimestampThread();
@@ -152,7 +162,12 @@
 
 aaudio_result_t AAudioServiceStreamBase::startDevice() {
     mClientHandle = AUDIO_PORT_HANDLE_NONE;
-    return mServiceEndpoint->startStream(this, &mClientHandle);
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    return endpoint->startStream(this, &mClientHandle);
 }
 
 /**
@@ -162,16 +177,11 @@
  */
 aaudio_result_t AAudioServiceStreamBase::start() {
     aaudio_result_t result = AAUDIO_OK;
+
     if (isRunning()) {
         return AAUDIO_OK;
     }
 
-    if (mServiceEndpoint == nullptr) {
-        ALOGE("AAudioServiceStreamBase::start() missing endpoint");
-        result = AAUDIO_ERROR_INVALID_STATE;
-        goto error;
-    }
-
     // Start with fresh presentation timestamps.
     mAtomicTimestamp.clear();
 
@@ -198,10 +208,6 @@
     if (!isRunning()) {
         return result;
     }
-    if (mServiceEndpoint == nullptr) {
-        ALOGE("AAudioServiceStreamShared::pause() missing endpoint");
-        return AAUDIO_ERROR_INVALID_STATE;
-    }
 
     // Send it now because the timestamp gets rounded up when stopStream() is called below.
     // Also we don't need the timestamps while we are shutting down.
@@ -213,7 +219,12 @@
         return result;
     }
 
-    result = mServiceEndpoint->stopStream(this, mClientHandle);
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    result = endpoint->stopStream(this, mClientHandle);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::pause() mServiceEndpoint returned %d", result);
         disconnect(); // TODO should we return or pause Base first?
@@ -230,11 +241,6 @@
         return result;
     }
 
-    if (mServiceEndpoint == nullptr) {
-        ALOGE("AAudioServiceStreamShared::stop() missing endpoint");
-        return AAUDIO_ERROR_INVALID_STATE;
-    }
-
     // Send it now because the timestamp gets rounded up when stopStream() is called below.
     // Also we don't need the timestamps while we are shutting down.
     sendCurrentTimestamp(); // warning - this calls a virtual function
@@ -244,8 +250,13 @@
         return result;
     }
 
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
     // TODO wait for data to be played out
-    result = mServiceEndpoint->stopStream(this, mClientHandle);
+    result = endpoint->stopStream(this, mClientHandle);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::stop() mServiceEndpoint returned %d", result);
         disconnect();
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index af435b4..815b1cc 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -233,7 +233,12 @@
     SimpleDoubleBuffer<Timestamp>  mAtomicTimestamp;
 
     android::AAudioService &mAudioService;
+
+    // The mServiceEndpoint variable can be accessed by multiple threads.
+    // So we access it by locally promoting a weak pointer to a smart pointer,
+    // which is thread-safe.
     android::sp<AAudioServiceEndpoint> mServiceEndpoint;
+    android::wp<AAudioServiceEndpoint> mServiceEndpointWeak;
 
 private:
     aaudio_handle_t         mHandle = -1;
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index 44ba1ca..2e7ff8b 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -70,14 +70,19 @@
         return result;
     }
 
-    result = mServiceEndpoint->registerStream(keep);
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+
+    result = endpoint->registerStream(keep);
     if (result != AAUDIO_OK) {
-        goto error;
+        return result;
     }
 
     setState(AAUDIO_STREAM_STATE_OPEN);
 
-error:
     return AAUDIO_OK;
 }
 
@@ -122,21 +127,37 @@
 
 aaudio_result_t AAudioServiceStreamMMAP::startClient(const android::AudioClient& client,
                                                        audio_port_handle_t *clientHandle) {
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
     // Start the client on behalf of the application. Generate a new porthandle.
-    aaudio_result_t result = mServiceEndpoint->startClient(client, clientHandle);
+    aaudio_result_t result = endpoint->startClient(client, clientHandle);
     return result;
 }
 
 aaudio_result_t AAudioServiceStreamMMAP::stopClient(audio_port_handle_t clientHandle) {
-    aaudio_result_t result = mServiceEndpoint->stopClient(clientHandle);
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    aaudio_result_t result = endpoint->stopClient(clientHandle);
     return result;
 }
 
 // Get free-running DSP or DMA hardware position from the HAL.
 aaudio_result_t AAudioServiceStreamMMAP::getFreeRunningPosition(int64_t *positionFrames,
                                                                   int64_t *timeNanos) {
-    sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP{
-            static_cast<AAudioServiceEndpointMMAP *>(mServiceEndpoint.get())};
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP =
+            static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
+
     aaudio_result_t result = serviceEndpointMMAP->getFreeRunningPosition(positionFrames, timeNanos);
     if (result == AAUDIO_OK) {
         Timestamp timestamp(*positionFrames, *timeNanos);
@@ -152,8 +173,15 @@
 // Get timestamp that was written by getFreeRunningPosition()
 aaudio_result_t AAudioServiceStreamMMAP::getHardwareTimestamp(int64_t *positionFrames,
                                                                 int64_t *timeNanos) {
-    sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP{
-            static_cast<AAudioServiceEndpointMMAP *>(mServiceEndpoint.get())};
+
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP =
+            static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
+
     // TODO Get presentation timestamp from the HAL
     if (mAtomicTimestamp.isValid()) {
         Timestamp timestamp = mAtomicTimestamp.read();
@@ -171,7 +199,12 @@
 aaudio_result_t AAudioServiceStreamMMAP::getAudioDataDescription(
         AudioEndpointParcelable &parcelable)
 {
-    sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP{
-            static_cast<AAudioServiceEndpointMMAP *>(mServiceEndpoint.get())};
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP =
+            static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
     return serviceEndpointMMAP->getDownDataDescription(parcelable);
 }
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index 084f996..f55c40d 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -128,6 +128,11 @@
 
     const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
 
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        result = AAUDIO_ERROR_INVALID_STATE;
+        goto error;
+    }
 
     // Is the request compatible with the shared endpoint?
     setFormat(configurationInput.getFormat());
@@ -141,20 +146,20 @@
 
     setSampleRate(configurationInput.getSampleRate());
     if (getSampleRate() == AAUDIO_UNSPECIFIED) {
-        setSampleRate(mServiceEndpoint->getSampleRate());
-    } else if (getSampleRate() != mServiceEndpoint->getSampleRate()) {
+        setSampleRate(endpoint->getSampleRate());
+    } else if (getSampleRate() != endpoint->getSampleRate()) {
         ALOGE("AAudioServiceStreamShared::open() mSampleRate = %d, need %d",
-              getSampleRate(), mServiceEndpoint->getSampleRate());
+              getSampleRate(), endpoint->getSampleRate());
         result = AAUDIO_ERROR_INVALID_RATE;
         goto error;
     }
 
     setSamplesPerFrame(configurationInput.getSamplesPerFrame());
     if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
-        setSamplesPerFrame(mServiceEndpoint->getSamplesPerFrame());
-    } else if (getSamplesPerFrame() != mServiceEndpoint->getSamplesPerFrame()) {
+        setSamplesPerFrame(endpoint->getSamplesPerFrame());
+    } else if (getSamplesPerFrame() != endpoint->getSamplesPerFrame()) {
         ALOGE("AAudioServiceStreamShared::open() mSamplesPerFrame = %d, need %d",
-              getSamplesPerFrame(), mServiceEndpoint->getSamplesPerFrame());
+              getSamplesPerFrame(), endpoint->getSamplesPerFrame());
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
     }
@@ -181,9 +186,9 @@
     }
 
     ALOGD("AAudioServiceStreamShared::open() actual rate = %d, channels = %d, deviceId = %d",
-          getSampleRate(), getSamplesPerFrame(), mServiceEndpoint->getDeviceId());
+          getSampleRate(), getSamplesPerFrame(), endpoint->getDeviceId());
 
-    result = mServiceEndpoint->registerStream(keep);
+    result = endpoint->registerStream(keep);
     if (result != AAUDIO_OK) {
         goto error;
     }
@@ -250,7 +255,13 @@
                                                                 int64_t *timeNanos) {
 
     int64_t position = 0;
-    aaudio_result_t result = mServiceEndpoint->getTimestamp(&position, timeNanos);
+    sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+    if (endpoint == nullptr) {
+        ALOGE("%s() has no endpoint", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+
+    aaudio_result_t result = endpoint->getTimestamp(&position, timeNanos);
     if (result == AAUDIO_OK) {
         int64_t offset = mTimestampPositionOffset.load();
         // TODO, do not go below starting value